draft-ietf-avtcore-multi-media-rtp-session-06

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AVTCORE WG M. Westerlund
Internet-Draft Ericsson
Updates: 3550, 3551 (if approved) C. Perkins
Intended status: Standards Track University of Glasgow
Expires: April 11, 2015 J. Lennox
 Vidyo
 October 08, 2014
 Sending Multiple Types of Media in a Single RTP Session
 draft-ietf-avtcore-multi-media-rtp-session-06
Abstract
 This document specifies how an RTP session can contain media streams
 with media from multiple media types such as audio, video, and text.
 This has been restricted by the RTP Specification, and thus this
 document updates RFC 3550 and RFC 3551 to enable this behaviour for
 applications that satisfy the applicability for using multiple media
 types in a single RTP session.
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
 Task Force (IETF). Note that other groups may also distribute
 working documents as Internet-Drafts. The list of current Internet-
 Drafts is at http://datatracker.ietf.org/drafts/current/.
 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."
 This Internet-Draft will expire on April 11, 2015.
Copyright Notice
 Copyright (c) 2014 IETF Trust and the persons identified as the
 document authors. All rights reserved.
 This document is subject to BCP 78 and the IETF Trust's Legal
 Provisions Relating to IETF Documents
 (http://trustee.ietf.org/license-info) in effect on the date of
 publication of this document. Please review these documents
 carefully, as they describe your rights and restrictions with respect
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 to this document. Code Components extracted from this document must
 include Simplified BSD License text as described in Section 4.e of
 the Trust Legal Provisions and are provided without warranty as
 described in the Simplified BSD License.
Table of Contents
 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
 2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3
 3. Motivation . . . . . . . . . . . . . . . . . . . . . . . . . 4
 4. Overview of Solution . . . . . . . . . . . . . . . . . . . . 5
 5. Applicability . . . . . . . . . . . . . . . . . . . . . . . . 6
 5.1. Usage of the RTP session . . . . . . . . . . . . . . . . 6
 5.2. Signalled Support . . . . . . . . . . . . . . . . . . . . 7
 5.3. Homogeneous Multi-party . . . . . . . . . . . . . . . . . 7
 5.4. Reduced number of Payload Types . . . . . . . . . . . . . 8
 5.5. Stream Differentiation . . . . . . . . . . . . . . . . . 8
 5.6. Non-compatible Extensions . . . . . . . . . . . . . . . . 8
 6. RTP Session Specification . . . . . . . . . . . . . . . . . . 9
 6.1. RTP Session . . . . . . . . . . . . . . . . . . . . . . . 9
 6.2. Sender Source Restrictions . . . . . . . . . . . . . . . 12
 6.3. Payload Type Applicability . . . . . . . . . . . . . . . 12
 6.4. RTCP Considerations . . . . . . . . . . . . . . . . . . . 12
 7. Extension Considerations . . . . . . . . . . . . . . . . . . 13
 7.1. RTP Retransmission . . . . . . . . . . . . . . . . . . . 13
 7.2. Generic FEC . . . . . . . . . . . . . . . . . . . . . . . 13
 8. Signalling . . . . . . . . . . . . . . . . . . . . . . . . . 14
 8.1. SDP-Based Signalling . . . . . . . . . . . . . . . . . . 15
 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15
 10. Security Considerations . . . . . . . . . . . . . . . . . . . 15
 11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 15
 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 15
 12.1. Normative References . . . . . . . . . . . . . . . . . . 15
 12.2. Informative References . . . . . . . . . . . . . . . . . 16
 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 17
1. Introduction
 When the Real-time Transport Protocol (RTP) [RFC3550] was designed,
 close to 20 years ago, IP networks were different to those deployed
 at the time of this writing. The virtually ubiquitous deployment of
 Network Address Translators (NAT) and Firewalls has since increased
 the cost and likely-hood of communication failure when using many
 different transport flows. Hence, there is pressure to reduce the
 number of concurrent transport flows used by RTP applications.
 The RTP specification recommends against sending several different
 types of media, for example audio and video, in a single RTP session.
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 The RTP profile for Audio and Video Conferences with Minimal Control
 (RTP/AVP) [RFC3551] mandates a similar restriction. The motivation
 for these limitations is partly to allow lower layer Quality of
 Service (QoS) mechanisms to be used, and partly due to limitations of
 the RTCP timing rules that assumes all media in a session to have
 similar bandwidth. The Session Description Protocol (SDP) [RFC4566]
 is one of the dominant signalling methods for establishing RTP
 sessions, and has enforced this rule by not allowing multiple media
 types for a given destination or set of ICE candidates.
 The fact that these limitations have been in place for so long, in
 addition to RFC 3550 being written without fully considering the use
 of multiple media types in an RTP session, results in a number of
 issues when allowing this behaviour. This memo updates [RFC3550] and
 [RFC3551] with important considerations regarding applicability and
 functionality when using multiple types of media in an RTP session,
 including normative specification of behaviour. This memo makes no
 changes to RTP behaviour when using multiple streams of media of the
 same type (e.g., multiple audio streams or multiple video streams) in
 a single RTP session.
 This memo is structured as follows. First, some basic definitions
 are provided. This is followed by a background that discusses the
 motivation in more detail. A overview of the solution of how to
 provide multiple media types in one RTP session is then presented.
 Next is the formal applicability this specification have followed by
 the normative specification. This is followed by a discussion how
 some RTP/RTCP Extensions are expected to function in the case of
 multiple media types in one RTP session. A specification of the
 requirements on signalling from this specification and a look how
 this is realized in SDP using Bundle
 [I-D.ietf-mmusic-sdp-bundle-negotiation]. The memo ends with the
 security considerations.
2. Definitions
 The following terms are used with supplied definitions:
 Endpoint: A single entity sending or receiving RTP packets. It can
 be decomposed into several functional blocks, but as long as it
 behaves as a single RTP stack entity it is classified as a single
 endpoint.
 Media Stream: A sequence of RTP packets using a single SSRC that
 together carries part or all of the content of a specific Media
 Type from a specific sender source within a given RTP session.
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 Media Type: Audio, video, text or application whose form and meaning
 are defined by a specific real-time application.
 QoS: Quality of Service, i.e. network mechanisms that intended to
 ensure that the packets within a flow or with a specific marking
 are transported with certain properties.
 RTP Session: As defined by [RFC3550], the endpoints belonging to the
 same RTP Session are those that share a single SSRC space. That
 is, those endpoints can see an SSRC identifier transmitted by any
 one of the other endpoints. An endpoint can receive an SSRC
 either as SSRC or as CSRC in RTP and RTCP packets. Thus, the RTP
 Session scope is decided by the endpoints' network interconnection
 topology, in combination with RTP and RTCP forwarding strategies
 deployed by endpoints and any interconnecting middle nodes.
 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 [RFC2119].
3. Motivation
 The existence of NATs and Firewalls at almost all Internet access has
 had implications on protocols like RTP that were designed to use
 multiple transport flows. First of all, the NAT/FW traversal
 solution needs to ensure that all these transport flows are
 established. This has three consequences:
 1. Increased delay to perform the transport flow establishment
 2. The more transport flows, the more state and the more resource
 consumption in the NAT and Firewalls. When the resource
 consumption in NAT/FWs reaches their limits, unexpected
 behaviours usually occur.
 3. More transport flows means a higher risk that some transport flow
 fails to be established, thus preventing the application to
 communicate.
 Using fewer transport flows reduces the risk of communication
 failure, improved establishment behaviour and less load on NAT and
 Firewalls.
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 Furthermore, we note that many RTP-using applications don't utilize
 any network level Quality of Service (QoS) functions. Nor do they
 expect or desire any separation in network treatment of its media
 packets, independent of whether they are audio, video or text. When
 an application has no such desire, it doesn't need to provide a
 transport flow structure that simplifies flow based QoS.
 For applications that don't require different lower-layer QoS for
 different media types, and that have no special requirements for RTP
 extensions or RTCP reporting, the requirement to separate different
 media into different RTP sessions might seem unnecessary. Provided
 the application accepts that all media flows will get similar RTCP
 reporting, using the same RTP session for several types of media at
 once appears a reasonable choice. The architecture ought to be
 agnostic about the type of media being carried in an RTP session to
 the extent possible given the constraints of the protocol.
4. Overview of Solution
 The goal of the solution is to enable each RTP session to contain
 more than just one media type. This includes having multiple RTP
 sessions containing a given media type, for example having three
 sessions containing both video and audio.
 The solution is quite straightforward. The first step is to override
 the SHOULD and SHOULD NOT language of the RTP specification
 [RFC3550]. Similar change is needed to a sentence in Section 6 of
 [RFC3551] that states that "different media types SHALL NOT be
 interleaved or multiplexed within a single RTP Session". This is
 resolved by appropriate exception clauses given that this
 specification and its applicability is followed.
 Within an RTP session where multiple media types have been configured
 for use, an SSRC can only send one type of media during its lifetime
 (i.e., it can switch between different audio codecs, since those are
 both the same type of media, but cannot switch between audio and
 video). Different SSRCs MUST be used for the different media
 sources, the same way multiple media sources of the same media type
 already have to do. The payload type will inform a receiver which
 media type the SSRC is being used for. Thus the payload type MUST be
 unique across all of the payload configurations independent of media
 type that is used in the RTP session.
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 Some few extra considerations within the RTP sessions also needs to
 be considered. RTCP bandwidth and regular reporting suppression (RTP
 /AVPF and RTP/SAVPF) SHOULD be configured to reduce the impact for
 bit-rate variations between streams and media types. It is also
 clarified how timeout calculations are to be done to avoid any
 issues. Certain payload types like FEC also need additional rules.
 The final important part of the solution to this is to use signalling
 and ensure that agreement on using multiple media types in an RTP
 session exists, and how that then is configured. This memo describes
 some existing requirements, while an external reference defines how
 this is accomplished in SDP.
5. Applicability
 This specification has limited applicability, and anyone intending to
 use it needs to ensure that their application and usage meets the
 below criteria.
5.1. Usage of the RTP session
 Before choosing to use this specification, an application implementer
 needs to ensure that they don't have a need for different RTP
 sessions between the media types for some reason. The main rule is
 that if one expects to have equal treatment of all media packets,
 then this specification might be suitable. The equal treatment
 include anything from network level up to RTCP reporting and
 feedback. The document Guidelines for using the Multiplexing
 Features of RTP [I-D.ietf-avtcore-multiplex-guidelines] gives more
 detailed guidance on aspects to consider when choosing how to use RTP
 and specifically sessions. RTP-using applications that need or would
 prefer multiple RTP sessions, but do not require the functionalities
 or behaviours that multiple transport flows give, can consider using
 Multiple RTP Sessions on a Single Lower-Layer Transport
 [I-D.westerlund-avtcore-transport-multiplexing].
 The second important consideration is the resulting behaviour when
 media flows to be sent within a single RTP session does not have
 similar RTCP requirements. There are limitations in the RTCP timing
 rules, and this implies a common RTCP reporting interval across all
 participants in a session. If an RTP session contains flows with
 very different RTCP requirements, for example due to media streams
 bandwidth consumption and packet rate, for example low-rate audio
 coupled with high-quality video, this can result in either excessive
 or insufficient RTCP for some flows, depending how the RTCP session
 bandwidth, and hence reporting interval, is configured. This is
 discussed further in Section 6.4.
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5.2. Signalled Support
 Usage of this specification is not compatible with anyone following
 RFC 3550 and intending to have different RTP sessions for each media
 type. Therefore there needs to be mutual agreement to use multiple
 media types in one RTP session by all participants within that RTP
 session. This agreement has to be determined using signalling in
 most cases.
 This requirement can be a problem for signalling solutions that can't
 negotiate with all participants. For declarative signalling
 solutions, mandating that the session is using multiple media types
 in one RTP session can be a way of attempting to ensure that all
 participants in the RTP session follow the requirement. However, for
 signalling solutions that lack methods for enforcing that a receiver
 supports a specific feature, this can still cause issues.
5.3. Homogeneous Multi-party
 In multiparty communication scenarios it is important to separate two
 different cases. One case is where the RTP session contains multiple
 participants in a common RTP session. This occurs for example in Any
 Source Multicast (ASM) and Transport Translator topologies as defined
 in RTP Topologies [RFC5117]. It can also occur in some
 implementations of RTP mixers that share the same SSRC/CSRC space
 across all participants. The second case is when the RTP session is
 terminated in a middlebox and the other participants sources are
 projected or switched into each RTP session and rewritten on RTP
 header level including SSRC mappings.
 For the first case, with a common RTP session or at least shared SSRC
 /CSRC values, all participants in multiparty communication are
 REQUIRED to support multiple media types in an RTP session. An
 participant using two or more RTP sessions towards a multiparty
 session can't be collapsed into a single session with multiple media
 types. The reason is that in case of multiple RTP sessions, the same
 SSRC value can be use in both RTP sessions without any issues, but
 when collapsed to a single session there is an SSRC collision. In
 addition some collisions can't be represented in the multiple
 separate RTP sessions. For example, in a session with audio and
 video, an SSRC value used for video will not show up in the Audio RTP
 session at the participant using multiple RTP sessions, and thus not
 trigger any collision handling. Thus any application using this type
 of RTP session structure MUST have a homogeneous support for multiple
 media types in one RTP session, or be forced to insert a translator
 node between that participant and the rest of the RTP session.
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 For the second case of separate RTP sessions for each multiparty
 participant and a central node it is possible to have a mix of single
 RTP session users and multiple RTP session users as long as one is
 willing to remap the SSRCs used by a participant with multiple RTP
 sessions into non-used values in the single RTP session SSRC space
 for each of the participants using a single RTP session with multiple
 media types. It can be noted that this type of implementation has to
 understand all types of RTP/RTCP extension being used in the RTP
 sessions to correctly be able to translate them between the RTP
 sessions. It might also suffer issues due to differencies in
 configured RTCP bandwidth and other parameters between the RTP
 sessions. It can also negatively impact the possibility for loop
 detection, as SSRC/CSRC can't be used to detect the loops, instead
 some other media stream identity name space that is common across all
 interconnect parts are needed.
5.4. Reduced number of Payload Types
 An RTP session with multiple media types in it have only a single
 7-bit Payload Type range for all its payload types. Within the 128
 available values, only 96 or less if "Multiplexing RTP Data and
 Control Packets on a Single Port" [RFC5761] is used, all the
 different RTP payload configurations for all the media types need to
 fit in the available space. For most applications this will not be a
 real problem, but the limitation exists and could be encountered.
5.5. Stream Differentiation
 If network level differentiation of the media streams of different
 media types are desired using this specification can cause severe
 limitations. All media streams in an RTP session, independent of the
 media type, will be sent over the same underlying transport flow.
 Any flow-based Quality of Service (QoS) mechanism will be unable to
 provide differentiated treatment between different media types, e.g.
 to prioritize audio over video. If differentiated treatment is
 desired using flow-based QoS, separate RTP sessions over different
 underlying transport flows needs to be used.
 Marking-based QoS scheme like DiffServ can be affected if network
 ingress is the one that performs markings based on flows. Endpoint
 marking where the network API supports marking on individual packet
 level will be unaffected by this specification. However, there exist
 limitations as discussed in [I-D.ietf-avtcore-multiplex-guidelines]
 exist for how different traffic classes can be applied on a single
 RTP media stream.
5.6. Non-compatible Extensions
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 There exist some RTP and RTCP extensions that rely on the existence
 of multiple RTP sessions. If the goal of using an RTP session with
 multiple media types is to have only a single RTP session, then these
 extensions can't be used. If one has no need to have different RTP
 sessions for the media types but is willing to have multiple RTP
 sessions, one for the main media transmission and one for the
 extension, they can be used. It is to be noted that this assumes
 that it is possible to get the extension working when the related RTP
 session contains multiple media types.
 Identified RTP/RTCP extensions that require multiple RTP Sessions
 are:
 RTP Retransmission: RTP Retransmission [RFC4588] has a session
 multiplexed mode. It also has a SSRC multiplexed mode that can be
 used instead. So use the mode that is suitable for the RTP
 application.
 XOR-Based FEC: The RTP Payload Format for Generic Forward Error
 Correction [RFC5109] and its predecessor [RFC2733] requires a
 separate RTP session unless the FEC data is carried in RTP Payload
 for Redundant Audio Data [RFC2198]. However, using the Generic
 FEC with the Redundancy payload has another set of restrictions,
 see Section 7.2.
 Note that the Source-Specific Media Attributes [RFC5576]
 specification defines an SDP syntax (the "FEC" semantic of the
 "ssrc-group" attribute) to signal FEC relationships between
 multiple media streams within a single RTP session. However, this
 can't be used as the FEC repair packets need to have the same SSRC
 value as the source packets being protected. [RFC5576] does not
 normatively update and resolve that restriction. There is ongoing
 work on an ULP extension to allow it be use FEC streams within the
 same RTP Session as the source stream
 [I-D.lennox-payload-ulp-ssrc-mux].
6. RTP Session Specification
 This section defines what needs to be done or avoided to make an RTP
 session with multiple media types function without issues.
6.1. RTP Session
 Section 5.2 of "RTP: A Transport Protocol for Real-Time Applications"
 [RFC3550] states:
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 For example, in a teleconference composed of audio and video media
 encoded separately, each medium SHOULD be carried in a separate
 RTP session with its own destination transport address.
 Separate audio and video streams SHOULD NOT be carried in a single
 RTP session and demultiplexed based on the payload type or SSRC
 fields.
 This specification changes both of these sentences. The first
 sentence is changed to:
 For example, in a teleconference composed of audio and video media
 encoded separately, each medium SHOULD be carried in a separate
 RTP session with its own destination transport address, unless
 specification [RFCXXXX] is followed and the application meets the
 applicability constraints.
 The second sentence is changed to:
 Separate audio and video streams SHOULD NOT be carried in a single
 RTP session and demultiplexed based on the payload type or SSRC
 fields, unless multiplexed based on both SSRC and payload type and
 usage meets what Multiple Media Types in an RTP Session [RFCXXXX]
 specifies.
 Second paragraph of Section 6 in RTP Profile for Audio and Video
 Conferences with Minimal Control [RFC3551] says:
 The payload types currently defined in this profile are assigned
 to exactly one of three categories or media types: audio only,
 video only and those combining audio and video. The media types
 are marked in Tables 4 and 5 as "A", "V" and "AV", respectively.
 Payload types of different media types SHALL NOT be interleaved or
 multiplexed within a single RTP session, but multiple RTP sessions
 MAY be used in parallel to send multiple media types. An RTP
 source MAY change payload types within the same media type during
 a session. See the section "Multiplexing RTP Sessions" of RFC
 3550 for additional explanation.
 This specifications purpose is to violate that existing SHALL NOT
 under certain conditions. Thus also this sentence has to be changed
 to allow for multiple media type's payload types in the same session.
 The above sentence is changed to:
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 Payload types of different media types SHALL NOT be interleaved or
 multiplexed within a single RTP session unless as specified and
 under the restriction in Multiple Media Types in an RTP Session
 [RFCXXXX]. Multiple RTP sessions MAY be used in parallel to send
 multiple media types.
 RFC-Editor Note: Please replace RFCXXXX with the RFC number of this
 specification when assigned.
 We can now go on and discuss the five bullets that are motivating the
 previous in Section 5.2 of the RTP Specification [RFC3550]. They are
 repeated here for the reader's convenience:
 1. If, say, two audio streams shared the same RTP session and the
 same SSRC value, and one were to change encodings and thus
 acquire a different RTP payload type, there would be no general
 way of identifying which stream had changed encodings.
 2. An SSRC is defined to identify a single timing and sequence
 number space. Interleaving multiple payload types would require
 different timing spaces if the media clock rates differ and would
 require different sequence number spaces to tell which payload
 type suffered packet loss.
 3. The RTCP sender and receiver reports (see Section 6.4 of RFC
 3550) can only describe one timing and sequence number space per
 SSRC and do not carry a payload type field.
 4. An RTP mixer would not be able to combine interleaved streams of
 incompatible media into one stream.
 5. Carrying multiple media in one RTP session precludes: the use of
 different network paths or network resource allocations if
 appropriate; reception of a subset of the media if desired, for
 example just audio if video would exceed the available bandwidth;
 and receiver implementations that use separate processes for the
 different media, whereas using separate RTP sessions permits
 either single- or multiple-process implementations.
 Bullets 1 to 3 are all related to that each media source has to use
 one or more unique SSRCs to avoid these issues as mandated below
 (Section 6.2). Bullet 4 can be served by two arguments, first of all
 each SSRC will be associated with a specific media type, communicated
 through the RTP payload type, allowing a middlebox to do media type
 specific operations. The second argument is that in many contexts
 blind combining without additional contexts are anyway not suitable.
 Regarding bullet 5 this is a understood and explicitly stated
 applicability limitations for the method described in this document.
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6.2. Sender Source Restrictions
 A SSRC in the RTP session MUST only send one media type (audio,
 video, text etc.) during the SSRC's lifetime. The main motivation
 is that a given SSRC has its own RTP timestamp and sequence number
 spaces. The same way that you can't send two streams of encoded
 audio on the same SSRC, you can't send one audio and one video
 encoding on the same SSRC. Each media encoding when made into an RTP
 stream needs to have the sole control over the sequence number and
 timestamp space. If not, one would not be able to detect packet loss
 for that particular stream. Nor can one easily determine which clock
 rate a particular SSRCs timestamp will increase with. For additional
 arguments why RTP payload type based multiplexing of multiple media
 streams doesn't work see Appendix A in
 [I-D.ietf-avtcore-multiplex-guidelines].
6.3. Payload Type Applicability
 Most Payload Types have a native media type, like an audio codec is
 natural belonging to the audio media type. However, there exist a
 number of RTP payload types that don't have a native media type. For
 example, transport robustness mechanisms like RTP Retransmission
 [RFC4588] and Generic FEC [RFC5109] inherit their media type from
 what they protect. RTP Retransmission is explicitly bound to the
 payload type it is protecting, and thus will inherit it. However
 Generic FEC is a excellent example of an RTP payload type that has no
 natural media type. The media type for what it protects is not
 relevant as it is the recovered RTP packets that have a particular
 media type, and thus Generic FEC is best categorized as an
 application media type.
 The above discussion is relevant to what limitations exist for RTP
 payload type usage within an RTP session that has multiple media
 types. In fact this document (Section 7.2) suggest that for usage of
 Generic FEC (XOR-based) as defined in RFC 5109 can actually use a
 single media type when used with independent RTP sessions for source
 and repair data.
 Note a particular SSRC carrying Generic FEC will clearly only
 protect a specific SSRC and thus that instance is bound to the
 SSRC's media type. For this specific case, it is possible to have
 one be applicable to both. However, in cases when the signalling
 is setup to enable fall back to using separate RTP sessions, then
 using a different media type, e.g. application, than the media
 being protected can create issues.
6.4. RTCP Considerations
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 Guidelines for handling RTCP when sending multiple media streams with
 disparate rates in a single RTP session are outlined in
 [I-D.ietf-avtcore-rtp-multi-stream]. These guidelines apply when
 sending multiple types of media in a single RTP session if the
 different types of media have different rates.
7. Extension Considerations
 This section discusses the impact on some RTP/RTCP extensions due to
 usage of multiple media types in on RTP session. Only extensions
 where something worth noting has been included.
7.1. RTP Retransmission
 SSRC-multiplexed RTP retransmission [RFC4588] is actually very
 straightforward. Each retransmission RTP payload type is explicitly
 connected to an associated payload type. If retransmission is only
 to be used with a subset of all payload types, this is not a problem,
 as it will be evident from the retransmission payload types which
 payload types that have retransmission enabled for them.
 Session-multiplexed RTP retransmission is also possible to use where
 an retransmission session contains the retransmissions of the
 associated payload types in the source RTP session. The only
 difference to previously is that the source RTP session is one which
 contains multiple media types. Thus it is even more likely that only
 a subset of the source RTP session's payload types and SSRCs are
 actually retransmitted.
 Open Issue: When using SDP to signal retransmission for one RTP
 session with multiple media types and one RTP session for the
 retransmission data will cause a situation where one will have
 multiple m= lines grouped using FID and the ones belonging to
 respective RTP session being grouped using BUNDLE. This usage might
 contradict both the FID semantics [RFC5888] and an assumption in the
 RTP retransmission specification [RFC4588].
7.2. Generic FEC
 The RTP Payload Format for Generic Forward Error Correction
 [RFC5109], and also its predecessor [RFC2733], requires some
 considerations, and they are different depending on what type of
 configuration of usage one has.
 Independent RTP Sessions, i.e. where source and repair data are sent
 in different RTP sessions. As this mode of configuration requires
 different RTP session, there has to be at least one RTP session for
 source data, this session can be one using multiple media types. The
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 repair session only needs one RTP Payload type indicating repair
 data, i.e. x/ulpfec or x/parityfec depending if RFC 5109 or RFC 2733
 is used. The media type in this session is not relevant and can in
 theory be any of the defined ones. It is RECOMMENDED that one uses
 "Application".
 In stream, using RTP Payload for Redundant Audio Data [RFC2198]
 combining repair and source data in the same packets. This is
 possible to use within a single RTP session. However, the usage and
 configuration of the payload types can create an issue. First of all
 it might be necessary to have one payload type per media type for the
 FEC repair data payload format, i.e. one for audio/ulpfec and one
 for text/ulpfec if audio and text are combined in an RTP session.
 Secondly each combination of source payload and its FEC repair data
 has to be an explicit configured payload type. This has potential
 for making the limitation of RTP payload types available into a real
 issue.
8. Signalling
 The Signalling requirements
 Establishing an RTP session with multiple media types requires
 signalling. This signalling needs to fulfil the following
 requirements:
 1. Ensure that any participant in the RTP session is aware that this
 is an RTP session with multiple media types.
 2. Ensure that the payload types in use in the RTP session are using
 unique values, with no overlap between the media types.
 3. Configure the RTP session level parameters, such as RTCP RR and
 RS bandwidth, AVPF trr-int, underlying transport, the RTCP
 extensions in use, and security parameters, commonly for the RTP
 session.
 4. RTP and RTCP functions that can be bound to a particular media
 type SHOULD be reused when possible also for other media types,
 instead of having to be configured for multiple code-points.
 Note: In some cases one will not have a choice but to use
 multiple configurations.
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Internet-Draft Multiple Media Types in an RTP Session October 2014
8.1. SDP-Based Signalling
 The signalling of multiple media types in one RTP session in SDP is
 specified in "Multiplexing Negotiation Using Session Description
 Protocol (SDP) Port Numbers"
 [I-D.ietf-mmusic-sdp-bundle-negotiation].
9. IANA Considerations
 This document makes no request of IANA.
 Note to RFC Editor: this section is to be removed on publication as
 an RFC.
10. Security Considerations
 Having an RTP session with multiple media types doesn't change the
 methods for securing a particular RTP session. One possible
 difference is that the different media have often had different
 security requirements. When combining multiple media types in one
 session, their security requirements also have to be combined by
 selecting the most demanding for each property. Thus having multiple
 media types can result in increased overhead for security for some
 media types to ensure that all requirements are meet.
 Otherwise, the recommendations for how to configure and RTP session
 do not add any additional requirements compared to normal RTP, except
 for the need to be able to ensure that the participants are aware
 that it is a multiple media type session. If not that is ensured it
 can cause issues in the RTP session for both the unaware and the
 aware one. Similar issues can also be produced in an normal RTP
 session by creating configurations for different end-points that
 doesn't match each other.
11. Acknowledgements
 The authors would like to thank Christer Holmberg, Gunnar Hellstroem,
 and Charles Eckel for the feedback on the document.
12. References
12.1. Normative References
 [I-D.ietf-avtcore-rtp-multi-stream]
 Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
 "Sending Multiple Media Streams in a Single RTP Session",
 draft-ietf-avtcore-rtp-multi-stream-05 (work in progress),
 July 2014.
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Internet-Draft Multiple Media Types in an RTP Session October 2014
 [I-D.ietf-mmusic-sdp-bundle-negotiation]
 Holmberg, C., Alvestrand, H., and C. Jennings,
 "Negotiating Media Multiplexing Using the Session
 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
 negotiation-11 (work in progress), September 2014.
 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
 Requirement Levels", BCP 14, RFC 2119, March 1997.
 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
 Jacobson, "RTP: A Transport Protocol for Real-Time
 Applications", STD 64, RFC 3550, July 2003.
 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
 Video Conferences with Minimal Control", STD 65, RFC 3551,
 July 2003.
12.2. Informative References
 [I-D.ietf-avtcore-multiplex-guidelines]
 Westerlund, M., Perkins, C., and H. Alvestrand,
 "Guidelines for using the Multiplexing Features of RTP to
 Support Multiple Media Streams", draft-ietf-avtcore-
 multiplex-guidelines-02 (work in progress), January 2014.
 [I-D.lennox-payload-ulp-ssrc-mux]
 Lennox, J., "Supporting Source-Multiplexing of the Real-
 Time Transport Protocol (RTP) Payload for Generic Forward
 Error Correction", draft-lennox-payload-ulp-ssrc-mux-00
 (work in progress), February 2013.
 [I-D.westerlund-avtcore-transport-multiplexing]
 Westerlund, M. and C. Perkins, "Multiplexing Multiple RTP
 Sessions onto a Single Lower-Layer Transport", draft-
 westerlund-avtcore-transport-multiplexing-07 (work in
 progress), October 2013.
 [RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
 Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
 Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
 September 1997.
 [RFC2733] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format
 for Generic Forward Error Correction", RFC 2733, December
 1999.
 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
 Description Protocol", RFC 4566, July 2006.
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Internet-Draft Multiple Media Types in an RTP Session October 2014
 [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
 Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
 July 2006.
 [RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
 Correction", RFC 5109, December 2007.
 [RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
 January 2008.
 [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
 Media Attributes in the Session Description Protocol
 (SDP)", RFC 5576, June 2009.
 [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
 Control Packets on a Single Port", RFC 5761, April 2010.
 [RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
 Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
Authors' Addresses
 Magnus Westerlund
 Ericsson
 Farogatan 6
 SE-164 80 Kista
 Sweden
 Phone: +46 10 714 82 87
 Email: magnus.westerlund@ericsson.com
 Colin Perkins
 University of Glasgow
 School of Computing Science
 Glasgow G12 8QQ
 United Kingdom
 Email: csp@csperkins.org
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Internet-Draft Multiple Media Types in an RTP Session October 2014
 Jonathan Lennox
 Vidyo, Inc.
 433 Hackensack Avenue
 Seventh Floor
 Hackensack, NJ 07601
 US
 Email: jonathan@vidyo.com
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