SIP: Session Initiation Protocol
draft-ietf-mmusic-sip-12

Internet Engineering Task Force MMUSIC WG
Internet Draft Handley/Schulzrinne/Schooler/Rosenberg
ietf-mmusic-sip-12.txt ISI/Columbia U./Caltech/Bell Labs.
January 15, 1999
Expires: July 1999
 SIP: Session Initiation Protocol
STATUS OF THIS MEMO
 This document is an Internet-Draft. Internet-Drafts are working
 documents of the Internet Engineering Task Force (IETF), its areas,
 and its working groups. Note that other groups may also distribute
 working documents as Internet-Drafts.
 Internet-Drafts are draft documents valid for a maximum of six months
 and may be updated, replaced, or obsoleted by other documents at any
 time. It is inappropriate to use Internet-Drafts as reference
 material or to cite them other than as ``work in progress''.
 To learn the current status of any Internet-Draft, please check the
 ``1id-abstracts.txt'' listing contained in the Internet-Drafts Shadow
 Directories on ftp.is.co.za (Africa), nic.nordu.net (Europe),
 munnari.oz.au (Pacific Rim), ftp.ietf.org (US East Coast), or
 ftp.isi.edu (US West Coast).
 Distribution of this document is unlimited.
 ABSTRACT
 The Session Initiation Protocol (SIP) is an application-
 layer control (signaling) protocol for creating,
 modifying and terminating sessions with one or more
 participants. These sessions include Internet multimedia
 conferences, Internet telephone calls and multimedia
 distribution. Members in a session can communicate via
 multicast or via a mesh of unicast relations, or a
 combination of these.
 SIP invitations used to create sessions carry session
 descriptions which allow participants to agree on a set
 of compatible media types. SIP supports user mobility by
 proxying and redirecting requests to the user's current
 location. Users can register their current location. SIP
 is not tied to any particular conference control
 protocol. SIP is designed to be independent of the
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 lower-layer transport protocol and can be extended with
 additional capabilities.
 This document is a product of the Multi-party Multimedia
 Session Control (MMUSIC) working group of the Internet
 Engineering Task Force. Comments are solicited and
 should be addressed to the working group's mailing list
 at confctrl@isi.edu and/or the authors.
1 Introduction
1.1 Overview of SIP Functionality
 The Session Initiation Protocol (SIP) is an application-layer control
 protocol that can establish, modify and terminate multimedia sessions
 or calls. These multimedia sessions include multimedia conferences,
 distance learning, Internet telephony and similar applications. SIP
 can invite both persons and "robots", such as a media storage
 service. SIP can invite parties to both unicast and multicast
 sessions; the initiator does not necessarily have to be a member of
 the session to which it is inviting. Media and participants can be
 added to an existing session.
 SIP can be used to initiate sessions as well as invite members to
 sessions that have been advertised and established by other means.
 Sessions can be advertised using multicast protocols such as SAP,
 electronic mail, news groups, web pages or directories (LDAP), among
 others.
 SIP transparently supports name mapping and redirection services,
 allowing the implementation of ISDN and Intelligent Network telephony
 subscriber services. These facilities also enable personal mobility.
 In the parlance of telecommunications intelligent network services,
 this is defined as: "Personal mobility is the ability of end users to
 originate and receive calls and access subscribed telecommunication
 services on any terminal in any location, and the ability of the
 network to identify end users as they move. Personal mobility is
 based on the use of a unique personal identity (i.e., personal
 number)." [1]. Personal mobility complements terminal mobility, i.e.,
 the ability to maintain communications when moving a single end
 system from one subnet to another.
 SIP supports five facets of establishing and terminating multimedia
 communications:
 User location: determination of the end system to be used for
 communication;
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 User capabilities: determination of the media and media parameters to
 be used;
 User availability: determination of the willingness of the called
 party to engage in communications;
 Call setup: "ringing", establishment of call parameters at both
 called and calling party;
 Call handling: including transfer and termination of calls.
 SIP can also initiate multi-party calls using a multipoint control
 unit (MCU) or fully-meshed interconnection instead of multicast.
 Internet telephony gateways that connect Public Switched Telephone
 Network (PSTN) parties can also use SIP to set up calls between them.
 SIP is designed as part of the overall IETF multimedia data and
 control architecture currently incorporating protocols such as RSVP
 (RFC 2205 [2]) for reserving network resources, the real-time
 transport protocol (RTP) (RFC 1889 [3]) for transporting real-time
 data and providing QOS feedback, the real-time streaming protocol
 (RTSP) (RFC 2326 [4]) for controlling delivery of streaming media,
 the session announcement protocol (SAP) [5] for advertising
 multimedia sessions via multicast and the session description
 protocol (SDP) (RFC 2327 [6]) for describing multimedia sessions.
 However, the functionality and operation of SIP does not depend on
 any of these protocols.
 SIP can also be used in conjunction with other call setup and
 signaling protocols. In that mode, an end system uses SIP exchanges
 to determine the appropriate end system address and protocol from a
 given address that is protocol-independent. For example, SIP could be
 used to determine that the party can be reached via H.323 [7], obtain
 the H.245 [8] gateway and user address and then use H.225.0 [9] to
 establish the call.
 In another example, SIP might be used to determine that the callee is
 reachable via the PSTN and indicate the phone number to be called,
 possibly suggesting an Internet-to-PSTN gateway to be used.
 SIP does not offer conference control services such as floor control
 or voting and does not prescribe how a conference is to be managed,
 but SIP can be used to introduce conference control protocols. SIP
 does not allocate multicast addresses.
 SIP can invite users to sessions with and without resource
 reservation. SIP does not reserve resources, but can convey to the
 invited system the information necessary to do this.
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1.2 Terminology
 In this document, the key words "MUST", "MUST NOT", "REQUIRED",
 "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
 and "OPTIONAL" are to be interpreted as described in RFC 2119 [10]
 and indicate requirement levels for compliant SIP implementations.
1.3 Definitions
 This specification uses a number of terms to refer to the roles
 played by participants in SIP communications. The definitions of
 client, server and proxy are similar to those used by the Hypertext
 Transport Protocol (HTTP) (RFC 2068 [11]). The terms and generic
 syntax of URI and URL are defined in RFC 2396 [12]. The following
 terms have special significance for SIP.
 Call: A call consists of all participants in a conference invited by
 a common source. A SIP call is identified by a globally unique
 call-id (Section 6.12). Thus, if a user is, for example, invited
 to the same multicast session by several people, each of these
 invitations will be a unique call. A point-to-point Internet
 telephony conversation maps into a single SIP call. In a
 multiparty conference unit (MCU) based call-in conference, each
 participant uses a separate call to invite himself to the MCU.
 Call leg: A call leg is identified by the combination of Call-ID, To
 and From.
 Client: An application program that sends SIP requests. Clients may
 or may not interact directly with a human user. User agents and
 proxies contain clients (and servers).
 Conference: A multimedia session (see below), identified by a common
 session description. A conference can have zero or more members
 and includes the cases of a multicast conference, a full-mesh
 conference and a two-party "telephone call", as well as
 combinations of these. Any number of calls can be used to
 create a conference.
 Downstream: Requests sent in the direction from the caller to the
 callee (i.e., user agent client to user agent server).
 Final response: A response that terminates a SIP transaction, as
 opposed to a provisional response that does not. All 2xx, 3xx,
 4xx, 5xx and 6xx responses are final.
 Initiator, calling party, caller: The party initiating a conference
 invitation. Note that the calling party does not have to be the
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 same as the one creating the conference.
 Invitation: A request sent to a user (or service) requesting
 participation in a session. A successful SIP invitation consists
 of two transactions: an INVITE request followed by an ACK
 request.
 Invitee, invited user, called party, callee: The person or service
 that the calling party is trying to invite to a conference.
 Isomorphic request or response: Two requests or responses are defined
 to be isomorphic for the purposes of this document if they have
 the same values for the Call-ID, To, From and CSeq header
 fields. In addition, requests have to have the same Request-URI.
 Location server: See location service.
 Location service: A location service is used by a SIP redirect or
 proxy server to obtain information about a callee's possible
 location(s). Location services are offered by location servers.
 Location servers MAY be co-located with a SIP server, but the
 manner in which a SIP server requests location services is
 beyond the scope of this document.
 Parallel search: In a parallel search, a proxy issues several
 requests to possible user locations upon receiving an incoming
 request. Rather than issuing one request and then waiting for
 the final response before issuing the next request as in a
 sequential search , a parallel search issues requests without
 waiting for the result of previous requests.
 Provisional response: A response used by the server to indicate
 progress, but that does not terminate a SIP transaction. 1xx
 responses are provisional, other responses are considered final.
 Proxy, proxy server: An intermediary program that acts as both a
 server and a client for the purpose of making requests on behalf
 of other clients. Requests are serviced internally or by passing
 them on, possibly after translation, to other servers. A proxy
 interprets, and, if necessary, rewrites a request message before
 forwarding it.
 Redirect server: A redirect server is a server that accepts a SIP
 request, maps the address into zero or more new addresses and
 returns these addresses to the client. Unlike a proxy server ,
 it does not initiate its own SIP request. Unlike a user agent
 server , it does not accept calls.
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 Registrar: A registrar is a server that accepts REGISTER requests. A
 registrar is typically co-located with a proxy or redirect
 server and MAY offer location services.
 Ringback: Ringback is the signaling tone produced by the calling
 client's application indicating that a called party is being
 alerted (ringing).
 Server: A server is an application program that accepts requests in
 order to service requests and sends back responses to those
 requests. Servers are either proxy, redirect or user agent
 servers or registrars.
 Session: From the SDP specification: "A multimedia session is a set
 of multimedia senders and receivers and the data streams flowing
 from senders to receivers. A multimedia conference is an example
 of a multimedia session." (RFC 2327 [6]) (A session as defined
 for SDP can comprise one or more RTP sessions.) As defined, a
 callee can be invited several times, by different calls, to the
 same session. If SDP is used, a session is defined by the
 concatenation of the user name , session id , network type ,
 address type and address elements in the origin field.
 (SIP) transaction: A SIP transaction occurs between a client and a
 server and comprises all messages from the first request sent
 from the client to the server up to a final (non-1xx) response
 sent from the server to the client. A transaction is identified
 by the CSeq sequence number (Section 6.17) within a single call
 leg. The ACK request has the same CSeq number as the
 corresponding INVITE request, but comprises a transaction of its
 own.
 Upstream: Responses sent in the direction from the user agent server
 to the user agent client.
 URL-encoded: A character string encoded according to RFC 1738,
 Section 2.2 [13].
 User agent client (UAC), calling user agent: A user agent client is a
 client application that initiates the SIP request.
 User agent server (UAS), called user agent: A user agent server is a
 server application that contacts the user when a SIP request is
 received and that returns a response on behalf of the user. The
 response accepts, rejects or redirects the request.
 An application program MAY be capable of acting both as a client and
 a server. For example, a typical multimedia conference control
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 application would act as a user agent client to initiate calls or to
 invite others to conferences and as a user agent server to accept
 invitations. The properties of the different SIP server types are
 summarized in Table 1.
 property redirect proxy user agent registrar
 server server server
 __________________________________________________________________
 also acts as a SIP client no yes no no
 returns 1xx status yes yes yes yes
 returns 2xx status no yes yes yes
 returns 3xx status yes yes yes yes
 returns 4xx status yes yes yes yes
 returns 5xx status yes yes yes yes
 returns 6xx status no yes yes no
 inserts Via header no yes no no
 accepts ACK yes yes yes no
 Table 1: Properties of the different SIP server types
1.4 Overview of SIP Operation
 This section explains the basic protocol functionality and operation.
 Callers and callees are identified by SIP addresses, described in
 Section 1.4.1. When making a SIP call, a caller first locates the
 appropriate server (Section 1.4.2) and then sends a SIP request
 (Section 1.4.3). The most common SIP operation is the invitation
 (Section 1.4.4). Instead of directly reaching the intended callee, a
 SIP request may be redirected or may trigger a chain of new SIP
 requests by proxies (Section 1.4.5). Users can register their
 location(s) with SIP servers (Section 4.2.6).
1.4.1 SIP Addressing
 The "objects" addressed by SIP are users at hosts, identified by a
 SIP URL. The SIP URL takes a form similar to a mailto or telnet URL,
 i.e., user@host. The user part is a user name or a telephone number.
 The host part is either a domain name or a numeric network address.
 See section 2 for a detailed discussion of SIP URL's.
 A user's SIP address can be obtained out-of-band, can be learned via
 existing media agents, can be included in some mailers' message
 headers, or can be recorded during previous invitation interactions.
 In many cases, a user's SIP URL can be guessed from their email
 address.
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 A SIP URL address can designate an individual (possibly located at
 one of several end systems), the first available person from a group
 of individuals or a whole group. The form of the address, for
 example, sip:sales@example.com , is not sufficient, in general, to
 determine the intent of the caller.
 If a user or service chooses to be reachable at an address that is
 guessable from the person's name and organizational affiliation, the
 traditional method of ensuring privacy by having an unlisted "phone"
 number is compromised. However, unlike traditional telephony, SIP
 offers authentication and access control mechanisms and can avail
 itself of lower-layer security mechanisms, so that client software
 can reject unauthorized or undesired call attempts.
1.4.2 Locating a SIP Server
 When a client wishes to send a request, the client either sends it to
 a locally configured SIP proxy server (as in HTTP), independent of
 the Request-URI, or sends it to the IP address and port corresponding
 to the Request-URI.
 For the latter case, the client must determine the protocol, port and
 IP address of a server to which to send the request. A client SHOULD
 follow the steps below to obtain this information, but MAY follow the
 alternative, optional procedure defined in Appendix D. At each step,
 unless stated otherwise, the client SHOULD try to contact a server at
 the port number listed in the Request-URI. If no port number is
 present in the Request-URI, the client uses port 5060. If the
 Request-URI specifies a protocol (TCP or UDP), the client contacts
 the server using that protocol. If no protocol is specified, the
 client tries UDP (if UDP is supported). If the attempt fails, or if
 the client doesn't support UDP but supports TCP, it then tries TCP.
 A client SHOULD be able to interpret explicit network notifications
 (such as ICMP messages) which indicate that a server is not
 reachable, rather than relying solely on timeouts. (For socket-based
 programs: For TCP, connect() returns ECONNREFUSED if the client could
 not connect to a server at that address. For UDP, the socket needs to
 be bound to the destination address using connect() rather than
 sendto() or similar so that a second write() fails with ECONNREFUSED
 if there is no server listening) If the client finds the server is
 not reachable at a particular address, it SHOULD behave as if it had
 received a 400-class error response to that request.
 The client tries to find one or more addresses for the SIP server by
 querying DNS. The procedure is as follows:
 1. If the host portion of the Request-URI is an IP address,
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 the client contacts the server at the given address.
 Otherwise, the client proceeds to the next step.
 2. The client queries the DNS server for address records for
 the host portion of the Request-URI. If the DNS server
 returns no address records, the client stops, as it has
 been unable to locate a server. By address record, we mean
 A RR's, AAAA RR's, or other similar address records, chosen
 according to the client's network protocol capabilities.
 There are no mandatory rules on how to select a host name
 for a SIP server. Users are encouraged to name their SIP
 servers using the sip.domainname (i.e., sip.example.com)
 convention, as specified in RFC 2219 [16]. Users may only
 know an email address instead of a full SIP URL for a
 callee, however. In that case, implementations may be able
 to increase the likelihood of reaching a SIP server for
 that domain by constructing a SIP URL from that email
 address by prefixing the host name with "sip.". In the
 future, this mechanism is likely to become unnecessary as
 better DNS techniques, such as the one in Appendix D,
 become widely available.
 A client MAY cache a successful DNS query result. A successful query
 is one which contained records in the answer, and a server was
 contacted at one of the addresses from the answer. When the client
 wishes to send a request to the same host, it MUST start the search
 as if it had just received this answer from the name server. The
 client MUST follow the procedures in RFC1035 [15] regarding DNS cache
 invalidation when the DNS time-to-live expires.
1.4.3 SIP Transaction
 Once the host part has been resolved to a SIP server, the client
 sends one or more SIP requwwests to that server and receives one or
 more responses from the server. A request (and its retransmissions)
 together with the responses triggered by that request make up a SIP
 transaction. All responses to a request contain the same values in
 the Call-ID, CSeq, To, and From fields (with the possible addition of
 a tag in the To field (section 6.37)). This allows responses to be
 matched with requests. The ACK request following an INVITE is not
 part of the transaction since it may traverse a different set of
 hosts.
 If TCP is used, request and responses within a single SIP transaction
 are carried over the same TCP connection (see Section 10). Several
 SIP requests from the same client to the same server MAY use the same
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 TCP connection or MAY use a new connection for each request.
 If the client sent the request via unicast UDP, the response is sent
 to the address contained in the next Via header field (Section 6.40)
 of the response. If the request is sent via multicast UDP, the
 response is directed to the same multicast address and destination
 port. For UDP, reliability is achieved using retransmission (Section
 10).
 The SIP message format and operation is independent of the transport
 protocol.
1.4.4 SIP Invitation
 A successful SIP invitation consists of two requests, INVITE followed
 by ACK. The INVITE (Section 4.2.1) request asks the callee to join a
 particular conference or establish a two-party conversation. After
 the callee has agreed to participate in the call, the caller confirms
 that it has received that response by sending an ACK (Section 4.2.2)
 request. If the caller no longer wants to participate in the call, it
 sends a BYE request instead of an ACK.
 The INVITE request typically contains a session description, for
 example written in SDP (RFC 2327 [6]) format, that provides the
 called party with enough information to join the session. For
 multicast sessions, the session description enumerates the media
 types and formats that are allowed to be distributed to that session.
 For a unicast session, the session description enumerates the media
 types and formats that the caller is willing to receive and where it
 wishes the media data to be sent. In either case, if the callee
 wishes to accept the call, it responds to the invitation by returning
 a similar description listing the media it wishes to receive. For a
 multicast session, the callee SHOULD only return a session
 description if it is unable to receive the media indicated in the
 caller's description or wants to receive data via unicast.
 The protocol exchanges for the INVITE method are shown in Fig. 1 for
 a proxy server and in Fig. 2 for a redirect server. (Note that the
 messages shown in the figures have been abbreviated slightly.) In
 Fig. 1, the proxy server accepts the INVITE request (step 1),
 contacts the location service with all or parts of the address (step
 2) and obtains a more precise location (step 3). The proxy server
 then issues a SIP INVITE request to the address(es) returned by the
 location service (step 4). The user agent server alerts the user
 (step 5) and returns a success indication to the proxy server (step
 6). The proxy server then returns the success result to the original
 caller (step 7). The receipt of this message is confirmed by the
 caller using an ACK request, which is forwarded to the callee (steps
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 8 and 9). Note that an ACK can also be sent directly to the callee,
 bypassing the proxy. All requests and responses have the same Call-
 ID.
 +....... cs.columbia.edu .......+
 : :
 : (~~~~~~~~~~) :
 : ( location ) :
 : ( service ) :
 : (~~~~~~~~~~) :
 : ^ | :
 : | hgs@lab :
 : 2| 3| :
 : | | :
 : henning | : 
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | \/ 4: INVITE 5: ring :
: cz@cs.tu-berlin.de ========================>(~~~~~~)=========>(~~~~~~) :
: <........................( )<.........( ) :
: : 7: 200 OK : ( )6: 200 OK ( ) :
: : : ( work ) ( lab ) :
: : 8: ACK : ( )9: ACK ( ) :
: ========================>(~~~~~~)=========>(~~~~~~) :
+.....................+ +...............................+
 ====> SIP request 
 ....> SIP response 
 
 ^
 | non-SIP protocols 
 |
 
 Figure 1: Example of SIP proxy server
 The redirect server shown in Fig. 2 accepts the INVITE request (step
 1), contacts the location service as before (steps 2 and 3) and,
 instead of contacting the newly found address itself, returns the
 address to the caller (step 4), which is then acknowledged via an ACK
 request (step 5). The caller issues a new request, with the same
 call-ID but a higher CSeq, to the address returned by the first
 server (step 6). In the example, the call succeeds (step 7). The
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 caller and callee complete the handshake with an ACK (step 8).
 The next section discusses what happens if the location service
 returns more than one possible alternative.
1.4.5 Locating a User
 A callee may move between a number of different end systems over
 time. These locations can be dynamically registered with the SIP
 server (Sections 1.4.7, 4.2.6). A location server MAY also use one or
 more other protocols, such as finger (RFC 1288 [17]), rwhois (RFC
 2167 [18]), LDAP (RFC 1777 [19]), multicast-based protocols [20] or
 operating-system dependent mechanisms to actively determine the end
 system where a user might be reachable. A location server MAY return
 several locations because the user is logged in at several hosts
 simultaneously or because the location server has (temporarily)
 inaccurate information. The SIP server combines the results to yield
 a list of a zero or more locations.
 The action taken on receiving a list of locations varies with the
 type of SIP server. A SIP redirect server returns the list to the
 client as Contact headers (Section 6.13). A SIP proxy server can
 sequentially or in parallel try the addresses until the call is
 successful (2xx response) or the callee has declined the call (6xx
 response). With sequential attempts, a proxy server can implement an
 "anycast" service.
 If a proxy server forwards a SIP request, it MUST add itself to the
 end of the list of forwarders noted in the Via (Section 6.40)
 headers. The Via trace ensures that replies can take the same path
 back, ensuring correct operation through compliant firewalls and
 avoiding request loops. On the response path, each host MUST remove
 its Via, so that routing internal information is hidden from the
 callee and outside networks. A proxy server MUST check that it does
 not generate a request to a host listed in the Via sent-by, via-
 received or via-maddr parameters (Section 6.40). (Note: If a host has
 several names or network addresses, this does not always work. Thus,
 each host also checks if it is part of the Via list.)
 A SIP invitation may traverse more than one SIP proxy server. If one
 of these "forks" the request, i.e., issues more than one request in
 response to receiving the invitation request, it is possible that a
 client is reached, independently, by more than one copy of the
 invitation request. Each of these copies bears the same Call-ID. The
 user agent MUST return the same status response returned in the first
 response. Duplicate requests are not an error.
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1.4.6 Changing an Existing Session
 In some circumstances, it is desirable to change the parameters of an
 existing session. This is done by re-issuing the INVITE, using the
 same Call-ID, but a new or different body or header fields to convey
 the new information. This re INVITE MUST have a higher CSeq than any
 previous request from the client to the server.
 For example, two parties may have been conversing and then want to
 add a third party, switching to multicast for efficiency. One of the
 participants invites the third party with the new multicast address
 and simultaneously sends an INVITE to the second party, with the new
 multicast session description, but with the old call identifier.
1.4.7 Registration Services
 The REGISTER request allows a client to let a proxy or redirect
 server know at which address(es) it can be reached. A client MAY also
 use it to install call handling features at the server.
1.5 Protocol Properties
1.5.1 Minimal State
 A single conference session or call involves one or more SIP
 request-response transactions. Proxy servers do not have to keep
 state for a particular call, however, they MAY maintain state for a
 single SIP transaction, as discussed in Section 12. For efficiency, a
 server MAY cache the results of location service requests.
1.5.2 Lower-Layer-Protocol Neutral
 SIP makes minimal assumptions about the underlying transport and
 network-layer protocols. The lower-layer can provide either a packet
 or a byte stream service, with reliable or unreliable service.
 In an Internet context, SIP is able to utilize both UDP and TCP as
 transport protocols, among others. UDP allows the application to more
 carefully control the timing of messages and their retransmission, to
 perform parallel searches without requiring TCP connection state for
 each outstanding request, and to use multicast. Routers can more
 readily snoop SIP UDP packets. TCP allows easier passage through
 existing firewalls.
 When TCP is used, SIP can use one or more connections to attempt to
 contact a user or to modify parameters of an existing conference.
 Different SIP requests for the same SIP call MAY use different TCP
 connections or a single persistent connection, as appropriate.
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 +....... cs.columbia.edu .......+
 : :
 : (~~~~~~~~~~) :
 : ( location ) :
 : ( service ) :
 : (~~~~~~~~~~) :
 : ^ | :
 : | hgs@lab :
 : 2| 3| :
 : | | :
 : henning| : 
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | \/ : 
: cz@cs.tu-berlin.de =======================>(~~~~~~) : 
: | ^ | <.......................( ) :
: | . | : 4: 302 Moved : ( ) :
: | . | : hgs@lab : ( work ) :
: | . | : : ( ) :
: | . | : 5: ACK : ( ) :
: | . | =======================>(~~~~~~) :
: | . | : : :
+.......|...|.........+ : :
 | . | : :
 | . | : :
 | . | : :
 | . | : :
 | . | 6: INVITE hgs@lab.cs.columbia.edu (~~~~~~) : 
 | . ==================================================> ( ) :
 | ..................................................... ( ) :
 | 7: 200 OK : ( lab ) : 
 | : ( ) :
 | 8: ACK : ( ) :
 ======================================================> (~~~~~~) :
 +...............................+ 
 
 ====> SIP request 
 ....> SIP response 
 
 ^
 | non-SIP protocols 
 |
 Figure 2: Example of SIP redirect server
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 For concreteness, this document will only refer to Internet
 protocols. However, SIP MAY also be used directly with protocols
 such as ATM AAL5, IPX, frame relay or X.25. The necessary naming
 conventions are beyond the scope of this document. User agents SHOULD
 implement both UDP and TCP transport. Proxy, registrar, and redirect
 servers MUST implement both UDP and TCP transport.
1.5.3 Text-Based
 SIP is text-based, using ISO 10646 in UTF-8 encoding throughout. This
 allows easy implementation in languages such as Java, Tcl and Perl,
 allows easy debugging, and most importantly, makes SIP flexible and
 extensible. As SIP is used for initiating multimedia conferences
 rather than delivering media data, it is believed that the additional
 overhead of using a text-based protocol is not significant.
2 SIP Uniform Resource Locators
 SIP URLs are used within SIP messages to indicate the originator
 (From), current destination (Request-URI) and final recipient (To) of
 a SIP request, and to specify redirection addresses (Contact). A SIP
 URL can also be embedded in web pages or other hyperlinks to indicate
 that a particular user or service can be called via SIP. When used as
 a hyperlink, the SIP URL indicates the use of the INVITE method.
 The SIP URL scheme is defined to allow setting SIP request-header
 fields and the SIP message-body.
 This corresponds to the use of mailto: URLs. It makes it
 possible, for example, to specify the subject, urgency or
 media types of calls initiated through a web page or as
 part of an email message.
 A SIP URL follows the guidelines of RFC 2396 [12] and has the syntax
 shown in Fig. 3. The syntax is described using Augmented Backus-Naur
 Form (See Section C). Note that reserved characters have to be
 escaped and that the "set of characters reserved within any given URI
 component is defined by that component. In general, a character is
 reserved if the semantics of the URI changes if the character is
 replaced with its escaped US-ASCII encoding" [12].
 The URI character classes referenced above are described in Appendix
 C.
 The components of the SIP URI have the following meanings.
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 SIP-URL = "sip:" [ userinfo "@" ] hostport
 url-parameters [ headers ]
 userinfo = user [ ":" password ]
 user = *( unreserved | escaped
 | "&" | "=" | "+" | "$" | "," )
 password = *( unreserved | escaped
 | "&" | "=" | "+" | "$" | "," )
 hostport = host [ ":" port ]
 host = hostname | IPv4address
 hostname = *( domainlabel "." ) toplabel [ "." ]
 domainlabel = alphanum | alphanum *( alphanum | "-" ) alphanum
 toplabel = alpha | alpha *( alphanum | "-" ) alphanum
 IPv4address = 1*digit "." 1*digit "." 1*digit "." 1*digit
 port = *digit
 url-parameters = *( ";" url-parameter )
 url-parameter = transport-param | user-param | method-param
 | ttl-param | maddr-param | other-param
 transport-param = "transport=" ( "udp" | "tcp" )
 ttl-param = "ttl=" ttl
 ttl = 1*3DIGIT ; 0 to 255
 maddr-param = "maddr=" host
 user-param = "user=" ( "phone" | "ip" )
 method-param = "method=" Method
 tag-param = "tag=" UUID
 UUID = 1*( hex | "-" )
 other-param = ( token | ( token "=" ( token | quoted-string )))
 headers = "?" header *( "&" header )
 header = hname "=" hvalue
 hname = 1*uric
 hvalue = *uric
 uric = reserved | unreserved | escaped
 reserved = ";" | "/" | "?" | ":" | "@" | "&" | "=" | "+" |
 "$" | ","
 digits = 1*DIGIT
 Figure 3: SIP URL syntax
 user: If the host is an Internet telephony gateway, the user field
 MAY also encode a telephone number using the notation of
 telephone-subscriber (Fig. 4). The telephone number is a special
 case of a user name and cannot be distinguished by a BNF. Thus,
 a URL parameter, user, is added to distinguish telephone numbers
 from user names. The phone identifier is to be used when
 connecting to a telephony gateway. Even without this parameter,
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Internet Draft SIP January 15, 1999
 telephone-subscriber = global-phone-number | local-phone-number
 global-phone-number = "+" 1*phonedigit [isdn-subaddress]
 [post-dial]
 local-phone-number = 1*(phonedigit | dtmf-digit | 
 pause-character) [isdn-subaddress] 
 [post-dial]
 isdn-subaddress = ";" "isub=" 1*phonedigit
 post-dial = ";" "postd=" 1*(phonedigit | dtmf-digit
 | pause-character)
 phonedigit = DIGIT | visual-separator
 visual-separator = "-" | "."
 pause-character = one-second-pause | wait-for-dial-tone
 one-second-pause = "p"
 wait-for-dial-tone = "w"
 dtmf-digit = "*" | "#" | "A" | "B" | "C" | "D"
 Figure 4: SIP URL syntax; telephone subscriber
 recipients of SIP URLs MAY interpret the pre-@ part as a phone
 number if local restrictions on the name space for user name
 allow it.
 password: The SIP scheme MAY use the format "user:password" in the
 userinfo field. The use of passwords in the userinfo is NOT
 RECOMMENDED, because the passing of authentication information
 in clear text (such as URIs) has proven to be a security risk in
 almost every case where it has been used.
 host: The mailto: URL and RFC 822 email addresses require that
 numeric host addresses ("host numbers") are enclosed in square
 brackets (presumably, since host names might be numeric), while
 host numbers without brackets are used for all other URLs. The
 SIP URL requires the latter form, without brackets.
 The issue of IPv6 literal addresses in URLs is being looked at
 elsewhere in the IETF. SIP implementers are advised to keep up to
 date on that activity.
 port: The port number to send a request to. If not present, the
 procedures outlined in Section 1.4.2 are used to determine the
 port number to send a request to.
 URL parameters: SIP URLs can define specific parameters of the
 request. URL parameters are added after the host component and
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 are separated by semi-colons. The transport parameter determines
 the the transport mechanism (UDP or TCP). UDP is to be assumed
 when no explicit transport parameter is included. The maddr
 parameter provides the server address to be contacted for this
 user, overriding the address supplied in the host field. This
 address is typically a multicast address, but could also be the
 address of a backup server. The ttl parameter determines the
 time-to-live value of the UDP multicast packet and MUST only be
 used if maddr is a multicast address and the transport protocol
 is UDP. The user parameter was described above. For example, to
 specify to call j.doe@big.com using multicast to 239.255.255.1
 with a ttl of 15, the following URL would be used:
 sip:j.doe@big.com;maddr=239.255.255.1;ttl=15
 The transport, maddr, and ttl parameters MUST NOT be used in the From
 and To header fields and the Request-URI; they are ignored if
 present.
 Headers: Headers of the SIP request can be defined with the "?"
 mechanism within a SIP URL. The special hname "body" indicates
 that the associated hvalue is the message-body of the SIP INVITE
 request. Headers MUST NOT be used in the From and To header
 fields and the Request-URI; they are ignored if present. hname
 and hvalue are encodings of a SIP header name and value,
 respectively. All URL reserved characters in the header names
 and values MUST be escaped.
 Method: The method of the SIP request can be specified with the
 method parameter. This parameter MUST NOT be used in the From
 and To header fields and the Request-URI; they are ignored if
 present.
 Table 2 summarizes where the components of the SIP URL can be used
 and what default values they assume if not present.
 Examples of SIP URLs are:
 sip:j.doe@big.com
 sip:j.doe:secret@big.com;transport=tcp
 sip:j.doe@big.com?subject=project
 sip:+1-212-555-1212:1234@gateway.com;user=phone
 sip:1212@gateway.com
 sip:alice@10.1.2.3
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 default Req.-URI To From Contact external
 user -- x x x x x
 password -- x x x x
 host mandatory x x x x x
 port 5060 x x x x x
 user-param ip x x x x x
 method INVITE x x
 maddr-param -- x x
 ttl-param 1 x x
 transp.-param -- x x
 headers -- x x
 Table 2: Use and default values of URL components for SIP headers,
 Request-URI and references
 sip:alice@example.com
 sip:alice
 sip:alice@registrar.com;method=REGISTER
 Within a SIP message, URLs are used to indicate the source and
 intended destination of a request, redirection addresses and the
 current destination of a request. Normally all these fields will
 contain SIP URLs.
 SIP URLs are case-insensitive, so that for example the two URLs
 sip:j.doe@example.com and SIP:J.Doe@Example.com are equivalent. All
 URL parameters are included when comparing SIP URLs for equality.
 SIP header fields MAY contain non-SIP URLs. As an example, if a call
 from a telephone is relayed to the Internet via SIP, the SIP From
 header field might contain a phone URL.
3 SIP Message Overview
 SIP is a text-based protocol and uses the ISO 10646 character set in
 UTF-8 encoding (RFC 2279 [21]). Senders MUST terminate lines with a
 CRLF, but receivers MUST also interpret CR and LF by themselves as
 line terminators.
 Except for the above difference in character sets, much of the
 message syntax is and header fields are identical to HTTP/1.1; rather
 than repeating the syntax and semantics here we use [HX.Y] to refer
 to Section X.Y of the current HTTP/1.1 specification (RFC 2068 [11]).
 In addition, we describe SIP in both prose and an augmented Backus-
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Internet Draft SIP January 15, 1999
 Naur form (ABNF). See section C for an overview of ABNF.
 Note, however, that SIP is not an extension of HTTP.
 Unlike HTTP, SIP MAY use UDP. When sent over TCP or UDP, multiple SIP
 transactions can be carried in a single TCP connection or UDP
 datagram. UDP datagrams, including all headers, SHOULD NOT be larger
 than the path maximum transmission unit (MTU) if the MTU is known, or
 1500 bytes if the MTU is unknown.
 The 1500 bytes accommodates encapsulation within the
 "typical" ethernet MTU without IP fragmentation. Recent
 studies [22] indicate that an MTU of 1500 bytes is a
 reasonable assumption. The next lower common MTU values are
 1006 bytes for SLIP and 296 for low-delay PPP (RFC 1191
 [23]). Thus, another reasonable value would be a message
 size of 950 bytes, to accommodate packet headers within the
 SLIP MTU without fragmentation.
 A SIP message is either a request from a client to a server, or a
 response from a server to a client.
 SIP-message = Request | Response
 Both Request (section 4) and Response (section 5) messages use the
 generic-message format of RFC 822 [24] for transferring entities (the
 body of the message). Both types of messages consist of a start-line,
 one or more header fields (also known as "headers"), an empty line
 (i.e., a line with nothing preceding the carriage-return line-feed
 (CRLF)) indicating the end of the header fields, and an optional
 message-body. To avoid confusion with similar-named headers in HTTP,
 we refer to the headers describing the message body as entity
 headers. These components are described in detail in the upcoming
 sections.
 generic-message = start-line
 *message-header
 CRLF
 [ message-body ]
 start-line = Request-Line | ;Section 4.1
 Status-Line ;Section 5.1
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Internet Draft SIP January 15, 1999
 message-header = ( general-header
 | request-header
 | response-header
 | entity-header )
 In the interest of robustness, any leading empty line(s) MUST be
 ignored. In other words, if the Request or Response message begins
 with one or more CRLF, CR, or LFs, these characters MUST be ignored.
4 Request
 The Request message format is shown below:
 Request = Request-Line ; Section 4.1
 *( general-header
 | request-header
 | entity-header )
 CRLF
 [ message-body ] ; Section 8
4.1 Request-Line
 The Request-Line begins with a method token, followed by the
 Request-URI and the protocol version, and ending with CRLF. The
 elements are separated by SP characters. No CR or LF are allowed
 except in the final CRLF sequence.
 Request-Line = Method SP Request-URI SP SIP-Version CRLF
4.2 Methods
 The methods are defined below. Methods that are not supported by a
 proxy or redirect server are treated by that server as if they were
 an OPTIONS method and forwarded accordingly. Methods that are not
 supported by a user agent server or registrar cause a 501 (Not
 Implemented) response to be returned (Section 7). As in HTTP, the
 Method token is case-sensitive.
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Internet Draft SIP January 15, 1999
 general-header = Accept ; Section 6.7
 | Accept-Encoding ; Section 6.8
 | Accept-Language ; Section 6.9
 | Call-ID ; Section 6.12
 | Contact ; Section 6.13
 | CSeq ; Section 6.17
 | Date ; Section 6.18
 | Encryption ; Section 6.19
 | Expires ; Section 6.20
 | From ; Section 6.21
 | Record-Route ; Section 6.29
 | Timestamp ; Section 6.36
 | To ; Section 6.37
 | Via ; Section 6.40
 entity-header = Content-Encoding ; Section 6.14
 | Content-Length ; Section 6.15
 | Content-Type ; Section 6.16
 request-header = Authorization ; Section 6.11
 | Contact ; Section 6.13
 | Hide ; Section 6.22
 | Max-Forwards ; Section 6.23
 | Organization ; Section 6.24
 | Priority ; Section 6.25
 | Proxy-Authorization ; Section 6.27
 | Proxy-Require ; Section 6.28
 | Route ; Section 6.33
 | Require ; Section 6.30
 | Response-Key ; Section 6.31
 | Subject ; Section 6.35
 | User-Agent ; Section 6.39
 response-header = Allow ; Section 6.10
 | Proxy-Authenticate ; Section 6.26
 | Retry-After ; Section 6.32
 | Server ; Section 6.34
 | Unsupported ; Section 6.38
 | Warning ; Section 6.41
 | WWW-Authenticate ; Section 6.42
 Table 3: SIP headers
 Method = "INVITE" | "ACK" | "OPTIONS" | "BYE"
 | "CANCEL" | "REGISTER"
4.2.1 INVITE
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 The INVITE method indicates that the user or service is being invited
 to participate in a session. The message body contains a description
 of the session to which the callee is being invited. For two-party
 calls, the caller indicates the type of media it is able to receive
 and possibly the media it is willing to send as well as their
 parameters such as network destination. A success response MUST
 indicate in its message body which media the callee wishes to receive
 and MAY indicate the media the callee is going to send.
 Not all session description formats have the ability to
 indicate sending media.
 A server MAY automatically respond to an invitation for a conference
 the user is already participating in, identified either by the SIP
 Call-ID or a globally unique identifier within the session
 description, with a 200 (OK) response.
 If a user agent receives an INVITE request for an existing call leg
 with a higher CSeq sequence number than any previous INVITE for the
 same Call-ID, it MUST check any version identifiers in the session
 description or, if there are no version identifiers, the content of
 the session description to see if it has changed. It MUST also
 inspect any other header fields for changes. If there is a change,
 the user agent MUST update any internal state or information
 generated as a result of that header. If the session description has
 changed, the user agent server MUST adjust the session parameters
 accordingly, possibly after asking the user for confirmation.
 (Versioning of the session description can be used to accommodate the
 capabilities of new arrivals to a conference, add or delete media or
 change from a unicast to a multicast conference.)
 This method MUST be supported by SIP proxy, redirect and user agent
 servers as well as clients.
4.2.2 ACK
 The ACK request confirms that the client has received a final
 response to an INVITE request. (ACK is used only with INVITE
 requests.) 2xx responses are acknowledged by client user agents, all
 other final responses by the first proxy or client user agent to
 receive the response. The Via is always initialized to the host that
 originates the ACK request, i.e., the client user agent after a 2xx
 response or the first proxy to receive a non-2xx final response. The
 ACK request is forwarded as the corresponding INVITE request, based
 on its Request-URI. See Section 10 for details.
 The ACK request MAY contain a message body with the final session
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Internet Draft SIP January 15, 1999
 description to be used by the callee. If the ACK message body is
 empty, the callee uses the session description in the INVITE request.
 A proxy server receiving an ACK request after having sent a 3xx, 4xx,
 5xx, or 6xx response must make a determination about whether the ACK
 is for it, or for some user agent or proxy server further downstream.
 This determination is made by examining the tag in the To field. If
 the tag in the ACK To header field matches the tag in the To header
 field of the response, and the From, CSeq and Call-ID header fields
 in the response match those in the ACK, the ACK is meant for the
 proxy server. Otherwise, the ACK SHOULD be proxied downstream as any
 other request.
 It is possible for a user agent client or proxy server to
 receive multiple 3xx, 4xx, 5xx, and 6xx responses to a
 request along a single branch. This can happen under
 various error conditions, typically when a forking proxy
 transitions from stateful to stateless before receiving all
 responses. The various responses will all be identical,
 except for the tag in the To field, which is different for
 each one. It can therefore be used as a means to
 disambiguate them.
 This method MUST be supported by SIP proxy, redirect and user agent
 servers as well as clients.
4.2.3 OPTIONS
 The server is being queried as to its capabilities. A server that
 believes it can contact the user, such as a user agent where the user
 is logged in and has been recently active, MAY respond to this
 request with a capability set. A called user agent MAY return a
 status reflecting how it would have responded to an invitation, e.g.,
 600 (Busy). Such a server SHOULD return an Allow header field
 indicating the methods that it supports. Proxy and redirect servers
 simply forward the request without indicating their capabilities.
 This method MUST be supported by SIP proxy, redirect and user agent
 servers, registrars and clients.
4.2.4 BYE
 The user agent client uses BYE to indicate to the server that it
 wishes to release the call. A BYE request is forwarded like an INVITE
 request and MAY be issued by either caller or callee. A party to a
 call SHOULD issue a BYE request before releasing a call ("hanging
 up"). A party receiving a BYE request MUST cease transmitting media
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 streams specifically directed at the party issuing the BYE request.
 If the INVITE request contained a Contact header, the callee SHOULD
 send a BYE request to that address rather than the From address.
 This method MUST be supported by proxy servers and SHOULD be
 supported by redirect and user agent SIP servers.
4.2.5 CANCEL
 The CANCEL request cancels a pending request with the same Call-ID,
 To, From and CSeq (sequence number only) header field values, but
 does not affect a completed request. (A request is considered
 completed if the server has returned a final status response.)
 A user agent client or proxy client MAY issue a CANCEL request at any
 time. A proxy, in particular, MAY choose to send a CANCEL to
 destinations that have not yet returned a final response after it has
 received a 2xx or 6xx response for one or more of the parallel-search
 requests. A proxy that receives a CANCEL request forwards the request
 to all destinations with pending requests.
 The Call-ID, To, the numeric part of CSeq and From headers in the
 CANCEL request are identical to those in the original request. This
 allows a CANCEL request to be matched with the request it cancels.
 However, to allow the client to distinguish responses to the CANCEL
 from those to the original request, the CSeq Method component is set
 to CANCEL. The Via header field is initialized to the proxy issuing
 the CANCEL request. (Thus, responses to this CANCEL request only
 reach the issuing proxy.)
 Once a user agent server has received a CANCEL, it MUST NOT issue a
 2xx response for the cancelled original request.
 A redirect or user agent server receiving a CANCEL request responds
 with a status of 200 (OK) if the transaction exists and a status of
 481 (Transaction Does Not Exist) if not, but takes no further action.
 In particular, any existing call is unaffected.
 The BYE request cannot be used to cancel branches of a
 parallel search, since several branches may, through
 intermediate proxies, find the same user agent server and
 then terminate the call. To terminate a call instead of
 just pending searches, the UAC must use BYE instead of or
 in addition to CANCEL. While CANCEL can terminate any
 pending request other than ACK or CANCEL, it is typically
 useful only for INVITE. 200 responses to INVITE and 200
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Internet Draft SIP January 15, 1999
 responses to CANCEL are distinguished by the method in the
 Cseq header field, so there is no ambiguity.
 This method MUST be supported by proxy servers and SHOULD be
 supported by all other SIP server types.
4.2.6 REGISTER
 A client uses the REGISTER method to register the address listed in
 the To header field with a SIP server.
 A user agent MAY register with a local server on startup by sending a
 REGISTER request to the well-known "all SIP servers" multicast
 address "sip.mcast.net" (224.0.1.75). This request SHOULD be scoped
 to ensure it is not forwarded beyond the boundaries of the
 administrative system. This MAY be done with either TTL or
 administrative scopes[25], depending on what is implemented in the
 network. SIP user agents MAY listen to that address and use it to
 become aware of the location of other local users [20]; however, they
 do not respond to the request. A user agent MAY also be configured
 with the address of a registrar server to which it sends a REGISTER
 request upon startup.
 Requests are processed in the order received. Clients SHOULD avoid
 sending a new registration (as opposed to a retransmission) until
 they have received the response from the server for the previous one.
 Clients may register from different locations, by necessity
 using different Call-ID values. Thus, the CSeq value cannot
 be used to enforce ordering. Since registrations are
 additive, ordering is less of a problem than if each
 REGISTER request completely replaced all earlier ones.
 The meaning of the REGISTER request-header fields is defined as
 follows. We define "address-of-record" as the SIP address that the
 registry knows the registrand, typically of the form "user@domain"
 rather than "user@host". In third-party registration, the entity
 issuing the request is different from the entity being registered.
 To: The To header field contains the address-of-record whose
 registration is to be created or updated.
 From: The From header field contains the address-of-record of the
 person responsible for the registration. For first-party
 registration, it is identical to the To header field value.
 Request-URI: The Request-URI names the destination of the
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 registration request, i.e., the domain of the registrar. The
 user name MUST be empty. Generally, the domains in the Request-
 URI and the To header field have the same value; however, it is
 possible to register as a "visitor", while maintaining one's
 name. For example, a traveller sip:alice@acme.com (To) might
 register under the Request-URI sip:atlanta.hiayh.org , with the
 former as the To header field and the latter as the Request-URI.
 The request is no longer forwarded once it reached the server
 whose authoritative domain is the one listed in the Request-URI.
 Call-ID: All registrations from a client SHOULD use the same Call-ID
 header value, at least within the same reboot cycle.
 Cseq: Registrations with the same Call-ID MUST have increasing CSeq
 header values. However, the server does not reject out-of-order
 requests.
 Contact: The request MAY contain a Contact header field; future non-
 REGISTER requests for the URI given in the To header field
 SHOULD be directed to the address(es) given in the Contact
 header.
 If the request does not contain a Contact header, the registration
 remains unchanged.
 This is useful to obtain the current list of registrations
 in the response. Registrations using SIP URIs that differ
 in one or more of host, port, transport-param or maddr-
 param (see Figure 3) from an existing registration are
 added to the list of registrations. Other URI types are
 compared according to the standard URI equivalency rules
 for the URI schema. If the URIs are equivalent to that of
 an existing registration, the new registration replaces the
 old one if it has a higher q value or, for the same value
 of q, if the ttl value is higher. All current registrations
 MUST share the same action value. Registrations that have
 a different action than current registrations for the same
 user MUST be rejected with status of 409 (Conflict).
 A proxy server ignores the q parameter when processing non-REGISTER
 requests, while a redirect server simply returns that parameter in
 its Contact response header field.
 Having the proxy server interpret the q parameter is not
 sufficient to guide proxy behavior, as it is not clear, for
 example, how long it is supposed to wait between trying
 addresses.
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Internet Draft SIP January 15, 1999
 If the registration is changed while a user agent or proxy server
 processes an invitation, the new information SHOULD be used.
 This allows a service known as "directed pick-up". In the
 telephone network, directed pickup permits a user at a
 remote station who hears his own phone ringing to pick up
 at that station, dial an access code, and be connected to
 the calling user as if he had answered his own phone.
 A server MAY choose any duration for the registration lifetime.
 Registrations not refreshed after this amount of time SHOULD be
 silently discarded. Responses to a registration SHOULD include an
 Expires header (Section 6.20) or expires Contact parameters (Section
 6.13), indicating the time at which the server will drop the
 registration. If none is present, one hour is assumed. Clients MAY
 request a registration lifetime by indicating the time in an Expires
 header in the request. A server SHOULD NOT use a higher lifetime than
 the one requested, but MAY use a lower one. A single address (if
 host-independent) MAY be registered from several different clients.
 A client cancels an existing registration by sending a REGISTER
 request with an expiration time (Expires) of zero seconds for a
 particular Contact or the wildcard Contact designated by a "*" for
 all registrations. Registrations are matched based on the user, host,
 port and maddr parameters.
 The server SHOULD return the current list of registrations in the 200
 response as Contact header fields.
 It is particularly important that REGISTER requests are authenticated
 since they allow to redirect future requests (see Section 13.2).
 Beyond its use as a simple location service, this method is
 needed if there are several SIP servers on a single host.
 In that case, only one of the servers can use the default
 port number.
 Support of this method is RECOMMENDED.
4.3 Request-URI
 The Request-URI is a SIP URL as described in Section 2 or a general
 URI. It indicates the user or service to which this request is being
 addressed. Unlike the To field, the Request-URI MAY be re-written by
 proxies.
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 When used as a Request-URI, a SIP-URL MUST NOT contain the
 transport-param, maddr-param, ttl-param, or headers elements. A
 server that receives a SIP-URL with these elements removes them
 before further processing.
 Typically, the UAC sets the Request-URI and To to the same
 SIP URL, presumed to remain unchanged over long time
 periods. However, if the UAC has cached a more direct path
 to the callee, e.g., from the Contact header field of a
 response to a previous request, the To would still contain
 the long-term, "public" address, while the Request-URI
 would be set to the cached address.
 Proxy and redirect servers MAY use the information in the Request-URI
 and request header fields to handle the request and possibly rewrite
 the Request-URI. For example, a request addressed to the generic
 address sip:sales@acme.com is proxied to the particular person, e.g.,
 sip:bob@ny.acme.com , with the To field remaining as
 sip:sales@acme.com. At ny.acme.com , Bob then designates Alice as
 the temporary substitute.
 The host part of the Request-URI typically agrees with one of the
 host names of the receiving server. If it does not, the server SHOULD
 proxy the request to the address indicated or return a 404 (Not
 Found) response if it is unwilling or unable to do so. For example,
 the Request-URI and server host name can disagree in the case of a
 firewall proxy that handles outgoing calls. This mode of operation is
 similar to that of HTTP proxies.
 If a SIP server receives a request with a URI indicating a scheme
 other than SIP which that server does not understand, the server MUST
 return a 400 (Bad Request) response. It MUST do this even if the To
 header field contains a scheme it does understand. This is because
 proxies are responsible for processing the Request-URI; the To field
 is of end-to-end significance.
4.3.1 SIP Version
 Both request and response messages include the version of SIP in use,
 and follow [H3.1] (with HTTP replaced by SIP, and HTTP/1.1 replaced
 by SIP/2.0) regarding version ordering, compliance requirements, and
 upgrading of version numbers. To be compliant with this
 specification, applications sending SIP messages MUST include a SIP-
 Version of "SIP/2.0".
4.4 Option Tags
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 Option tags are unique identifiers used to designate new options in
 SIP. These tags are used in Require (Section 6.30) and Unsupported
 (Section 6.38) fields.
 Syntax:
 option-tag = token
 See Section C for a definition of token. The creator of a new SIP
 option MUST either prefix the option with their reverse domain name
 or register the new option with the Internet Assigned Numbers
 Authority (IANA). For example, "com.foo.mynewfeature" is an apt name
 for a feature whose inventor can be reached at "foo.com". Individual
 organizations are then responsible for ensuring that option names
 don't collide. Options registered with IANA have the prefix
 "org.iana.sip.", options described in RFCs have the prefix
 "org.ietf.rfc.N", where N is the RFC number. Option tags are case-
 insensitive.
4.4.1 Registering New Option Tags with IANA
 When registering a new SIP option, the following information MUST be
 provided:
 o Name and description of option. The name MAY be of any length,
 but SHOULD be no more than twenty characters long. The name
 MUST consist of alphanum (See Figure 3) characters only;
 o Indication of who has change control over the option (for
 example, IETF, ISO, ITU-T, other international standardization
 bodies, a consortium or a particular company or group of
 companies);
 o A reference to a further description, if available, for
 example (in order of preference) an RFC, a published paper, a
 patent filing, a technical report, documented source code or a
 computer manual;
 o Contact information (postal and email address);
 Registrations should be sent to iana@iana.org.
 This procedure has been borrowed from RTSP [4] and the RTP
 AVP [26].
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5 Response
 After receiving and interpreting a request message, the recipient
 responds with a SIP response message. The response message format is
 shown below:
 Response = Status-Line ; Section 5.1
 *( general-header
 | response-header
 | entity-header )
 CRLF
 [ message-body ] ; Section 8
 SIP's structure of responses is similar to [H6], but is defined
 explicitly here.
5.1 Status-Line
 The first line of a Response message is the Status-Line, consisting
 of the protocol version (Section 4.3.1) followed by a numeric
 Status-Code and its associated textual phrase, with each element
 separated by SP characters. No CR or LF is allowed except in the
 final CRLF sequence.
 Status-Line = SIP-version SP Status-Code SP Reason-Phrase CRLF
5.1.1 Status Codes and Reason Phrases
 The Status-Code is a 3-digit integer result code that indicates the
 outcome of the attempt to understand and satisfy the request. The
 Reason-Phrase is intended to give a short textual description of the
 Status-Code. The Status-Code is intended for use by automata, whereas
 the Reason-Phrase is intended for the human user. The client is not
 required to examine or display the Reason-Phrase.
 Status-Code = Informational ;Fig. 5
 | Success ;Fig. 5
 | Redirection ;Fig. 6
 | Client-Error ;Fig. 7
 | Server-Error ;Fig. 8
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 | Global-Failure ;Fig. 9
 | extension-code
 extension-code = 3DIGIT
 Reason-Phrase = *<TEXT-UTF8, excluding CR, LF>
 We provide an overview of the Status-Code below, and provide full
 definitions in Section 7. The first digit of the Status-Code defines
 the class of response. The last two digits do not have any
 categorization role. SIP/2.0 allows 6 values for the first digit:
 1xx: Informational -- request received, continuing to process the
 request;
 2xx: Success -- the action was successfully received, understood, and
 accepted;
 3xx: Redirection -- further action needs to be taken in order to
 complete the request;
 4xx: Client Error -- the request contains bad syntax or cannot be
 fulfilled at this server;
 5xx: Server Error -- the server failed to fulfill an apparently valid
 request;
 6xx: Global Failure -- the request cannot be fulfilled at any server.
 Figures 5 through 9 present the individual values of the numeric
 response codes, and an example set of corresponding reason phrases
 for SIP/2.0. These reason phrases are only recommended; they may be
 replaced by local equivalents without affecting the protocol. Note
 that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response
 codes in the range starting at x80 to avoid conflicts with newly
 defined HTTP response codes, and adds a new class, 6xx, of response
 codes.
 SIP response codes are extensible. SIP applications are not required
 to understand the meaning of all registered response codes, though
 such understanding is obviously desirable. However, applications MUST
 understand the class of any response code, as indicated by the first
 digit, and treat any unrecognized response as being equivalent to the
 x00 response code of that class, with the exception that an
 unrecognized response MUST NOT be cached. For example, if a client
 receives an unrecognized response code of 431, it can safely assume
 that there was something wrong with its request and treat the
 response as if it had received a 400 (Bad Request) response code. In
 such cases, user agents SHOULD present to the user the message body
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 returned with the response, since that message body is likely to
 include human-readable information which will explain the unusual
 status.
 Informational = "100" ; Trying
 | "180" ; Ringing
 | "181" ; Call Is Being Forwarded
 | "182" ; Queued
 Success = "200" ; OK
 Figure 5: Informational and success status codes
 Redirection = "300" ; Multiple Choices
 | "301" ; Moved Permanently
 | "302" ; Moved Temporarily
 | "303" ; See Other
 | "305" ; Use Proxy
 | "380" ; Alternative Service
 Figure 6: Redirection status codes
6 Header Field Definitions
 SIP header fields are similar to HTTP header fields in both syntax
 and semantics. In particular, SIP header fields follow the syntax for
 message-header as described in [H4.2]. The rules for extending header
 fields over multiple lines, and use of multiple message-header fields
 with the same field-name, described in [H4.2] also apply to SIP. The
 rules in [H4.2] regarding ordering of header fields apply to SIP,
 with the exception of Via fields, see below, whose order matters.
 Additionally, header fields which are hop-by-hop MUST appear before
 any header fields which are end-to-end. Proxies SHOULD NOT reorder
 header fields. Proxies add Via header fields and MAY add other hop-
 by-hop header fields. They can modify certain header fields, such as
 Max-Forwards 6.23 and "fix up" the Via header fields with "received"
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 Client-Error = "400" ; Bad Request
 | "401" ; Unauthorized
 | "402" ; Payment Required
 | "403" ; Forbidden
 | "404" ; Not Found
 | "405" ; Method Not Allowed
 | "406" ; Not Acceptable
 | "407" ; Proxy Authentication Required
 | "408" ; Request Timeout
 | "409" ; Conflict
 | "410" ; Gone
 | "411" ; Length Required
 | "413" ; Request Entity Too Large
 | "414" ; Request-URI Too Large
 | "415" ; Unsupported Media Type
 | "420" ; Bad Extension
 | "480" ; Temporarily not available
 | "481" ; Call Leg/Transaction Does Not Exist
 | "482" ; Loop Detected
 | "483" ; Too Many Hops
 | "484" ; Address Incomplete
 | "485" ; Ambiguous
 | "486" ; Busy Here
 Figure 7: Client error status codes
 Server-Error = "500" ; Internal Server Error
 | "501" ; Not Implemented
 | "502" ; Bad Gateway
 | "503" ; Service Unavailable
 | "504" ; Gateway Time-out
 | "505" ; SIP Version not supported
 Figure 8: Server error status codes
 parameters as described in Section 6.40.1. Proxies MUST NOT alter any
 fields that are authenticated (see Section 13.2).
 The header fields required, optional and not applicable for each
 method are listed in Table 4 and Table 5. The table uses "o" to
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 Global-Failure | "600" ; Busy Everywhere
 | "603" ; Decline
 | "604" ; Does not exist anywhere
 | "606" ; Not Acceptable
 Figure 9: Global failure status codes
 indicate optional, "m" mandatory and "-" for not applicable. A "*"
 indicates that the header fields are needed only if message body is
 not empty. See sections 6.15, 6.16 and 8 for details.
 The "where" column describes the request and response types with
 which the header field can be used. "R" refers to header fields that
 can be used in requests (that is, request and general header fields).
 "r" designates a response or general-header field as applicable to
 all responses, while a list of numeric values indicates the status
 codes with which the header field can be used. "g" and "e" designate
 general (Section 6.1) and entity header (Section 6.2) fields,
 respectively. If a header field is marked "c", it is copied from the
 request to the response.
 The "enc." column describes whether this message header field MAY be
 encrypted end-to-end. A "n" designates fields that MUST NOT be
 encrypted, while "c" designates fields that SHOULD be encrypted if
 encryption is used.
 The "e-e" column has a value of "e" for end-to-end and a value of "h"
 for hop-by-hop header fields.
 Other header fields can be added as required; a server MUST ignore
 header fields not defined in this specification that it does not
 understand. A proxy MUST NOT remove or modify header fields not
 defined in this specification that it does not understand. A compact
 form of these header fields is also defined in Section 9 for use over
 UDP when the request has to fit into a single packet and size is an
 issue.
 Table 6 in Appendix A lists those header fields that different client
 and server types MUST be able to parse.
6.1 General Header Fields
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 where enc. e-e ACK BYE CAN INV OPT REG
 __________________________________________________________
 Accept R e - - - o o o
 Accept 415 e - - - o o o
 Accept-Encoding R e - - - o o o
 Accept-Encoding 415 e - - - o o o
 Accept-Language R e - o o o o o
 Accept-Language 415 e - o o o o o
 Allow 200 e - - - - m -
 Allow 405 e o o o o o o
 Authorization R e o o o o o o
 Call-ID gc n e m m m m m m
 Contact R e o - - o o o
 Contact 1xx e - - - o o -
 Contact 2xx e - - - o o o
 Contact 3xx e - o - o o o
 Contact 485 e - o - o o o
 Content-Encoding e e o - - o o o
 Content-Length e e o - - o o o
 Content-Type e e * - - * * *
 CSeq gc n e m m m m m m
 Date g e o o o o o o
 Encryption g n e o o o o o o
 Expires g e - - - o - o
 From gc n e m m m m m m
 Hide R n h o o o o o o
 Max-Forwards R n e o o o o o o
 Organization g c h - - - o o o
 Table 4: Summary of header fields, A--O
 General header fields apply to both request and response messages.
 The "general-header" field names can be extended reliably only in
 combination with a change in the protocol version. However, new or
 experimental header fields MAY be given the semantics of general
 header fields if all parties in the communication recognize them to
 be "general-header" fields. Unrecognized header fields are treated as
 "entity-header" fields.
6.2 Entity Header Fields
 The "entity-header" fields define meta-information about the
 message-body or, if no body is present, about the resource identified
 by the request. The term "entity header" is an HTTP 1.1 term where
 the response body can contain a transformed version of the message
 body. The original message body is referred to as the "entity". We
 retain the same terminology for header fields but usually refer to
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 where enc. e-e ACK BYE CAN INV OPT REG
 ___________________________________________________________________
 Proxy-Authenticate 407 n h o o o o o o
 Proxy-Authorization R n h o o o o o o
 Proxy-Require R n h o o o o o o
 Priority R c e - - - o - -
 Require R e o o o o o o
 Retry-After R c e - - - - - o
 Retry-After 404,480,486 c e o o o o o o
 503 c e o o o o o o
 600,603 c e o o o o o o
 Response-Key R c e - o o o o o
 Record-Route R h o o o o o o
 Record-Route 2xx h o o o o o o
 Route R h - o o o o o
 Server r c e o o o o o o
 Subject R c e - - - o - -
 Timestamp g e o o o o o o
 To gc(1) n e m m m m m m
 Unsupported 420 e o o o o o o
 User-Agent g c e o o o o o o
 Via gc(2) n e m m m m m m
 Warning r e o o o o o o
 WWW-Authenticate 401 c e o o o o o o
 Table 5: Summary of header fields, P--Z; (1): copied with possible
 addition of tag; (2): UAS removes first Via header field
 the "message body" rather then the entity as the two are the same in
 SIP.
6.3 Request Header Fields
 The "request-header" fields allow the client to pass additional
 information about the request, and about the client itself, to the
 server. These fields act as request modifiers, with semantics
 equivalent to the parameters of a programming language method
 invocation.
 The "request-header" field names can be extended reliably only in
 combination with a change in the protocol version. However, new or
 experimental header fields MAY be given the semantics of "request-
 header" fields if all parties in the communication recognize them to
 be request-header fields. Unrecognized header fields are treated as
 "entity-header" fields.
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6.4 Response Header Fields
 The "response-header" fields allow the server to pass additional
 information about the response which cannot be placed in the Status-
 Line. These header fields give information about the server and about
 further access to the resource identified by the Request-URI.
 Response-header field names can be extended reliably only in
 combination with a change in the protocol version. However, new or
 experimental header fields MAY be given the semantics of "response-
 header" fields if all parties in the communication recognize them to
 be "response-header" fields. Unrecognized header fields are treated
 as "entity-header" fields.
6.5 End-to-end and Hop-by-hop Headers
 End-to-end headers MUST be transmitted unmodified across all proxies,
 while hop-by-hop headers MAY be modified or added by proxies.
6.6 Header Field Format
 Header fields ("general-header", "request-header", "response-header",
 and "entity-header") follow the same generic header format as that
 given in Section 3.1 of RFC 822 [24]. Each header field consists of a
 name followed by a colon (":") and the field value. Field names are
 case-insensitive. The field value MAY be preceded by any amount of
 leading white space (LWS), though a single space (SP) is preferred.
 Header fields can be extended over multiple lines by preceding each
 extra line with at least one SP or horizontal tab (HT). Applications
 MUST follow HTTP "common form" when generating these constructs,
 since there might exist some implementations that fail to accept
 anything beyond the common forms.
 message-header = field-name ":" [ field-value ] CRLF
 field-name = token
 field-value = *( field-content | LWS )
 field-content = < the OCTETs making up the field-value
 and consisting of either *TEXT-UTF8
 or combinations of token,
 tspecials, and quoted-string>
 The relative order of header fields with different field names is not
 significant. Multiple header fields with the same field-name may be
 present in a message if and only if the entire field-value for that
 header field is defined as a comma-separated list (i.e., #(values)).
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 It MUST be possible to combine the multiple header fields into one
 "field-name: field-value" pair, without changing the semantics of the
 message, by appending each subsequent field-value to the first, each
 separated by a comma. The order in which header fields with the same
 field-name are received is therefore significant to the
 interpretation of the combined field value, and thus a proxy MUST NOT
 change the order of these field values when a message is forwarded.
 Field names are not case-sensitive, although their values may be.
6.7 Accept
 The Accept header follows the syntax defined in [H14.1]. The
 semantics are also identical, with the exception that if no Accept
 header is present, the server SHOULD assume a default value of
 application/sdp.
 This request-header field is used only with the INVITE, OPTIONS and
 REGISTER request methods to indicate what media types are acceptable
 in the response.
 Example:
 Accept: application/sdp;level=1, application/x-private, text/html
6.8 Accept-Encoding
 The Accept-Encoding request-header field is similar to Accept, but
 restricts the content-codings [H3.4.1] that are acceptable in the
 response. See [H14.3]. The syntax of this header is defined in
 [H14.3]. The semantics in SIP are identical to those defined in
 [H14.3].
6.9 Accept-Language
 The Accept-Language header follows the syntax defined in [H14.4]. The
 rules for ordering the languages based on the q parameter apply to
 SIP as well. When used in SIP, the Accept-Language request-header
 field can be used to allow the client to indicate to the server in
 which language it would prefer to receive reason phrases, session
 descriptions or status responses carried as message bodies. A proxy
 MAY use this field to help select the destination for the call, for
 example, a human operator conversant in a language spoken by the
 caller.
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 Example:
 Accept-Language: da, en-gb;q=0.8, en;q=0.7
6.10 Allow
 The Allow entity-header field lists the set of methods supported by
 the resource identified by the Request-URI. The purpose of this field
 is strictly to inform the recipient of valid methods associated with
 the resource. An Allow header field MUST be present in a 405 (Method
 Not Allowed) response and SHOULD be present in an OPTIONS response.
 Allow = "Allow" ":" 1#Method
6.11 Authorization
 A user agent that wishes to authenticate itself with a server --
 usually, but not necessarily, after receiving a 401 response -- MAY
 do so by including an Authorization request-header field with the
 request. The Authorization field value consists of credentials
 containing the authentication information of the user agent for the
 realm of the resource being requested.
 Section 13.2 overviews the use of the Authorization header, and
 section 15 describes the syntax and semantics when used with PGP
 based authentication.
6.12 Call-ID
 The Call-ID general-header field uniquely identifies a particular
 invitation or all registrations of a particular client. Note that a
 single multimedia conference can give rise to several calls with
 different Call-IDs, e.g., if a user invites a single individual
 several times to the same (long-running) conference.
 For an INVITE request, a callee user agent server SHOULD NOT alert
 the user if the user has responded previously to the Call-ID in the
 INVITE request. If the user is already a member of the conference and
 the conference parameters contained in the session description have
 not changed, a callee user agent server MAY silently accept the call,
 regardless of the Call-ID. An invitation for an existing Call-ID or
 session can change the parameters of the conference. A client
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 application MAY decide to simply indicate to the user that the
 conference parameters have been changed and accept the invitation
 automatically or it MAY require user confirmation.
 A user may be invited to the same conference or call using several
 different Call-IDs. If desired, the client MAY use identifiers within
 the session description to detect this duplication. For example, SDP
 contains a session id and version number in the origin (o) field.
 The REGISTER and OPTIONS methods use the Call-ID value to
 unambiguously match requests and responses. All REGISTER requests
 issued by a single client SHOULD use the same Call-ID, at least
 within the same boot cycle.
 Since the Call-ID is generated by and for SIP, there is no
 reason to deal with the complexity of URL-encoding and
 case-ignoring string comparison.
 Call-ID = ( "Call-ID" | "i" ) ":" local-id "@" host
 local-id = 1*uric
 "host" SHOULD be either a fully qualified domain name or a globally
 routable IP address. If this is the case, the "local-id" SHOULD be an
 identifier consisting of URI characters that is unique within "host".
 Use of cryptographically random identifiers [27] is RECOMMENDED. If,
 however, host is not an FQDN or globally routable IP address (such as
 a net 10 address), the local-id MUST be globally unique, as opposed
 to unique within host. These rules guarantee overall global
 uniqueness of the Call-ID. The value for Call-ID MUST NOT be reused
 for a different call. Call-IDs are case-sensitive.
 Using cryptographically random identifiers provides some
 protection against session hijacking. Call-ID, To and From
 are needed to identify a call leg. The distinction between
 call and call leg matters in calls with third-party
 control.
 For systems which have tight bandwidth constraints, many of the
 mandatory SIP headers have a compact form, as discussed in Section 9.
 These are alternate names for the headers which occupy less space in
 the message. In the case of Call-ID, the compact form is i.
 For example, both of the following are valid:
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 Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com
 or
 i:f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com
6.13 Contact
 The Contact general-header field can appear in INVITE, ACK, and
 REGISTER requests, and in 1xx, 2xx, 3xx, and 485 responses. In
 general, it provides a URL where the user can be reached for further
 communications.
 INVITE and ACK requests: INVITE and ACK requests MAY contain Contact
 headers indicating from which location the request is
 originating.
 This allows the callee to send future requests, such as
 BYE, directly to the caller instead of through a series of
 proxies. The Via header is not sufficient since the
 desired address may be that of a proxy.
 INVITE 2xx responses: A user agent server sending a definitive,
 positive response (2xx) MAY insert a Contact response header
 field indicating the SIP address under which it is reachable
 most directly for future SIP requests, such as ACK, within the
 same Call-ID. The Contact header field contains the address of
 the server itself or that of a proxy, e.g., if the host is
 behind a firewall. The value of this Contact header is copied
 into the Request-URI of subsequent requests for this call. If
 the response also contains a Record-Route header field, the
 address in the Contact header field is added as the last item in
 the Route header field. See Section 6.29 for details.
 The Contact value SHOULD NOT be cached across calls, as it
 may not represent the most desirable location for a
 particular destination address.
 INVITE 1xx responses: A UAS sending a provisional response (1xx) MAY
 insert a Contact response header. It has the same semantics in a
 1xx response as a 2xx INVITE response. Note that CANCEL requests
 MUST NOT be sent to that address, but rather follow the same
 path as the original request.
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 REGISTER requests: REGISTER requests MAY contain a Contact header
 field indicating at which locations the user is reachable. The
 REGISTER request defines a wildcard Contact field, "*", which
 MUST only be used with Expires: 0 to remove all registrations
 for a particular user. An optional "expires" parameter indicates
 the desired expiration time of the registration. If a Contact
 entry does not have an "expires" parameter, the Expires header
 field is used as the default value. If neither of these
 mechanisms is used, SIP URIs are assumed to expire after one
 hour. Other URI schemes have no expiration times.
 REGISTER 2xx responses: A REGISTER response MAY return all locations
 at which the user is currently reachable. An optional "expires"
 parameter indicates the expiration time of the registration. If
 a Contact entry does not have an "expires" parameter, the value
 of the Expires header field indicates the expiration time. If
 neither mechanism is used, the expiration time specified in the
 request, explicitly or by default, is used.
 3xx and 485 responses: The Contact response-header field can be used
 with a 3xx or 485 (Ambiguous) response codes to indicate one or
 more alternate addresses to try. It can appear in responses to
 BYE, INVITE and OPTIONS methods. The Contact header field
 contains URIs giving the new locations or user names to try, or
 may simply specify additional transport parameters. A 300
 (Multiple Choices), 301 (Moved Permanently), 302 (Moved
 Temporarily) or 485 (Ambiguous) response SHOULD contain a
 Contact field containing URIs of new addresses to be tried. A
 301 or 302 response may also give the same location and username
 that was being tried but specify additional transport parameters
 such as a different server or multicast address to try or a
 change of SIP transport from UDP to TCP or vice versa. The
 client copies the "user", "password", "host", "port" and "user-
 param" elements of the Contact URI into the Request-URI of the
 redirected request and directs the request to the address
 specified by the "maddr" and "port" parameters, using the
 transport protocol given in the "transport" parameter. If
 "maddr" is a multicast address, the value of "ttl" is used as
 the time-to-live value.
 Note that the Contact header field MAY also refer to a different
 entity than the one originally called. For example, a SIP call
 connected to GSTN gateway may need to deliver a special information
 announcement such as "The number you have dialed has been changed."
 A Contact response header field can contain any suitable URI
 indicating where the called party can be reached, not limited to SIP
 URLs. For example, it could contain URL's for phones, fax, or irc (if
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Internet Draft SIP January 15, 1999
 they were defined) or a mailto: (RFC 2368, [28]) URL.
 The following parameters are defined. Additional parameters may be
 defined in other specifications.
 q: The "qvalue" indicates the relative preference among the locations
 given. "qvalue" values are decimal numbers from 0 to 1, with
 higher values indicating higher preference.
 action: The "action" parameter is used only when registering with the
 REGISTER request. It indicates whether the client wishes that
 the server proxy or redirect future requests intended for the
 client. If this parameter is not specified the action taken
 depends on server configuration. In its response, the registrar
 SHOULD indicate the mode used. This parameter is ignored for
 other requests.
 expires: The "expires" parameter indicates how long the URI is valid.
 The parameter is either a number indicating seconds or a quoted
 string containing a SIP-date. If this parameter is not provided,
 the value of the Expires header field determines how long the
 URI is valid. Implementations MAY treat values larger than
 2**32-1 (4294967295 or 136 years) as equivalent to 2**32-1.
Contact = ( "Contact" | "m" ) ":" ("*" | (1# ( name-addr | addr-spec )
 [ *( ";" contact-params ) ] [ comment ] ))
name-addr = [ display-name ] "<" addr-spec ">"
addr-spec = SIP-URL | URI
display-name = *token | quoted-string
contact-params = "q" "=" qvalue
 | "action" "=" "proxy" | "redirect"
 | "expires" "=" delta-seconds | <"> SIP-date <">
 | extension-attribute
extension-attribute = extension-name [ "=" extension-value ]
 Even if the "display-name" is empty, the "name-addr" form MUST be
 used if the "addr-spec" contains a comma, semicolon or question mark.
 The Contact header field fulfills functionality similar to
 the Location header field in HTTP. However, the HTTP header
 only allows one address, unquoted. Since URIs can contain
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 commas and semicolons as reserved characters, they can be
 mistaken for header or parameter delimiters, respectively.
 The current syntax corresponds to that for the To and From
 header, which also allows the use of display names.
 Example:
 Contact: "Mr. Watson" <sip:watson@worcester.bell-telephone.com>
 ;q=0.7; expires=3600,
 "Mr. Watson" <mailto:watson@bell-telephone.com> ;q=0.1
6.14 Content-Encoding
 Content-Encoding = ( "Content-Encoding" | "e" ) ":"
 1#content-coding
 The Content-Encoding entity-header field is used as a modifier to the
 "media-type". When present, its value indicates what additional
 content codings have been applied to the entity-body, and thus what
 decoding mechanisms MUST be applied in order to obtain the media-type
 referenced by the Content-Type header field. Content-Encoding is
 primarily used to allow a body to be compressed without losing the
 identity of its underlying media type.
 If multiple encodings have been applied to an entity, the content
 codings MUST be listed in the order in which they were applied.
 All content-coding values are case-insensitive. The Internet Assigned
 Numbers Authority (IANA) acts as a registry for content-coding value
 tokens. See [3.5] for a definition of the syntax for content-coding.
 Clients MAY apply content encodings to the body in requests. If the
 server is not capable of decoding the body, or does not recognize any
 of the content-coding values, it MUST send a 415 "Unsupported Media
 Type" response, listing acceptable encodings in the Accept-Encoding
 header. A server MAY apply content encodings to the bodies in
 responses. The server MUST only use encodings listed in the Accept-
 Encoding header in the request.
6.15 Content-Length
 The Content-Length entity-header field indicates the size of the
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 message-body, in decimal number of octets, sent to the recipient.
 Content-Length = ( "Content-Length" | "l" ) ":" 1*DIGIT
 An example is
 Content-Length: 3495
 Applications SHOULD use this field to indicate the size of the
 message-body to be transferred, regardless of the media type of the
 entity. Any Content-Length greater than or equal to zero is a valid
 value. If no body is present in a message, then the Content-Length
 header field MUST be set to zero. If a server receives a UDP request
 without Content-Length, it MUST assume that the request encompasses
 the remainder of the packet. If a server receives a UDP request with
 a Content-Length, but the value is larger than the size of the body
 sent in the request, the client SHOULD generate a 400 class response.
 If there is additional data in the UDP packet after the last byte of
 the body has been read, the server MUST treat the remaining data as a
 separate message. This allows several messages to be placed in a
 single UDP packet.
 If a response does not contain a Content-Length, the client assumes
 that it encompasses the remainder of the UDP packet or the data until
 the TCP connection is closed, as applicable. Section 8 describes how
 to determine the length of the message body.
6.16 Content-Type
 The Content-Type entity-header field indicates the media type of the
 message-body sent to the recipient. The "media-type" element is
 defined in [H3.7].
 Content-Type = ( "Content-Type" | "c" ) ":" media-type
 Examples of this header field are
 Content-Type: application/sdp
 Content-Type: text/html; charset=ISO-8859-4
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6.17 CSeq
 Clients MUST add the CSeq (command sequence) general-header field to
 every request. A CSeq header field in a request contains the request
 method and a single decimal sequence number chosen by the requesting
 client, unique within a single value of Call-ID. The sequence number
 MUST be expressible as a 32-bit unsigned integer. The initial value
 of the sequence number is arbitrary, but MUST be less than 2**31.
 Consecutive requests that differ in request method, headers or body,
 but have the same Call-ID MUST contain strictly monotonically
 increasing and contiguous sequence numbers; sequence numbers do not
 wrap around. Retransmissions of the same request carry the same
 sequence number, but an INVITE with a different message body or
 different header fields (a "re-invitation") acquires a new, higher
 sequence number. A server MUST echo the CSeq value from the request
 in its response. If the Method value is missing in the received CSeq
 header field, the server fills it in appropriately.
 The ACK and CANCEL requests MUST contain the same CSeq value as the
 INVITE request that it refers to, while a BYE request cancelling an
 invitation MUST have a higher sequence number. A BYE request with a
 CSeq that is not higher should cause a 400 response to be generated.
 A user agent server MUST remember the highest sequence number for any
 INVITE request with the same Call-ID value. The server MUST respond
 to, and then discard, any INVITE request with a lower sequence
 number.
 All requests spawned in a parallel search have the same CSeq value as
 the request triggering the parallel search.
 CSeq = "CSeq" ":" 1*DIGIT Method
 Strictly speaking, CSeq header fields are needed for any
 SIP request that can be cancelled by a BYE or CANCEL
 request or where a client can issue several requests for
 the same Call-ID in close succession. Without a sequence
 number, the response to an INVITE could be mistaken for the
 response to the cancellation (BYE or CANCEL). Also, if the
 network duplicates packets or if an ACK is delayed until
 the server has sent an additional response, the client
 could interpret an old response as the response to a re-
 invitation issued shortly thereafter. Using CSeq also makes
 it easy for the server to distinguish different versions of
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 an invitation, without comparing the message body.
 The Method value allows the client to distinguish the response to an
 INVITE request from that of a CANCEL response. CANCEL requests can be
 generated by proxies; if they were to increase the sequence number,
 it might conflict with a later request issued by the user agent for
 the same call.
 With a length of 32 bits, a server could generate, within a single
 call, one request a second for about 136 years before needing to wrap
 around. The initial value of the sequence number is chosen so that
 subsequent requests within the same call will not wrap around. A
 non-zero initial value allows to use a time-based initial sequence
 number, if the client desires. A client could, for example, choose
 the 31 most significant bits of a 32-bit second clock as an initial
 sequence number.
 Forked requests MUST have the same CSeq as there would be ambiguity
 otherwise between these forked requests and later BYE issued by the
 client user agent.
 Example:
 CSeq: 4711 INVITE
6.18 Date
 Date is a general-header field. Its syntax is:
 SIP-date = rfc1123-date
 See [H14.19] for a definition of rfc1123-date. Note that unlike
 HTTP/1.1, SIP only supports the most recent RFC1123 [29] formatting
 for dates.
 The Date header field reflects the time when the request or response
 is first sent. Thus, retransmissions have the same Date header field
 value as the original.
 The Date header field can be used by simple end systems
 without a battery-backed clock to acquire a notion of
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 current time.
6.19 Encryption
 The Encryption general-header field specifies that the content has
 been encrypted. Section 13 describes the overall SIP security
 architecture and algorithms. This header field is intended for end-
 to-end encryption of requests and responses. Requests are encrypted
 based on the public key belonging to the entity named in the To
 header field. Responses are encrypted based on the public key
 conveyed in the Response-Key header field. Note that the public keys
 themselves may not be used for the encryption. This depends on the
 particular algorithms used.
 For any encrypted message, at least the message body and possibly
 other message header fields are encrypted. An application receiving a
 request or response containing an Encryption header field decrypts
 the body and then concatenates the plaintext to the request line and
 headers of the original message. Message headers in the decrypted
 part completely replace those with the same field name in the
 plaintext part. (Note: If only the body of the message is to be
 encrypted, the body has to be prefixed with CRLF to allow proper
 concatenation.) Note that the request method and Request-URI cannot
 be encrypted.
 Encryption only provides privacy; the recipient has no
 guarantee that the request or response came from the party
 listed in the From message header, only that the sender
 used the recipient's public key. However, proxies will not
 be able to modify the request or response.
 Encryption = "Encryption" ":" encryption-scheme 1*SP
 #encryption-params
 encryption-scheme = token
 encryption-params = token "=" ( token | quoted-string )
 The token indicates the form of encryption used; it is
 described in section 13.
 The example in Figure 10 shows a message encrypted with ASCII-armored
 PGP that was generated by applying "pgp -ea" to the payload to be
 encrypted.
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 INVITE sip:watson@boston.bell-telephone.com SIP/2.0
 Via: SIP/2.0/UDP 169.130.12.5
 From: <sip:a.g.bell@bell-telephone.com>
 To: T. A. Watson <sip:watson@bell-telephone.com>
 Call-ID: 187602141351@worcester.bell-telephone.com
 Content-Length: 885
 Encryption: PGP version=2.6.2,encoding=ascii
 hQEMAxkp5GPd+j5xAQf/ZDIfGD/PDOM1wayvwdQAKgGgjmZWe+MTy9NEX8O25Red
 h0/pyrd/+DV5C2BYs7yzSOSXaj1C/tTK/4do6rtjhP8QA3vbDdVdaFciwEVAcuXs
 ODxlNAVqyDi1RqFC28BJIvQ5KfEkPuACKTK7WlRSBc7vNPEA3nyqZGBTwhxRSbIR
 RuFEsHSVojdCam4htcqxGnFwD9sksqs6LIyCFaiTAhWtwcCaN437G7mUYzy2KLcA
 zPVGq1VQg83b99zPzIxRdlZ+K7+bAnu8Rtu+ohOCMLV3TPXbyp+err1YiThCZHIu
 X9dOVj3CMjCP66RSHa/ea0wYTRRNYA/G+kdP8DSUcqYAAAE/hZPX6nFIqk7AVnf6
 IpWHUPTelNUJpzUp5Ou+q/5P7ZAsn+cSAuF2YWtVjCf+SQmBR13p2EYYWHoxlA2/
 GgKADYe4M3JSwOtqwU8zUJF3FIfk7vsxmSqtUQrRQaiIhqNyG7KxJt4YjWnEjF5E
 WUIPhvyGFMJaeQXIyGRYZAYvKKklyAJcm29zLACxU5alX4M25lHQd9FR9Zmq6Jed
 wbWvia6cAIfsvlZ9JGocmQYF7pcuz5pnczqP+/yvRqFJtDGD/v3s++G2R+ViVYJO
 z/lxGUZaM4IWBCf+4DUjNanZM0oxAE28NjaIZ0rrldDQmO8V9FtPKdHxkqA5iJP+
 6vGOFti1Ak4kmEz0vM/Nsv7kkubTFhRl05OiJIGr9S1UhenlZv9l6RuXsOY/EwH2
 z8X9N4MhMyXEVuC9rt8/AUhmVQ==
 =bOW+
 Figure 10: PGP Encryption Example
 Since proxies can base their forwarding decision on any combination
 of SIP header fields, there is no guarantee that an encrypted request
 "hiding" header fields will reach the same destination as an
 otherwise identical un-encrypted request.
6.20 Expires
 The Expires entity-header field gives the date and time after which
 the message content expires.
 This header field is currently defined only for the REGISTER and
 INVITE methods. For REGISTER, it is a request and response-header
 field. In a REGISTER request, the client indicates how long it wishes
 the registration to be valid. In the response, the server indicates
 the earliest expiration time of all registrations. The server MAY
 choose a shorter time interval than that requested by the client, but
 SHOULD NOT choose a longer one.
 For INVITE requests, it is a request and response-header field. In a
 request, the caller can limit the validity of an invitation, for
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 example, if a client wants to limit the time duration of a search or
 a conference invitation. A user interface MAY take this as a hint to
 leave the invitation window on the screen even if the user is not
 currently at the workstation. This also limits the duration of a
 search. If the request expires before the search completes, the proxy
 returns a 408 (Request Timeout) status. In a 302 (Moved Temporarily)
 response, a server can advise the client of the maximal duration of
 the redirection.
 The value of this field can be either a SIP-date or an integer number
 of seconds (in decimal), measured from the receipt of the request.
 The latter approach is preferable for short durations, as it does not
 depend on clients and servers sharing a synchronized clock.
 Implementations MAY treat values larger than 2**32-1 (4294967295 or
 136 years) as equivalent to 2**32-1.
 Expires = "Expires" ":" ( SIP-date | delta-seconds )
 Two examples of its use are
 Expires: 1994年12月01日 16:00:00 GMT
 Expires: 5
6.21 From
 Requests and responses MUST contain a From general-header field,
 indicating the initiator of the request. The From field MAY contain
 the "tag" parameter. The server copies the From header field from the
 request to the response. The optional "display-name" is meant to be
 rendered by a human-user interface. A system SHOULD use the display
 name "Anonymous" if the identity of the client is to remain hidden.
 The SIP-URL MUST NOT contain the "transport-param", "maddr-param",
 "ttl-param", or "headers" elements. A server that receives a SIP-URL
 with these elements removes them before further processing.
 Even if the "display-name" is empty, the "name-addr" form MUST be
 used if the "addr-spec" contains a comma, question mark, or
 semicolon.
 From = ( "From" | "f" ) ":" ( name-addr | addr-spec )
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 *( ";" addr-params )
 addr-params = tag-param
 tag-param = "tag=" UUID
 UUID = 1*( hex | "-" )
 Examples:
 From: "A. G. Bell" <sip:agb@bell-telephone.com>
 From: sip:+12125551212@server.phone2net.com
 From: Anonymous <sip:c8oqz84zk7z@privacy.org>
 The "tag" MAY appear in the From field of a request. It MUST be
 present when it is possible that two instances of a user sharing a
 SIP address can make call invitations with the same Call-ID.
 The "tag" value MUST be globally unique and cryptographically random
 with at least 32 bits of randomness. A single user maintains the same
 tag throughout the call identified by the Call-ID.
 Call-ID, To and From are needed to identify a call leg.
 The distinction between call and call leg matters in calls
 with multiple responses to a forked request. The format is
 similar to the equivalent RFC 822 [24] header, but with a
 URI instead of just an email address.
6.22 Hide
 A client uses the Hide request header field to indicate that it wants
 the path comprised of the Via header fields (Section 6.40) to be
 hidden from subsequent proxies and user agents. It can take two
 forms: Hide: route and Hide: hop. Hide header fields are typically
 added by the client user agent, but MAY be added by any proxy along
 the path.
 If a request contains the "Hide: route" header field, all following
 proxies SHOULD hide their previous hop. If a request contains the
 "Hide: hop" header field, only the next proxy SHOULD hide the
 previous hop and then remove the Hide option unless it also wants to
 remain anonymous.
 A server hides the previous hop by encrypting the "host" and "port"
 parts of the top-most Via header field with an algorithm of its
 choice. Servers SHOULD add additional "salt" to the "host" and "port"
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 information prior to encryption to prevent malicious downstream
 proxies from guessing earlier parts of the path based on seeing
 identical encrypted Via headers. Hidden Via fields are marked with
 the "hidden" Via option, as described in Section 6.40.
 A server that is capable of hiding Via headers MUST attempt to
 decrypt all Via headers marked as "hidden" to perform loop detection.
 Servers that are not capable of hiding can ignore hidden Via fields
 in their loop detection algorithm.
 If hidden headers were not marked, a proxy would have to
 decrypt all headers to detect loops, just in case one was
 encrypted, as the Hide: Hop option may have been removed
 along the way.
 A host MUST NOT add such a "Hide: hop" header field unless it can
 guarantee it will only send a request for this destination to the
 same next hop. The reason for this is that it is possible that the
 request will loop back through this same hop from a downstream proxy.
 The loop will be detected by the next hop if the choice of next hop
 is fixed, but could loop an arbitrary number of times otherwise.
 A client requesting "Hide: route" can only rely on keeping the
 request path private if it sends the request to a trusted proxy.
 Hiding the route of a SIP request is of limited value if the request
 results in data packets being exchanged directly between the calling
 and called user agent.
 The use of Hide header fields is discouraged unless path privacy is
 truly needed; Hide fields impose extra processing costs and
 restrictions for proxies and can cause requests to generate 482 (Loop
 Detected) responses that could otherwise be avoided.
 The encryption of Via header fields is described in more detail in
 Section 13.
 The Hide header field has the following syntax:
 Hide = "Hide" ":" ( "route" | "hop" )
6.23 Max-Forwards
 The Max-Forwards request-header field may be used with any SIP method
 to limit the number of proxies or gateways that can forward the
 request to the next downstream server. This can also be useful when
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 the client is attempting to trace a request chain which appears to be
 failing or looping in mid-chain.
 Max-Forwards = "Max-Forwards" ":" 1*DIGIT
 The Max-Forwards value is a decimal integer indicating the remaining
 number of times this request message is allowed to be forwarded.
 Each proxy or gateway recipient of a request containing a Max-
 Forwards header field MUST check and update its value prior to
 forwarding the request. If the received value is zero (0), the
 recipient MUST NOT forward the request. Instead, for the OPTIONS and
 REGISTER methods, it MUST respond as the final recipient. For all
 other methods, the server returns 483 (Too many hops).
 If the received Max-Forwards value is greater than zero, then the
 forwarded message MUST contain an updated Max-Forwards field with a
 value decremented by one (1).
 Example:
 Max-Forwards: 6
6.24 Organization
 The Organization general-header field conveys the name of the
 organization to which the entity issuing the request or response
 belongs. It MAY also be inserted by proxies at the boundary of an
 organization.
 The field MAY be used by client software to filter calls.
 Organization = "Organization" ":" *TEXT-UTF8
6.25 Priority
 The Priority request-header field indicates the urgency of the
 request as perceived by the client.
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 Priority = "Priority" ":" priority-value
 priority-value = "emergency" | "urgent" | "normal"
 | "non-urgent"
 It is RECOMMENDED that the value of "emergency" only be used when
 life, limb or property are in imminent danger.
 Examples:
 Subject: A tornado is heading our way!
 Priority: emergency
 Subject: Weekend plans
 Priority: non-urgent
 These are the values of RFC 2076 [30], with the addition of
 "emergency".
6.26 Proxy-Authenticate
 The Proxy-Authenticate response-header field MUST be included as part
 of a 407 (Proxy Authentication Required) response. The field value
 consists of a challenge that indicates the authentication scheme and
 parameters applicable to the proxy for this Request-URI.
 Unlike its usage within HTTP, the Proxy-Authenticate header MUST be
 passed upstream in the response to tha UAC. In SIP, only UAC's can
 authenticate themselves to proxies.
 The syntax for this header is defined in [H14.33]. See 14 for further
 details on its usage.
 A client SHOULD cache the credentials used for a particular proxy
 server and realm for the next request to that server. Credentials
 are, in general, valid for a specific value of the Request-URI at a
 particular proxy server. If a client contacts a proxy server that has
 required authentication in the past, but the client does not have
 credentials for the particular Request-URI, it MAY attempt to use the
 most-recently used credential. The server responds with 401
 (Unauthorized) if the client guessed wrong.
 This suggested caching behavior is motivated by proxies
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 restricting phone calls to authenticated users. It seems
 likely that in most cases, all destinations require the
 same password. Note that end-to-end authentication is
 likely to be destination-specific.
6.27 Proxy-Authorization
 The Proxy-Authorization request-header field allows the client to
 identify itself (or its user) to a proxy which requires
 authentication. The Proxy-Authorization field value consists of
 credentials containing the authentication information of the user
 agent for the proxy and/or realm of the resource being requested.
 Unlike Authorization, the Proxy-Authorization header field applies
 only to the next outbound proxy that demanded authentication using
 the Proxy- Authenticate field. When multiple proxies are used in a
 chain, the Proxy-Authorization header field is consumed by the first
 outbound proxy that was expecting to receive credentials. A proxy MAY
 relay the credentials from the client request to the next proxy if
 that is the mechanism by which the proxies cooperatively authenticate
 a given request.
 See [H14.34] for a definition of the syntax, and section 14 for a
 discussion of its usage.
6.28 Proxy-Require
 The Proxy-Require header field is used to indicate proxy-sensitive
 features that MUST be supported by the proxy. Any Proxy-Require
 header field features that are not supported by the proxy MUST be
 negatively acknowledged by the proxy to the client if not supported.
 Servers treat this field identically to the Require field.
 See Section 6.30 for more details on the mechanics of this message
 and a usage example.
6.29 Record-Route
 The Record-Route request and response header field is added to a
 request by any proxy that insists on being in the path of subsequent
 requests for the same call leg. It contains a globally reachable
 Request-URI that identifies the proxy server. Each proxy server adds
 its Request-URI to the beginning of the list.
 The server copies the Record-Route header field unchanged into the
 response. (Record-Route is only relevant for 2xx responses.)
 The calling user agent client copies the Record-Route header into a
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 Route header field of subsequent requests within the same call leg,
 reversing the order of requests, so that the first entry is closest
 to the user agent client. If the response contained a Contact header
 field, the calling user agent adds its content as the last Route
 header. Unless this would cause a loop, any client MUST send any
 subsequent requests for this call leg to the first Request-URI in the
 Route request header field and remove that entry.
 The calling user agent MUST NOT use the Record-Route header field in
 requests that contain Route header fields.
 Some proxies, such as those controlling firewalls or in an
 automatic call distribution (ACD) system, need to maintain
 call state and thus need to receive any BYE and ACK packets
 for the call.
 The Record-Route header field has the following syntax:
 Record-Route = "Record-Route" ":" 1# name-addr
 Proxy servers SHOULD use the "maddr" URL parameter containing their
 address to ensure that subsequent requests are guaranteed to reach
 exactly the same server.
 Example for a request that has traversed the hosts ieee.org and
 bell-telephone.com , in that order:
 Record-Route: <sip:a.g.bell@bell-telephone.com>,
 <sip:a.bell@ieee.org>
6.30 Require
 The Require request-header field is used by clients to tell user
 agent servers about options that the client expects the server to
 support in order to properly process the request. If a server does
 not understand the option, it MUST respond by returning status code
 420 (Bad Extension) and list those options it does not understand in
 the Unsupported header.
 Require = "Require" ":" 1#option-tag
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 Example:
 C->S: INVITE sip:watson@bell-telephone.com SIP/2.0
 Require: com.example.billing
 Payment: sheep_skins, conch_shells
 S->C: SIP/2.0 420 Bad Extension
 Unsupported: com.example.billing
 This is to make sure that the client-server interaction
 will proceed without delay when all options are understood
 by both sides, and only slow down if options are not
 understood (as in the example above). For a well-matched
 client-server pair, the interaction proceeds quickly,
 saving a round-trip often required by negotiation
 mechanisms. In addition, it also removes ambiguity when the
 client requires features that the server does not
 understand. Some features, such as call handling fields,
 are only of interest to end systems.
 Proxy and redirect servers MUST ignore features that are not
 understood. If a particular extension requires that intermediate
 devices support it, the extension MUST be tagged in the Proxy-Require
 field as well (see Section 6.28).
6.31 Response-Key
 The Response-Key request-header field can be used by a client to
 request the key that the called user agent SHOULD use to encrypt the
 response with. The syntax is:
 Response-Key = "Response-Key" ":" key-scheme 1*SP #key-param
 key-scheme = token
 key-param = token "=" ( token | quoted-string )
 The "key-scheme" gives the type of encryption to be used for the
 response. Section 13 describes security schemes.
 If the client insists that the server return an encrypted response,
 it includes a
 Require: org.ietf.sip.encrypt-response
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 header field in its request. If the server cannot encrypt for
 whatever reason, it MUST follow normal Require header field
 procedures and return a 420 (Bad Extension) response. If this Require
 header field is not present, a server SHOULD still encrypt if it can.
6.32 Retry-After
 The Retry-After general-header field can be used with a 503 (Service
 Unavailable) response to indicate how long the service is expected to
 be unavailable to the requesting client and with a 404 (Not Found),
 600 (Busy), or 603 (Decline) response to indicate when the called
 party anticipates being available again. The value of this field can
 be either an SIP-date or an integer number of seconds (in decimal)
 after the time of the response.
 A REGISTER request MAY include this header field when deleting
 registrations with "Contact: * ;expires: 0". The Retry-After value
 then indicates when the user might again be reachable. The registrar
 MAY then include this information in responses to future calls.
 An optional comment can be used to indicate additional information
 about the time of callback. An optional "duration" parameter
 indicates how long the called party will be reachable starting at the
 initial time of availability. If no duration parameter is given, the
 service is assumed to be available indefinitely.
 Retry-After = "Retry-After" ":" ( SIP-date | delta-seconds )
 [ comment ] [ ";" "duration" "=" delta-seconds ]
 Examples of its use are
 Retry-After: 1997年7月21日 18:48:34 GMT (I'm in a meeting)
 Retry-After: 9999年1月01日 00:00:00 GMT
 (Dear John: Don't call me back, ever)
 Retry-After: 1997年9月26日 21:00:00 GMT;duration=3600
 Retry-After: 120
 In the third example, the callee is reachable for one hour starting
 at 21:00 GMT. In the last example, the delay is 2 minutes.
6.33 Route
 The Route request-header field determines the route taken by a
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 request. Each host removes the first entry and then proxies the
 request to the host listed in that entry, also using it as the
 Request-URI. The operation is further described in Section 6.29.
 The Route header field has the following syntax:
 Route = "Route" ":" 1# name-addr
6.34 Server
 The Server response-header field contains information about the
 software used by the user agent server to handle the request. The
 syntax for this field is defined in [H14.39].
6.35 Subject
 This is intended to provide a summary, or to indicate the nature, of
 the call, allowing call filtering without having to parse the session
 description. (Also, the session description does not have to use the
 same subject indication as the invitation.)
 Subject = ( "Subject" | "s" ) ":" *TEXT-UTF8
 Example:
 Subject: Tune in - they are talking about your work!
6.36 Timestamp
 The timestamp general-header field describes when the client sent the
 request to the server. The value of the timestamp is of significance
 only to the client and MAY use any timescale. The server MUST echo
 the exact same value and MAY, if it has accurate information about
 this, add a floating point number indicating the number of seconds
 that have elapsed since it has received the request. The timestamp is
 used by the client to compute the round-trip time to the server so
 that it can adjust the timeout value for retransmissions.
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 Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
 delay = *(DIGIT) [ "." *(DIGIT) ]
 Note that there MUST NOT be any LWS between a DIGIT and the decimal
 point.
6.37 To
 The To general-header field specifies recipient of the request, with
 the same SIP URL syntax as the From field.
 To = ( "To" | "t" ) ":" ( name-addr | addr-spec )
 *( ";" addr-params )
 Requests and responses MUST contain a To general-header field,
 indicating the desired recipient of the request. The optional
 "display-name" is meant to be rendered by a human-user interface.
 The UAS or redirect server copies the To header field into its
 response, and MUST add a "tag" parameter if the request contained
 more than one Via header field.
 If there was more than one Via header field, the request
 was handled by at least one proxy server. Since the
 receiver cannot know whether any of the proxy servers
 forked the request, it is safest to assume that they might
 have.
 The SIP-URL MUST NOT contain the "transport-param", "maddr-param",
 "ttl-param", or "headers" elements. A server that receives a SIP-URL
 with these elements removes them before further processing.
 The "tag" parameter serves as a general mechanism to distinguish
 multiple instances of a user identified by a single SIP URL. As
 proxies can fork requests, the same request can reach multiple
 instances of a user (mobile and home phones, for example). As each
 can respond, there needs to be a means to distinguish the responses
 from each at the caller. The situation also arises with multicast
 requests. The tag in the To header field serves to distinguish
 responses at the UAC. It MUST be placed in the To field of the
 response by each instance when there is a possibility that the
 request was forked at an intermediate proxy. The "tag" MUST be added
 by UAS, registrars and redirect servers, but MUST NOT be inserted
 into responses forwarded upstream by proxies. The "tag" is added for
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 all definitive responses for all methods, and MAY be added for
 informational responses from a UAS or redirect server. All subsequent
 transactions between two entities MUST include the "tag" parameter,
 as described in Section 11.
 See Section 6.21 for details of the "tag" parameter.
 The "tag" parameter in To headers is ignored when matching responses
 to requests that did not contain a "tag" in their To header.
 A SIP server returns a 400 (Bad Request) response if it receives a
 request with a To header field containing a URI with a scheme it does
 not recognize.
 Even if the "display-name" is empty, the "name-addr" form MUST be
 used if the "addr-spec" contains a comma, question mark, or
 semicolon.
 The following are examples of valid To headers:
 To: The Operator <sip:operator@cs.columbia.edu>;tag=287447
 To: sip:+12125551212@server.phone2net.com
 Call-ID, To and From are needed to identify a call leg.
 The distinction between call and call leg matters in calls
 with multiple responses from a forked request. The "tag" is
 added to the To header field in the response to allow
 forking of future requests for the same call by proxies,
 while addressing only one of the possibly several
 responding user agent servers. It also allows several
 instances of the callee to send requests that can be
 distinguished.
6.38 Unsupported
 The Unsupported response-header field lists the features not
 supported by the server. See Section 6.30 for a usage example and
 motivation.
 Syntax:
 Unsupported = "Unsupported" ":" 1#option-tag
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6.39 User-Agent
 The User-Agent general-header field contains information about the
 client user agent originating the request. The syntax and semantics
 are defined in [H14.42].
6.40 Via
 The Via field indicates the path taken by the request so far. This
 prevents request looping and ensures replies take the same path as
 the requests, which assists in firewall traversal and other unusual
 routing situations.
6.40.1 Requests
 The client originating the request MUST insert into the request a Via
 field containing its host name or network address and, if not the
 default port number, the port number at which it wishes to receive
 responses. (Note that this port number can differ from the UDP source
 port number of the request.) A fully-qualified domain name is
 RECOMMENDED. Each subsequent proxy server that sends the request
 onwards MUST add its own additional Via field before any existing Via
 fields. A proxy that receives a redirection (3xx) response and then
 searches recursively, MUST use the same Via headers as on the
 original proxied request.
 A proxy SHOULD check the top-most Via header field to ensure that it
 contains the sender's correct network address, as seen from that
 proxy. If the sender's address is incorrect, the proxy MUST add an
 additional "received" attribute, as described 6.40.2.
 A host behind a network address translator (NAT) or
 firewall may not be able to insert a network address into
 the Via header that can be reached by the next hop beyond
 the NAT. Use of the received attribute allows SIP requests
 to traverse NAT's which only modify the source IP address.
 NAT's which modify port numbers, called Network Address
 Port Translator's (NAPT) will not properly pass SIP when
 transported on UDP, in which case an application layer
 gateway is required. When run over TCP, SIP stands a
 better chance of traversing NAT's, since its behavior is
 similar to HTTP in this case (but of course on different
 ports).
 A proxy sending a request to a multicast address MUST add the "maddr"
 parameter to its Via header field, and SHOULD add the "ttl"
 parameter. If a server receives a request which contained an "maddr"
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 parameter in the topmost Via field, it SHOULD send the response to
 the multicast address listed in the "maddr" parameter.
 If a proxy server receives a request which contains its own address
 in the Via header value, it MUST respond with a 482 (Loop Detected)
 status code.
 A proxy server MUST NOT forward a request to a multicast group which
 already appears in any of the Via headers.
 This prevents a malfunctioning proxy server from causing
 loops. Also, it cannot be guaranteed that a proxy server
 can always detect that the address returned by a location
 service refers to a host listed in the Via list, as a
 single host may have aliases or several network interfaces.
6.40.2 Receiver-tagged Via Header Fields
 Normally, every host that sends or forwards a SIP message adds a Via
 field indicating the path traversed. However, it is possible that
 Network Address Translators (NAT) changes the source address and port
 of the request (e.g., from net-10 to a globally routable address), in
 which case the Via header field cannot be relied on to route replies.
 To prevent this, a proxy SHOULD check the top-most Via header field
 to ensure that it contains the sender's correct network address, as
 seen from that proxy. If the sender's address is incorrect, the proxy
 MUST add a "received" parameter to the Via header field inserted by
 the previous hop. Such a modified Via header field is known as a
 receiver-tagged Via header field. An example is:
 Via: SIP/2.0/UDP erlang.bell-telephone.com:5060
 Via: SIP/2.0/UDP 10.0.0.1:5060 ;received=199.172.136.3
 In this example, the message originated from 10.0.0.1 and traversed a
 NAT with the external address border.ieee.org (199.172.136.3) to
 reach erlang.bell-telephone.com. The latter noticed the mismatch,
 and added a parameter to the previous hop's Via header field,
 containing the address that the packet actually came from. (Note that
 the NAT border.ieee.org is not a SIP server.)
6.40.3 Responses
 Via header fields in responses are processed by a proxy or UAC
 according to the following rules:
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 1. The first Via header field should indicate the proxy or
 client processing this response. If it does not, discard
 the message. Otherwise, remove this Via field.
 2. If there is no second Via header field, this response is
 destined for this client. Otherwise, the processing depends
 on whether the Via field contains a "maddr" parameter or is
 a receiver-tagged field:
 - If the second Via header field contains a "maddr"
 parameter, send the response to the multicast address
 listed there, using the port indicated in "sent-by", or
 port 5060 if none is present. The response SHOULD be sent
 using the TTL indicated in the "ttl" parameter, or with a
 TTL of 1 if that parameter is not present. For
 robustness, responses MUST be sent to the address
 indicated in the "maddr" parameter even if it is not a
 multicast address.
 - If the second Via header field does not contain a "maddr"
 parameter and is a receiver-tagged field (Section
 6.40.2), send the message to the address in the
 "received" parameter, using the port indicated in the
 "sent-by" value, or using port 5060 if none is present.
 - If neither of the previous cases apply, send the message
 to the address indicated by the "sent-by" value in the
 second Via header field.
6.40.4 User Agent and Redirect Servers
 A UAS or redirect server sends a response based on one of the
 following rules:
 o If the first Via header field in the request contains a
 "maddr" parameter, send the response to the multicast address
 listed there, using the port indicated in "sent-by", or port
 5060 if none is present. The response SHOULD be sent using the
 TTL indicated in the "ttl" parameter, or with a TTL of 1 if
 that parameter is not present. For robustness, responses MUST
 be sent to the address indicated in the "maddr" parameter even
 if it is not a multicast address.
 o If the address in the "sent-by" value of the first Via field
 differs from the source address of the packet, send the
 response to the actual packet source address, similar to the
 treatment for receiver-tagged Via header fields (Section
 6.40.2).
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 o If neither of these conditions is true, send the response to
 the address contained in the "sent-by" value. If the request
 was sent using TCP, use the existing TCP connection if
 available.
6.40.5 Syntax
 The format for a Via header field is shown in Fig. 11. The defaults
 for "protocol-name" and "transport" are "SIP" and "UDP",
 respectively. The "maddr" parameter, designating the multicast
 address, and the "ttl" parameter, designating the time-to-live (TTL)
 value, are included only if the request was sent via multicast. The
 "received" parameter is added only for receiver-added Via fields
 (Section 6.40.2). For reasons of privacy, a client or proxy may wish
 to hide its Via information by encrypting it (see Section 6.22). The
 "hidden" parameter is included if this header field was hidden by the
 upstream proxy (see 6.22). Note that privacy of the proxy relies on
 the cooperation of the next hop, as the next-hop proxy will, by
 necessity, know the IP address and port number of the source host.
 Via = ( "Via" | "v") ":" 1#( sent-protocol sent-by
 *( ";" via-params ) [ comment ] )
 via-params = via-hidden | via-ttl | via-maddr 
 | via-received | via-branch
 via-hidden = "hidden"
 via-ttl = "ttl" "=" ttl
 via-maddr = "maddr" "=" maddr
 via-received = "received" "=" host
 via-branch = "branch" "=" token
 sent-protocol = protocol-name "/" protocol-version "/" transport
 protocol-name = "SIP" | token
 protocol-version = token
 transport = "UDP" | "TCP" | token
 sent-by = ( host [ ":" port ] ) | ( concealed-host )
 concealed-host = token
 ttl = 1*3DIGIT ; 0 to 255
 Figure 11: Syntax of Via header field
 The "branch" parameter is included by every forking proxy. The token
 MUST be unique for each distinct request generated when a proxy
 forks. CANCEL requests MUST have the same branch value as the
 corresponding forked request. When a response arrives at the proxy
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 it can use the branch value to figure out which branch the response
 corresponds to. A proxy which generates a single request (non-
 forking) MAY also insert the "branch" parameter. The identifier has
 to be unique only within a set of isomorphic requests.
 Via: SIP/2.0/UDP first.example.com:4000;ttl=16
 ;maddr=224.2.0.1 ;branch=a7c6a8dlze (Example)
 Via: SIP/2.0/UDP adk8
6.41 Warning
 The Warning response-header field is used to carry additional
 information about the status of a response. Warning headers are sent
 with responses and have the following format:
 Warning = "Warning" ":" 1#warning-value
 warning-value = warn-code SP warn-agent SP warn-text
 warn-code = 3DIGIT
 warn-agent = ( host [ ":" port ] ) | pseudonym
 ; the name or pseudonym of the server adding
 ; the Warning header, for use in debugging
 warn-text = quoted-string
 A response MAY carry more than one Warning header.
 The "warn-text" should be in a natural language that is most likely
 to be intelligible to the human user receiving the response. This
 decision can be based on any available knowledge, such as the
 location of the cache or user, the Accept-Language field in a
 request, or the Content-Language field in a response. The default
 language is i-default [31].
 Any server MAY add Warning headers to a response. Proxy servers MUST
 place additional Warning headers before any Authorization headers.
 Within that constraint, Warning headers MUST be added after any
 existing Warning headers not covered by a signature. A proxy server
 MUST NOT delete any Warning header field that it received with a
 response.
 When multiple Warning headers are attached to a response, the user
 agent SHOULD display as many of them as possible, in the order that
 they appear in the response. If it is not possible to display all of
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 the warnings, the user agent first displays warnings that appear
 early in the response.
 The warn-code consists of three digits. A first digit of "3"
 indicates warnings specific to SIP.
 This is a list of the currently-defined "warn-code"s, each with a
 recommended warn-text in English, and a description of its meaning.
 Note that these warnings describe failures induced by the session
 description.
 Warnings 300 through 329 are reserved for indicating problems with
 keywords in the session description, 330 through 339 are warnings
 related to basic network services requested in the session
 description, 370 through 379 are warnings related to quantitative QoS
 parameters requested in the session description, and 390 through 399
 are miscellaneous warnings that do not fall into one of the above
 categories.
 300 Incompatible network protocol: One or more network protocols
 contained in the session description are not available.
 301 Incompatible network address formats: One or more network address
 formats contained in the session description are not available.
 302 Incompatible transport protocol: One or more transport protocols
 described in the session description are not available.
 303 Incompatible bandwidth units: One or more bandwidth measurement
 units contained in the session description were not understood.
 304 Media type not available: One or more media types contained in
 the session description are not available.
 305 Incompatible media format: One or more media formats contained in
 the session description available.
 306 Attribute not understood: One or more of the media attributes in
 the session description are not supported.
 307 Session description parameter not understood: A parameter other
 than those listed above was not understood.
 330 Multicast not available: The site where the user is located does
 not support multicast.
 331 Unicast not available: The site where the user is located does
 not support unicast communication (usually due to the presence
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 of a firewall).
 370 Insufficient bandwidth: The bandwidth specified in the session
 description or defined by the media exceeds that known to be
 available.
 399 Miscellaneous warning: The warning text can include arbitrary
 information to be presented to a human user, or logged. A system
 receiving this warning MUST NOT take any automated action.
 1xx and 2xx have been taken by HTTP/1.1.
 Additional "warn-code"s, as in the example below, can be defined
 through IANA.
 Examples:
 Warning: 307 isi.edu "Session parameter 'foo' not understood"
 Warning: 301 isi.edu "Incompatible network address type 'E.164'"
6.42 WWW-Authenticate
 The WWW-Authenticate response-header field MUST be included in 401
 (Unauthorized) response messages. The field value consists of at
 least one challenge that indicates the authentication scheme(s) and
 parameters applicable to the Request-URI. See [H14.46] for a
 definition of the syntax, and section 14 for an overview of usage.
 The content of the "realm" parameter SHOULD be displayed to the user.
 A user agent SHOULD cache the authorization credentials for a given
 value of the destination (To header) and "realm" and attempt to re-
 use these values on the next request for that destination.
 In addition to the "basic" and "digest" authentication schemes
 defined in the specifications cited above, SIP defines a new scheme,
 PGP (RFC 2015, [32]), Section 15. Other schemes, such as S/MIME, are
 for further study.
7 Status Code Definitions
 The response codes are consistent with, and extend, HTTP/1.1 response
 codes. Not all HTTP/1.1 response codes are appropriate, and only
 those that are appropriate are given here. Other HTTP/1.1 response
 codes SHOULD NOT be used. Response codes not defined by HTTP/1.1 have
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 codes x80 upwards to avoid clashes with future HTTP response codes.
 Also, SIP defines a new class, 6xx. The default behavior for unknown
 response codes is given for each category of codes.
7.1 Informational 1xx
 Informational responses indicate that the server or proxy contacted
 is performing some further action and does not yet have a definitive
 response. The client SHOULD wait for a further response from the
 server, and the server SHOULD send such a response without further
 prompting. A server SHOULD send a 1xx response if it expects to take
 more than 200 ms to obtain a final response. A server MAY issue zero
 or more 1xx responses, with no restriction on their ordering or
 uniqueness. Note that 1xx responses are not transmitted reliably,
 that is, they do not cause the client to send an ACK. Servers are
 free to retransmit informational responses and clients can inquire
 about the current state of call processing by re-sending the request.
7.1.1 100 Trying
 Some unspecified action is being taken on behalf of this call (e.g.,
 a database is being consulted), but the user has not yet been
 located.
7.1.2 180 Ringing
 The called user agent has located a possible location where the user
 has registered recently and is trying to alert the user.
7.1.3 181 Call Is Being Forwarded
 A proxy server MAY use this status code to indicate that the call is
 being forwarded to a different set of destinations.
7.1.4 182 Queued
 The called party is temporarily unavailable, but the callee has
 decided to queue the call rather than reject it. When the callee
 becomes available, it will return the appropriate final status
 response. The reason phrase MAY give further details about the status
 of the call, e.g., "5 calls queued; expected waiting time is 15
 minutes". The server MAY issue several 182 responses to update the
 caller about the status of the queued call.
7.2 Successful 2xx
 The request was successful and MUST terminate a search.
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7.2.1 200 OK
 The request has succeeded. The information returned with the response
 depends on the method used in the request, for example:
 BYE: The call has been terminated. The message body is empty.
 CANCEL: The search has been cancelled. The message body is empty.
 INVITE: The callee has agreed to participate; the message body
 indicates the callee's capabilities.
 OPTIONS: The callee has agreed to share its capabilities, included in
 the message body.
 REGISTER: The registration has succeeded. The client treats the
 message body according to its Content-Type.
7.3 Redirection 3xx
 3xx responses give information about the user's new location, or
 about alternative services that might be able to satisfy the call.
 They SHOULD terminate an existing search, and MAY cause the initiator
 to begin a new search if appropriate.
 Any redirection (3xx) response MUST NOT suggest any of the addresses
 in the Via (Section 6.40) path of the request in the Contact header
 field. (Addresses match if their host and port number match.)
 To avoid forwarding loops, a user agent client or proxy MUST check
 whether the address returned by a redirect server equals an address
 tried earlier.
7.3.1 300 Multiple Choices
 The address in the request resolved to several choices, each with its
 own specific location, and the user (or user agent) can select a
 preferred communication end point and redirect its request to that
 location.
 The response SHOULD include an entity containing a list of resource
 characteristics and location(s) from which the user or user agent can
 choose the one most appropriate, if allowed by the Accept request
 header. The entity format is specified by the media type given in the
 Content-Type header field. The choices SHOULD also be listed as
 Contact fields (Section 6.13). Unlike HTTP, the SIP response MAY
 contain several Contact fields or a list of addresses in a Contact
 field. User agents MAY use the Contact header field value for
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 automatic redirection or MAY ask the user to confirm a choice.
 However, this specification does not define any standard for such
 automatic selection.
 This status response is appropriate if the callee can be
 reached at several different locations and the server
 cannot or prefers not to proxy the request.
7.3.2 301 Moved Permanently
 The user can no longer be found at the address in the Request-URI and
 the requesting client SHOULD retry at the new address given by the
 Contact header field (Section 6.13). The caller SHOULD update any
 local directories, address books and user location caches with this
 new value and redirect future requests to the address(es) listed.
7.3.3 302 Moved Temporarily
 The requesting client SHOULD retry the request at the new address(es)
 given by the Contact header field (Section 6.13). The duration of
 the redirection can be indicated through an Expires (Section 6.20)
 header. If there is no explicit expiration time, the address is only
 valid for this call and MUST NOT be cached for future calls.
7.3.4 305 Use Proxy
 The requested resource MUST be accessed through the proxy given by
 the Contact field. The Contact field gives the URI of the proxy. The
 recipient is expected to repeat this single request via the proxy.
 305 responses MUST only be generated by user agent servers.
7.3.5 380 Alternative Service
 The call was not successful, but alternative services are possible.
 The alternative services are described in the message body of the
 response. Formats for such bodies are not defined here, and may be
 the subject of future standardization.
7.4 Request Failure 4xx
 4xx responses are definite failure responses from a particular
 server. The client SHOULD NOT retry the same request without
 modification (e.g., adding appropriate authorization). However, the
 same request to a different server might be successful.
7.4.1 400 Bad Request
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 The request could not be understood due to malformed syntax.
7.4.2 401 Unauthorized
 The request requires user authentication.
7.4.3 402 Payment Required
 Reserved for future use.
7.4.4 403 Forbidden
 The server understood the request, but is refusing to fulfill it.
 Authorization will not help, and the request SHOULD NOT be repeated.
7.4.5 404 Not Found
 The server has definitive information that the user does not exist at
 the domain specified in the Request-URI. This status is also returned
 if the domain in the Request-URI does not match any of the domains
 handled by the recipient of the request.
7.4.6 405 Method Not Allowed
 The method specified in the Request-Line is not allowed for the
 address identified by the Request-URI. The response MUST include an
 Allow header field containing a list of valid methods for the
 indicated address.
7.4.7 406 Not Acceptable
 The resource identified by the request is only capable of generating
 response entities which have content characteristics not acceptable
 according to the accept headers sent in the request.
7.4.8 407 Proxy Authentication Required
 This code is similar to 401 (Unauthorized), but indicates that the
 client MUST first authenticate itself with the proxy. The proxy MUST
 return a Proxy-Authenticate header field (section 6.26) containing a
 challenge applicable to the proxy for the requested resource. The
 client MAY repeat the request with a suitable Proxy-Authorization
 header field (section 6.27). SIP access authentication is explained
 in section 13.2 and 14.
 This status code is used for applications where access to the
 communication channel (e.g., a telephony gateway) rather than the
 callee requires authentication.
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7.4.9 408 Request Timeout
 The server could not produce a response, e.g., a user location,
 within the time indicated in the Expires request-header field. The
 client MAY repeat the request without modifications at any later
 time.
7.4.10 409 Conflict
 The request could not be completed due to a conflict with the current
 state of the resource. This response is returned if the action
 parameter in a REGISTER request conflicts with existing
 registrations.
7.4.11 410 Gone
 The requested resource is no longer available at the server and no
 forwarding address is known. This condition is expected to be
 considered permanent. If the server does not know, or has no facility
 to determine, whether or not the condition is permanent, the status
 code 404 (Not Found) SHOULD be used instead.
7.4.12 411 Length Required
 The server refuses to accept the request without a defined Content-
 Length. The client MAY repeat the request if it adds a valid
 Content-Length header field containing the length of the message-body
 in the request message.
7.4.13 413 Request Entity Too Large
 The server is refusing to process a request because the request
 entity is larger than the server is willing or able to process. The
 server MAY close the connection to prevent the client from continuing
 the request.
 If the condition is temporary, the server SHOULD include a Retry-
 After header field to indicate that it is temporary and after what
 time the client MAY try again.
7.4.14 414 Request-URI Too Long
 The server is refusing to service the request because the Request-URI
 is longer than the server is willing to interpret.
7.4.15 415 Unsupported Media Type
 The server is refusing to service the request because the message
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 body of the request is in a format not supported by the requested
 resource for the requested method. The server SHOULD return a list of
 acceptable formats using the Accept, Accept-Encoding and Accept-
 Language header fields.
7.4.16 420 Bad Extension
 The server did not understand the protocol extension specified in a
 Require (Section 6.30) header field.
7.4.17 480 Temporarily Unavailable
 The callee's end system was contacted successfully but the callee is
 currently unavailable (e.g., not logged in or logged in in such a
 manner as to preclude communication with the callee). The response
 MAY indicate a better time to call in the Retry-After header. The
 user could also be available elsewhere (unbeknownst to this host),
 thus, this response does not terminate any searches. The reason
 phrase SHOULD indicate a more precise cause as to why the callee is
 unavailable. This value SHOULD be setable by the user agent. Status
 486 (Busy Here) MAY be used to more precisely indicate a particular
 reason for the call failure.
 This status is also returned by a redirect server that recognizes the
 user identified by the Request-URI, but does not currently have a
 valid forwarding location for that user.
7.4.18 481 Call Leg/Transaction Does Not Exist
 This status is returned under two conditions: The server received a
 BYE request that does not match any existing call leg or the server
 received a CANCEL request that does not match any existing
 transaction. (A server simply discards an ACK referring to an unknown
 transaction.)
7.4.19 482 Loop Detected
 The server received a request with a Via (Section 6.40) path
 containing itself.
7.4.20 483 Too Many Hops
 The server received a request that contains more Via entries (hops)
 (Section 6.40) than allowed by the Max-Forwards (Section 6.23) header
 field.
7.4.21 484 Address Incomplete
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 The server received a request with a To (Section 6.37) address or
 Request-URI that was incomplete. Additional information SHOULD be
 provided.
 This status code allows overlapped dialing. With overlapped
 dialing, the client does not know the length of the dialing
 string. It sends strings of increasing lengths, prompting
 the user for more input, until it no longer receives a 484
 status response.
7.4.22 485 Ambiguous
 The callee address provided in the request was ambiguous. The
 response MAY contain a listing of possible unambiguous addresses in
 Contact headers.
 Revealing alternatives can infringe on privacy concerns of the user
 or the organization. It MUST be possible to configure a server to
 respond with status 404 (Not Found) or to suppress the listing of
 possible choices if the request address was ambiguous.
 Example response to a request with the URL lee@example.com :
 485 Ambiguous SIP/2.0
 Contact: Carol Lee <sip:carol.lee@example.com>
 Contact: Ping Lee <sip:p.lee@example.com>
 Contact: Lee M. Foote <sip:lee.foote@example.com>
 Some email and voice mail systems provide this
 functionality. A status code separate from 3xx is used
 since the semantics are different: for 300, it is assumed
 that the same person or service will be reached by the
 choices provided. While an automated choice or sequential
 search makes sense for a 3xx response, user intervention is
 required for a 485 response.
7.4.23 486 Busy Here
 The callee's end system was contacted successfully but the callee is
 currently not willing or able to take additional calls. The response
 MAY indicate a better time to call in the Retry-After header. The
 user could also be available elsewhere, such as through a voice mail
 service, thus, this response does not terminate any searches. Status
 600 (Busy Everywhere) SHOULD be used if the client knows that no
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 other end system will be able to accept this call.
7.5 Server Failure 5xx
 5xx responses are failure responses given when a server itself has
 erred. They are not definitive failures, and MUST NOT terminate a
 search if other possible locations remain untried.
7.5.1 500 Server Internal Error
 The server encountered an unexpected condition that prevented it from
 fulfilling the request. The client MAY display the specific error
 condition, and MAY retry the request after several seconds.
7.5.2 501 Not Implemented
 The server does not support the functionality required to fulfill the
 request. This is the appropriate response when the server does not
 recognize the request method and is not capable of supporting it for
 any user.
7.5.3 502 Bad Gateway
 The server, while acting as a gateway or proxy, received an invalid
 response from the downstream server it accessed in attempting to
 fulfill the request.
7.5.4 503 Service Unavailable
 The server is currently unable to handle the request due to a
 temporary overloading or maintenance of the server. The implication
 is that this is a temporary condition which will be alleviated after
 some delay. If known, the length of the delay MAY be indicated in a
 Retry-After header. If no Retry-After is given, the client MUST
 handle the response as it would for a 500 response.
 Note: The existence of the 503 status code does not imply that a
 server has to use it when becoming overloaded. Some servers MAY wish
 to simply refuse the connection.
7.5.5 504 Gateway Time-out
 The server, while acting as a gateway, did not receive a timely
 response from the server (e.g., a location server) it accessed in
 attempting to complete the request.
7.5.6 505 Version Not Supported
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 The server does not support, or refuses to support, the SIP protocol
 version that was used in the request message. The server is
 indicating that it is unable or unwilling to complete the request
 using the same major version as the client, other than with this
 error message. The response MAY contain an entity describing why that
 version is not supported and what other protocols are supported by
 that server. The format for such an entity is not defined here and
 may be the subject of future standardization.
7.6 Global Failures 6xx
 6xx responses indicate that a server has definitive information about
 a particular user, not just the particular instance indicated in the
 Request-URI. All further searches for this user are doomed to failure
 and pending searches SHOULD be terminated.
7.6.1 600 Busy Everywhere
 The callee's end system was contacted successfully but the callee is
 busy and does not wish to take the call at this time. The response
 MAY indicate a better time to call in the Retry-After header. If the
 callee does not wish to reveal the reason for declining the call, the
 callee uses status code 603 (Decline) instead. This status response
 is returned only if the client knows that no other end point (such as
 a voice mail system) will answer the request. Otherwise, 486 (Busy
 Here) should be returned.
7.6.2 603 Decline
 The callee's machine was successfully contacted but the user
 explicitly does not wish to or cannot participate. The response MAY
 indicate a better time to call in the Retry-After header.
7.6.3 604 Does Not Exist Anywhere
 The server has authoritative information that the user indicated in
 the To request field does not exist anywhere. Searching for the user
 elsewhere will not yield any results.
7.6.4 606 Not Acceptable
 The user's agent was contacted successfully but some aspects of the
 session description such as the requested media, bandwidth, or
 addressing style were not acceptable.
 A 606 (Not Acceptable) response means that the user wishes to
 communicate, but cannot adequately support the session described. The
 606 (Not Acceptable) response MAY contain a list of reasons in a
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 Warning header field describing why the session described cannot be
 supported. Reasons are listed in Section 6.41. It is hoped that
 negotiation will not frequently be needed, and when a new user is
 being invited to join an already existing conference, negotiation may
 not be possible. It is up to the invitation initiator to decide
 whether or not to act on a 606 (Not Acceptable) response.
8 SIP Message Body
8.1 Body Inclusion
 Requests MAY contain message bodies unless otherwise noted. Within
 this specification, the BYE request MUST NOT contain a message body.
 For ACK, INVITE and OPTIONS, the message body is always a session
 description. The use of message bodies for REGISTER requests is for
 further study.
 For response messages, the request method and the response status
 code determine the type and interpretation of any message body. All
 responses MAY include a body. Message bodies for 1xx responses
 contain advisory information about the progress of the request. 2xx
 responses to INVITE requests contain session descriptions. In 3xx
 responses, the message body MAY contain the description of
 alternative destinations or services, as described in Section 7.3.
 For responses with status 400 or greater, the message body MAY
 contain additional, human-readable information about the reasons for
 failure. It is RECOMMENDED that information in 1xx and 300 and
 greater responses be of type text/plain or text/html
8.2 Message Body Type
 The Internet media type of the message body MUST be given by the
 Content-Type header field. If the body has undergone any encoding
 (such as compression) then this MUST be indicated by the Content-
 Encoding header field, otherwise Content-Encoding MUST be omitted. If
 applicable, the character set of the message body is indicated as
 part of the Content-Type header-field value.
8.3 Message Body Length
 The body length in bytes SHOULD be given by the Content-Length header
 field. Section 6.15 describes the behavior in detail.
 The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.
 (Note: The chunked encoding modifies the body of a message in order
 to transfer it as a series of chunks, each with its own size
 indicator.)
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9 Compact Form
 When SIP is carried over UDP with authentication and a complex
 session description, it may be possible that the size of a request or
 response is larger than the MTU. To address this problem, a more
 compact form of SIP is also defined by using abbreviations for the
 common header fields listed below:
 short field name long field name note
 c Content-Type
 e Content-Encoding
 f From
 i Call-ID
 m Contact from "moved"
 l Content-Length
 s Subject
 t To
 v Via
 Thus, the message in section 16.2 could also be written:
 INVITE sip:schooler@vlsi.caltech.edu SIP/2.0
 v:SIP/2.0/UDP 131.215.131.131;maddr=239.128.16.254;ttl=16
 v:SIP/2.0/UDP 128.16.64.19
 f:sip:mjh@isi.edu
 t:sip:schooler@cs.caltech.edu
 i:62729-27@128.16.64.19
 c:application/sdp
 CSeq: 4711 INVITE
 l:187
 v=0
 o=user1 53655765 2353687637 IN IP4 128.3.4.5
 s=Mbone Audio
 i=Discussion of Mbone Engineering Issues
 e=mbone@somewhere.com
 c=IN IP4 224.2.0.1/127
 t=0 0
 m=audio 3456 RTP/AVP 0
 Clients MAY mix short field names and long field names within the
 same request. Servers MUST accept both short and long field names for
 requests. Proxies MAY change header fields between their long and
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 short forms, but this MUST NOT be done to fields following an
 Authorization header.
10 Behavior of SIP Clients and Servers
10.1 General Remarks
 SIP is defined so it can use either UDP (unicast or multicast) or TCP
 as a transport protocol; it provides its own reliability mechanism.
10.1.1 Requests
 Servers discard isomorphic requests, but first retransmit the
 appropriate response. (SIP requests are said to be idempotent , i.e.,
 receiving more than one copy of a request does not change the server
 state.)
 After receiving a CANCEL request from an upstream client, a stateful
 proxy server MAY send a CANCEL on all branches where it has not yet
 received a final response.
 When a user agent receives a request, it checks the Call-ID against
 those of in-progress calls. If the Call-ID was found, it compares the
 tag value of To with the user's tag and rejects the request if the
 two do not match. If the From header, including any tag value,
 matches the value for an existing call leg, the server compares the
 CSeq header field value. If less than or equal to the current
 sequence number, the request is a retransmission. Otherwise, it is a
 new request. If the From header does not match an existing call leg,
 a new call leg is created.
 If the Call-ID was not found, a new call leg is created, with entries
 for the To, From and Call-ID headers. In this case, the To header
 field should not have contained a tag. The server returns a response
 containing the same To value, but with a unique tag added. The tag
 MAY be omitted if the request contained only one Via header field.
10.1.2 Responses
 A server MAY issue one or more provisional responses at any time
 before sending a final response. If a stateful proxy, user agent
 server, redirect server or registrar cannot respond to a request with
 a final response within 200 ms, it SHOULD issue a provisional (1xx)
 response as soon as possible. Stateless proxies MUST NOT issue
 provisional responses on their own.
 Responses are mapped to requests by the matching To, From, Call-ID,
 CSeq headers and the branch parameter of the first Via header.
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 Responses terminate request retransmissions even if they have Via
 headers that cause them to be delivered to an upstream client.
 A stateful proxy may receive a response that it does not have state
 for, that is, where it has no a record of an associated request. If
 the Via header field indicates that the upstream server used TCP, the
 proxy actively opens a TCP connection to that address. Thus, proxies
 have to be prepared to receive responses on the incoming side of
 passive TCP connections, even though most responses will arrive on
 the incoming side of an active connection. (An active connection is a
 TCP connection initiated by the proxy, a passive connection is one
 accepted by the proxy, but initiated by another entity.)
 100 responses SHOULD NOT be forwarded, other 1xx responses MAY be
 forwarded, possibly after the server eliminates responses with status
 codes that had already been sent earlier. 2xx responses are forwarded
 according to the Via header. Once a stateful proxy has received a 2xx
 response, it MUST NOT forward non-2xx final responses. Responses
 with status 300 and higher are retransmitted by each stateful proxy
 until the next upstream proxy sends an ACK (see below for timing
 details) or CANCEL.
 A stateful proxy SHOULD maintain state for at least 32 seconds after
 the receipt of the first definitive non-200 response, in order to
 handle retransmissions of the response.
 The 32 second window is given by the maximum retransmission
 duration of 200-class responses using the default timers,
 in case the ACK is lost somewhere on the way to the called
 user agent or the next stateful proxy.
10.2 Source Addresses, Destination Addresses and Connections
10.2.1 Unicast UDP
 Responses are returned to the address listed in the Via header field
 (Section 6.40), not the source address of the request.
 Recall that responses are not generated by the next-hop
 stateless server, but generated by either a proxy server or
 the user agent server. Thus, the stateless proxy can only
 use the Via header field to forward the response.
10.2.2 Multicast UDP
 Requests MAY be multicast; multicast requests likely feature a host-
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 independent Request-URI. This request SHOULD be scoped to ensure it
 is not forwarded beyond the boundaries of the administrative system.
 This MAY be done with either TTL or administrative scopes[25],
 depending on what is implemented in the network.
 A client receiving a multicast query does not have to check whether
 the host part of the Request-URI matches its own host or domain name.
 If the request was received via multicast, the response is also
 returned via multicast. Responses to multicast requests are multicast
 with the same TTL as the request, where the TTL is derived from the
 ttl parameter in the Via header (Section 6.40).
 To avoid response implosion, servers MUST NOT answer multicast
 requests with a status code other than 2xx or 6xx. The server delays
 its response by a random interval uniformly distributed between zero
 and one second. Servers MAY suppress responses if they hear a lower-
 numbered or 6xx response from another group member prior to sending.
 Servers do not respond to CANCEL requests received via multicast to
 avoid request implosion. A proxy or UAC SHOULD send a CANCEL on
 receiving the first 2xx or 6xx response to a multicast request.
 Server response suppression is a MAY since it requires a
 server to violate some basic message processing rules. Lets
 say A sends a multicast request, and it is received by B,C,
 and D. B sends a 200 response. The topmost Via field in the
 response will contain the address of A. C will also receive
 this response, and could use it to suppress its own
 response. However, C would normally not examine this
 response, as the topmost Via is not its own. Normally, a
 response received with an incorrect topmost Via MUST be
 dropped, but not in this case. To distinguish this packet
 from a misrouted or multicast looped packet is fairly
 complex, and for this reason the procedure is a MAY. The
 CANCEL, instead, provides a simpler and more standard way
 to perform response suppression. It is for this reason that
 the use of CANCEL here is a SHOULD
10.3 TCP
 A single TCP connection can serve one or more SIP transactions. A
 transaction contains zero or more provisional responses followed by
 one or more final responses. (Typically, transactions contain exactly
 one final response, but there are exceptional circumstances, where,
 for example, multiple 200 responses can be generated.)
 The client SHOULD keep the connection open at least until the first
 final response arrives. If the client closes or resets the TCP
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 connection prior to receiving the first final response, the server
 treats this action as equivalent to a CANCEL request.
 This behavior makes it less likely that malfunctioning
 clients cause a proxy server to keep connection state
 indefinitely.
 The server SHOULD NOT close the TCP connection until it has sent its
 final response, at which point it MAY close the TCP connection if it
 wishes to. However, normally it is the client's responsibility to
 close the connection.
 If the server leaves the connection open, and if the client so
 desires it MAY re-use the connection for further SIP requests or for
 requests from the same family of protocols (such as HTTP or stream
 control commands).
 If a server needs to return a response to a client and no longer has
 a connection open to that client, it MAY open a connection to the
 address listed in the Via header. Thus, a proxy or user agent MUST be
 prepared to receive both requests and responses on a "passive"
 connection.
10.4 Reliability for BYE, CANCEL, OPTIONS, REGISTER Requests
10.4.1 UDP
 A SIP client using UDP SHOULD retransmit a BYE, CANCEL, OPTIONS, or
 REGISTER request with an exponential backoff, starting at a T1 second
 interval, doubling the interval for each packet, and capping off at a
 T2 second interval. This means that after the first packet is sent,
 the second is sent T1 seconds later, the next 2*T1 seconds after
 that, the next 4*T1 seconds after that, and so on, until the interval
 hits T2. Subsequent retransmissions are spaced by T2 seconds. If the
 client receives a provisional response, it continues to retransmit
 the request, but with an interval of T2 seconds. Retransmissions
 cease when the client has sent a total of eleven packets, or receives
 a definitive response. Default values for T1 and T2 are 500 ms and 4
 s, respectively. Clients MAY use larger values, but SHOULD NOT use
 smaller ones. Servers retransmit the response upon receipt of a
 request retransmission. After the server sends a final response, it
 cannot be sure the client has received the response, and thus SHOULD
 cache the results for at least 10*T2 seconds to avoid having to, for
 example, contact the user or location server again upon receiving a
 request retransmission.
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 Use of the exponential backoff is for congestion control
 purposes. However, the back-off must cap off, since request
 retransmissions are used to trigger response
 retransmissions at the server. Without a cap, the loss of a
 single response could significantly increase transaction
 latencies.
 The value of the initial retransmission timer is smaller than that
 that for TCP since it is expected that network paths suitable for
 interactive communications have round-trip times smaller than 500 ms.
 For congestion control purposes, the retransmission count has to be
 bounded. Given that most transactions are expected to consist of one
 request and a few responses, round-trip time estimation is not likely
 to be very useful. If RTT estimation is desired to more quickly
 discover a missing final response, each request retransmission needs
 to be labeled with its own Timestamp (Section 6.36), returned in the
 response. The server caches the result until it can be sure that the
 client will not retransmit the same request again.
 Each server in a proxy chain generates its own final response to a
 CANCEL request. The server responds immediately upon receipt of the
 CANCEL request rather than waiting until it has received final
 responses from the CANCEL requests it generates.
 BYE and OPTIONS final responses are generated by redirect and user
 agent servers; REGISTER final responses are generated by registrars.
 Note that in contrast to the reliability mechanism described in
 Section 10.5, responses to these requests are not retransmitted
 periodically and not acknowledged via ACK.
10.4.2 TCP
 Clients using TCP do not need to retransmit requests.
10.5 Reliability for INVITE Requests
 Special considerations apply for the INVITE method.
 1. After receiving an invitation, considerable time can elapse
 before the server can determine the outcome. For example,
 if the called party is "rung" or extensive searches are
 performed, delays between the request and a definitive
 response can reach several tens of seconds. If either
 caller or callee are automated servers not directly
 controlled by a human being, a call attempt could be
 unbounded in time.
 2. If a telephony user interface is modeled or if we need to
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 interface to the PSTN, the caller's user interface will
 provide "ringback", a signal that the callee is being
 alerted. (The status response 180 (Ringing) MAY be used to
 initiate ringback.) Once the callee picks up, the caller
 needs to know so that it can enable the voice path and stop
 ringback. The callee's response to the invitation could get
 lost. Unless the response is transmitted reliably, the
 caller will continue to hear ringback while the callee
 assumes that the call exists.
 3. The client has to be able to terminate an on-going request,
 e.g., because it is no longer willing to wait for the
 connection or search to succeed. The server will have to
 wait several retransmission intervals to interpret the lack
 of request retransmissions as the end of a call. If the
 call succeeds shortly after the caller has given up, the
 callee will "pick up the phone" and not be "connected".
10.5.1 UDP
 For UDP, A SIP client SHOULD retransmit a SIP INVITE request with an
 interval that starts at T1 seconds, and doubles after each packet
 transmission. The client ceases retransmissions if it receives a
 provisional or definitive response, or once it has sent a total of 7
 request packets.
 A server which transmits a provisional response should retransmit it
 upon reception of a duplicate request. A server which transmits a
 final response should retransmit it with an interval that starts at
 T1 seconds, and doubles for each subsequent packet. Response
 retransmissions cease when any one of the following occurs:
 1. An ACK request for the same transaction is received;
 2. a BYE request for the same call leg is received;
 3. a CANCEL request for the same call leg is received and the
 final response status was equal or greater to 300;
 4. the response has been transmitted 7 times.
 Only the user agent client generates an ACK for 2xx final responses,
 If the response contained a Contact header field, the ACK MAY be sent
 to the address listed in that Contact header field. If the response
 did not contain a Contact header, the client uses the same To header
 field and Request-URI as for the INVITE request and sends the ACK to
 the same destination as the original INVITE request. ACKs for final
 responses other than 2xx are sent to the same server that the
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 original request was sent to, using the same Request-URI as the
 original request. Note, however, that the To header field in the ACK
 is copied from the response being acknowledged, not the request, and
 thus MAY additionally contain the tag parameter. Also note than
 unlike 2xx final responses, a proxy generates an ACK for non-2xx
 final responses.
 The ACK request MUST NOT be acknowledged to prevent a response-ACK
 feedback loop. Fig. 12 and 13 show the client and server state
 diagram for invitations.
 The mechanism in Sec. 10.4 would not work well for INVITE
 because of the long delays between INVITE and a final
 response. If the 200 response were to get lost, the callee
 would believe the call to exist, but the voice path would
 be dead since the caller does not know that the callee has
 picked up. Thus, the INVITE retransmission interval would
 have to be on the order of a second or two to limit the
 duration of this state confusion. Retransmitting the
 response with an exponential back-off helps ensure that the
 response is received, without placing an undue burden on
 the network.
10.5.2 TCP
 A user agent using TCP MUST NOT retransmit requests, but uses the
 same algorithm as for UDP (Section 10.5.1) to retransmit responses
 until it receives an ACK.
 It is necessary to retransmit 2xx responses as their
 reliability is assured end-to-end only. If the chain of
 proxies has a UDP link in the middle, it could lose the
 response, with no possibility of recovery. For simplicity,
 we also retransmit non-2xx responses, although that is not
 strictly necessary.
10.6 Reliability for ACK Requests
 The ACK request does not generate responses. It is only generated
 when a response to an INVITE request arrives (see Section 10.5). This
 behavior is independent of the transport protocol. Note that the ACK
 request MAY take a different path than the original INVITE request,
 and MAY even cause a new TCP connection to be opened in order to send
 it.
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 +===========+
 * *
 ...........>* Initial *<;;;;;;;;;;
 : 7 INVITE * * ;
 : sent +===========+ ;
 : | ;
 : | - ;
 : | INVITE ;
 : | ;
 : v ;
 : ************* ;
 : T1*2^n <--* * ;
 : INVITE -->* Calling *--------+ ;
 : * * | ;
 : ************* | ;
 : : | | ;
 :.............: | 1xx xxx | ;
 | - ACK | ;
 | | ;
 v | ; 
 ************* | ;
 * * | ;
 * Ringing *<->1xx | ; 
 * * | ;
 ************* | ;
 | | ;
 |<-------------+ ; 
 | ;
 v ;
 ************* ;
 xxx <--* * ;
 ACK -->* Completed * ;
 * * ;
 ************* ;
 ; 32s (for proxy);
 ;;;;;;;;;;;;;;;;;;
 event (xxx=status)
 message 
 Figure 12: State transition diagram of client for INVITE method
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Internet Draft SIP January 15, 1999
 7 pkts sent +===============+ 
+-------------->* *
| * Initial *<...............
|;;;;;;;;;;;;;;>* *
|; +===============+ :
|; CANCEL ! :
|; 200 ! :
|; ! INVITE :
|; ! 1xx :
|; ! :
|; v :
|; ***************** BYE :
|; INVITE -->* * 200 :
|; 1xx <--* Call proceed. *..............>:
|; * * :
|;;;;;;;;;;;;;;;***************** :
|; ! ! :
|: ! ! :
|; failure ! ! picks up :
|; >= 300 ! ! 200 :
|; +-------+ +-------+ :
|; v v :
|; *********** *********** :
|;INVITE<* *<T1*2^n->* *>INVITE :
|;status>* failure *>status<-* success *<status :
|; * * * * :
|;;;;;;;;*********** *********** :
| ! : | | ! : :
| ! : | | ! : :
+-------------!-:-+------------+ ! : :
 ! :.................!..:.........>:
 ! ! BYE : 
 +---------+---------+ 200 :
 ! ACK :
 ! : 
 v :
 ***************** :
 V---* * :
 ACK * Confirmed * :
 |-->* * :
 ***************** . 
 : : 
 :......................>:
 event
 message sent
 Figure 13: State transition diagram of server for INVITE method
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10.7 ICMP Handling
 Handling of ICMP messages in the case of UDP messages is
 straightforward. For requests, a host, network, port, or protocol
 unreachable error SHOULD be treated as if a 400-class response was
 received. For responses, these errors SHOULD cause the server to
 cease retransmitting the response.
 Source quench ICMP messages SHOULD be ignored. TTL exceeded errors
 SHOULD be ignored. Parameter problem errors SHOULD be treated as if a
 400-class response was received.
11 Behavior of SIP User Agents
 This section describes the rules for user agent client and servers
 for generating and processing requests and responses.
11.1 Caller Issues Initial INVITE Request
 When a user agent client desires to initiate a call, it formulates an
 INVITE request. The To field in the request contains the address of
 the callee. The Request-URI contains the same address. The From field
 contains the address of the caller. If the From address can appear
 in requests generated by other user agent clients for the same call,
 the caller MUST insert the tag parameter in the From field. A UAC MAY
 optionally add a Contact header containing an address where it would
 like to be contacted for transactions from the callee back to the
 caller.
11.2 Callee Issues Response
 When the initial INVITE request is received at the callee, the callee
 can accept, redirect, or reject the call. In all of these cases, it
 formulates a response. The response MUST copy the To, From, Call-ID,
 CSeq and Via fields from the request. Additionally, the responding
 UAS MUST add the tag parameter to the To field in the response if the
 request contained more than one Via header field. Since a request
 from a UAC may fork and arrive at multiple hosts, the tag parameter
 serves to distinguish, at the UAC, multiple responses from different
 UAS's. The UAS MAY add a Contact header field in the response. It
 contains an address where the callee would like to be contacted for
 subsequent transactions, including the ACK for the current INVITE.
 The UAS stores the values of the To and From field, including any
 tags. These become the local and remote addresses of the call leg,
 respectively.
11.3 Caller Receives Response to Initial Request
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 Multiple responses may arrive at the UAC for a single INVITE request,
 due to a forking proxy. Each response is distinguished by the "tag"
 parameter in the To header field, and each represents a distinct call
 leg. The caller MAY choose to acknowledge or terminate the call with
 each responding UAS. To acknowledge, it sends an ACK request, and to
 terminate it sends a BYE request. The To header field in the ACK or
 BYE MUST be the same as the To field in the 200 response, including
 any tag. The From header field MUST be the same as the From header
 field in the 200 (OK) response, including any tag. The Request-URI of
 the ACK or BYE request MAY be set to whatever address was found in
 the Contact header field in the 200 (OK) response, if present.
 Alternately, a UAC may copy the address from the To header field into
 the Request-URI. The UAC also notes the value of the To and From
 header fields in each response. For each call leg, the To header
 field becomes the remote address, and the From header field becomes
 the local address.
11.4 Caller or Callee Generate Subsequent Requests
 Once the call has been established, either the caller or callee MAY
 generate INVITE or BYE requests to change or terminate the call.
 Regardless of whether the caller or callee is generating the new
 request, the header fields in the request are set as follows. For the
 desired call leg, the To header field is set to the remote address,
 and the From header field is set to the local address (both including
 any tags). The Contact header field MAY be different than the Contact
 header field sent in a previous response or request. The Request-URI
 MAY be set to the value of the Contact header field received in a
 previous request or response from the remote party, or to the value
 of the remote address.
11.5 Receiving Subsequent Requests
 When a request is received subsequently, the following checks are
 made:
 1. If the Call-ID is new, the request is for a new call,
 regardless of the values of the To and From header fields.
 2. If the Call-ID exists, the request is for an existing call.
 If the To, From, Call-ID, and CSeq values exactly match
 (including tags) those of any requests received previously,
 the request is a retransmission.
 3. If there was no match to the previous step, the To and From
 fields are compared against existing call leg local and
 remote addresses. If there is a match, and the CSeq in the
 request is higher than the last CSeq received on that leg,
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 the request is a new transaction for an existing call leg.
12 Behavior of SIP Proxy and Redirect Servers
 This section describes behavior of SIP redirect and proxy servers in
 detail. Proxy servers can "fork" connections, i.e., a single incoming
 request spawns several outgoing (client) requests.
12.1 Redirect Server
 A redirect server does not issue any SIP requests of its own. After
 receiving a request other than CANCEL, the server gathers the list of
 alternative locations and returns a final response of class 3xx or it
 refuses the request. For well-formed CANCEL requests, it SHOULD
 return a 2xx response. This response ends the SIP transaction. The
 redirect server maintains transaction state for the whole SIP
 transaction. It is up to the client to detect forwarding loops
 between redirect servers.
12.2 User Agent Server
 User agent servers behave similarly to redirect servers, except that
 they also accept requests and can return a response of class 2xx.
12.3 Proxy Server
 This section outlines processing rules for proxy servers. A proxy
 server can either be stateful or stateless. When stateful, a proxy
 remembers the incoming request which generated outgoing requests, and
 the outgoing requests. A stateless proxy forgets all information once
 an outgoing request is generated. A forking proxy SHOULD be stateful.
 Proxies that accept TCP connections MUST be stateful.
 Otherwise, if the proxy were to lose a request, the TCP
 client would never retransmit it.
 A stateful proxy SHOULD NOT become stateless until after it sends a
 definitive response upstream, and at least 32 seconds after it
 received a definitive response.
 A stateful proxy acts as a virtual UAS/UAC. It implements the server
 state machine when receiving requests, and the client state machine
 for generating outgoing requests, with the exception of receiving a
 2xx response to an INVITE. Instead of generating an ACK, the 2xx
 response is always forwarded upstream towards the caller.
 Furthermore, ACK's for 200 responses to INVITE's are always proxied
 downstream towards the UAS, as they would be for a stateless proxy.
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 A stateless proxy does not act as a virtual UAS/UAC (as this would
 require state). Rather, a stateless proxy forwards every request it
 receives downstream, and every response it receives upstream.
12.3.1 Proxying Requests
 To prevent loops, a server MUST check if its own address is already
 contained in the Via header field of the incoming request.
 The To, From, Call-ID, and Contact tags are copied exactly from the
 original request. The proxy SHOULD change the Request-URI to indicate
 the server where it intends to send the request.
 A proxy server always inserts a Via header field containing its own
 address into those requests that are caused by an incoming request.
 Each proxy MUST insert a "branch" parameter (Section 6.40).
12.3.2 Proxying Responses
 A proxy only processes a response if the topmost Via field matches
 one of its addresses. A response with a non-matching top Via field
 MUST be dropped.
12.3.3 Stateless Proxy: Proxying Responses
 A stateless proxy removes its own Via field, and checks the address
 in the next Via field. In the case of UDP, the response is sent to
 the address listed in the "maddr" tag if present, otherwise to the
 "received" tag if present, and finally to the address in the "sent-
 by" field. A proxy MUST remain stateful when handling requests
 received via TCP.
 A stateless proxy MUST NOT generate its own provisional responses.
12.3.4 Stateful Proxy: Receiving Requests
 When a stateful proxy receives a request, it checks the To, From
 (including tags), Call-ID and CSeq against existing request records.
 If the tuple exists, the request is a retransmission. The provisional
 or final response sent previously is retransmitted, as per the server
 state machine. If the tuple does not exist, the request corresponds
 to a new transaction, and the request should be proxied.
 A stateful proxy server MAY generate its own provisional (1xx)
 responses.
12.3.5 Stateful Proxy: Receiving ACKs
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 When an ACK request is received, it is either processed locally or
 proxied. To make this determination, the To, From, CSeq and Call-ID
 fields are compared against those in previous requests. If there is
 no match, the ACK request is proxied as if it were an INVITE request.
 If there is a match, and if the server had ever sent a 200 response
 upstream, the ACK is proxied. If the server had never sent any
 responses upstream, the ACK is also proxied. If the server had sent a
 3xx, 4xx, 5xx or 6xx response, but no 2xx response, the ACK is
 processed locally if the tag in the To field of the ACK matches the
 tag sent by the proxy in the response.
12.3.6 Stateful Proxy: Receiving Responses
 When a proxy server receives a response that has passed the Via
 checks, the proxy server checks the To (without the tag), From
 (including the tag), Call-ID and CSeq against values seen in previous
 requests. If there is no match, the response is forwarded upstream to
 the address listed in the Via field. If there is a match, the
 "branch" tag in the Via field is examined. If it matches a known
 branch identifier, the response is for the given branch, and
 processed by the virtual client for the given branch. Otherwise, the
 response is dropped.
 A stateful proxy should obey the rules in Section 12.4 to determine
 if the response should be proxied upstream. If it is to be proxied,
 the same rules for stateless proxies above are followed, with the
 following addition for TCP. If a request was received via TCP
 (indicated by the protocol in the top Via header), the proxy checks
 to see if it has a connection currently open to that address. If so,
 the response is sent on that connection. Otherwise, a new TCP
 connection is opened to the address and port in the Via field, and
 the response is sent there. Note that this implies that a UAC or
 proxy MUST be prepared to receive responses on the incoming side of a
 TCP connection. Definitive non 200-class responses MUST be
 retransmitted by the proxy, even over a TCP connection.
12.3.7 Stateless, Non-Forking Proxy
 Proxies in this category issue at most a single unicast request for
 each incoming SIP request, that is, they do not "fork" requests.
 However, servers MAY choose to always operate in a mode that allows
 issuing of several requests, as described in Section 12.4.
 The server can forward the request and any responses. It does not
 have to maintain any state for the SIP transaction. Reliability is
 assured by the next redirect or stateful proxy server in the server
 chain.
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 A proxy server SHOULD cache the result of any address translations
 and the response to speed forwarding of retransmissions. After the
 cache entry has been expired, the server cannot tell whether an
 incoming request is actually a retransmission of an older request.
 The server will treat it as a new request and commence another
 search.
12.4 Forking Proxy
 The server MUST respond to the request immediately with a 100
 (Trying) response.
 Successful responses to an INVITE request MAY contain a Contact
 header field so that the following ACK or BYE bypasses the proxy
 search mechanism. If the proxy requires future requests to be routed
 through it, it adds a Record-Route header to the request (Section
 6.29).
 The following C-code describes the behavior of a proxy server issuing
 several requests in response to an incoming INVITE request. The
 function request(r, a, b) sends a SIP request of type r to address a,
 with branch id b. await_response() waits until a response is received
 and returns the response. close(a) closes the TCP connection to
 client with address a. response(r) sends a response to the client.
 ismulticast() returns 1 if the location is a multicast address and
 zero otherwise. The variable timeleft indicates the amount of time
 left until the maximum response time has expired. The variable
 recurse indicates whether the server will recursively try addresses
 returned through a 3xx response. A server MAY decide to recursively
 try only certain addresses, e.g., those which are within the same
 domain as the proxy server. Thus, an initial multicast request can
 trigger additional unicast requests.
 /* request type */
 typedef enum {INVITE, ACK, BYE, OPTIONS, CANCEL, REGISTER} Method;
 process_request(Method R, int N, address_t address[])
 {
 struct {
 int branch; /* branch id */
 int done; /* has responded */
 } outgoing[];
 int done[]; /* address has responded */
 char *location[]; /* list of locations */
 int heard = 0; /* number of sites heard from */
 int class; /* class of status code */
 int timeleft = 120; /* sample timeout value */
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 int loc = 0; /* number of locations */
 struct { /* response */
 int status; /* response: CANCEL=-1 */
 int locations; /* number of redirect locations */
 char *location[]; /* redirect locations */
 address_t a; /* address of respondent */
 int branch; /* branch identifier */
 } r, best; /* response, best response */
 int i;
 best.status = 1000;
 for (i = 0; i < N; i++) {
 request(R, address[i], i);
 outgoing[i].done = 0;
 outgoing[i].branch = i;
 }
 while (timeleft > 0 && heard < N) {
 r = await_response();
 class = r.status / 100;
 /* If final response, mark branch as done. */
 if (class >= 2) {
 heard++;
 for (i = 0; i < N; i++) {
 if (r.branch == outgoing[i].branch) {
 outgoing[i].done = 1;
 break;
 }
 }
 }
 /* CANCEL: respond, fork and wait for responses */
 else if (class < 0) {
 best.status = 200;
 response(best);
 for (i = 0; i < N; i++) {
 if (!outgoing[i].done)
 request(CANCEL, address[i], outgoing[i].branch);
 }
 best.status = -1;
 }
 /* Send an ACK */
 if (class != 2) {
 if (R == INVITE) request(ACK, r.a, r.branch);
 }
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 if (class == 2) {
 if (r.status < best.status) best = r;
 break;
 }
 else if (class == 3) {
 /* A server MAY optionally recurse. The server MUST check
 * whether it has tried this location before and whether the
 * location is part of the Via path of the incoming request.
 * This check is omitted here for brevity. Multicast locations
 * MUST NOT be returned to the client if the server is not
 * recursing.
 */
 if (recurse) {
 multicast = 0;
 N += r.locations;
 for (i = 0; i < r.locations; i++) {
 request(R, r.location[i]);
 }
 } else if (!ismulticast(r.location)) {
 best = r;
 }
 }
 else if (class == 4) {
 if (best.status >= 400) best = r;
 }
 else if (class == 5) {
 if (best.status >= 500) best = r;
 }
 else if (class == 6) {
 best = r;
 break;
 }
 }
 /* We haven't heard anything useful from anybody. */
 if (best.status == 1000) {
 best.status = 404;
 }
 if (best.status/100 != 3) loc = 0;
 response(best);
 }
 Responses are processed as follows. The process completes (and state
 can be freed) when all requests have been answered by final status
 responses (for unicast) or 60 seconds have elapsed (for multicast). A
 proxy MAY send a CANCEL to all branches and return a 408 (Timeout) to
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 the client after 60 seconds or more.
 1xx: The proxy MAY forward the response upstream towards the client.
 2xx: The proxy MUST forward the response upstream towards the client,
 without sending an ACK downstream. After receiving a 2xx, the
 server MAY terminate all other pending requests by sending a
 CANCEL request and closing the TCP connection, if applicable.
 (Terminating pending requests is advisable as searches consume
 resources. Also, INVITE requests could "ring" on a number of
 workstations if the callee is currently logged in more than
 once.)
 3xx: The proxy MUST send an ACK and MAY recurse on the listed Contact
 addresses. Otherwise, the lowest-numbered response is returned
 if there were no 2xx responses.
 Location lists are not merged as that would prevent
 forwarding of authenticated responses. Also, responses can
 have message bodies, so that merging is not feasible.
 4xx, 5xx: The proxy MUST send an ACK and remember the response if it
 has a lower status code than any previous 4xx and 5xx responses.
 On completion, the lowest-numbered response is returned if there
 were no 2xx or 3xx responses.
 6xx: The proxy MUST forward the response to the client and send an
 ACK. Other pending requests MAY be terminated with CANCEL as
 described for 2xx responses.
 A proxy server forwards any response for Call-IDs for which it does
 not have a pending transaction according to the response's Via
 header. User agent servers respond to BYE requests for unknown call
 legs with status code 481 (Transaction Does Not Exist); they drop ACK
 requests with unknown call legs silently.
 Special considerations apply for choosing forwarding destinations for
 ACK and BYE requests. In most cases, these requests will bypass
 proxies and reach the desired party directly, keeping proxies from
 having to make forwarding decisions.
 A proxy MAY maintain call state for a period of its choosing. If a
 proxy still has list of destinations that it forwarded the last
 INVITE to, it SHOULD direct ACK requests only to those downstream
 servers.
13 Security Considerations
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13.1 Confidentiality and Privacy: Encryption
13.1.1 End-to-End Encryption
 SIP requests and responses can contain sensitive information about
 the communication patterns and communication content of individuals.
 The SIP message body MAY also contain encryption keys for the session
 itself. SIP supports three complementary forms of encryption to
 protect privacy:
 o End-to-end encryption of the SIP message body and certain
 sensitive header fields;
 o hop-by-hop encryption to prevent eavesdropping that tracks who
 is calling whom;
 o hop-by-hop encryption of Via fields to hide the route a
 request has taken.
 Not all of the SIP request or response can be encrypted end-to-end
 because header fields such as To and Via need to be visible to
 proxies so that the SIP request can be routed correctly. Hop-by-hop
 encryption encrypts the entire SIP request or response on the wire so
 that packet sniffers or other eavesdroppers cannot see who is calling
 whom. Hop-by-hop encryption can also encrypt requests and responses
 that have been end-to-end encrypted. Note that proxies can still see
 who is calling whom, and this information is also deducible by
 performing a network traffic analysis, so this provides a very
 limited but still worthwhile degree of protection.
 SIP Via fields are used to route a response back along the path taken
 by the request and to prevent infinite request loops. However, the
 information given by them can also provide useful information to an
 attacker. Section 6.22 describes how a sender can request that Via
 fields be encrypted by cooperating proxies without compromising the
 purpose of the Via field.
 End-to-end encryption relies on keys shared by the two user agents
 involved in the request. Typically, the message is sent encrypted
 with the public key of the recipient, so that only that recipient can
 read the message. All implementations SHOULD support PGP-based
 encryption [33] and MAY implement other schemes.
 A SIP request (or response) is end-to-end encrypted by splitting the
 message to be sent into a part to be encrypted and a short header
 that will remain in the clear. Some parts of the SIP message, namely
 the request line, the response line and certain header fields marked
 with "n" in the "enc." column in Table 4 and 5 need to be read and
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 returned by proxies and thus MUST NOT be encrypted end-to-end.
 Possibly sensitive information that needs to be made available as
 plaintext include destination address (To) and the forwarding path
 (Via) of the call. The Authorization header field MUST remain in the
 clear if it contains a digital signature as the signature is
 generated after encryption, but MAY be encrypted if it contains
 "basic" or "digest" authentication. The From header field SHOULD
 normally remain in the clear, but MAY be encrypted if required, in
 which case some proxies MAY return a 401 (Unauthorized) status if
 they require a From field.
 Other header fields MAY be encrypted or MAY travel in the clear as
 desired by the sender. The Subject, Allow and Content-Type header
 fields will typically be encrypted. The Accept, Accept-Language,
 Date, Expires, Priority, Require, Call-ID, Cseq, and Timestamp header
 fields will remain in the clear.
 All fields that will remain in the clear MUST precede those that will
 be encrypted. The message is encrypted starting with the first
 character of the first header field that will be encrypted and
 continuing through to the end of the message body. If no header
 fields are to be encrypted, encrypting starts with the second CRLF
 pair after the last header field, as shown below. Carriage return and
 line feed characters have been made visible as "$", and the encrypted
 part of the message is outlined.
 INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
 Via: SIP/2.0/UDP 169.130.12.5$
 To: T. A. Watson <sip:watson@bell-telephone.com>$
 From: A. Bell <sip:a.g.bell@bell-telephone.com>$
 Encryption: PGP version=5.0$
 Content-Length: 224$
 Call-ID: 187602141351@worcester.bell-telephone.com$
 CSeq: 488$
 $
 *******************************************************
 * Subject: Mr. Watson, come here.$ *
 * Content-Type: application/sdp$ *
 * $ *
 * v=0$ *
 * o=bell 53655765 2353687637 IN IP4 128.3.4.5$ *
 * c=IN IP4 135.180.144.94$ *
 * m=audio 3456 RTP/AVP 0 3 4 5$ *
 *******************************************************
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 An Encryption header field MUST be added to indicate the encryption
 mechanism used. A Content-Length field is added that indicates the
 length of the encrypted body. The encrypted body is preceded by a
 blank line as a normal SIP message body would be.
 Upon receipt by the called user agent possessing the correct
 decryption key, the message body as indicated by the Content-Length
 field is decrypted, and the now-decrypted body is appended to the
 clear-text header fields. There is no need for an additional
 Content-Length header field within the encrypted body because the
 length of the actual message body is unambiguous after decryption.
 Had no SIP header fields required encryption, the message would have
 been as below. Note that the encrypted body MUST then include a blank
 line (start with CRLF) to disambiguate between any possible SIP
 header fields that might have been present and the SIP message body.
 INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
 Via: SIP/2.0/UDP 169.130.12.5$
 To: T. A. Watson <sip:watson@bell-telephone.com>$
 From: A. Bell <a.g.bell@bell-telephone.com>$
 Encryption: PGP version=5.0$
 Content-Type: application/sdp$
 Content-Length: 107$
 $
 *************************************************
 * $ *
 * v=0$ *
 * o=bell 53655765 2353687637 IN IP4 128.3.4.5$ *
 * c=IN IP4 135.180.144.94$ *
 * m=audio 3456 RTP/AVP 0 3 4 5$ *
 *************************************************
13.1.2 Privacy of SIP Responses
 SIP requests can be sent securely using end-to-end encryption and
 authentication to a called user agent that sends an insecure
 response. This is allowed by the SIP security model, but is not a
 good idea. However, unless the correct behaviour is explicit, it
 would not always be possible for the called user agent to infer what
 a reasonable behaviour was. Thus when end-to-end encryption is used
 by the request originator, the encryption key to be used for the
 response SHOULD be specified in the request. If this were not done,
 it might be possible for the called user agent to incorrectly infer
 an appropriate key to use in the response. Thus, to prevent key-
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Internet Draft SIP January 15, 1999
 guessing becoming an acceptable strategy, we specify that a called
 user agent receiving a request that does not specify a key to be used
 for the response SHOULD send that response unencrypted.
 Any SIP header fields that were encrypted in a request SHOULD also be
 encrypted in an encrypted response. Contact response fields MAY be
 encrypted if the information they contain is sensitive, or MAY be
 left in the clear to permit proxies more scope for localized
 searches.
13.1.3 Encryption by Proxies
 Normally, proxies are not allowed to alter end-to-end header fields
 and message bodies. Proxies MAY, however, encrypt an unsigned request
 or response with the key of the call recipient.
 Proxies need to encrypt a SIP request if the end system
 cannot perform encryption or to enforce organizational
 security policies.
13.1.4 Hop-by-Hop Encryption
 SIP requests and responses MAY also be protected by security
 mechanisms at the transport or network layer. No particular mechanism
 is defined or recommended here. Two possibilities are IPSEC [34] or
 TLS [35]. The use of a particular mechanism will generally need to be
 specified out of band, through manual configuration, for example.
13.1.5 Via field encryption
 When Via header fields are to be hidden, a proxy that receives a
 request containing an appropriate "Hide: hop" header field (as
 specified in section 6.22) SHOULD encrypt the header field. As only
 the proxy that encrypts the field will decrypt it, the algorithm
 chosen is entirely up to the proxy implementor. Two methods satisfy
 these requirements:
 o The server keeps a cache of Via header fields and the
 associated To header field, and replaces the Via header field
 with an index into the cache. On the reverse path, take the
 Via header field from the cache rather than the message.
 This is insufficient to prevent message looping, and so an
 additional ID MUST be added so that the proxy can detect loops.
 This SHOULD NOT normally be the address of the proxy as the goal
 is to hide the route, so instead a sufficiently large random
 number SHOULD be used by the proxy and maintained in the cache.
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 It is possible for replies to get directed to the wrong
 originator if the cache entry gets reused, so great care needs
 to be taken to ensure this does not happen.
 o The server MAY use a secret key to encrypt the Via field, a
 timestamp and an appropriate checksum in any such message with
 the same secret key. The checksum is needed to detect whether
 successful decoding has occurred, and the timestamp is
 required to prevent possible replay attacks and to ensure that
 no two requests from the same previous hop have the same
 encrypted Via field. This is the preferred solution.
13.2 Message Integrity and Access Control: Authentication
 Protective measures need to be taken to prevent an active attacker
 from modifying and replaying SIP requests and responses. The same
 cryptographic measures that are used to ensure the authenticity of
 the SIP message also serve to authenticate the originator of the
 message. However, the "basic" and "digest" authentication mechanism
 offer authentication only, without message integrity.
 Transport-layer or network-layer authentication MAY be used for hop-
 by-hop authentication. SIP also extends the HTTP WWW-Authenticate
 (Section 6.42) and Authorization (Section 6.11) header field and
 their Proxy counterparts to include cryptographically strong
 signatures. SIP also supports the HTTP "basic" and "digest" schemes
 (see Section 14) and other HTTP authentication schemes to be defined
 that offer a rudimentary mechanism of ascertaining the identity of
 the caller.
 Since SIP requests are often sent to parties with which no
 prior communication relationship has existed, we do not
 specify authentication based on shared secrets.
 SIP requests MAY be authenticated using the Authorization header
 field to include a digital signature of certain header fields, the
 request method and version number and the payload, none of which are
 modified between client and called user agent. The Authorization
 header field is used in requests to authenticate the request
 originator end-to-end to proxies and the called user agent, and in
 responses to authenticate the called user agent or proxies returning
 their own failure codes. If required, hop-by-hop authentication can
 be provided, for example, by the IPSEC Authentication Header.
 SIP does not dictate which digital signature scheme is used for
 authentication, but does define how to provide authentication using
 PGP in Section 15. As indicated above, SIP implementations MAY also
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Internet Draft SIP January 15, 1999
 use "basic" and "digest" authentication and other authentication
 mechanisms defined for HTTP. Note that "basic" authentication has
 severe security limitations. The following does not apply to these
 schemes.
 To cryptographically sign a SIP request, the order of the SIP header
 fields is important. When an Authorization header field is present,
 it indicates that all header fields following the Authorization
 header field have been included in the signature. Therefore, hop-
 by-hop header fields which MUST or SHOULD be modified by proxies MUST
 precede the Authorization header field as they will generally be
 modified or added-to by proxy servers. Hop-by-hop header fields
 which MAY be modified by a proxy MAY appear before or after the
 Authorization header. When they appear before, they MAY be modified
 by a proxy. When they appear after, they MUST NOT be modified by a
 proxy. To sign a request, a client constructs a message from the
 request method (in upper case) followed, without LWS, by the SIP
 version number, followed, again without LWS, by the request headers
 to be signed and the message body. The message thus constructed is
 then signed.
 For example, if the SIP request is to be:
 INVITE sip:watson@boston.bell-telephone.com SIP/2.0
 Via: SIP/2.0/UDP 169.130.12.5
 Authorization: PGP version=5.0, signature=...
 From: A. Bell <sip:a.g.bell@bell-telephone.com>
 To: T. A. Watson <sip:watson@bell-telephone.com>
 Call-ID: 187602141351@worcester.bell-telephone.com
 Subject: Mr. Watson, come here.
 Content-Type: application/sdp
 Content-Length: ...
 v=0
 o=bell 53655765 2353687637 IN IP4 128.3.4.5
 c=IN IP4 135.180.144.94
 m=audio 3456 RTP/AVP 0 3 4 5
 Then the data block that is signed is:
 INVITESIP/2.0From: A. Bell <sip:a.g.bell@bell-telephone.com>
 To: T. A. Watson <sip:watson@bell-telephone.com>
 Call-ID: 187602141351@worcester.bell-telephone.com
 Subject: Mr. Watson, come here.
 Content-Type: application/sdp
 Content-Length: ...
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Internet Draft SIP January 15, 1999
 v=0
 o=bell 53655765 2353687637 IN IP4 128.3.4.5
 c=IN IP4 135.180.144.94
 m=audio 3456 RTP/AVP 0 3 4 5
 Clients wishing to authenticate requests MUST construct the portion
 of the message below the Authorization header using a canonical form.
 This allows a proxy to parse the message, take it apart, and
 reconstruct it, without causing an authentication failure due to
 extra white space, for example. Canonical form consists of the
 following rules:
 o No short form header fields
 o Header field names are capitalized as shown in this document
 o No white space between the header name and the colon
 o A single space after the colon
 o Line termination with a CRLF
 o No line folding
 o No comma separated lists of header values; each must appear as
 a separate header
 o Only a single SP between tokens, between tokens and quoted
 strings, and between quoted strings; no SP after last token or
 quoted string
 o No LWS between tokens and separators, except as described
 above for after the colon in header fields
 Note that if a message is encrypted and authenticated using a digital
 signature, when the message is generated encryption is performed
 before the digital signature is generated. On receipt, the digital
 signature is checked before decryption.
 A client MAY require that a server sign its response by including a
 Require: org.ietf.sip.signed-response request header field. The
 client indicates the desired authentication method via the WWW-
 Authenticate header.
 The correct behaviour in handling unauthenticated responses to a
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Internet Draft SIP January 15, 1999
 request that requires authenticated responses is described in section
 13.2.1.
13.2.1 Trusting responses
 There is the possibility that an eavesdropper listens to requests and
 then injects unauthenticated responses that terminate, redirect or
 otherwise interfere with a call. (Even encrypted requests contain
 enough information to fake a response.)
 Clients need to be particularly careful with 3xx redirection
 responses. Thus a client receiving, for example, a 301 (Moved
 Permanently) which was not authenticated when the public key of the
 called user agent is known to the client, and authentication was
 requested in the request SHOULD be treated as suspicious. The correct
 behaviour in such a case would be for the called-user to form a dated
 response containing the Contact field to be used, to sign it, and
 give this signed stub response to the proxy that will provide the
 redirection. Thus the response can be authenticated correctly. A
 client SHOULD NOT automatically redirect such a request to the new
 location without alerting the user to the authentication failure
 before doing so.
 Another problem might be responses such as 6xx failure responses
 which would simply terminate a search, or "4xx" and "5xx" response
 failures.
 If TCP is being used, a proxy SHOULD treat 4xx and 5xx responses as
 valid, as they will not terminate a search. However, fake 6xx
 responses from a rogue proxy terminate a search incorrectly. 6xx
 responses SHOULD be authenticated if requested by the client, and
 failure to do so SHOULD cause such a client to ignore the 6xx
 response and continue a search.
 With UDP, the same problem with 6xx responses exists, but also an
 active eavesdropper can generate 4xx and 5xx responses that might
 cause a proxy or client to believe a failure occurred when in fact it
 did not. Typically 4xx and 5xx responses will not be signed by the
 called user agent, and so there is no simple way to detect these
 rogue responses. This problem is best prevented by using hop-by-hop
 encryption of the SIP request, which removes any additional problems
 that UDP might have over TCP.
 These attacks are prevented by having the client require response
 authentication and dropping unauthenticated responses. A server user
 agent that cannot perform response authentication responds using the
 normal Require response of 420 (Bad Extension).
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13.3 Callee Privacy
 User location and SIP-initiated calls can violate a callee's privacy.
 An implementation SHOULD be able to restrict, on a per-user basis,
 what kind of location and availability information is given out to
 certain classes of callers.
13.4 Known Security Problems
 With either TCP or UDP, a denial of service attack exists by a rogue
 proxy sending 6xx responses. Although a client SHOULD choose to
 ignore such responses if it requested authentication, a proxy cannot
 do so. It is obliged to forward the 6xx response back to the client.
 The client can then ignore the response, but if it repeats the
 request it will probably reach the same rogue proxy again, and the
 process will repeat.
14 SIP Authentication using HTTP Basic and Digest Schemes
 SIP implementations MAY use HTTP's basic and digest authentication
 mechanisms to provide a rudimentary form of security. This section
 overviews usage of these mechanisms in SIP. The basic operation is
 almost completely identical to that for HTTP [36]. This section
 outlines this operation, pointing to [36] for details, and noting the
 differences when used in SIP.
14.1 Framework
 The framework for SIP authentication parallels that for HTTP [36]. In
 particular, the BNF for auth-scheme, auth-param, challenge, realm,
 realm-value, and credentials is identical. The 401 response is used
 by user agent servers in SIP to challenge the authorization of a user
 agent client. Additionally, registrars and redirect servers MAY make
 use of 401 responses for authorization, but proxies MUST NOT, and
 instead MAY use the 407 response. The requirements for inclusion of
 the Proxy-Authenticate, Proxy-Authorization, WWW-Authenticate, and
 Authorization in the various messages is identical to [36].
 Since SIP does not have the concept of a canonical root URL, the
 notion of protections spaces are interpreted differently for SIP. The
 realm is a protection domain for all SIP URIs with the same value for
 the userinfo, host and port part of the SIP Request-URI. For example:
 INVITE sip:alice.wonderland@example.com SIP/2.0
 WWW-Authenticate: Basic realm="business"
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 and
 INVITE sip:aw@example.com SIP/2.0
 WWW-Authenticate: Basic realm="business"
 define different protection realms according to this rule.
 When a UAC resubmits a request with its credentials after receiving a
 401 or 407 response, it MUST increment the CSeq header field as it
 would normally do when sending an updated request.
14.2 Basic Authentication
 The rules for basic authentication follow those defined in [36], but
 with the words "origin server" replaced with "user agent server,
 redirect server , or registrar".
 Since SIP URIs are not hierarchical, the paragraph in [36] that
 states that "all paths at or deeper than the depth of the last
 symbolic element in the path field of the Request-URI also are within
 the protection space specified by the Basic realm value of the
 current challenge" does not apply for SIP. SIP clients MAY
 preemptively send the corresponding Authorization header with
 requests for SIP URIs within the same protection realm (as defined
 above) without receipt of another challenge from the server.
14.3 Digest Authentication
 The rules for digest authentication follow those defined in [36],
 with "HTTP 1.1" replaced by "SIP/2.0" in addition to the following
 differences:
 1. The URI included in the challenge has the following BNF:
 URI = SIP-URL
 2. The BNF for digest-uri-value is:
 digest-uri-value = Request-URI ;as defined in Section
 4.3
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 3. The example procedure for choosing a nonce based on Etag
 does not work for SIP.
 4. The Authentication-Info and Proxy-Authentication-Info
 fields are not used in SIP.
 5. The text in [36] regarding cache operation does not apply
 to SIP.
 6. [36] requires that a server check that the URI in the
 request line, and the URI included in the Authorization
 header, point to the same resource. In a SIP context, these
 two URI's may actually refer to different users, due to
 forwarding at some proxy. Therefore, in SIP, a server MAY
 check that the request-uri in the Authorization header
 corresponds to a user that the server is willing to accept
 forwarded or direct calls for.
14.4 Proxy-Authentication
 The use of the Proxy-Authentication and Proxy-Authorization parallel
 that as described in [36], with one difference. Proxies MUST NOT add
 the Proxy-Authorization header. 407 responses MUST be forwarded
 upstream towards the client following the procedures for any other
 response. It is the client's responsibility to add the Proxy-
 Authorization header containing credentials for the proxy which has
 asked for authentication.
 If a proxy were to resubmit a request with a Proxy-
 Authorization header field, it would need to increment the
 CSeq in the new request. However, this would mean that the
 UAC which submitted the original request would discard a
 response from the UAS, as the CSeq value would be
 different.
 See sections 6.26 and 6.27 for additional information on usage of
 these fields as they apply to SIP.
15 SIP Security Using PGP
15.1 PGP Authentication Scheme
 The "pgp" authentication scheme is based on the model that the client
 authenticates itself with a request signed with the client's private
 key. The server can then ascertain the origin of the request if it
 has access to the public key, preferably signed by a trusted third
 party.
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15.1.1 The WWW-Authenticate Response Header
 WWW-Authenticate = "WWW-Authenticate" ":" "pgp" pgp-challenge
 pgp-challenge = * (";" pgp-params )
 pgp-params = realm | pgp-version | pgp-algorithm | nonce
 realm = "realm" "=" realm-value
 realm-value = quoted-string
 pgp-version =
 "version" "=" <"> digit *( "." digit ) *letter <">
 pgp-algorithm = "algorithm" "=" ( "md5" | "sha1" | token )
 nonce = "nonce" "=" nonce-value
 nonce-value = quoted-string
 The meanings of the values of the parameters used above are as
 follows:
 realm: A string to be displayed to users so they know which identity
 to use. This string SHOULD contain at least the name of the host
 performing the authentication and MAY additionally indicate the
 collection of users who might have access. An example might be "
 Users with call-out privileges ".
 pgp-algorithm: The value of this parameter indicates the PGP message
 integrity check (MIC) to be used to produce the signature. If
 this not present it is assumed to be "md5". The currently
 defined values are "md5" for the MD5 checksum, and "sha1" for
 the SHA.1 algorithm.
 pgp-version: The version of PGP that the client MUST use. Common
 values are "2.6.2" and "5.0". The default is 5.0.
 nonce: A server-specified data string which should be uniquely
 generated each time a 401 response is made. It is RECOMMENDED
 that this string be base64 or hexadecimal data. Specifically,
 since the string is passed in the header lines as a quoted
 string, the double-quote character is not allowed. The contents
 of the nonce are implementation dependent. The quality of the
 implementation depends on a good choice. Since the nonce is used
 only to prevent replay attacks and is signed, a time stamp in
 units convenient to the server is sufficient.
 Replay attacks within the duration of the call setup are of
 limited interest, so that timestamps with a resolution of a
 few seconds are often should be sufficient. In that case,
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 the server does not have to keep a record of the nonces.
 Example:
 WWW-Authenticate: pgp ;version="5.0"
 ;realm="Your Startrek identity, please" ;algorithm=md5
 ;nonce="913082051"
15.1.2 The Authorization Request Header
 The client is expected to retry the request, passing an Authorization
 header line, which is defined as follows.
 Authorization = "Authorization" ":" "pgp" *( ";" pgp-response )
 pgp-response = realm | pgp-version | pgp-signature
 | signed-by | nonce
 pgp-signature = "signature" "=" quoted-string
 signed-by = "signed-by" "=" <"> URI <">
 The client MUST increment the CSeq header before resubmitting the
 request. The signature MUST correspond to the From header of the
 request unless the signed-by parameter is provided.
 pgp-signature: The PGP ASCII-armored signature [33], as it appears
 between the "BEGIN PGP MESSAGE" and "END PGP MESSAGE"
 delimiters, without the version indication. The signature is
 included without any linebreaks.
 The signature is computed across the nonce (if present), request
 method, request version and header fields following the Authorization
 header and the message body, in the same order as they appear in the
 message. The request method and version are prepended to the header
 fields without any white space. The signature is computed across the
 headers as sent, and the terminating CRLF. The CRLF following the
 Authorization header is NOT included in the signature.
 A server MAY be configured not to generate nonces only if replay
 attacks are not a concern.
 Not generating nonces avoids the additional set of request,
 401 response and possibly ACK messages and reduces delay by
 one round-trip time.
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 Using the ASCII-armored version is about 25% less space-
 efficient than including the binary signature, but it is
 significantly easier for the receiver to piece together.
 Versions of the PGP program always include the full
 (compressed) signed text in their output unless ASCII-
 armored mode ( -sta ) is specified. Typical signatures are
 about 200 bytes long. -- The PGP signature mechanism allows
 the client to simply pass the request to an external PGP
 program. This relies on the requirement that proxy servers
 are not allowed to reorder or change header fields.
 realm: The realm is copied from the corresponding WWW-Authenticate
 header field parameter.
 signed-by: If and only if the request was not signed by the entity
 listed in the From header, the signed-by header indicates the
 name of the signing entity, expressed as a URI.
 Receivers of signed SIP messages SHOULD discard any end-to-end header
 fields above the Authorization header, as they may have been
 maliciously added en route by a proxy.
 Example:
 Authorization: pgp version="5.0"
 ;realm="Your Startrek identity, please"
 ;nonce="913082051"
 ;signature="iQB1AwUBNNJiUaYBnHmiiQh1AQFYsgL/Wt3dk6TWK81/b0gcNDf
 VAUGU4rhEBW972IPxFSOZ94L1qhCLInTPaqhHFw1cb3lB01rA0RhpV4t5yCdUt
 SRYBSkOK29o5e1KlFeW23EzYPVUm2TlDAhbcjbMdfC+KLFX
 =aIrx"
15.2 PGP Encryption Scheme
 The PGP encryption scheme uses the following syntax:
 Encryption = "Encryption" ":" "pgp" pgp-eparams
 pgp-eparams = 1# ( pgp-version | pgp-encoding )
 pgp-encoding = "encoding" "=" "ascii" | token
 encoding: Describes the encoding or "armor" used by PGP. The value
 "ascii" refers to the standard PGP ASCII armor, without the
 lines containing "BEGIN PGP MESSAGE" and "END PGP MESSAGE" and
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 without the version identifier. By default, the encrypted part
 is included as binary.
 Example:
 Encryption: pgp version="2.6.2", encoding="ascii"
15.3 Response-Key Header Field for PGP
 Response-Key = "Response-Key" ":" "pgp" pgp-eparams
 pgp-eparams = 1# ( pgp-version | pgp-encoding | pgp-key)
 pgp-key = "key" "=" quoted-string
 If ASCII encoding has been requested via the encoding parameter, the
 key parameter contains the user's public key as extracted from the
 pgp key ring with the "pgp -kxa user ".
 Example:
 Response-Key: pgp version="2.6.2", encoding="ascii",
 key="mQBtAzNWHNYAAAEDAL7QvAdK2utY05wuUG+ItYK5tCF8HNJM60sU4rLaV+eUnkMk
 mOmJWtc2wXcZx1XaXb2lkydTQOesrUR75IwNXBuZXPEIMThEa5WLsT7VLme7njnx
 sE86SgWmAZx5ookIdQAFEbQxSGVubmluZyBTY2h1bHpyaW5uZSA8c2NodWx6cmlu
 bmVAY3MuY29sdW1iaWEuZWR1Pg==
 =+y19"
16 Examples
 In the following examples, we often omit the message body and the
 corresponding Content-Length and Content-Type headers for brevity.
16.1 Registration
 A user at host saturn.bell-tel.com registers on start-up, via
 multicast, with the local SIP server named bell-tel.com. In the
 example, the user agent on saturn expects to receive SIP requests on
 UDP port 3890.
 C->S: REGISTER sip:bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP saturn.bell-tel.com
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 From: sip:watson@bell-tel.com
 To: sip:watson@bell-tel.com
 Call-ID: 70710@saturn.bell-tel.com
 CSeq: 1 REGISTER
 Contact: <sip:watson@saturn.bell-tel.com:3890;transport=udp>
 Expires: 7200
 The registration expires after two hours. Any future invitations for
 watson@bell-tel.com arriving at sip.bell-tel.com will now be
 redirected to watson@saturn.bell-tel.com, UDP port 3890.
 If Watson wants to be reached elsewhere, say, an on-line service he
 uses while traveling, he updates his reservation after first
 cancelling any existing locations:
 C->S: REGISTER sip:bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP saturn.bell-tel.com
 From: sip:watson@bell-tel.com
 To: sip:watson@bell-tel.com
 Call-ID: 70710@saturn.bell-tel.com
 CSeq: 2 REGISTER
 Contact: *
 Expires: 0
 C->S: REGISTER sip:bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP saturn.bell-tel.com
 From: sip:watson@bell-tel.com
 To: sip:watson@bell-tel.com
 Call-ID: 70710@saturn.bell-tel.com
 CSeq: 3 REGISTER
 Contact: sip:tawatson@example.com
 Now, the server will forward any request for Watson to the server at
 example.com, using the Request-URI tawatson@example.com. For the
 server at example.com to reach Watson, he will need to send a
 REGISTER there, or inform the server of his current location through
 some other means.
 It is possible to use third-party registration. Here, the secretary
 jon.diligent registers his boss, T. Watson:
 C->S: REGISTER sip:bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP pluto.bell-tel.com
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 From: sip:jon.diligent@bell-tel.com
 To: sip:watson@bell-tel.com
 Call-ID: 17320@pluto.bell-tel.com
 CSeq: 1 REGISTER
 Contact: sip:tawatson@example.com
 The request could be sent to either the registrar at bell-tel.com or
 the server at example.com. In the latter case, the server at
 example.com would proxy the request to the address indicated in the
 Request-URI. Then, Max-Forwards header could be used to restrict the
 registration to that server.
16.2 Invitation to a Multicast Conference
 The first example invites schooler@vlsi.cs.caltech.edu to a multicast
 session. All examples use the Session Description Protocol (SDP) (RFC
 2327 [6]) as the session description format.
16.2.1 Request
 C->S: INVITE sip:schooler@cs.caltech.edu SIP/2.0
 Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348
 ;maddr=239.128.16.254;ttl=16
 Via: SIP/2.0/UDP north.east.isi.edu
 From: Mark Handley <sip:mjh@isi.edu>
 To: Eve Schooler <sip:schooler@caltech.edu>
 Call-ID: 2963313058@north.east.isi.edu
 CSeq: 1 INVITE
 Subject: SIP will be discussed, too
 Content-Type: application/sdp
 Content-Length: 187
 v=0
 o=user1 53655765 2353687637 IN IP4 128.3.4.5
 s=Mbone Audio
 i=Discussion of Mbone Engineering Issues
 e=mbone@somewhere.com
 c=IN IP4 224.2.0.1/127
 t=0 0
 m=audio 3456 RTP/AVP 0
 The From request header above states that the request was initiated
 by mjh@isi.edu and addressed to schooler@caltech.edu (From header
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 fields). The Via fields list the hosts along the path from invitation
 initiator (the last element of the list) towards the callee. In the
 example above, the message was last multicast to the administratively
 scoped group 239.128.16.254 with a ttl of 16 from the host
 csvax.cs.caltech.edu. The second Via header field indicates that it
 was originally sent from the host north.east.isi.edu. The Request-URI
 indicates that the request is currently being being addressed to
 schooler@cs.caltech.edu, the local address that csvax looked up for
 the callee.
 In this case, the session description is using the Session
 Description Protocol (SDP), as stated in the Content-Type header.
 The header is terminated by an empty line and is followed by a
 message body containing the session description.
16.2.2 Response
 The called user agent, directly or indirectly through proxy servers,
 indicates that it is alerting ("ringing") the called party:
 S->C: SIP/2.0 180 Ringing
 Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348
 ;maddr=239.128.16.254;ttl=16
 Via: SIP/2.0/UDP north.east.isi.edu
 From: Mark Handley <sip:mjh@isi.edu>
 To: Eve Schooler <sip:schooler@caltech.edu> ;tag=9883472
 Call-ID: 2963313058@north.east.isi.edu
 CSeq: 1 INVITE
 A sample response to the invitation is given below. The first line of
 the response states the SIP version number, that it is a 200 (OK)
 response, which means the request was successful. The Via headers are
 taken from the request, and entries are removed hop by hop as the
 response retraces the path of the request. A new authentication field
 MAY be added by the invited user's agent if required. The Call-ID is
 taken directly from the original request, along with the remaining
 fields of the request message. The original sense of From field is
 preserved (i.e., it is the session initiator).
 In addition, the Contact header gives details of the host where the
 user was located, or alternatively the relevant proxy contact point
 which should be reachable from the caller's host.
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 S->C: SIP/2.0 200 OK
 Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348
 ;maddr=239.128.16.254;ttl=16
 Via: SIP/2.0/UDP north.east.isi.edu
 From: Mark Handley <sip:mjh@isi.edu>
 To: Eve Schooler <sip:schooler@caltech.edu> ;tag=9883472
 Call-ID: 2963313058@north.east.isi.edu
 CSeq: 1 INVITE
 Contact: sip:es@jove.cs.caltech.edu
 The caller confirms the invitation by sending an ACK request to the
 location named in the Contact header:
 C->S: ACK sip:es@jove.cs.caltech.edu SIP/2.0
 Via: SIP/2.0/UDP north.east.isi.edu
 From: Mark Handley <sip:mjh@isi.edu>
 To: Eve Schooler <sip:schooler@caltech.edu> ;tag=9883472
 Call-ID: 2963313058@north.east.isi.edu
 CSeq: 1 ACK
16.3 Two-party Call
 For two-party Internet phone calls, the response must contain a
 description of where to send the data. In the example below, Bell
 calls Watson. Bell indicates that he can receive RTP audio codings 0
 (PCMU), 3 (GSM), 4 (G.723) and 5 (DVI4).
 C->S: INVITE sip:watson@boston.bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP kton.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:watson@bell-tel.com>
 Call-ID: 3298420296@kton.bell-tel.com
 CSeq: 1 INVITE
 Subject: Mr. Watson, come here.
 Content-Type: application/sdp
 Content-Length: ...
 v=0
 o=bell 53655765 2353687637 IN IP4 128.3.4.5
 s=Mr. Watson, come here.
 c=IN IP4 kton.bell-tel.com
 m=audio 3456 RTP/AVP 0 3 4 5
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 S->C: SIP/2.0 100 Trying
 Via: SIP/2.0/UDP kton.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
 Call-ID: 3298420296@kton.bell-tel.com
 CSeq: 1 INVITE
 Content-Length: 0
 S->C: SIP/2.0 180 Ringing
 Via: SIP/2.0/UDP kton.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
 Call-ID: 3298420296@kton.bell-tel.com
 CSeq: 1 INVITE
 Content-Length: 0
 S->C: SIP/2.0 182 Queued, 2 callers ahead
 Via: SIP/2.0/UDP kton.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
 Call-ID: 3298420296@kton.bell-tel.com
 CSeq: 1 INVITE
 Content-Length: 0
 S->C: SIP/2.0 182 Queued, 1 caller ahead
 Via: SIP/2.0/UDP kton.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
 Call-ID: 3298420296@kton.bell-tel.com
 CSeq: 1 INVITE
 Content-Length: 0
 S->C: SIP/2.0 200 OK
 Via: SIP/2.0/UDP kton.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: <sip:watson@bell-tel.com> ;tag=37462311
 Call-ID: 3298420296@kton.bell-tel.com
 CSeq: 1 INVITE
 Contact: sip:watson@boston.bell-tel.com
 Content-Type: application/sdp
 Content-Length: ...
 v=0
 o=watson 4858949 4858949 IN IP4 192.1.2.3
 s=I'm on my way
 c=IN IP4 boston.bell-tel.com
 m=audio 5004 RTP/AVP 0 3
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 The example illustrates the use of informational status responses.
 Here, the reception of the call is confirmed immediately (100), then,
 possibly after some database mapping delay, the call rings (180) and
 is then queued, with periodic status updates.
 Watson can only receive PCMU and GSM. Note that Watson's list of
 codecs may or may not be a subset of the one offered by Bell, as each
 party indicates the data types it is willing to receive. Watson will
 send audio data to port 3456 at c.bell-tel.com, Bell will send to
 port 5004 at boston.bell-tel.com.
 By default, the media session is one RTP session. Watson will receive
 RTCP packets on port 5005, while Bell will receive them on port 3457.
 Since the two sides have agreed on the set of media, Bell confirms
 the call without enclosing another session description:
 C->S: ACK sip:watson@boston.bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP kton.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
 Call-ID: 3298420296@kton.bell-tel.com
 CSeq: 1 ACK
16.4 Terminating a Call
 To terminate a call, caller or callee can send a BYE request:
 C->S: BYE sip:watson@boston.bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP kton.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. A. Watson <sip:watson@bell-tel.com> ;tag=37462311
 Call-ID: 3298420296@kton.bell-tel.com
 CSeq: 2 BYE
 If the callee wants to abort the call, it simply reverses the To and
 From fields. Note that it is unlikely that a BYE from the callee will
 traverse the same proxies as the original INVITE.
16.5 Forking Proxy
 In this example, Bell (a.g.bell@bell-tel.com) (C), currently seated
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 at host c.bell-tel.com wants to call Watson (t.watson@ieee.org). At
 the time of the call, Watson is logged in at two workstations,
 t.watson@x.bell-tel.com (X) and watson@y.bell-tel.com (Y), and has
 registered with the IEEE proxy server (P) called sip.ieee.org. The
 IEEE server also has a registration for the home machine of Watson,
 at watson@h.bell-tel.com (H), as well as a permanent registration at
 watson@acm.org (A). For brevity, the examples omit the session
 description and Via header fields.
 Bell's user agent sends the invitation to the SIP server for the
 ieee.org domain:
 C->P: INVITE sip:t.watson@ieee.org SIP/2.0
 Via: SIP/2.0/UDP c.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 INVITE
 The SIP server at ieee.org tries the four addresses in parallel. It
 sends the following message to the home machine:
 P->H: INVITE sip:watson@h.bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP sip.ieee.org ;branch=1
 Via: SIP/2.0/UDP c.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 INVITE
 This request immediately yields a 404 (Not Found) response, since
 Watson is not currently logged in at home:
 H->P: SIP/2.0 404 Not Found
 Via: SIP/2.0/UDP sip.ieee.org ;branch=1
 Via: SIP/2.0/UDP c.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>;tag=87454273
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 INVITE
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 The proxy ACKs the response so that host H can stop retransmitting
 it:
 P->H: ACK sip:watson@h.bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP sip.ieee.org ;branch=1
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>;tag=87454273
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 ACK
 Also, P attempts to reach Watson through the ACM server:
 P->A: INVITE sip:watson@acm.org SIP/2.0
 Via: SIP/2.0/UDP sip.ieee.org ;branch=2
 Via: SIP/2.0/UDP c.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 INVITE
 In parallel, the next attempt proceeds, with an INVITE to X and Y:
 P->X: INVITE sip:t.watson@x.bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP sip.ieee.org ;branch=3
 Via: SIP/2.0/UDP c.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 INVITE
 P->Y: INVITE sip:watson@y.bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP sip.ieee.org ;branch=4
 Via: SIP/2.0/UDP c.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 INVITE
 As it happens, both Watson at X and a colleague in the other lab at
 host Y hear the phones ringing and pick up. Both X and Y return 200s
 via the proxy to Bell.
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 X->P: SIP/2.0 200 OK
 Via: SIP/2.0/UDP sip.ieee.org ;branch=3
 Via: SIP/2.0/UDP c.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org> ;tag=192137601
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 INVITE
 Contact: sip:t.watson@x.bell-tel.com
 Y->P: SIP/2.0 200 OK
 Via: SIP/2.0/UDP sip.ieee.org ;branch=4
 Via: SIP/2.0/UDP c.bell-tel.com
 Contact: sip:t.watson@y.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org> ;tag=35253448
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 INVITE
 Both responses are forwarded to Bell, using the Via information. At
 this point, the ACM server is still searching its database. P can now
 cancel this attempt:
 P->A: CANCEL sip:watson@acm.org SIP/2.0
 Via: SIP/2.0/UDP sip.ieee.org ;branch=2
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 CANCEL
 The ACM server gladly stops its neural-network database search and
 responds with a 200. The 200 will not travel any further, since P is
 the last Via stop.
 A->P: SIP/2.0 200 OK
 Via: SIP/2.0/UDP sip.ieee.org ;branch=2
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 CANCEL
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 Bell gets the two 200 responses from X and Y in short order. Bell's
 reaction now depends on his software. He can either send an ACK to
 both if human intelligence is needed to determine who he wants to
 talk to or he can automatically reject one of the two calls. Here, he
 acknowledges both, separately and directly to the final destination:
 C->X: ACK sip:t.watson@x.bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP c.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>;tag=192137601
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 ACK
 C->Y: ACK sip:watson@y.bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP c.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>;tag=35253448
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 ACK
 After a brief discussion between Bell with X and Y, it becomes clear
 that Watson is at X. (Note that this is not a three-way call; only
 Bell can talk to X and Y, but X and Y cannot talk to each other.)
 Thus, Bell sends a BYE to Y, which is replied to:
 C->Y: BYE sip:watson@y.bell-tel.com SIP/2.0
 Via: SIP/2.0/UDP c.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>;tag=35253448
 Call-ID: 31415@c.bell-tel.com
 CSeq: 2 BYE
 Y->C: SIP/2.0 200 OK
 Via: SIP/2.0/UDP c.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>;tag=35253448
 Call-ID: 31415@c.bell-tel.com
 CSeq: 2 BYE
16.6 Redirects
 Replies with status codes 301 (Moved Permanently) or 302 (Moved
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 Temporarily) specify another location using the Contact field.
 Continuing our earlier example, the server P at ieee.org decides to
 redirect rather than proxy the request:
 P->C: SIP/2.0 302 Moved temporarily
 Via: SIP/2.0/UDP c.bell-tel.com
 From: A. Bell <sip:a.g.bell@bell-tel.com>
 To: T. Watson <sip:t.watson@ieee.org>;tag=72538263
 Call-ID: 31415@c.bell-tel.com
 CSeq: 1 INVITE
 Contact: sip:watson@h.bell-tel.com,
 sip:watson@acm.org, sip:t.watson@x.bell-tel.com,
 sip:watson@y.bell-tel.com
 CSeq: 1 INVITE
 As another example, assume Alice (A) wants to delegate her calls to
 Bob (B) while she is on vacation until July 29th, 1998. Any calls
 meant for her will reach Bob with Alice's To field, indicating to him
 what role he is to play. Charlie (C) calls Alice (A), whose server
 returns:
 A->C: SIP/2.0 302 Moved temporarily
 From: Charlie <sip:charlie@caller.com>
 To: Alice <sip:alice@anywhere.com> ;tag=2332462
 Call-ID: 27182@caller.com
 Contact: sip:bob@anywhere.com
 Expires: 1998年7月29日 9:00:00 GMT
 CSeq: 1 INVITE
 Charlie then sends the following request to the SIP server of the
 anywhere.com domain. Note that the server at anywhere.com forwards
 the request to Bob based on the Request-URI.
 C->B: INVITE sip:bob@anywhere.com SIP/2.0
 From: sip:charlie@caller.com
 To: sip:alice@anywhere.com
 Call-ID: 27182@caller.com
 CSeq: 2 INVITE
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 In the third redirection example, we assume that all outgoing
 requests are directed through a local firewall F at caller.com, with
 Charlie again inviting Alice:
 C->F: INVITE sip:alice@anywhere.com SIP/2.0
 From: sip:charlie@caller.com
 To: Alice <sip:alice@anywhere.com>
 Call-ID: 27182@caller.com
 CSeq: 1 INVITE
 The local firewall at caller.com happens to be overloaded and thus
 redirects the call from Charlie to a secondary server S:
 F->C: SIP/2.0 302 Moved temporarily
 From: sip:charlie@caller.com
 To: Alice <sip:alice@anywhere.com>
 Call-ID: 27182@caller.com
 CSeq: 1 INVITE
 Contact: <sip:alice@anywhere.com:5080;maddr=spare.caller.com>
 Based on this response, Charlie directs the same invitation to the
 secondary server spare.caller.com at port 5080, but maintains the
 same Request-URI as before:
 C->S: INVITE sip:alice@anywhere.com SIP/2.0
 From: sip:charlie@caller.com
 To: Alice <sip:alice@anywhere.com>
 Call-ID: 27182@caller.com
 CSeq: 2 INVITE
16.7 Negotiation
 An example of a 606 (Not Acceptable) response is:
 S->C: SIP/2.0 606 Not Acceptable
 From: sip:mjh@isi.edu
 To: <sip:schooler@cs.caltech.edu> ;tag=7434264
 Call-ID: 14142@north.east.isi.edu
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 CSeq: 1 INVITE
 Contact: sip:mjh@north.east.isi.edu
 Warning: 370 "Insufficient bandwidth (only have ISDN)",
 305 "Incompatible media format",
 330 "Multicast not available"
 Content-Type: application/sdp
 Content-Length: 50
 v=0
 s=Let's talk
 b=CT:128
 c=IN IP4 north.east.isi.edu
 m=audio 3456 RTP/AVP 5 0 7
 m=video 2232 RTP/AVP 31
 In this example, the original request specified a bandwidth that was
 higher than the access link could support, requested multicast, and
 requested a set of media encodings. The response states that only 128
 kb/s is available and that (only) DVI, PCM or LPC audio could be
 supported in order of preference.
 The response also states that multicast is not available. In such a
 case, it might be appropriate to set up a transcoding gateway and
 re-invite the user.
16.8 OPTIONS Request
 A caller Alice can use an OPTIONS request to find out the
 capabilities of a potential callee Bob, without "ringing" the
 designated address. Bob returns a description indicating that he is
 capable of receiving audio encodings PCM Ulaw (payload type 0), 1016
 (payload type 1), GSM (payload type 3), and SX7300/8000 (dynamic
 payload type 99), and video encodings H.261 (payload type 31) and
 H.263 (payload type 34).
 C->S: OPTIONS sip:bob@example.com SIP/2.0
 From: Alice <sip:alice@anywhere.org>
 To: Bob <sip:bob@example.com>
 Call-ID: 6378@host.anywhere.org
 CSeq: 1 OPTIONS
 Accept: application/sdp
 S->C: SIP/2.0 200 OK
 From: Alice <sip:alice@anywhere.org>
 To: Bob <sip:bob@example.com> ;tag=376364382
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 Call-ID: 6378@host.anywhere.org
 Content-Length: 81
 Content-Type: application/sdp
 v=0
 m=audio 0 RTP/AVP 0 1 3 99
 m=video 0 RTP/AVP 31 34
 a=rtpmap:99 SX7300/8000
A Minimal Implementation
A.1 Client
 All clients MUST be able to generate the INVITE and ACK requests.
 Clients MUST generate and parse the Call-ID, Content-Length,
 Content-Type, CSeq, From and To headers. Clients MUST also parse the
 Require header. A minimal implementation MUST understand SDP (RFC
 2327, [6]). It MUST be able to recognize the status code classes 1
 through 6 and act accordingly.
 The following capability sets build on top of the minimal
 implementation described in the previous paragraph. In general, each
 capability listed below builds on the ones above it:
 Basic: A basic implementation adds support for the BYE method to
 allow the interruption of a pending call attempt. It includes a
 User-Agent header in its requests and indicate its preferred
 language in the Accept-Language header.
 Redirection: To support call forwarding, a client needs to be able to
 understand the Contact header, but only the SIP-URL part, not
 the parameters.
 Firewall-friendly: A firewall-friendly client understands the Route
 and Record-Route header fields and can be configured to use a
 local proxy for all outgoing requests.
 Negotiation: A client MUST be able to request the OPTIONS method and
 understand the 380 (Alternative Service) status and the Contact
 parameters to participate in terminal and media negotiation. It
 SHOULD be able to parse the Warning response header to provide
 useful feedback to the caller.
 Authentication: If a client wishes to invite callees that require
 caller authentication, it MUST be able to recognize the 401
 (Unauthorized) status code, MUST be able to generate the
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 Authorization request header and MUST understand the WWW-
 Authenticate response header.
 If a client wishes to use proxies that require caller authentication,
 it MUST be able to recognize the 407 (Proxy Authentication Required)
 status code, MUST be able to generate the Proxy-Authorization request
 header and understand the Proxy-Authenticate response header.
A.2 Server
 A minimally compliant server implementation MUST understand the
 INVITE, ACK, OPTIONS and BYE requests. A proxy server MUST also
 understand CANCEL. It MUST parse and generate, as appropriate, the
 Call-ID, Content-Length, Content-Type, CSeq, Expires, From, Max-
 Forwards, Require, To and Via headers. It MUST echo the CSeq and
 Timestamp headers in the response. It SHOULD include the Server
 header in its responses.
A.3 Header Processing
 Table 6 lists the headers that different implementations support. UAC
 refers to a user-agent client (calling user agent), UAS to a user-
 agent server (called user-agent).
 The fields in the table have the following meaning. Type is as in
 Table 4 and 5. "-" indicates the field is not meaningful to this
 system (although it might be generated by it). "m" indicates the
 field MUST be understood. "b" indicates the field SHOULD be
 understood by a Basic implementation. "r" indicates the field SHOULD
 be understood if the system claims to understand redirection. "a"
 indicates the field SHOULD be understood if the system claims to
 support authentication. "e" indicates the field SHOULD be understood
 if the system claims to support encryption. "o" indicates support of
 the field is purely optional. Headers whose support is optional for
 all implementations are not shown.
B Usage of the Session Description Protocol (SDP)
 This section describes the use of the Session Description Protocol
 (SDP) (RFC 2327 [6]).
B.1 Configuring Media Streams
 The caller and callee align their media descriptions so that the nth
 media stream ("m=" line) in the caller's session description
 corresponds to the nth media stream in the callee's description.
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 type UAC proxy UAS registrar
 _____________________________________________________
 Accept R - o m m
 Accept-Encoding R - - m m
 Accept-Language R - b b b
 Allow 405 o - - -
 Authorization R a o a a
 Call-ID g m m m m
 Content-Encoding g m - m m
 Content-Length g m m m m
 Content-Type g m - m m
 CSeq g m m m m
 Encryption g e - e e
 Expires g - o o m
 From g m o m m
 Hide R - m - -
 Contact R - - - m
 Contact r r r - -
 Max-Forwards R - b - -
 Proxy-Authenticate 407 a - - -
 Proxy-Authorization R - a - -
 Proxy-Require R - m - -
 Require R m - m m
 Response-Key R - - e e
 Route R - m - -
 Timestamp g o o m m
 To g m m m m
 Unsupported r b b - -
 User-Agent g b - b -
 Via g m m m m
 WWW-Authenticate 401 a - - -
 Table 6: Header Field Processing Requirements
 All media descriptions SHOULD contain "a=rtpmap" mappings from RTP
 payload types to encodings.
 This allows easier migration away from static payload
 types.
 If the callee wants to neither send nor receive a stream offered by
 the caller, the callee sets the port number of that stream to zero in
 its media description.
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 There currently is no other way than port zero for the
 callee to refuse a bidirectional stream offered by the
 caller. Both caller and callee need to be aware what media
 tools are to be started.
 For example, assume that the caller Alice has included the following
 description in her INVITE request. It includes an audio stream and
 two bidirectional video streams, using H.261 (payload type 31) and
 MPEG (payload type 32).
 v=0
 o=alice 2890844526 2890844526 IN IP4 host.anywhere.com
 c=IN IP4 host.anywhere.com
 m=audio 49170 RTP/AVP 0
 a=rtpmap:0 PCMU/8000
 m=video 51372 RTP/AVP 31
 a=rtpmap:31 H261/90000
 m=video 53000 RTP/AVP 32
 a=rtpmap:32 MPV/90000
 The callee, Bob, does not want to receive or send the first video
 stream, so it returns the media description below:
 v=0
 o=bob 2890844730 2890844730 IN IP4 host.example.com
 c=IN IP4 host.example.com
 m=audio 47920 RTP/AVP 0 1
 a=rtpmap:0 PCMU/8000
 a=rtpmap:1 1016/8000
 m=video 0 RTP/AVP 31
 m=video 53000 RTP/AVP 32
 a=rtpmap:32 MPV/90000
B.2 Setting SDP Values for Unicast
 If a session description from a caller contains a media stream which
 is listed as send (receive) only, it means that the caller is only
 willing to send (receive) this stream, not receive (send). The same
 is true for the callee.
 For receive-only and send-or-receive streams, the port number and
 address in the session description indicate where the media stream
 should be sent to by the recipient of the session description, either
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 caller or callee. For send-only streams, the address and port number
 have no significance and SHOULD be set to zero.
 The list of payload types for each media stream conveys two pieces of
 information, namely the set of codecs that the caller or callee is
 capable of sending or receiving, and the RTP payload type numbers
 used to identify those codecs. For receive-only or send-and-receive
 media streams, a caller SHOULD list all of the codecs it is capable
 of supporting in the session description in an INVITE or ACK. For
 send-only streams, the caller SHOULD indicate only those it wishes to
 send for this session. For receive-only streams, the payload type
 numbers indicate the value of the payload type field in RTP packets
 the caller is expecting to receive for that codec type. For send-only
 streams, the payload type numbers indicate the value of the payload
 type field in RTP packets the caller is planning to send for that
 codec type. For send-and-receive streams, the payload type numbers
 indicate the value of the payload type field the caller expects to
 both send and receive.
 If a media stream is listed as receive-only by the caller, the callee
 lists, in the response, those codecs it intends to use from among the
 ones listed in the request. If a media stream is listed as send-only
 by the caller, the callee lists, in the response, those codecs it is
 willing to receive among the ones listed in the the request. If the
 media stream is listed as both send and receive, the callee lists
 those codecs it is capable of sending or receiving among the ones
 listed by the caller in the INVITE. The actual payload type numbers
 in the callee's session description corresponding to a particular
 codec MUST be the same as the caller's session description.
 If caller and callee have no media formats in common for a particular
 stream, the callee MUST return a session description containing the
 particular "m=" line, but with the port number set to zero, and no
 payload types listed.
 If there are no media formats in common for all streams, the callee
 SHOULD return a 400 response, with a 304 Warning header field.
B.3 Multicast Operation
 The interpretation of send-only and receive-only for multicast media
 sessions differs from that for unicast sessions. For multicast,
 send-only means that the recipient of the session description (caller
 or callee) SHOULD only send media streams to the address and port
 indicated. Receive-only means that the recipient of the session
 description SHOULD only receive media on the address and port
 indicated.
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 For multicast, receive and send multicast addresses are the same and
 all parties use the same port numbers to receive media data. If the
 session description provided by the caller is acceptable to the
 callee, the callee can choose not to include a session description or
 MAY echo the description in the response.
 A callee MAY, in the response, return a session description with some
 of the payload types removed, or port numbers set to zero (but no
 other value). This indicates to the caller that the callee does not
 support the given stream or media types which were removed. A callee
 MUST NOT change whether a given stream is send-only, receive-only, or
 send-and-receive.
 If a callee does not support multicast at all, it SHOULD return a 400
 status response and include a 330 Warning.
B.4 Delayed Media Streams
 In some cases, a caller may not know the set of media formats which
 it can support at the time it would like to issue an invitation. This
 is the case when the caller is actually a gateway to another protocol
 which performs media format negotiation after call setup. When this
 occurs, a caller MAY issue an INVITE with a session description that
 contains no media lines. The callee SHOULD interpret this to mean
 that the caller wishes to participate in a multimedia session
 described by the session description, but that the media streams are
 not yet known. The callee SHOULD return a session description
 indicating the streams and media formats it is willing to support,
 however. The caller MAY update the session description either in the
 ACK request or in a re-INVITE at a later time, once the streams are
 known.
B.5 Putting Media Streams on Hold
 If a party in a call wants to put the other party "on hold", i.e.,
 request that it temporarily stops sending one or more media streams,
 a party re-invites the other by sending an INVITE request with a
 modified session description. The session description is the same as
 in the original invitation (or response), but the "c" destination
 addresses for the media streams to be put on hold are set to zero
 (0.0.0.0).
B.6 Subject and SDP "s=" Line
 The SDP "s=" line and the SIP Subject header field have different
 meanings when inviting to a multicast session. The session
 description line describes the subject of the multicast session,
 while the SIP Subject header field describes the reason for the
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 invitation. The example in Section 16.2 illustrates this point. For
 invitations to two-party sessions, the SDP "s=" line MAY be left
 empty.
B.7 The SDP "o=" Line
 The "o=" line is not strictly necessary for two-party sessions, but
 MUST be present to allow re-use of SDP-based tools.
C Summary of Augmented BNF
 All of the mechanisms specified in this document are described in
 both prose and an augmented Backus-Naur Form (BNF) similar to that
 used by RFC 822 [9]. Implementors will need to be familiar with the
 notation in order to understand this specification. The augmented BNF
 includes the following constructs:
 name = definition
 The name of a rule is simply the name itself (without any enclosing
 "<" and ">") and is separated from its definition by the equal "="
 character. White space is only significant in that indentation of
 continuation lines is used to indicate a rule definition that spans
 more than one line. Certain basic rules are in uppercase, such as SP,
 LWS, HT, CRLF, DIGIT, ALPHA, etc. Angle brackets are used within
 definitions whenever their presence will facilitate discerning the
 use of rule names.
 "literal"
 Quotation marks surround literal text. Unless stated otherwise, the
 text is case-insensitive.
 rule1 | rule2
 Elements separated by a bar ("|") are alternatives, e.g., "yes | no"
 will accept yes or no.
 (rule1 rule2)
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 Elements enclosed in parentheses are treated as a single element.
 Thus, "(elem (foo | bar) elem)" allows the token sequences "elem foo
 elem" and "elem bar elem".
 *rule
 The character "*" preceding an element indicates repetition. The full
 form is "<n>*<m>element" indicating at least <n> and at most <m>
 occurrences of element. Default values are 0 and infinity so that
 "*(element)" allows any number, including zero; "1*element" requires
 at least one; and "1*2element" allows one or two.
 [rule]
 Square brackets enclose optional elements; "[foo bar]" is equivalent
 to "*1(foo bar)".
 N rule
 Specific repetition: "<n>(element)" is equivalent to
 "<n>*<n>(element)"; that is, exactly <n> occurrences of (element).
 Thus 2DIGIT is a 2-digit number, and 3ALPHA is a string of three
 alphabetic characters.
 #rule
 A construct "#" is defined, similar to "*", for defining lists of
 elements. The full form is "<n>#<m> element" indicating at least <n>
 and at most <m> elements, each separated by one or more commas (",")
 and OPTIONAL linear white space (LWS). This makes the usual form of
 lists very easy; a rule such as
 ( *LWS element *( *LWS "," *LWS element ))
 can be shown as 1# element. Wherever this construct is used, null
 elements are allowed, but do not contribute to the count of elements
 present. That is, "(element), , (element)" is permitted, but counts
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 as only two elements. Therefore, where at least one element is
 required, at least one non-null element MUST be present. Default
 values are 0 and infinity so that "#element" allows any number,
 including zero; "1#element" requires at least one; and "1#2element"
 allows one or two.
 ; comment
 A semi-colon, set off some distance to the right of rule text, starts
 a comment that continues to the end of line. This is a simple way of
 including useful notes in parallel with the specifications.
 implied *LWS
 The grammar described by this specification is word-based. Except
 where noted otherwise, linear white space (LWS) can be included
 between any two adjacent words (token or quoted-string), and between
 adjacent tokens and separators, without changing the interpretation
 of a field. At least one delimiter (LWS and/or separators) MUST exist
 between any two tokens (for the definition of "token" below), since
 they would otherwise be interpreted as a single token.
C.1 Basic Rules
 The following rules are used throughout this specification to
 describe basic parsing constructs. The US-ASCII coded character set
 is defined by ANSI X3.4-1986.
 OCTET = <any 8-bit sequence of data>
 CHAR = <any US-ASCII character (octets 0 - 127)>
 upalpha = "A" | "B" | "C" | "D" | "E" | "F" | "G" | "H" | "I" |
 "J" | "K" | "L" | "M" | "N" | "O" | "P" | "Q" | "R" |
 "S" | "T" | "U" | "V" | "W" | "X" | "Y" | "Z"
 lowalpha = "a" | "b" | "c" | "d" | "e" | "f" | "g" | "h" | "i" |
 "j" | "k" | "l" | "m" | "n" | "o" | "p" | "q" | "r" |
 "s" | "t" | "u" | "v" | "w" | "x" | "y" | "z"
 alpha = lowalpha | upalpha
 digit = "0" | "1" | "2" | "3" | "4" | "5" | "6" | "7" |
 "8" | "9"
 alphanum = alpha | digit
 CTL = <any US-ASCII control character
 (octets 0 -- 31) and DEL (127)>
 CR = %d13 ; US-ASCII CR, carriage return character
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Internet Draft SIP January 15, 1999
 LF = %d10 ; US-ASCII LF, line feed character
 SP = %d32 ; US-ASCII SP, space character
 HT = %d09 ; US-ASCII HT, horizontal tab character
 CRLF = CR LF ; typically the end of a line
 The following are defined in RFC 2396 [12] for the SIP URI:
 unreserved = alphanum | mark
 mark = "-" | "_" | "." | "!" | "~" | "*" | "'"
 | "(" | ")"
 escaped = "%" hex hex
 SIP header field values can be folded onto multiple lines if the
 continuation line begins with a space or horizontal tab. All linear
 white space, including folding, has the same semantics as SP. A
 recipient MAY replace any linear white space with a single SP before
 interpreting the field value or forwarding the message downstream.
 LWS = [CRLF] 1*( SP | HT ) ; linear whitespace
 The TEXT-UTF8 rule is only used for descriptive field contents and
 values that are not intended to be interpreted by the message parser.
 Words of *TEXT-UTF8 contain characters from the UTF-8 character set
 (RFC 2279 [21]). In this regard, SIP differs from HTTP, which uses
 the ISO 8859-1 character set.
 TEXT-UTF8 = <any UTF-8 character encoding, except CTLs,
 but including LWS>
 A CRLF is allowed in the definition of TEXT-UTF8 only as part of a
 header field continuation. It is expected that the folding LWS will
 be replaced with a single SP before interpretation of the TEXT-UTF8
 value.
 Hexadecimal numeric characters are used in several protocol elements.
 hex = "A" | "B" | "C" | "D" | "E" | "F"
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 | "a" | "b" | "c" | "d" | "e" | "f" | digit
 Many SIP header field values consist of words separated by LWS or
 special characters. These special characters MUST be in a quoted
 string to be used within a parameter value.
 token = 1*< any CHAR except CTL's or separators>
 separators = "(" | ")" | "<" | ">" | "@" |
 "," | ";" | ":" | "\" | <"> |
 "/" | "[" | "]" | "?" | "=" |
 "{" | "}" | SP | HT
 Comments can be included in some SIP header fields by surrounding the
 comment text with parentheses. Comments are only allowed in fields
 containing "comment" as part of their field value definition. In all
 other fields, parentheses are considered part of the field value.
 comment = "(" *(ctext | quoted-pair | comment) ")"
 ctext = < any TEXT-UTF8 excluding "(" and ")">
 A string of text is parsed as a single word if it is quoted using
 double-quote marks.
 quoted-string = ( <"> *(qdtext | quoted-pair ) <"> )
 qdtext = <any TEXT-UTF8 except <">>
 The backslash character ("\") MAY be used as a single-character
 quoting mechanism only within quoted-string and comment constructs.
 quoted-pair = " \ " CHAR
D Using SRV DNS Records
 The following procedure is experimental and relies on DNS SRV records
 (RFC 2052 [14]). The steps listed below are used in place of the two
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 steps in section 1.4.2.
 If a step elicits no addresses, the client continues to the next
 step. However if a step elicits one or more addresses, but no SIP
 server at any of those addresses responds, then the client concludes
 the server is down and doesn't continue on to the next step.
 When SRV records are to be used, the protocol to use when querying
 for the SRV record is "sip". SRV records contain port numbers for
 servers, in addition to IP addresses; the client always uses this
 port number when contacting the SIP server. Otherwise, the port
 number in the SIP URI is used, if present. If there is no port in the
 URI, the default port, 5060, is used.
 1. If the host portion of the Request-URI is an IP address,
 the client contacts the server at the given address. If the
 host portion of the Request-URI is not an IP address, the
 client proceeds to the next step.
 2. The Request-URI is examined. If it contains an explicit
 port number, the next two steps are skipped.
 3. The Request-URI is examined. If it does not specify a
 protocol (TCP or UDP), the client queries the name server
 for SRV records for both UDP (if supported by the client)
 and TCP (if supported by the client) SIP servers. The
 format of these queries is defined in RFC 2052 [14]. The
 results of the query or queries are merged together and
 ordered based on priority. Then, the searching technique
 outlined in RFC 2052 [14] is used to select servers in
 order. If DNS doesn't return any records, the user goes to
 the last step. Otherwise, the user attempts to contact each
 server in the order listed. If no server is contacted, the
 user gives up.
 4. If the Request-URI specifies a protocol (TCP or UDP) that
 is supported by the client, the client queries the name
 server for SRV records for SIP servers of that protocol
 type only. If the client does not support the protocol
 specified in the Request-URI, it gives up. The searching
 technique outlined in RFC 2052 [14] is used to select
 servers from the DNS response in order. If DNS doesn't
 return any records, the user goes to the last step.
 Otherwise, the user attempts to contact each server in the
 order listed. If no server is contacted, the user gives up.
 5. The client queries the name server for address records for
 the host portion of the Request-URI. If there were no
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 address records, the client stops, as it has been unable to
 locate a server. By address record, we mean A RR's, AAAA
 RR's, or their most modern equivalent.
 A client MAY cache a successful DNS query result. A successful query
 is one which contained records in the answer, and a server was
 contacted at one of the addresses from the answer. When the client
 wishes to send a request to the same host, it starts the search as if
 it had just received this answer from the name server. The server
 uses the procedures specified in RFC1035 [15] regarding cache
 invalidation when the time-to-live of the DNS result expires. If the
 client does not find a SIP server among the addresses listed in the
 cached answer, it starts the search at the beginning of the sequence
 described above.
 For example, consider a client that wishes to send a SIP request. The
 Request-URI for the destination is sip:user@company.com. The client
 only supports UDP. It would follow these steps:
 1. The host portion is not an IP address, so the client goes
 to step 2 above.
 2. The client does a DNS query of QNAME="sip.udp.company.com",
 QCLASS=IN, QTYPE=SRV. Since it doesn't support TCP, it
 omits the TCP query. There were no addresses in the DNS
 response, so the client goes to the next step.
 3. The client does a DNS query for A records for
 "company.com". An address is found, so that client attempts
 to contact a server at that address at port 5060.
E IANA Considerations
 Section 4.4 describes a name space and mechanism for registering SIP
 options.
 Section 6.41 describes the name space for registering SIP warn-codes.
F Changes in Version -12
 Since version -11, the following changes have been made. These
 changes reflect IESG comments.
 o Section 16.3 was missing Content-Type header.
 o The DNS search procedure has changed. Reference to CNAME
 lookups has been removed since their usage is implicit in
 normal DNS procedures. Also, automatically appending a sip. to
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 the domain name in the URL before lookup has been removed. A
 note has been added discussing the creating of SIP URL's from
 email addresses and encourages the usage of rfc2219.
 o Semicolon removed from user and password BNF in SIP URL.
 o An email address for IANA registrations is now listed.
 o For the Moved Temporarily redirect response, no default value
 for the expiration of this address is specified. This has been
 clarified; the redirected address is only valid for the
 duration of the call. This is consistent with exising text
 under the Contact header description which indicates that the
 values should not be cached.
 o Clarification that CANCEL and ACK's should have the same Via
 branch parameter as the request they cancel or acknowledge.
G Acknowledgments
 We wish to thank the members of the IETF MMUSIC WG for their comments
 and suggestions. Detailed comments were provided by Anders
 Kristensen, Jim Buller, Dave Devanathan, Yaron Goland, Christian
 Huitema, Gadi Karmi, Jonathan Lennox, Keith Moore, Vern Paxson, Moshe
 J. Sambol, and Eric Tremblay.
 This work is based, inter alia, on [37,38].
H Authors' Addresses
 Mark Handley
 USC Information Sciences Institute
 c/o MIT Laboratory for Computer Science
 545 Technology Square
 Cambridge, MA 02139
 USA
 electronic mail: mjh@isi.edu
 Henning Schulzrinne
 Dept. of Computer Science
 Columbia University
 1214 Amsterdam Avenue
 New York, NY 10027
 USA
 electronic mail: schulzrinne@cs.columbia.edu
 Eve Schooler
 Computer Science Department 256-80
Handley/Schulzrinne/Schooler/Rosenberg [Page 140]
Internet Draft SIP January 15, 1999
 California Institute of Technology
 Pasadena, CA 91125
 USA
 electronic mail: schooler@cs.caltech.edu
 Jonathan Rosenberg
 Lucent Technologies, Bell Laboratories
 Rm. 4C-526
 101 Crawfords Corner Road
 Holmdel, NJ 07733
 USA
 electronic mail: jdrosen@bell-labs.com
I Bibliography
 [1] R. Pandya, "Emerging mobile and personal communication systems,"
 IEEE Communications Magazine , vol. 33, pp. 44--52, June 1995.
 [2] B. Braden, L. Zhang, S. Berson, S. Herzog, and S. Jamin,
 "Resource ReSerVation protocol (RSVP) -- version 1 functional
 specification," RFC 2205, Internet Engineering Task Force, Oct. 1997.
 [3] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: a
 transport protocol for real-time applications," RFC 1889, Internet
 Engineering Task Force, Jan. 1996.
 [4] H. Schulzrinne, R. Lanphier, and A. Rao, "Real time streaming
 protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.
 1998.
 [5] M. Handley, "SAP: Session announcement protocol," Internet Draft,
 Internet Engineering Task Force, Nov. 1996. Work in progress.
 [6] M. Handley and V. Jacobson, "SDP: session description protocol,"
 RFC 2327, Internet Engineering Task Force, Apr. 1998.
 [7] International Telecommunication Union, "Visual telephone systems
 and equipment for local area networks which provide a non-guaranteed
 quality of service," Recommendation H.323, Telecommunication
 Standardization Sector of ITU, Geneva, Switzerland, May 1996.
 [8] International Telecommunication Union, "Control protocol for
 multimedia communication," Recommendation H.245, Telecommunication
 Standardization Sector of ITU, Geneva, Switzerland, Feb. 1998.
 [9] International Telecommunication Union, "Media stream
 packetization and synchronization on non-guaranteed quality of
 service LANs," Recommendation H.225.0, Telecommunication
Handley/Schulzrinne/Schooler/Rosenberg [Page 141]
Internet Draft SIP January 15, 1999
 Standardization Sector of ITU, Geneva, Switzerland, Nov. 1996.
 [10] S. Bradner, "Key words for use in RFCs to indicate requirement
 levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.
 [11] R. Fielding, J. Gettys, J. Mogul, H. Nielsen, and T. Berners-
 Lee, "Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet
 Engineering Task Force, Jan. 1997.
 [12] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource
 identifiers (URI): generic syntax," RFC 2396, Internet Engineering
 Task Force, Aug. 1998.
 [13] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource
 locators (URL)," RFC 1738, Internet Engineering Task Force, Dec.
 1994.
 [14] A. Gulbrandsen and P. Vixie, "A DNS RR for specifying the
 location of services (DNS SRV)," RFC 2052, Internet Engineering Task
 Force, Oct. 1996.
 [15] P. Mockapetris, "Domain names - implementation and
 specification," RFC STD 13, 1035, Internet Engineering Task Force,
 Nov. 1987.
 [16] M. Hamilton and R. Wright, "Use of DNS aliases for network
 services," RFC 2219, Internet Engineering Task Force, Oct. 1997.
 [17] D. Zimmerman, "The finger user information protocol," RFC 1288,
 Internet Engineering Task Force, Dec. 1991.
 [18] S. Williamson, M. Kosters, D. Blacka, J. Singh, and K. Zeilstra,
 "Referral whois (rwhois) protocol V1.5," RFC 2167, Internet
 Engineering Task Force, June 1997.
 [19] W. Yeong, T. Howes, and S. Kille, "Lightweight directory access
 protocol," RFC 1777, Internet Engineering Task Force, Mar. 1995.
 [20] E. M. Schooler, "A multicast user directory service for
 synchronous rendezvous," Master's Thesis CS-TR-96-18, Department of
 Computer Science, California Institute of Technology, Pasadena,
 California, Aug. 1996.
 [21] F. Yergeau, "UTF-8, a transformation format of ISO 10646," RFC
 2279, Internet Engineering Task Force, Jan. 1998.
 [22] W. R. Stevens, TCP/IP illustrated: the protocols , vol. 1.
 Reading, Massachusetts: Addison-Wesley, 1994.
Handley/Schulzrinne/Schooler/Rosenberg [Page 142]
Internet Draft SIP January 15, 1999
 [23] J. Mogul and S. Deering, "Path MTU discovery," RFC 1191,
 Internet Engineering Task Force, Nov. 1990.
 [24] D. Crocker, "Standard for the format of ARPA internet text
 messages," RFC STD 11, 822, Internet Engineering Task Force, Aug.
 1982.
 [25] D. Meyer, "Administratively scoped IP multicast," RFC 2365,
 Internet Engineering Task Force, July 1998.
 [26] H. Schulzrinne, "RTP profile for audio and video conferences
 with minimal control," RFC 1890, Internet Engineering Task Force,
 Jan. 1996.
 [27] D. Eastlake, S. Crocker, and J. Schiller, "Randomness
 recommendations for security," RFC 1750, Internet Engineering Task
 Force, Dec. 1994.
 [28] P. Hoffman, L. Masinter, and J. Zawinski, "The mailto URL
 scheme," RFC 2368, Internet Engineering Task Force, July 1998.
 [29] B. Braden, "Requirements for internet hosts - application and
 support," RFC STD 3, 1123, Internet Engineering Task Force, Oct.
 1989.
 [30] J. Palme, "Common internet message headers," RFC 2076, Internet
 Engineering Task Force, Feb. 1997.
 [31] H. Alvestrand, "IETF policy on character sets and languages,"
 RFC 2277, Internet Engineering Task Force, Jan. 1998.
 [32] M. Elkins, "MIME security with pretty good privacy (PGP)," RFC
 2015, Internet Engineering Task Force, Oct. 1996.
 [33] D. Atkins, W. Stallings, and P. Zimmermann, "PGP message
 exchange formats," RFC 1991, Internet Engineering Task Force, Aug.
 1996.
 [34] R. Atkinson, "Security architecture for the internet protocol,"
 RFC 1825, Internet Engineering Task Force, Aug. 1995.
 [35] C. Allen and T. Dierks, "The TLS protocol version 1.0," Internet
 Draft, Internet Engineering Task Force, Nov. 1997. Work in progress.
 [36] J. Franks, P. Hallam-Baker, J. Hostetler, S. Lawrence, P. Leach,
 A. Luotonen, and L. Stewart, "HTTP authentication: Basic and digest
 access authentication," Internet Draft, Internet Engineering Task
 Force, Sept. 1998. Work in progress.
Handley/Schulzrinne/Schooler/Rosenberg [Page 143]
Internet Draft SIP January 15, 1999
 [37] E. M. Schooler, "Case study: multimedia conference control in a
 packet-switched teleconferencing system," Journal of Internetworking:
 Research and Experience , vol. 4, pp. 99--120, June 1993. ISI
 reprint series ISI/RS-93-359.
 [38] H. Schulzrinne, "Personal mobility for multimedia services in
 the Internet," in European Workshop on Interactive Distributed
 Multimedia Systems and Services (IDMS) , (Berlin, Germany), Mar.
 1996.
 Full Copyright Statement
 Copyright (c) The Internet Society (1999). All Rights Reserved.
 This document and translations of it may be copied and furnished to
 others, and derivative works that comment on or otherwise explain it
 or assist in its implementation may be prepared, copied, published
 and distributed, in whole or in part, without restriction of any
 kind, provided that the above copyright notice and this paragraph are
 included on all such copies and derivative works. However, this
 document itself may not be modified in any way, such as by removing
 the copyright notice or references to the Internet Society or other
 Internet organizations, except as needed for the purpose of
 developing Internet standards in which case the procedures for
 copyrights defined in the Internet Standards process must be
 followed, or as required to translate it into languages other than
 English.
 The limited permissions granted above are perpetual and will not be
 revoked by the Internet Society or its successors or assigns.
 This document and the information contained herein is provided on an
 "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
 TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
 BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
 HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
 MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
 Table of Contents
 1 Introduction ........................................ 2
 1.1 Overview of SIP Functionality ....................... 2
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 1.2 Terminology ......................................... 4
 1.3 Definitions ......................................... 4
 1.4 Overview of SIP Operation ........................... 7
 1.4.1 SIP Addressing ...................................... 7
 1.4.2 Locating a SIP Server ............................... 8
 1.4.3 SIP Transaction ..................................... 9
 1.4.4 SIP Invitation ...................................... 10
 1.4.5 Locating a User ..................................... 12
 1.4.6 Changing an Existing Session ........................ 13
 1.4.7 Registration Services ............................... 13
 1.5 Protocol Properties ................................. 13
 1.5.1 Minimal State ....................................... 13
 1.5.2 Lower-Layer-Protocol Neutral ........................ 13
 1.5.3 Text-Based .......................................... 15
 2 SIP Uniform Resource Locators ....................... 15
 3 SIP Message Overview ................................ 19
 4 Request ............................................. 21
 4.1 Request-Line ........................................ 21
 4.2 Methods ............................................. 21
 4.2.1 INVITE .............................................. 22
 4.2.2 ACK ................................................. 23
 4.2.3 OPTIONS ............................................. 24
 4.2.4 BYE ................................................. 24
 4.2.5 CANCEL .............................................. 25
 4.2.6 REGISTER ............................................ 26
 4.3 Request-URI ......................................... 28
 4.3.1 SIP Version ......................................... 29
 4.4 Option Tags ......................................... 29
 4.4.1 Registering New Option Tags with IANA ............... 30
 5 Response ............................................ 31
 5.1 Status-Line ......................................... 31
 5.1.1 Status Codes and Reason Phrases ..................... 31
 6 Header Field Definitions ............................ 33
 6.1 General Header Fields ............................... 35
 6.2 Entity Header Fields ................................ 36
 6.3 Request Header Fields ............................... 37
 6.4 Response Header Fields .............................. 38
 6.5 End-to-end and Hop-by-hop Headers ................... 38
 6.6 Header Field Format ................................. 38
 6.7 Accept .............................................. 39
 6.8 Accept-Encoding ..................................... 39
 6.9 Accept-Language ..................................... 39
 6.10 Allow ............................................... 40
 6.11 Authorization ....................................... 40
 6.12 Call-ID ............................................. 40
 6.13 Contact ............................................. 42
 6.14 Content-Encoding .................................... 45
 6.15 Content-Length ...................................... 45
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 6.16 Content-Type ........................................ 46
 6.17 CSeq ................................................ 47
 6.18 Date ................................................ 48
 6.19 Encryption .......................................... 49
 6.20 Expires ............................................. 50
 6.21 From ................................................ 51
 6.22 Hide ................................................ 52
 6.23 Max-Forwards ........................................ 53
 6.24 Organization ........................................ 54
 6.25 Priority ............................................ 54
 6.26 Proxy-Authenticate .................................. 55
 6.27 Proxy-Authorization ................................. 56
 6.28 Proxy-Require ....................................... 56
 6.29 Record-Route ........................................ 56
 6.30 Require ............................................. 57
 6.31 Response-Key ........................................ 58
 6.32 Retry-After ......................................... 59
 6.33 Route ............................................... 59
 6.34 Server .............................................. 60
 6.35 Subject ............................................. 60
 6.36 Timestamp ........................................... 60
 6.37 To .................................................. 61
 6.38 Unsupported ......................................... 62
 6.39 User-Agent .......................................... 63
 6.40 Via ................................................. 63
 6.40.1 Requests ............................................ 63
 6.40.2 Receiver-tagged Via Header Fields ................... 64
 6.40.3 Responses ........................................... 64
 6.40.4 User Agent and Redirect Servers ..................... 65
 6.40.5 Syntax .............................................. 66
 6.41 Warning ............................................. 67
 6.42 WWW-Authenticate .................................... 69
 7 Status Code Definitions ............................. 69
 7.1 Informational 1xx ................................... 70
 7.1.1 100 Trying .......................................... 70
 7.1.2 180 Ringing ......................................... 70
 7.1.3 181 Call Is Being Forwarded ......................... 70
 7.1.4 182 Queued .......................................... 70
 7.2 Successful 2xx ...................................... 70
 7.2.1 200 OK .............................................. 71
 7.3 Redirection 3xx ..................................... 71
 7.3.1 300 Multiple Choices ................................ 71
 7.3.2 301 Moved Permanently ............................... 72
 7.3.3 302 Moved Temporarily ............................... 72
 7.3.4 305 Use Proxy ....................................... 72
 7.3.5 380 Alternative Service ............................. 72
 7.4 Request Failure 4xx ................................. 72
 7.4.1 400 Bad Request ..................................... 72
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 7.4.2 401 Unauthorized .................................... 73
 7.4.3 402 Payment Required ................................ 73
 7.4.4 403 Forbidden ....................................... 73
 7.4.5 404 Not Found ....................................... 73
 7.4.6 405 Method Not Allowed .............................. 73
 7.4.7 406 Not Acceptable .................................. 73
 7.4.8 407 Proxy Authentication Required ................... 73
 7.4.9 408 Request Timeout ................................. 74
 7.4.10 409 Conflict ........................................ 74
 7.4.11 410 Gone ............................................ 74
 7.4.12 411 Length Required ................................. 74
 7.4.13 413 Request Entity Too Large ........................ 74
 7.4.14 414 Request-URI Too Long ............................ 74
 7.4.15 415 Unsupported Media Type .......................... 74
 7.4.16 420 Bad Extension ................................... 75
 7.4.17 480 Temporarily Unavailable ......................... 75
 7.4.18 481 Call Leg/Transaction Does Not Exist ............. 75
 7.4.19 482 Loop Detected ................................... 75
 7.4.20 483 Too Many Hops ................................... 75
 7.4.21 484 Address Incomplete .............................. 75
 7.4.22 485 Ambiguous ....................................... 76
 7.4.23 486 Busy Here ....................................... 76
 7.5 Server Failure 5xx .................................. 77
 7.5.1 500 Server Internal Error ........................... 77
 7.5.2 501 Not Implemented ................................. 77
 7.5.3 502 Bad Gateway ..................................... 77
 7.5.4 503 Service Unavailable ............................. 77
 7.5.5 504 Gateway Time-out ................................ 77
 7.5.6 505 Version Not Supported ........................... 77
 7.6 Global Failures 6xx ................................. 78
 7.6.1 600 Busy Everywhere ................................. 78
 7.6.2 603 Decline ......................................... 78
 7.6.3 604 Does Not Exist Anywhere ......................... 78
 7.6.4 606 Not Acceptable .................................. 78
 8 SIP Message Body .................................... 79
 8.1 Body Inclusion ...................................... 79
 8.2 Message Body Type ................................... 79
 8.3 Message Body Length ................................. 79
 9 Compact Form ........................................ 80
 10 Behavior of SIP Clients and Servers ................. 81
 10.1 General Remarks ..................................... 81
 10.1.1 Requests ............................................ 81
 10.1.2 Responses ........................................... 81
 10.2 Source Addresses, Destination Addresses and
 Connections .................................................... 82
 10.2.1 Unicast UDP ......................................... 82
 10.2.2 Multicast UDP ....................................... 82
 10.3 TCP ................................................. 83
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 10.4 Reliability for BYE, CANCEL, OPTIONS, REGISTER
 Requests ....................................................... 84
 10.4.1 UDP ................................................. 84
 10.4.2 TCP ................................................. 85
 10.5 Reliability for INVITE Requests ..................... 85
 10.5.1 UDP ................................................. 86
 10.5.2 TCP ................................................. 87
 10.6 Reliability for ACK Requests ........................ 87
 10.7 ICMP Handling ....................................... 90
 11 Behavior of SIP User Agents ......................... 90
 11.1 Caller Issues Initial INVITE Request ................ 90
 11.2 Callee Issues Response .............................. 90
 11.3 Caller Receives Response to Initial Request ......... 90
 11.4 Caller or Callee Generate Subsequent Requests ....... 91
 11.5 Receiving Subsequent Requests ....................... 91
 12 Behavior of SIP Proxy and Redirect Servers .......... 92
 12.1 Redirect Server ..................................... 92
 12.2 User Agent Server ................................... 92
 12.3 Proxy Server ........................................ 92
 12.3.1 Proxying Requests ................................... 93
 12.3.2 Proxying Responses .................................. 93
 12.3.3 Stateless Proxy: Proxying Responses ................. 93
 12.3.4 Stateful Proxy: Receiving Requests .................. 93
 12.3.5 Stateful Proxy: Receiving ACKs ...................... 93
 12.3.6 Stateful Proxy: Receiving Responses ................. 94
 12.3.7 Stateless, Non-Forking Proxy ........................ 94
 12.4 Forking Proxy ....................................... 95
 13 Security Considerations ............................. 98
 13.1 Confidentiality and Privacy: Encryption ............. 99
 13.1.1 End-to-End Encryption ............................... 99
 13.1.2 Privacy of SIP Responses ............................ 101
 13.1.3 Encryption by Proxies ............................... 102
 13.1.4 Hop-by-Hop Encryption ............................... 102
 13.1.5 Via field encryption ................................ 102
 13.2 Message Integrity and Access Control:
 Authentication ................................................. 103
 13.2.1 Trusting responses .................................. 106
 13.3 Callee Privacy ...................................... 107
 13.4 Known Security Problems ............................. 107
 14 SIP Authentication using HTTP Basic and Digest
 Schemes ........................................................ 107
 14.1 Framework ........................................... 107
 14.2 Basic Authentication ................................ 108
 14.3 Digest Authentication ............................... 108
 14.4 Proxy-Authentication ................................ 109
 15 SIP Security Using PGP .............................. 109
 15.1 PGP Authentication Scheme ........................... 109
 15.1.1 The WWW-Authenticate Response Header ................ 110
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 15.1.2 The Authorization Request Header .................... 111
 15.2 PGP Encryption Scheme ............................... 112
 15.3 Response-Key Header Field for PGP ................... 113
 16 Examples ............................................ 113
 16.1 Registration ........................................ 113
 16.2 Invitation to a Multicast Conference ................ 115
 16.2.1 Request ............................................. 115
 16.2.2 Response ............................................ 116
 16.3 Two-party Call ...................................... 117
 16.4 Terminating a Call .................................. 119
 16.5 Forking Proxy ....................................... 119
 16.6 Redirects ........................................... 123
 16.7 Negotiation ......................................... 125
 16.8 OPTIONS Request ..................................... 126
 A Minimal Implementation .............................. 127
 A.1 Client .............................................. 127
 A.2 Server .............................................. 128
 A.3 Header Processing ................................... 128
 B Usage of the Session Description Protocol (SDP)
 ................................................................ 128
 B.1 Configuring Media Streams ........................... 128
 B.2 Setting SDP Values for Unicast ...................... 130
 B.3 Multicast Operation ................................. 131
 B.4 Delayed Media Streams ............................... 132
 B.5 Putting Media Streams on Hold ....................... 132
 B.6 Subject and SDP "s=" Line ........................... 132
 B.7 The SDP "o=" Line ................................... 133
 C Summary of Augmented BNF ............................ 133
 C.1 Basic Rules ......................................... 135
 D Using SRV DNS Records ............................... 137
 E IANA Considerations ................................. 139
 F Changes in Version -12 .............................. 139
 G Acknowledgments ..................................... 140
 H Authors' Addresses .................................. 140
 I Bibliography ........................................ 141
Handley/Schulzrinne/Schooler/Rosenberg [Page 149]