Sending Multiple Types of Media in a Single RTP Session
draft-ietf-avtcore-multi-media-rtp-session-13

Abstract

This document specifies how an RTP session can contain RTP 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

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This Internet-Draft will expire on June 20, 2016.

Copyright Notice

Copyright (c) 2015 IETF Trust and the persons identified as the document authors. All rights reserved.

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Table of Contents

• 1. Introduction
• 2. Terminology
• 3. Background and Motivation
• 4. Applicability
• 5. Using Multiple Media Types in a Single RTP Session
• 5.1. Allowing Multiple Media Types in an RTP Session
• 5.2. Demultiplexing media types within an RTP session
• 5.3. Per-SSRC Media Type Restrictions
• 5.4. RTCP Considerations
• 6. Extension Considerations
• 6.1. RTP Retransmission Payload Format
• 6.2. RTP Payload Format for Generic FEC
• 6.3. RTP Payload Format for Redundant Audio
• 7. Signalling
• 8. Security Considerations
• 9. IANA Considerations
• 10. Acknowledgements
• 11. References
• 11.1. Normative References
• 11.2. Informative References
• Authors' Addresses

1. Introduction

The Real-time Transport Protocol [RFC3550] was designed to use separate RTP sessions to transport different types of media. This implies that different transport layer flows are used for different RTP streams. For example, a video conferencing application might send audio and video traffic RTP flows on separate UDP ports. With increased use of network address/port translation, firewalls, and other middleboxes it is, however, becoming difficult to establish multiple transport layer flows between endpoints. Hence, there is pressure to reduce the number of concurrent transport flows used by RTP applications.

This memo updates [RFC3550] and [RFC3551] to allow multiple media types to be sent in a single RTP session in certain cases, thereby reducing the number of transport layer flows that are needed. It makes no changes to RTP behaviour when using multiple RTP streams containing media of the same type (e.g., multiple audio streams or multiple video streams) in a single RTP session. However [I-D.ietf-avtcore-rtp-multi-stream] provides important clarifications to RTP behaviour in that case.

This memo is structured as follows. Section 2 defines terminology. Section 3 further describes the background to, and motivation for, this memo and Section 4 describes the scenarios where this memo is applicable. Section 5 discusses issues arising from the base RTP and RTCP specification when using multiple types of media in a single RTP session, while Section 6 considers the impact of RTP extensions. We discuss signalling in Section 7. Finally, security considerations are discussed in Section 8.

2. Terminology

The terms Encoded Stream, Endpoint, Media Source, RTP Session, and RTP Stream are used as defined in [RFC7656]. We also define the following terms:

Media Type:

The general type of media data used by a real-time application. The media type corresponds to the value used in the <media> field of an SDP m= line. The media types defined at the time of this writing are "audio", "video", "text", "image", "application", and "message". [RFC4566] [RFC6466]

Quality of Service (QoS):

Network mechanisms that are intended to ensure that the packets within a flow or with a specific marking are transported with certain properties.

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in [RFC2119].

3. Background and Motivation

RTP was designed to support multimedia sessions, containing multiple types of media sent simultaneously, by using multiple transport layer flows. The existence of network address translators, firewalls, and other middleboxes complicates this, however, since a mechanism is needed to ensure that all the transport layer flows needed by the application can be established. This has three consequences:

1. increased delay to establish a complete session, since each of the transport layer flows needs to be negotiated and established;
2. increased state and resource consumption in the middleboxes that can lead to unexpected behaviour when middlebox resource limits are reached; and
3. increased risk that a subset of the transport layer flows will fail to be established, thus preventing the application from communicating.

Using fewer transport layer flows can hence be seen to reduce the risk of communication failure, and can lead to improved reliability and performance.

One of the benefits of using multiple transport layer flows is that it makes it easy to use network layer quality of service (QoS) mechanisms to give differentiated performance for different flows. However, we note that many RTP-using application don't use network QoS features, and don't expect or desire any separation in network treatment of their 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.

Given the above issues, it might seem appropriate for RTP-based applications to send all their RTP streams bundled into one RTP session, running over a single transport layer flow. However, this is prohibited by the RTP specification, because the design of RTP makes certain assumptions that can be incompatible with sending multiple media types in a single RTP session. Specifically, the RTP control protocol (RTCP) timing rules assume that all RTP media flows in a single RTP session have broadly similar RTCP reporting and feedback requirements, which can be problematic when different types of media are multiplexed together. Various RTP extensions also make assumptions about SSRC use and RTCP reporting that are incompatible with sending different media types in a single RTP session.

This memo updates [RFC3550] and [RFC3551] to allow RTP sessions to contain more than one media type in certain circumstances, and gives guidance on when it is safe to send multiple media types in a single RTP session.

4. Applicability

This specification has limited applicability, and anyone intending to use it needs to ensure that their application and use case meets the following criteria:

Equal treatment of media:

The use of a single RTP session normally results in similar network treatment for all types of media used within the session. Applications that require significantly different network quality of service (QoS) or RTCP configuration for different RTP streams are better suited by sending those RTP streams in separate RTP session, using separate transport layer flows for each, since that gives greater flexibility. Further guidance on how to provide differential treatment for some media is given in [I-D.ietf-avtcore-multiplex-guidelines] and [RFC7657].

Compatible RTCP Behaviour:

The RTCP timing rules enforce a single RTCP reporting interval for all participants in an RTP session. Flows with very different media sending rate or RTCP feedback requirements cannot be multiplexed together, since this leads to either excessive or insufficient RTCP for some flows, depending on how the RTCP session bandwidth, and hence reporting interval, is configured. For example, it is likely infeasible to find a single RTCP configuration that simultaneously suits both a low-rate audio flow with no feedback, and a high-quality video flow with sophisticated RTCP-based feedback. Thus, combining these into a single RTP session is difficult and/or inadvisable.

Signalled Support:

The extensions defined in this memo are not compatible with unmodified [RFC3550]-compatible endpoints. Their use requires signalling and mutual agreement by all participants within an RTP session. 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.

Consistent support for multiparty RTP sessions:

If it is desired to send multiple types of media in a multiparty RTP session, then all participants in that session need to support sending multiple type of media in a single RTP session. It is not possible, in the general case, to implement a gateway that can interconnect an endpoint using multiple types of media sent using separate RTP sessions, with one or more endpoints that send multiple types of media in a single RTP session.

One reason for this is that the same SSRC value can safely be used for different streams in multiple RTP sessions, but when collapsed to a single RTP session there is an SSRC collision. This would not be an issue, since SSRC collision detection will resolve the conflict, except that some RTP payload formats and extensions use matching SSRCs to identify related flows, and break when a single RTP session is used.

A middlebox that remaps SSRC values when combining multiple RTP sessions into one also needs to be aware of all possible RTCP packet types that might be used, so that it can remap the SSRC values in those packets. This is impossible to do without restricting the set of RTCP packet types that can be used to those that are known by the middlebox. Such a middlebox might also have difficulty due to differences in configured RTCP bandwidth and other parameters between the RTP sessions.

Finally, the use of a middlebox that translates SSRC values can negatively impact the possibility for loop detection, as SSRC/CSRC can't be used to detect the loops; instead some other RTP stream or media source identity name space that is common across all interconnect parts is needed.

Ability to operate with limited payload type space:

An RTP session has only a single 7-bit payload type space for all its payload type numbers. Some applications might find this space limiting when using different media types and RTP payload formats within a single RTP session.

Avoids incompatible Extensions:

Some RTP and RTCP extensions rely on the existence of multiple RTP sessions and relate RTP streams between sessions. Others report on particular media types, and cannot be used with other media types. Applications that send multiple types of media into a single RTP session need to avoid such extensions.

5. Using Multiple Media Types in a Single RTP Session

This section defines what needs to be done or avoided to make an RTP session with multiple media types function without issues.

5.1. Allowing Multiple Media Types in an RTP Session

Section 5.2 of "RTP: A Transport Protocol for Real-Time Applications" [RFC3550] states:

• 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 media sources SHOULD NOT be carried in a single RTP session, unless the guidelines specified in [RFCXXXX] are followed.

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 specification's purpose is to override that existing SHALL NOT under certain conditions. Thus this sentence also has to be changed to allow for multiple media type's payload types in the same session. The sentence containing "SHALL NOT" in the above paragraph is changed to:

• Payload types of different media types SHALL NOT be interleaved or multiplexed within a single RTP session unless [RFCXXXX] is used, and the application conforms to the applicability constraints. 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.

5.2. Demultiplexing media types within an RTP session

When receiving packets from a transport layer flow, an endpoint will first separate the RTP and RTCP packets from the non-RTP packets, and pass them to the RTP/RTCP protocol handler. The RTP and RTCP packets are then demultiplexed based on their SSRC into the different RTP streams. For each RTP stream, incoming RTCP packets are processed, and the RTP payload type is used to select the appropriate media decoder. This process remains the same irrespective of whether multiple media types are sent in a single RTP session or not.

As explained below, it is important to note that the RTP payload type is never used to distinguish RTP streams. The RTP packets are demultiplexed into RTP streams based on their SSRC, then the RTP payload type is used to select the correct media decoding pathway for each RTP stream.

5.3. Per-SSRC Media Type Restrictions

An SSRC in an RTP session can change between media formats of the same type, subject to certain restrictions [RFC7160], but MUST NOT change media type during its lifetime. For example, an SSRC can change between different audio formats, but cannot start sending audio then change to sending video. The lifetime of an SSRC ends when an RTCP BYE packet for that SSRC is sent, or when it ceases transmission for long enough that it times out for the other participants in the session.

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 encoded streams of audio with the same SSRC, you can't send one encoded audio and one encoded video stream with the same SSRC. Each encoded stream 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 encoded 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 sources doesn't work, see [I-D.ietf-avtcore-multiplex-guidelines].

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.

5.4. RTCP Considerations

When sending multiple types of media that have different rates in a single RTP session, endpoints MUST follow the guidelines for handling RTCP described in Section 7 of [I-D.ietf-avtcore-rtp-multi-stream].

6. Extension Considerations

This section outlines known issues and incompatibilities with RTP and RTCP extensions when multiple media types are used in a single RTP sessions. Future extensions to RTP and RTCP need to consider, and document, any potential incompatibility.

6.1. RTP Retransmission Payload Format

The RTP Retransmission Payload Format [RFC4588] can operate in either SSRC-multiplexed mode or session-multiplex mode.

In SSRC-multiplexed mode, retransmitted RTP packets are sent in the same RTP session as the original packets, but use a different SSRC with the same RTCP SDES CNAME. If each endpoint sends only a single original RTP stream and a single retransmission RTP stream in the session, this is sufficient. If an endpoint sends multiple original and retransmission RTP streams, as would occur when sending multiple media types in a single RTP session, then each original RTP stream and the retransmission RTP stream have to be associated using heuristics. By having retransmission requests outstanding for only one SSRC not yet mapped, a receiver can determine the binding between original and retransmission RTP stream. Another alternative is the use of different RTP payload types, allowing the signalled "apt" (associated payload type) parameter of the RTP retransmission payload format to be used to associate retransmitted and original packets.

Session-multiplexed mode sends the retransmission RTP stream in a separate RTP session to the original RTP stream, but using the same SSRC for each, with association being done by matching SSRCs between the two sessions. This is unaffected by the use of multiple media types in a single RTP session, since each media type will be sent using a different SSRC in the original RTP session, and the same SSRCs can be used in the retransmission session, allowing the streams to be associated. This can be signalled using SDP with the BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation] and FID grouping [RFC5888] extensions. These SDP extensions require each "m=" line to only be included in a single FID group, but the RTP retransmission payload format uses FID groups to indicate the m= lines that form an original and retransmission pair. Accordingly, when using the BUNDLE extension to allow multiple media types to be sent in a single RTP session, each original media source (m= line) that is retransmitted needs a corresponding m= line in the retransmission RTP session. In case there are multiple media lines for retransmission, these media lines will form an independent BUNDLE group from the BUNDLE group with the source streams.

An example SDP fragment showing the grouping structures is provided in Figure 1. This example is not legal SDP and only the most important attributes have been left in place. Note that this SDP is not an initial BUNDLE offer. As can be seen there are two bundle groups, one for the source RTP session and one for the retransmissions. Then each of the media sources are grouped with its retransmission flow using FID, resulting in three more groupings.

       a=group:BUNDLE foo bar fiz
       a=group:BUNDLE zoo kelp glo
       a=group:FID foo zoo
       a=group:FID bar kelp
       a=group:FID fiz glo
       m=audio 10000 RTP/AVP 0
       a=mid:foo
       a=rtpmap:0 PCMU/8000
       m=video 10000 RTP/AVP 31
       a=mid:bar
       a=rtpmap:31 H261/90000
       m=video 10000 RTP/AVP 31
       a=mid:fiz
       a=rtpmap:31 H261/90000
       m=audio 40000 RTP/AVPF 99
       a=rtpmap:99 rtx/90000
       a=fmtp:99 apt=0;rtx-time=3000
       a=mid:zoo
       m=video 40000 RTP/AVPF 100
       a=rtpmap:100 rtx/90000
       a=fmtp:199 apt=31;rtx-time=3000
       a=mid:kelp
       m=video 40000 RTP/AVPF 100
       a=rtpmap:100 rtx/90000
       a=fmtp:199 apt=31;rtx-time=3000
       a=mid:glo