- Pike Programming Language

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Internet Engineering Task Force (IETF)                        C. Perkins
Request for Comments: 8834                         University of Glasgow
Category: Standards Track                                  M. Westerlund
ISSN: 2070-1721                                                 Ericsson
                                                                  J. Ott
                                             Technical University Munich
                                                            January 2021

Media Transport and Use of RTP in WebRTC

Abstract

The framework for Web Real-Time Communication (WebRTC) provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. between two peers' web browsers. This memo describes the media transport aspects of the WebRTC framework. It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context and gives requirements for which RTP features, profiles, and extensions need to be supported.

Status of This Memo

This is an Internet Standards Track document.

This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Further information on Internet Standards is available in Section 2 of RFC 7841.

Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at https://www.rfc-editor.org/info/rfc8834.

Copyright Notice

This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://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 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.

   1.  Introduction
   2.  Rationale
   3.  Terminology
   4.  WebRTC Use of RTP: Core Protocols
     4.1.  RTP and RTCP
     4.2.  Choice of the RTP Profile
     4.3.  Choice of RTP Payload Formats
     4.4.  Use of RTP Sessions
     4.5.  RTP and RTCP Multiplexing
     4.6.  Reduced Size RTCP
     4.7.  Symmetric RTP/RTCP
     4.8.  Choice of RTP Synchronization Source (SSRC)
     4.9.  Generation of the RTCP Canonical Name (CNAME)
     4.10. Handling of Leap Seconds
   5.  WebRTC Use of RTP: Extensions
     5.1.  Conferencing Extensions and Topologies
       5.1.1.  Full Intra Request (FIR)
       5.1.2.  Picture Loss Indication (PLI)
       5.1.3.  Slice Loss Indication (SLI)
       5.1.4.  Reference Picture Selection Indication (RPSI)
       5.1.5.  Temporal-Spatial Trade-Off Request (TSTR)
       5.1.6.  Temporary Maximum Media Stream Bit Rate Request (TMMBR)
     5.2.  Header Extensions
       5.2.1.  Rapid Synchronization
       5.2.2.  Client-to-Mixer Audio Level
       5.2.3.  Mixer-to-Client Audio Level
       5.2.4.  Media Stream Identification
       5.2.5.  Coordination of Video Orientation
   6.  WebRTC Use of RTP: Improving Transport Robustness
     6.1.  Negative Acknowledgements and RTP Retransmission
     6.2.  Forward Error Correction (FEC)
   7.  WebRTC Use of RTP: Rate Control and Media Adaptation
     7.1.  Boundary Conditions and Circuit Breakers
     7.2.  Congestion Control Interoperability and Legacy Systems
   8.  WebRTC Use of RTP: Performance Monitoring
   9.  WebRTC Use of RTP: Future Extensions
   10. Signaling Considerations
   11. WebRTC API Considerations
   12. RTP Implementation Considerations
     12.1.  Configuration and Use of RTP Sessions
       12.1.1.  Use of Multiple Media Sources within an RTP Session
       12.1.2.  Use of Multiple RTP Sessions
       12.1.3.  Differentiated Treatment of RTP Streams
     12.2.  Media Source, RTP Streams, and Participant Identification
       12.2.1.  Media Source Identification
       12.2.2.  SSRC Collision Detection
       12.2.3.  Media Synchronization Context
   13. Security Considerations
   14. IANA Considerations
   15. References
     15.1.  Normative References
     15.2.  Informative References
   Acknowledgements
   Authors' Addresses

1. Introduction

The Real-time Transport Protocol (RTP) [RFC3550] provides a framework for delivery of audio and video teleconferencing data and other real- time media applications. Previous work has defined the RTP protocol, along with numerous profiles, payload formats, and other extensions. When combined with appropriate signaling, these form the basis for many teleconferencing systems.

The Web Real-Time Communication (WebRTC) framework provides the protocol building blocks to support direct, interactive, real-time communication using audio, video, collaboration, games, etc. between two peers' web browsers. This memo describes how the RTP framework is to be used in the WebRTC context. It proposes a baseline set of RTP features that are to be implemented by all WebRTC endpoints, along with suggested extensions for enhanced functionality.

This memo specifies a protocol intended for use within the WebRTC framework but is not restricted to that context. An overview of the WebRTC framework is given in [RFC8825].

The structure of this memo is as follows. Section 2 outlines our rationale for preparing this memo and choosing these RTP features. Section 3 defines terminology. Requirements for core RTP protocols are described in Section 4, and suggested RTP extensions are described in Section 5. Section 6 outlines mechanisms that can increase robustness to network problems, while Section 7 describes congestion control and rate adaptation mechanisms. The discussion of mandated RTP mechanisms concludes in Section 8 with a review of performance monitoring and network management tools. Section 9 gives some guidelines for future incorporation of other RTP and RTP Control Protocol (RTCP) extensions into this framework. Section 10 describes requirements placed on the signaling channel. Section 11 discusses the relationship between features of the RTP framework and the WebRTC application programming interface (API), and Section 12 discusses RTP implementation considerations. The memo concludes with security considerations (Section 13) and IANA considerations (Section 14).

2. Rationale

The RTP framework comprises the RTP data transfer protocol, the RTP control protocol, and numerous RTP payload formats, profiles, and extensions. This range of add-ons has allowed RTP to meet various needs that were not envisaged by the original protocol designers and support many new media encodings, but it raises the question of what extensions are to be supported by new implementations. The development of the WebRTC framework provides an opportunity to review the available RTP features and extensions and define a common baseline RTP feature set for all WebRTC endpoints. This builds on the past 20 years of RTP development to mandate the use of extensions that have shown widespread utility, while still remaining compatible with the wide installed base of RTP implementations where possible.

RTP and RTCP extensions that are not discussed in this document can be implemented by WebRTC endpoints if they are beneficial for new use cases. However, they are not necessary to address the WebRTC use cases and requirements identified in [RFC7478].

While the baseline set of RTP features and extensions defined in this memo is targeted at the requirements of the WebRTC framework, it is expected to be broadly useful for other conferencing-related uses of RTP. In particular, it is likely that this set of RTP features and extensions will be appropriate for other desktop or mobile video- conferencing systems, or for room-based high-quality telepresence applications.

3. Terminology

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 BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here. Lower- or mixed-case uses of these key words are not to be interpreted as carrying special significance in this memo.

We define the following additional terms:

   WebRTC MediaStream:  The MediaStream concept defined by the W3C in
   
      the WebRTC API [W3C.WD-mediacapture-streams].  A MediaStream
      consists of zero or more MediaStreamTracks.
   
   MediaStreamTrack:  Part of the MediaStream concept defined by the W3C
      in the WebRTC API [W3C.WD-mediacapture-streams].  A
      MediaStreamTrack is an individual stream of media from any type of
      media source such as a microphone or a camera, but conceptual
      sources such as an audio mix or a video composition are also
      possible.
   
   Transport-layer flow:  A unidirectional flow of transport packets
      that are identified by a particular 5-tuple of source IP address,
      source port, destination IP address, destination port, and
      transport protocol.
   
   Bidirectional transport-layer flow:  A bidirectional transport-layer
      flow is a transport-layer flow that is symmetric.  That is, the
      transport-layer flow in the reverse direction has a 5-tuple where
      the source and destination address and ports are swapped compared
      to the forward path transport-layer flow, and the transport
      protocol is the same.

This document uses the terminology from [RFC7656] and [RFC8825]. Other terms are used according to their definitions from the RTP specification [RFC3550]. In particular, note the following frequently used terms: RTP stream, RTP session, and endpoint.

4. WebRTC Use of RTP: Core Protocols

The following sections describe the core features of RTP and RTCP that need to be implemented, along with the mandated RTP profiles. Also described are the core extensions providing essential features that all WebRTC endpoints need to implement to function effectively on today's networks.

4.1. RTP and RTCP

The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be implemented as the media transport protocol for WebRTC. RTP itself comprises two parts: the RTP data transfer protocol and the RTP Control Protocol (RTCP). RTCP is a fundamental and integral part of RTP and MUST be implemented and used in all WebRTC endpoints.

The following RTP and RTCP features are sometimes omitted in limited- functionality implementations of RTP, but they are REQUIRED in all WebRTC endpoints:

Support for use of multiple simultaneous synchronization source (SSRC) values in a single RTP session, including support for RTP endpoints that send many SSRC values simultaneously, following [RFC3550] and [RFC8108]. The RTCP optimizations for multi-SSRC sessions defined in [RFC8861] MAY be supported; if supported, the usage MUST be signaled.

Random choice of SSRC on joining a session; collision detection and resolution for SSRC values (see also Section 4.8).

Support for reception of RTP data packets containing contributing source (CSRC) lists, as generated by RTP mixers, and RTCP packets relating to CSRCs.

Sending correct synchronization information in the RTCP Sender Reports, to allow receivers to implement lip synchronization; see Section 5.2.1 regarding support for the rapid RTP synchronization extensions.

Support for multiple synchronization contexts. Participants that send multiple simultaneous RTP packet streams SHOULD do so as part of a single synchronization context, using a single RTCP CNAME for all streams and allowing receivers to play the streams out in a synchronized manner. For compatibility with potential future versions of this specification, or for interoperability with non- WebRTC devices through a gateway, receivers MUST support multiple synchronization contexts, indicated by the use of multiple RTCP CNAMEs in an RTP session. This specification mandates the usage of a single CNAME when sending RTP streams in some circumstances; see Section 4.9.

Support for sending and receiving RTCP Sender Report (SR), Receiver Report (RR), Source Description (SDES), and BYE packet types. Note that support for other RTCP packet types is OPTIONAL unless mandated by other parts of this specification. Note that additional RTCP packet types are used by the RTP/SAVPF profile (Section 4.2) and the other RTCP extensions (Section 5). WebRTC endpoints that implement the Session Description Protocol (SDP) bundle negotiation extension will use the SDP Grouping Framework "mid" attribute to identify media streams. Such endpoints MUST implement the RTCP SDES media identification (MID) item described in [RFC8843].

Support for multiple endpoints in a single RTP session, and for scaling the RTCP transmission interval according to the number of participants in the session; support for randomized RTCP transmission intervals to avoid synchronization of RTCP reports; support for RTCP timer reconsideration (Section 6.3.6 of [RFC3550]) and reverse reconsideration (Section 6.3.4 of [RFC3550]).

Support for configuring the RTCP bandwidth as a fraction of the media bandwidth, and for configuring the fraction of the RTCP bandwidth allocated to senders -- e.g., using the SDP "b=" line [RFC4566] [RFC3556].

Support for the reduced minimum RTCP reporting interval described in Section 6.2 of [RFC3550]. When using the reduced minimum RTCP reporting interval, the fixed (nonreduced) minimum interval MUST be used when calculating the participant timeout interval (see Sections 6.2 and 6.3.5 of [RFC3550]). The delay before sending the initial compound RTCP packet can be set to zero (see Section 6.2 of [RFC3550] as updated by [RFC8108]).

Support for discontinuous transmission. RTP allows endpoints to pause and resume transmission at any time. When resuming, the RTP sequence number will increase by one, as usual, while the increase in the RTP timestamp value will depend on the duration of the pause. Discontinuous transmission is most commonly used with some audio payload formats, but it is not audio specific and can be used with any RTP payload format.

Ignore unknown RTCP packet types and RTP header extensions. This is to ensure robust handling of future extensions, middlebox behaviors, etc., that can result in receiving RTP header extensions or RTCP packet types that were not signaled. If a compound RTCP packet that contains a mixture of known and unknown RTCP packet types is received, the known packet types need to be processed as usual, with only the unknown packet types being discarded.

It is known that a significant number of legacy RTP implementations, especially those targeted at systems with only Voice over IP (VoIP), do not support all of the above features and in some cases do not support RTCP at all. Implementers are advised to consider the requirements for graceful degradation when interoperating with legacy implementations.

Other implementation considerations are discussed in Section 12.

4.2. Choice of the RTP Profile

The complete specification of RTP for a particular application domain requires the choice of an RTP profile. For WebRTC use, the extended secure RTP profile for RTCP-based feedback (RTP/SAVPF) [RFC5124], as extended by [RFC7007], MUST be implemented. The RTP/SAVPF profile is the combination of the basic RTP/AVP profile [RFC3551], the RTP profile for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP profile (RTP/SAVP) [RFC3711].

The RTCP-based feedback extensions [RFC4585] are needed for the improved RTCP timer model. This allows more flexible transmission of RTCP packets in response to events, rather than strictly according to bandwidth, and is vital for being able to report congestion signals as well as media events. These extensions also allow saving RTCP bandwidth, and an endpoint will commonly only use the full RTCP bandwidth allocation if there are many events that require feedback. The timer rules are also needed to make use of the RTP conferencing extensions discussed in Section 5.1.

      |  Note: The enhanced RTCP timer model defined in the RTP/AVPF
      |  profile is backwards compatible with legacy systems that
      |  implement only the RTP/AVP or RTP/SAVP profile, given some
      |  constraints on parameter configuration such as the RTCP
      |  bandwidth value and "trr-int".  The most important factor for
      |  interworking with RTP/(S)AVP endpoints via a gateway is to set
      |  the "trr-int" parameter to a value representing 4 seconds; see
      |  Section 7.1.3 of [RFC8108].

The secure RTP (SRTP) profile extensions [RFC3711] are needed to provide media encryption, integrity protection, replay protection, and a limited form of source authentication. WebRTC endpoints MUST NOT send packets using the basic RTP/AVP profile or the RTP/AVPF profile; they MUST employ the full RTP/SAVPF profile to protect all RTP and RTCP packets that are generated. In other words, implementations MUST use SRTP and Secure RTCP (SRTCP). The RTP/SAVPF profile MUST be configured using the cipher suites, DTLS-SRTP protection profiles, keying mechanisms, and other parameters described in [RFC8827].

4.3. Choice of RTP Payload Formats

Mandatory-to-implement audio codecs and RTP payload formats for WebRTC endpoints are defined in [RFC7874]. Mandatory-to-implement video codecs and RTP payload formats for WebRTC endpoints are defined in [RFC7742]. WebRTC endpoints MAY additionally implement any other codec for which an RTP payload format and associated signaling has been defined.

WebRTC endpoints cannot assume that the other participants in an RTP session understand any RTP payload format, no matter how common. The mapping between RTP payload type numbers and specific configurations of particular RTP payload formats MUST be agreed before those payload types/formats can be used. In an SDP context, this can be done using the "a=rtpmap:" and "a=fmtp:" attributes associated with an "m=" line, along with any other SDP attributes needed to configure the RTP payload format.

Endpoints can signal support for multiple RTP payload formats or multiple configurations of a single RTP payload format, as long as each unique RTP payload format configuration uses a different RTP payload type number. As outlined in Section 4.8, the RTP payload type number is sometimes used to associate an RTP packet stream with a signaling context. This association is possible provided unique RTP payload type numbers are used in each context. For example, an RTP packet stream can be associated with an SDP "m=" line by comparing the RTP payload type numbers used by the RTP packet stream with payload types signaled in the "a=rtpmap:" lines in the media sections of the SDP. This leads to the following considerations:

If RTP packet streams are being associated with signaling contexts based on the RTP payload type, then the assignment of RTP payload type numbers MUST be unique across signaling contexts.

If the same RTP payload format configuration is used in multiple contexts, then a different RTP payload type number has to be assigned in each context to ensure uniqueness.

If the RTP payload type number is not being used to associate RTP packet streams with a signaling context, then the same RTP payload type number can be used to indicate the exact same RTP payload format configuration in multiple contexts.

A single RTP payload type number MUST NOT be assigned to different RTP payload formats, or different configurations of the same RTP payload format, within a single RTP session (note that the "m=" lines in an SDP BUNDLE group [RFC8843] form a single RTP session).

An endpoint that has signaled support for multiple RTP payload formats MUST be able to accept data in any of those payload formats at any time, unless it has previously signaled limitations on its decoding capability. This requirement is constrained if several types of media (e.g., audio and video) are sent in the same RTP session. In such a case, a source (SSRC) is restricted to switching only between the RTP payload formats signaled for the type of media that is being sent by that source; see Section 4.4. To support rapid rate adaptation by changing codecs, RTP does not require advance signaling for changes between RTP payload formats used by a single SSRC that were signaled during session setup.

If performing changes between two RTP payload types that use different RTP clock rates, an RTP sender MUST follow the recommendations in Section 4.1 of [RFC7160]. RTP receivers MUST follow the recommendations in Section 4.3 of [RFC7160] in order to support sources that switch between clock rates in an RTP session. These recommendations for receivers are backwards compatible with the case where senders use only a single clock rate.

4.4. Use of RTP Sessions

An association amongst a set of endpoints communicating using RTP is known as an RTP session [RFC3550]. An endpoint can be involved in several RTP sessions at the same time. In a multimedia session, each type of media has typically been carried in a separate RTP session (e.g., using one RTP session for the audio and a separate RTP session using a different transport-layer flow for the video). WebRTC endpoints are REQUIRED to implement support for multimedia sessions in this way, separating each RTP session using different transport- layer flows for compatibility with legacy systems (this is sometimes called session multiplexing).

In modern-day networks, however, with the widespread use of network address/port translators (NAT/NAPT) and firewalls, it is desirable to reduce the number of transport-layer flows used by RTP applications. This can be done by sending all the RTP packet streams in a single RTP session, which will comprise a single transport-layer flow. This will prevent the use of some quality-of-service mechanisms, as discussed in Section 12.1.3. Implementations are therefore also REQUIRED to support transport of all RTP packet streams, independent of media type, in a single RTP session using a single transport-layer flow, according to [RFC8860] (this is sometimes called SSRC multiplexing). If multiple types of media are to be used in a single RTP session, all participants in that RTP session MUST agree to this usage. In an SDP context, the mechanisms described in [RFC8843] can be used to signal such a bundle of RTP packet streams forming a single RTP session.

Further discussion about the suitability of different RTP session structures and multiplexing methods to different scenarios can be found in [RFC8872].

4.5. RTP and RTCP Multiplexing

Historically, RTP and RTCP have been run on separate transport-layer flows (e.g., two UDP ports for each RTP session, one for RTP and one for RTCP). With the increased use of Network Address/Port Translation (NAT/NAPT), this has become problematic, since maintaining multiple NAT bindings can be costly. It also complicates firewall administration, since multiple ports need to be opened to allow RTP traffic. To reduce these costs and session setup times, implementations are REQUIRED to support multiplexing RTP data packets and RTCP control packets on a single transport-layer flow [RFC5761]. Such RTP and RTCP multiplexing MUST be negotiated in the signaling channel before it is used. If SDP is used for signaling, this negotiation MUST use the mechanism defined in [RFC5761]. Implementations can also support sending RTP and RTCP on separate transport-layer flows, but this is OPTIONAL to implement. If an implementation does not support RTP and RTCP sent on separate transport-layer flows, it MUST indicate that using the mechanism defined in [RFC8858].

Note that the use of RTP and RTCP multiplexed onto a single transport-layer flow ensures that there is occasional traffic sent on that port, even if there is no active media traffic. This can be useful to keep NAT bindings alive [RFC6263].

4.6. Reduced Size RTCP

RTCP packets are usually sent as compound RTCP packets, and [RFC3550] requires that those compound packets start with an SR or RR packet. When using frequent RTCP feedback messages under the RTP/AVPF profile [RFC4585], these statistics are not needed in every packet, and they unnecessarily increase the mean RTCP packet size. This can limit the frequency at which RTCP packets can be sent within the RTCP bandwidth share.

To avoid this problem, [RFC5506] specifies how to reduce the mean RTCP message size and allow for more frequent feedback. Frequent feedback, in turn, is essential to make real-time applications quickly aware of changing network conditions and to allow them to adapt their transmission and encoding behavior. Implementations MUST support sending and receiving noncompound RTCP feedback packets [RFC5506]. Use of noncompound RTCP packets MUST be negotiated using the signaling channel. If SDP is used for signaling, this negotiation MUST use the attributes defined in [RFC5506]. For backwards compatibility, implementations are also REQUIRED to support the use of compound RTCP feedback packets if the remote endpoint does not agree to the use of noncompound RTCP in the signaling exchange.

4.7. Symmetric RTP/RTCP

To ease traversal of NAT and firewall devices, implementations are REQUIRED to implement and use symmetric RTP [RFC4961]. The reason for using symmetric RTP is primarily to avoid issues with NATs and firewalls by ensuring that the send and receive RTP packet streams, as well as RTCP, are actually bidirectional transport-layer flows. This will keep alive the NAT and firewall pinholes and help indicate consent that the receive direction is a transport-layer flow the intended recipient actually wants. In addition, it saves resources, specifically ports at the endpoints, but also in the network, because the NAT mappings or firewall state is not unnecessarily bloated. The amount of per-flow QoS state kept in the network is also reduced.

4.8. Choice of RTP Synchronization Source (SSRC)

Implementations are REQUIRED to support signaled RTP synchronization source (SSRC) identifiers. If SDP is used, this MUST be done using the "a=ssrc:" SDP attribute defined in Sections 4.1 and 5 of [RFC5576] and the "previous-ssrc" source attribute defined in Section 6.2 of [RFC5576]; other per-SSRC attributes defined in [RFC5576] MAY be supported.

While support for signaled SSRC identifiers is mandated, their use in an RTP session is OPTIONAL. Implementations MUST be prepared to accept RTP and RTCP packets using SSRCs that have not been explicitly signaled ahead of time. Implementations MUST support random SSRC assignment and MUST support SSRC collision detection and resolution, according to [RFC3550]. When using signaled SSRC values, collision detection MUST be performed as described in Section 5 of [RFC5576].

It is often desirable to associate an RTP packet stream with a non- RTP context. For users of the WebRTC API, a mapping between SSRCs and MediaStreamTracks is provided per Section 11. For gateways or other usages, it is possible to associate an RTP packet stream with an "m=" line in a session description formatted using SDP. If SSRCs are signaled, this is straightforward (in SDP, the "a=ssrc:" line will be at the media level, allowing a direct association with an "m=" line). If SSRCs are not signaled, the RTP payload type numbers used in an RTP packet stream are often sufficient to associate that packet stream with a signaling context. For example, if RTP payload type numbers are assigned as described in Section 4.3 of this memo, the RTP payload types used by an RTP packet stream can be compared with values in SDP "a=rtpmap:" lines, which are at the media level in SDP and so map to an "m=" line.

4.9. Generation of the RTCP Canonical Name (CNAME)

The RTCP Canonical Name (CNAME) provides a persistent transport-level identifier for an RTP endpoint. While the SSRC identifier for an RTP endpoint can change if a collision is detected or when the RTP application is restarted, its RTCP CNAME is meant to stay unchanged for the duration of an RTCPeerConnection [W3C.WebRTC], so that RTP endpoints can be uniquely identified and associated with their RTP packet streams within a set of related RTP sessions.

Each RTP endpoint MUST have at least one RTCP CNAME, and that RTCP CNAME MUST be unique within the RTCPeerConnection. RTCP CNAMEs identify a particular synchronization context -- i.e., all SSRCs associated with a single RTCP CNAME share a common reference clock. If an endpoint has SSRCs that are associated with several unsynchronized reference clocks, and hence different synchronization contexts, it will need to use multiple RTCP CNAMEs, one for each synchronization context.

Taking the discussion in Section 11 into account, a WebRTC endpoint MUST NOT use more than one RTCP CNAME in the RTP sessions belonging to a single RTCPeerConnection (that is, an RTCPeerConnection forms a synchronization context). RTP middleboxes MAY generate RTP packet streams associated with more than one RTCP CNAME, to allow them to avoid having to resynchronize media from multiple different endpoints that are part of a multiparty RTP session.

The RTP specification [RFC3550] includes guidelines for choosing a unique RTP CNAME, but these are not sufficient in the presence of NAT devices. In addition, long-term persistent identifiers can be problematic from a privacy viewpoint (Section 13). Accordingly, a WebRTC endpoint MUST generate a new, unique, short-term persistent RTCP CNAME for each RTCPeerConnection, following [RFC7022], with a single exception; if explicitly requested at creation, an RTCPeerConnection MAY use the same CNAME as an existing RTCPeerConnection within their common same-origin context.

A WebRTC endpoint MUST support reception of any CNAME that matches the syntax limitations specified by the RTP specification [RFC3550] and cannot assume that any CNAME will be chosen according to the form suggested above.

4.10. Handling of Leap Seconds

The guidelines given in [RFC7164] regarding handling of leap seconds to limit their impact on RTP media play-out and synchronization SHOULD be followed.

5. WebRTC Use of RTP: Extensions

There are a number of RTP extensions that are either needed to obtain full functionality, or extremely useful to improve on the baseline performance, in the WebRTC context. One set of these extensions is related to conferencing, while others are more generic in nature. The following subsections describe the various RTP extensions mandated or suggested for use within WebRTC.

5.1. Conferencing Extensions and Topologies

RTP is a protocol that inherently supports group communication. Groups can be implemented by having each endpoint send its RTP packet streams to an RTP middlebox that redistributes the traffic, by using a mesh of unicast RTP packet streams between endpoints, or by using an IP multicast group to distribute the RTP packet streams. These topologies can be implemented in a number of ways as discussed in [RFC7667].

While the use of IP multicast groups is popular in IPTV systems, the topologies based on RTP middleboxes are dominant in interactive video-conferencing environments. Topologies based on a mesh of unicast transport-layer flows to create a common RTP session have not seen widespread deployment to date. Accordingly, WebRTC endpoints are not expected to support topologies based on IP multicast groups or mesh-based topologies, such as a point-to-multipoint mesh configured as a single RTP session ("Topo-Mesh" in the terminology of [RFC7667]). However, a point-to-multipoint mesh constructed using several RTP sessions, implemented in WebRTC using independent RTCPeerConnections [W3C.WebRTC], can be expected to be used in WebRTC and needs to be supported.

WebRTC endpoints implemented according to this memo are expected to support all the topologies described in [RFC7667] where the RTP endpoints send and receive unicast RTP packet streams to and from some peer device, provided that peer can participate in performing congestion control on the RTP packet streams. The peer device could be another RTP endpoint, or it could be an RTP middlebox that redistributes the RTP packet streams to other RTP endpoints. This limitation means that some of the RTP middlebox-based topologies are not suitable for use in WebRTC. Specifically:

   *  Video-switching Multipoint Control Units (MCUs) (Topo-Video-
      switch-MCU) SHOULD NOT be used, since they make the use of RTCP
      for congestion control and quality-of-service reports problematic
      (see Section 3.8 of [RFC7667]).
   
   *  The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
      SHOULD NOT be used, because its safe use requires a congestion
      control algorithm or RTP circuit breaker that handles point to
      multipoint, which has not yet been standardized.

The following topology can be used, however it has some issues worth noting:

   *  Content-modifying MCUs with RTCP termination (Topo-RTCP-
      terminating-MCU) MAY be used.  Note that in this RTP topology, RTP
      loop detection and identification of active senders is the
      responsibility of the WebRTC application; since the clients are
      isolated from each other at the RTP layer, RTP cannot assist with
      these functions (see Section 3.9 of [RFC7667]).

The RTP extensions described in Sections 5.1.1 to 5.1.6 are designed to be used with centralized conferencing, where an RTP middlebox (e.g., a conference bridge) receives a participant's RTP packet streams and distributes them to the other participants. These extensions are not necessary for interoperability; an RTP endpoint that does not implement these extensions will work correctly but might offer poor performance. Support for the listed extensions will greatly improve the quality of experience; to provide a reasonable baseline quality, some of these extensions are mandatory to be supported by WebRTC endpoints.

The RTCP conferencing extensions are defined in "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/ AVPF)" [RFC4585] and "Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)" [RFC5104]; they are fully usable by the secure variant of this profile (RTP/SAVPF) [RFC5124].

5.1.1. Full Intra Request (FIR)

The Full Intra Request message is defined in Sections 3.5.1 and 4.3.1 of Codec Control Messages [RFC5104]. It is used to make the mixer request a new Intra picture from a participant in the session. This is used when switching between sources to ensure that the receivers can decode the video or other predictive media encoding with long prediction chains. WebRTC endpoints that are sending media MUST understand and react to FIR feedback messages they receive, since this greatly improves the user experience when using centralized mixer-based conferencing. Support for sending FIR messages is OPTIONAL.

5.1.2. Picture Loss Indication (PLI)

The Picture Loss Indication message is defined in Section 6.3.1 of the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the sending encoder that it lost the decoder context and would like to have it repaired somehow. This is semantically different from the Full Intra Request above, as there could be multiple ways to fulfill the request. WebRTC endpoints that are sending media MUST understand and react to PLI feedback messages as a loss-tolerance mechanism. Receivers MAY send PLI messages.

5.1.3. Slice Loss Indication (SLI)

The Slice Loss Indication message is defined in Section 6.3.2 of the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the encoder that it has detected the loss or corruption of one or more consecutive macro blocks and would like to have these repaired somehow. It is RECOMMENDED that receivers generate SLI feedback messages if slices are lost when using a codec that supports the concept of macro blocks. A sender that receives an SLI feedback message SHOULD attempt to repair the lost slice(s).

5.1.4. Reference Picture Selection Indication (RPSI)

Reference Picture Selection Indication (RPSI) messages are defined in Section 6.3.3 of the RTP/AVPF profile [RFC4585]. Some video-encoding standards allow the use of older reference pictures than the most recent one for predictive coding. If such a codec is in use, and if the encoder has learned that encoder-decoder synchronization has been lost, then a known-as-correct reference picture can be used as a base for future coding. The RPSI message allows this to be signaled. Receivers that detect that encoder-decoder synchronization has been lost SHOULD generate an RPSI feedback message if the codec being used supports reference-picture selection. An RTP packet-stream sender that receives such an RPSI message SHOULD act on that messages to change the reference picture, if it is possible to do so within the available bandwidth constraints and with the codec being used.

5.1.5. Temporal-Spatial Trade-Off Request (TSTR)

The temporal-spatial trade-off request and notification are defined in Sections 3.5.2 and 4.3.2 of [RFC5104]. This request can be used to ask the video encoder to change the trade-off it makes between temporal and spatial resolution -- for example, to prefer high spatial image quality but low frame rate. Support for TSTR requests and notifications is OPTIONAL.

5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR)

The Temporary Maximum Media Stream Bit Rate Request (TMMBR) feedback message is defined in Sections 3.5.4 and 4.2.1 of Codec Control Messages [RFC5104]. This request and its corresponding Temporary Maximum Media Stream Bit Rate Notification (TMMBN) message [RFC5104] are used by a media receiver to inform the sending party that there is a current limitation on the amount of bandwidth available to this receiver. There can be various reasons for this: for example, an RTP mixer can use this message to limit the media rate of the sender being forwarded by the mixer (without doing media transcoding) to fit the bottlenecks existing towards the other session participants. WebRTC endpoints that are sending media are REQUIRED to implement support for TMMBR messages and MUST follow bandwidth limitations set by a TMMBR message received for their SSRC. The sending of TMMBR messages is OPTIONAL.

5.2. Header Extensions

The RTP specification [RFC3550] provides the capability to include RTP header extensions containing in-band data, but the format and semantics of the extensions are poorly specified. The use of header extensions is OPTIONAL in WebRTC, but if they are used, they MUST be formatted and signaled following the general mechanism for RTP header extensions defined in [RFC8285], since this gives well-defined semantics to RTP header extensions.

As noted in [RFC8285], the requirement from the RTP specification that header extensions are "designed so that the header extension may be ignored" [RFC3550] stands. To be specific, header extensions MUST only be used for data that can safely be ignored by the recipient without affecting interoperability and MUST NOT be used when the presence of the extension has changed the form or nature of the rest of the packet in a way that is not compatible with the way the stream is signaled (e.g., as defined by the payload type). Valid examples of RTP header extensions might include metadata that is additional to the usual RTP information but that can safely be ignored without compromising interoperability.

5.2.1. Rapid Synchronization

Many RTP sessions require synchronization between audio, video, and other content. This synchronization is performed by receivers, using information contained in RTCP SR packets, as described in the RTP specification [RFC3550]. This basic mechanism can be slow, however, so it is RECOMMENDED that the rapid RTP synchronization extensions described in [RFC6051] be implemented in addition to RTCP SR-based synchronization.