Network Working Group H. Alvestrand
Internet-Draft Google
Intended status: Standards Track October 22, 2014
Expires: April 25, 2015
Transports for WebRTC
draft-ietf-rtcweb-transports-07
Abstract
This document describes the data transport protocols used by WebRTC,
including the protocols used for interaction with intermediate boxes
such as firewalls, relays and NAT boxes.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
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This Internet-Draft will expire on April 25, 2015.
Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved.
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described in the Simplified BSD License.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Requirements language . . . . . . . . . . . . . . . . . . . . 3
3. Transport and Middlebox specification . . . . . . . . . . . . 3
3.1. System-provided interfaces . . . . . . . . . . . . . . . 3
3.2. Ability to use IPv4 and IPv6 . . . . . . . . . . . . . . 3
3.3. Usage of temporary IPv6 addresses . . . . . . . . . . . . 4
3.4. Middle box related functions . . . . . . . . . . . . . . 4
3.5. Transport protocols implemented . . . . . . . . . . . . . 6
4. Media Prioritization . . . . . . . . . . . . . . . . . . . . 6
4.1. Usage of Quality of Service - DSCP and Multiplexing . . . 6
4.2. Local prioritization . . . . . . . . . . . . . . . . . . 8
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
6. Security Considerations . . . . . . . . . . . . . . . . . . . 9
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 9
8. References . . . . . . . . . . . . . . . . . . . . . . . . . 9
8.1. Normative References . . . . . . . . . . . . . . . . . . 9
8.2. Informative References . . . . . . . . . . . . . . . . . 12
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 12
A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 12
A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 13
A.3. Changes from -02 to -03 . . . . . . . . . . . . . . . . . 13
A.4. Changes from -03 to -04 . . . . . . . . . . . . . . . . . 14
A.5. Changes from -04 to -05 . . . . . . . . . . . . . . . . . 14
A.6. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 14
A.7. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 14
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 15
1. Introduction
WebRTC is a protocol suite aimed at real time multimedia exchange
between browsers, and between browsers and other entities.
WebRTC is described in the WebRTC overview document,
[I-D.ietf-rtcweb-overview], which also defines terminology used in
this document.
This document focuses on the data transport protocols that are used
by conforming implementations, including the protocols used for
interaction with intermediate boxes such as firewalls, relays and NAT
boxes.
This protocol suite intends to satisfy the security considerations
described in the WebRTC security documents,
[I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security-arch].
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This document describes requirements that apply to all WebRTC
devices. When there are requirements that apply only to WebRTC User
Agents (also called browsers) , this is called out.
The form "WebRTC endpoint" is used as a synonym for "WebRTC device"
in contexts where other text talks about endpoints.
2. Requirements language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
3. Transport and Middlebox specification
3.1. System-provided interfaces
The protocol specifications used here assume that the following
protocols are available to the WebRTC devices:
o UDP. This is the protocol assumed by most protocol elements
described.
o TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL
and ICE-TCP.
For both protocols, IPv4 and IPv6 support is assumed.
For UDP, this specification assumes the ability to set the DSCP code
point of the sockets opened on a per-packet basis, in order to
achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos]
(see Section 4.1) when multiple media types are multiplexed. It does
not assume that the DSCP codepoints will be honored, and does assume
that they may be zeroed or changed, since this is a local
configuration issue.
Platforms that do not give access to these interfaces will not be
able to support a conforming WebRTC implementation.
This specification does not assume that the implementation will have
access to ICMP or raw IP.
3.2. Ability to use IPv4 and IPv6
Web applications running in a WebRTC browser MUST be able to utilize
both IPv4 and IPv6 where available - that is, when two peers have
only IPv4 connectivity to each other, or they have only IPv6
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connectivity to each other, applications running in the WebRTC
browser MUST be able to communicate.
WebRTC devices, when attached to networks with appropriate protocol
support MUST also be able to communicate using IPv6 and IPv4.
When TURN is used, and the TURN server has IPv4 or IPv6 connectivity
to the peer or its TURN server, candidates of the appropriate types
MUST be supported. The "Happy Eyeballs" specification for ICE
[I-D.reddy-mmusic-ice-happy-eyeballs] SHOULD be supported.
3.3. Usage of temporary IPv6 addresses
The IPv6 default address selection specification [RFC6724] specifies
that temporary addresses [RFC4941] are to be preferred over permanent
addresses. This is a change from the rules specified by [RFC3484].
For applications that select a single address, this is usually done
by the IPV6_PREFER_SRC_TMP preference flag specified in [RFC5014].
However, this rule is not completely obvious in the ICE scope. This
is therefore clarified as follows:
When a WebRTC endpoint gathers all IPv6 addresses on a host, and both
temporary addresses and permanent addresses of the same scope are
present, the client SHOULD discard the permanent addresses before
forming pairs. This is consistent with the default policy described
in [RFC6724].
3.4. Middle box related functions
Except when called out, all requirements in this section apply to all
WebRTC devices.
The primary mechanism to deal with middle boxes is ICE, which is an
appropriate way to deal with NAT boxes and firewalls that accept
traffic from the inside, but only from the outside if it is in
response to inside traffic (simple stateful firewalls).
WebRTC endpoints MUST support ICE [RFC5245]. The implementation MUST
be a full ICE implementation, not ICE-Lite. A full ICE
implementation allows interworking with both ICE and ICE-Lite
implementations when they are deployed appropriately.
In order to deal with situations where both parties are behind NATs
of the type that perform endpoint-dependent mapping (as defined in
[RFC5128] section 2.4), WebRTC endpoints MUST support TURN [RFC5766].
WebRTC browsers MUST support configuration of STUN and TURN servers,
both from browser configuration and from an application.
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In order to deal with firewalls that block all UDP traffic, the mode
of TURN that uses TCP between the client and the server MUST be
supported, and the mode of TURN that uses TLS over TCP between the
client and the server MUST be supported. See [RFC5766] section 2.1
for details.
In order to deal with situations where one party is on an IPv4
network and the other party is on an IPv6 network, TURN extensions
for IPv6 [RFC6156] MUST be supported.
TURN TCP candidates, where the connection from the client's TURN
server to the peer is a TCP connection, [RFC6062] MAY be supported.
However, such candidates are not seen as providing any significant
benefit, for the following reasons.
First, use of TURN TCP candidates would only be relevant in cases
which both peers are required to use TCP to establish a
PeerConnection.
Second, that use case is supported in a different way by both sides
establishing UDP relay candidates using TURN over TCP to connect to
their respective relay servers.
Third, using TCP only between the endpoint and its relay may result
in less issues with TCP in regards to real-time constraints, e.g. due
to head of line blocking.
ICE-TCP candidates [RFC6544] MUST be supported; this may allow
applications to communicate to peers with public IP addresses across
UDP-blocking firewalls without using a TURN server.
If ICE-TCP connections are used, RTP framing according to [RFC4571]
MUST be used for all content that doesn't have its own framing
mechanism.
The ALTERNATE-SERVER mechanism specified in [RFC5389] (STUN) section
11 (300 Try Alternate) MUST be supported.
In order to deal with the scenario in which the media must traverse a
HTTP Proxy, WebRTC browser MUST support the HTTP CONNECT request
(Section 4.3.6 of [RFC7231]). WebRTC devices SHOULD support this
request.
The HTTP Proxy may require authentication and therefore, if HTTP
CONNECT request is supported, proxy authentication as described in
Section 4.3.6 of [RFC7231] and [RFC7235] MUST also be supported.
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In addition, the HTTP CONNECT MUST include an indication of the
protocol being used with the HTTP CONNECT initiated tunnel as
described in [I-D.ietf-httpbis-tunnel-protocol]
3.5. Transport protocols implemented
For transport of media, secure RTP is used. The details of the
profile of RTP used are described in "RTP Usage"
[I-D.ietf-rtcweb-rtp-usage]. Key exchange MUST be done using DTLS-
SRTP, as described in [I-D.ietf-rtcweb-security-arch].
For data transport over the WebRTC data channel
[I-D.ietf-rtcweb-data-channel], WebRTC endpoints MUST support SCTP
over DTLS over ICE. This encapsulation is specified in
[I-D.ietf-tsvwg-sctp-dtls-encaps]. Negotiation of this transport in
SDP is defined in [I-D.ietf-mmusic-sctp-sdp]. The SCTP extension for
NDATA, [I-D.ietf-tsvwg-sctp-ndata], MUST be supported.
The setup protocol for WebRTC data channels is described in
[I-D.jesup-rtcweb-data-protocol].
WebRTC devices MUST support multiplexing of DTLS and RTP over the
same port pair, as described in the DTLS_SRTP specification
[RFC5764], section 5.1.2. All application layer protocol payloads
over this DTLS connection are SCTP packets.
Protocol identification MUST be supplied as part of the DTLS
handshake, as specified in [I-D.thomson-rtcweb-alpn].
4. Media Prioritization
The WebRTC prioritization model is that the application tells the
WebRTC browser about the priority of media and data flows through an
API.
The priority associated with a media or data flow is classified as
"normal", "below normal", "high" or "very high". There are only four
priority levels at the API.
The priority settings affect two pieces of behavior: Packet markings
and packet send sequence decisions. Each is described in its own
section below.
4.1. Usage of Quality of Service - DSCP and Multiplexing
WebRTC endpoints SHOULD attempt to set QoS on the packets sent,
according to the guidelines in [I-D.ietf-tsvwg-rtcweb-qos]. It is
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appropriate to depart from this recommendation when running on
platforms where QoS marking is not implemented.
The WebRTC endpoint MAY turn off use of DSCP markings if it detects
symptoms of unexpected behaviour like priority inversion or blocking
of packets with certain DSCP markings. The detection of these
conditions is implementation dependent. (Question: Does there need
to be an API knob to turn off DSCP markings?)
All packets carrying data from the SCTP association supporting the
data channels MUST use a single DSCP code point.
All packets on one TCP connection, no matter what it carries, MUST
use a single DSCP code point.
More advice on the use of DSCP code points with RTP is given in
[I-D.ietf-dart-dscp-rtp].
There exist a number of schemes for achieving quality of service that
do not depend solely on DSCP code points. Some of these schemes
depend on classifying the traffic into flows based on 5-tuple (source
address, source port, protocol, destination address, destination
port) or 6-tuple (5-tuple + DSCP code point). Under differing
conditions, it may therefore make sense for a sending application to
choose any of the configurations:
o Each media stream carried on its own 5-tuple
o Media streams grouped by media type into 5-tuples (such as
carrying all audio on one 5-tuple)
o All media sent over a single 5-tuple, with or without
differentiation into 6-tuples based on DSCP code points
In each of the configurations mentioned, data channels may be carried
in its own 5-tuple, or multiplexed together with one of the media
flows.
More complex configurations, such as sending a high priority video
stream on one 5-tuple and sending all other video streams multiplexed
together over another 5-tuple, can also be envisioned. More
information on mapping media flows to 5-tuples can be found in
[I-D.ietf-rtcweb-rtp-usage].
A sending WebRTC endpoint MUST be able to support the following
configurations:
o multiplex all media and data on a single 5-tuple (fully bundled)
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o send each media stream on its own 5-tuple and data on its own
5-tuple (fully unbundled)
o bundle each media type (audio, video or data) into its own 5-tuple
(bundling by media type)
It MAY choose to support other configurations.
Sending data over multiple 5-tuples is not supported.
A receiving WebRTC endpoint MUST be able to receive media and data in
all these configurations.
4.2. Local prioritization
When an WebRTC endpoint has packets to send on multiple streams (with
each media stream and each data channel considered as one "stream"
for this purpose) that are congestion-controlled under the same
congestion controller, the WebRTC endpoint SHOULD cause data to be
emitted in such a way that each stream at each level of priority is
being given approximately twice the transmission capacity (measured
in payload bytes) of the level below.
Thus, when congestion occurs, a "very high" priority flow will have
the ability to send 8 times as much data as a "below normal" flow if
both have data to send. This prioritization is independent of the
media type. The details of which packet to send first are
implementation defined.
For example: If there is a very high priority audio flow sending 100
byte packets, and a normal priority video flow sending 1000 byte
packets, and outgoing capacity exists for sending >5000 payload
bytes, it would be appropriate to send 4000 bytes (40 packets) of
audio and 1000 bytes (one packet) of video as the result of a single
pass of sending decisions.
Conversely, if the audio flow is marked normal priority and the video
flow is marked very high priority, the scheduler may decide to send 2
video packets (2000 bytes) and 5 audio packets (500 bytes) when
outgoing capacity exists for sending > 2500 payload bytes.
If there are two very high priority audio flows, each will be able to
send 4000 bytes in the same period where a normal priority video flow
is able to send 1000 bytes.
Two example implementation strategies are:
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o When the available bandwidth is known from the congestion control
algorithm, configure each codec and each data channel with a
target send rate that is appropriate to its share of the available
bandwidth.
o When congestion control indicates that a specified number of
packets can be sent, send packets that are available to send using
a weighted round robin scheme across the connections.
Any combination of these, or other schemes that have the same effect,
is valid, as long as the distribution of transmission capacity is
approximately correct.
For media, it is usually inappropriate to use deep queues for
sending; it is more useful to, for instance, skip intermediate frames
that have no dependencies on them in order to achieve a lower
bitrate. For reliable data, queues are useful.
5. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
6. Security Considerations
Security considerations are enumerated in [I-D.ietf-rtcweb-security].
7. Acknowledgements
This document is based on earlier versions embedded in
[I-D.ietf-rtcweb-overview], which were the results of contributions
from many RTCWEB WG members.
Special thanks for reviews of earlier versions of this draft go to
Eduardo Gueiros, Magnus Westerlund, Markus Isomaki and Dan Wing; the
contributions from Andrew Hutton also deserve special mention.
8. References
8.1. Normative References
[I-D.ietf-httpbis-tunnel-protocol]
Hutton, A., Uberti, J., and M. Thomson, "The Tunnel-
Protocol HTTP Request Header Field", draft-ietf-httpbis-
tunnel-protocol-00 (work in progress), August 2014.
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[I-D.ietf-mmusic-sctp-sdp]
Loreto, S. and G. Camarillo, "Stream Control Transmission
Protocol (SCTP)-Based Media Transport in the Session
Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-07
(work in progress), July 2014.
[I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-11 (work in
progress), July 2014.
[I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-15 (work in progress), May
2014.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-07 (work in progress), July 2014.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-10 (work in progress), July 2014.
[I-D.ietf-tsvwg-rtcweb-qos]
Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J.
Polk, "DSCP and other packet markings for RTCWeb QoS",
draft-ietf-tsvwg-rtcweb-qos-02 (work in progress), June
2014.
[I-D.ietf-tsvwg-sctp-dtls-encaps]
Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS
Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp-
dtls-encaps-05 (work in progress), July 2014.
[I-D.ietf-tsvwg-sctp-ndata]
Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann,
"Stream Schedulers and a New Data Chunk for the Stream
Control Transmission Protocol", draft-ietf-tsvwg-sctp-
ndata-01 (work in progress), July 2014.
[I-D.reddy-mmusic-ice-happy-eyeballs]
Reddy, T., Patil, P., and P. Martinsen, "Happy Eyeballs
Extension for ICE", draft-reddy-mmusic-ice-happy-
eyeballs-07 (work in progress), June 2014.
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[I-D.thomson-rtcweb-alpn]
Thomson, M., "Application Layer Protocol Negotiation for
Web Real-Time Communications (WebRTC)", draft-thomson-
rtcweb-alpn-00 (work in progress), April 2014.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
and RTP Control Protocol (RTCP) Packets over Connection-
Oriented Transport", RFC 4571, July 2006.
[RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy
Extensions for Stateless Address Autoconfiguration in
IPv6", RFC 4941, September 2007.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, April
2010.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
October 2008.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.
[RFC6062] Perreault, S. and J. Rosenberg, "Traversal Using Relays
around NAT (TURN) Extensions for TCP Allocations", RFC
6062, November 2010.
[RFC6156] Camarillo, G., Novo, O., and S. Perreault, "Traversal
Using Relays around NAT (TURN) Extension for IPv6", RFC
6156, April 2011.
[RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach,
"TCP Candidates with Interactive Connectivity
Establishment (ICE)", RFC 6544, March 2012.
[RFC6724] Thaler, D., Draves, R., Matsumoto, A., and T. Chown,
"Default Address Selection for Internet Protocol Version 6
(IPv6)", RFC 6724, September 2012.
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[RFC7231] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol
(HTTP/1.1): Semantics and Content", RFC 7231, June 2014.
[RFC7235] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol
(HTTP/1.1): Authentication", RFC 7235, June 2014.
8.2. Informative References
[I-D.ietf-dart-dscp-rtp]
Black, D. and P. Jones, "Differentiated Services
(DiffServ) and Real-time Communication", draft-ietf-dart-
dscp-rtp-08 (work in progress), October 2014.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-10
(work in progress), June 2014.
[I-D.jesup-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Protocol", draft-jesup-rtcweb-data-protocol-04 (work in
progress), February 2013.
[RFC3484] Draves, R., "Default Address Selection for Internet
Protocol version 6 (IPv6)", RFC 3484, February 2003.
[RFC5014] Nordmark, E., Chakrabarti, S., and J. Laganier, "IPv6
Socket API for Source Address Selection", RFC 5014,
September 2007.
[RFC5128] Srisuresh, P., Ford, B., and D. Kegel, "State of Peer-to-
Peer (P2P) Communication across Network Address
Translators (NATs)", RFC 5128, March 2008.
Appendix A. Change log
This section should be removed before publication as an RFC.
A.1. Changes from -00 to -01
o Clarified DSCP requirements, with reference to -qos-
o Clarified "symmetric NAT" -> "NATs which perform endpoint-
dependent mapping"
o Made support of TURN over TCP mandatory
o Made support of TURN over TLS a MAY, and added open question
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o Added an informative reference to -firewalls-
o Called out that we don't make requirements on HTTP proxy
interaction (yet
A.2. Changes from -01 to -02
o Required support for 300 Alternate Server from STUN.
o Separated the ICE-TCP candidate requirement from the TURN-TCP
requirement.
o Added new sections on using QoS functions, and on multiplexing
considerations.
o Removed all mention of RTP profiles. Those are the business of
the RTP usage draft, not this one.
o Required support for TURN IPv6 extensions.
o Removed reference to the TURN URI scheme, as it was unnecessary.
o Made an explicit statement that multiplexing (or not) is an
application matter.
.
A.3. Changes from -02 to -03
o Added required support for draft-ietf-tsvwg-sctp-ndata
o Removed discussion of multiplexing, since this is present in rtp-
usage.
o Added RFC 4571 reference for framing RTP packets over TCP.
o Downgraded TURN TCP candidates from SHOULD to MAY, and added more
language discussing TCP usage.
o Added language on IPv6 temporary addresses.
o Added language describing multiplexing choices.
o Added a separate section detailing what it means when we say that
an WebRTC implementation MUST support both IPv4 and IPv6.
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A.4. Changes from -03 to -04
o Added a section on prioritization, moved the DSCP section into it,
and added a section on local prioritization, giving a specific
algorithm for interpreting "priority" in local prioritization.
o ICE-TCP candidates was changed from MAY to MUST, in recognition of
the sense of the room at the London IETF.
A.5. Changes from -04 to -05
o Reworded introduction
o Removed all references to "WebRTC". It now uses only the term
RTCWEB.
o Addressed a number of clarity / language comments
o Rewrote the prioritization to cover data channels and to describe
multiple ways of prioritizing flows
o Made explicit reference to "MUST do DTLS-SRTP", and referred to
security-arch for details
A.6. Changes from -05 to -06
o Changed all references to "RTCWEB" to "WebRTC", except one
reference to the working group
o Added reference to the httpbis "connect" protocol (being adopted
by HTTPBIS)
o Added reference to the ALPN header (being adopted by RTCWEB)
o Added reference to the DART RTP document
o Said explicitly that SCTP for data channels has a single DSCP
codepoint
A.7. Changes from -06 to -07
o Updated terminology in accordance with -overview. Got rid of all
occurences of "WebRTC implementation".
o Modified description of ICE-TCP encapsulation in accordance with
list discussion.
o Added HTTP CONNECT requirement in accordance with list discussion.
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Author's Address
Harald Alvestrand
Google
Email: harald@alvestrand.no
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