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Media Transport and Use of RTP in WebRTC
RFC 8834

Document Type RFC - Proposed Standard (January 2021)
Authors Colin Perkins , Magnus Westerlund , Joerg Ott
Last updated 2021-01-18
RFC stream Internet Engineering Task Force (IETF)
Additional resources Mailing list discussion
IESG Responsible AD Alissa Cooper
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RFC 8834

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


   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

Copyright Notice

   Copyright (c) 2021 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
   ( 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.

Table of Contents

   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
   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

3.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "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

   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

   *  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

   *  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

   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

   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

   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

   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

   *  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

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

   This header extension uses the generic header extension framework
   described in [RFC8285] and so needs to be negotiated before it can be

5.2.2.  Client-to-Mixer Audio Level

   The client-to-mixer audio level extension [RFC6464] is an RTP header
   extension used by an endpoint to inform a mixer about the level of
   audio activity in the packet to which the header is attached.  This
   enables an RTP middlebox to make mixing or selection decisions
   without decoding or detailed inspection of the payload, reducing the
   complexity in some types of mixers.  It can also save decoding
   resources in receivers, which can choose to decode only the most
   relevant RTP packet streams based on audio activity levels.

   The client-to-mixer audio level header extension [RFC6464] MUST be
   implemented.  It is REQUIRED that implementations be capable of
   encrypting the header extension according to [RFC6904], since the
   information contained in these header extensions can be considered
   sensitive.  The use of this encryption is RECOMMENDED; however, usage
   of the encryption can be explicitly disabled through API or

   This header extension uses the generic header extension framework
   described in [RFC8285] and so needs to be negotiated before it can be

5.2.3.  Mixer-to-Client Audio Level

   The mixer-to-client audio level header extension [RFC6465] provides
   an endpoint with the audio level of the different sources mixed into
   a common source stream by an RTP mixer.  This enables a user
   interface to indicate the relative activity level of each session
   participant, rather than just being included or not based on the CSRC
   field.  This is a pure optimization of non-critical functions and is
   hence OPTIONAL to implement.  If this header extension is
   implemented, it is REQUIRED that implementations be capable of
   encrypting the header extension according to [RFC6904], since the
   information contained in these header extensions can be considered
   sensitive.  It is further RECOMMENDED that this encryption be used,
   unless the encryption has been explicitly disabled through API or

   This header extension uses the generic header extension framework
   described in [RFC8285] and so needs to be negotiated before it can be

5.2.4.  Media Stream Identification

   WebRTC endpoints that implement the SDP bundle negotiation extension
   will use the SDP Grouping Framework "mid" attribute to identify media
   streams.  Such endpoints MUST implement the RTP MID header extension
   described in [RFC8843].

   This header extension uses the generic header extension framework
   described in [RFC8285] and so needs to be negotiated before it can be

5.2.5.  Coordination of Video Orientation

   WebRTC endpoints that send or receive video MUST implement the
   coordination of video orientation (CVO) RTP header extension as
   described in Section 4 of [RFC7742].

   This header extension uses the generic header extension framework
   described in [RFC8285] and so needs to be negotiated before it can be

6.  WebRTC Use of RTP: Improving Transport Robustness

   There are tools that can make RTP packet streams robust against
   packet loss and reduce the impact of loss on media quality.  However,
   they generally add some overhead compared to a non-robust stream.
   The overhead needs to be considered, and the aggregate bitrate MUST
   be rate controlled to avoid causing network congestion (see
   Section 7).  As a result, improving robustness might require a lower
   base encoding quality but has the potential to deliver that quality
   with fewer errors.  The mechanisms described in the following
   subsections can be used to improve tolerance to packet loss.

6.1.  Negative Acknowledgements and RTP Retransmission

   As a consequence of supporting the RTP/SAVPF profile, implementations
   can send negative acknowledgements (NACKs) for RTP data packets
   [RFC4585].  This feedback can be used to inform a sender of the loss
   of particular RTP packets, subject to the capacity limitations of the
   RTCP feedback channel.  A sender can use this information to optimize
   the user experience by adapting the media encoding to compensate for
   known lost packets.

   RTP packet stream senders are REQUIRED to understand the generic NACK
   message defined in Section 6.2.1 of [RFC4585], but they MAY choose to
   ignore some or all of this feedback (following Section 4.2 of
   [RFC4585]).  Receivers MAY send NACKs for missing RTP packets.
   Guidelines on when to send NACKs are provided in [RFC4585].  It is
   not expected that a receiver will send a NACK for every lost RTP
   packet; rather, it needs to consider the cost of sending NACK
   feedback and the importance of the lost packet to make an informed
   decision on whether it is worth telling the sender about a packet-
   loss event.

   The RTP retransmission payload format [RFC4588] offers the ability to
   retransmit lost packets based on NACK feedback.  Retransmission needs
   to be used with care in interactive real-time applications to ensure
   that the retransmitted packet arrives in time to be useful, but it
   can be effective in environments with relatively low network RTT.
   (An RTP sender can estimate the RTT to the receivers using the
   information in RTCP SR and RR packets, as described at the end of
   Section 6.4.1 of [RFC3550]).  The use of retransmissions can also
   increase the forward RTP bandwidth and can potentially cause
   increased packet loss if the original packet loss was caused by
   network congestion.  Note, however, that retransmission of an
   important lost packet to repair decoder state can have lower cost
   than sending a full intra frame.  It is not appropriate to blindly
   retransmit RTP packets in response to a NACK.  The importance of lost
   packets and the likelihood of them arriving in time to be useful need
   to be considered before RTP retransmission is used.

   Receivers are REQUIRED to implement support for RTP retransmission
   packets [RFC4588] sent using SSRC multiplexing and MAY also support
   RTP retransmission packets sent using session multiplexing.  Senders
   MAY send RTP retransmission packets in response to NACKs if support
   for the RTP retransmission payload format has been negotiated and the
   sender believes it is useful to send a retransmission of the
   packet(s) referenced in the NACK.  Senders do not need to retransmit
   every NACKed packet.

6.2.  Forward Error Correction (FEC)

   The use of Forward Error Correction (FEC) can provide an effective
   protection against some degree of packet loss, at the cost of steady
   bandwidth overhead.  There are several FEC schemes that are defined
   for use with RTP.  Some of these schemes are specific to a particular
   RTP payload format, and others operate across RTP packets and can be
   used with any payload format.  Note that using redundant encoding or
   FEC will lead to increased play-out delay, which needs to be
   considered when choosing FEC schemes and their parameters.

   WebRTC endpoints MUST follow the recommendations for FEC use given in
   [RFC8854].  WebRTC endpoints MAY support other types of FEC, but
   these MUST be negotiated before they are used.

7.  WebRTC Use of RTP: Rate Control and Media Adaptation

   WebRTC will be used in heterogeneous network environments using a
   variety of link technologies, including both wired and wireless
   links, to interconnect potentially large groups of users around the
   world.  As a result, the network paths between users can have widely
   varying one-way delays, available bitrates, load levels, and traffic
   mixtures.  Individual endpoints can send one or more RTP packet
   streams to each participant, and there can be several participants.
   Each of these RTP packet streams can contain different types of
   media, and the type of media, bitrate, and number of RTP packet
   streams as well as transport-layer flows can be highly asymmetric.
   Non-RTP traffic can share the network paths with RTP transport-layer
   flows.  Since the network environment is not predictable or stable,
   WebRTC endpoints MUST ensure that the RTP traffic they generate can
   adapt to match changes in the available network capacity.

   The quality of experience for users of WebRTC is very dependent on
   effective adaptation of the media to the limitations of the network.
   Endpoints have to be designed so they do not transmit significantly
   more data than the network path can support, except for very short
   time periods; otherwise, high levels of network packet loss or delay
   spikes will occur, causing media quality degradation.  The limiting
   factor on the capacity of the network path might be the link
   bandwidth, or it might be competition with other traffic on the link
   (this can be non-WebRTC traffic, traffic due to other WebRTC flows,
   or even competition with other WebRTC flows in the same session).

   An effective media congestion control algorithm is therefore an
   essential part of the WebRTC framework.  However, at the time of this
   writing, there is no standard congestion control algorithm that can
   be used for interactive media applications such as WebRTC's flows.
   Some requirements for congestion control algorithms for
   RTCPeerConnections are discussed in [RFC8836].  If a standardized
   congestion control algorithm that satisfies these requirements is
   developed in the future, this memo will need to be updated to mandate
   its use.

7.1.  Boundary Conditions and Circuit Breakers

   WebRTC endpoints MUST implement the RTP circuit breaker algorithm
   that is described in [RFC8083].  The RTP circuit breaker is designed
   to enable applications to recognize and react to situations of
   extreme network congestion.  However, since the RTP circuit breaker
   might not be triggered until congestion becomes extreme, it cannot be
   considered a substitute for congestion control, and applications MUST
   also implement congestion control to allow them to adapt to changes
   in network capacity.  The congestion control algorithm will have to
   be proprietary until a standardized congestion control algorithm is
   available.  Any future RTP congestion control algorithms are expected
   to operate within the envelope allowed by the circuit breaker.

   The session-establishment signaling will also necessarily establish
   boundaries to which the media bitrate will conform.  The choice of
   media codecs provides upper and lower bounds on the supported
   bitrates that the application can utilize to provide useful quality,
   and the packetization choices that exist.  In addition, the signaling
   channel can establish maximum media bitrate boundaries using, for
   example, the SDP "b=AS:" or "b=CT:" lines and the RTP/AVPF TMMBR
   messages (see Section 5.1.6 of this memo).  Signaled bandwidth
   limitations, such as SDP "b=AS:" or "b=CT:" lines received from the
   peer, MUST be followed when sending RTP packet streams.  A WebRTC
   endpoint receiving media SHOULD signal its bandwidth limitations.
   These limitations have to be based on known bandwidth limitations,
   for example the capacity of the edge links.

7.2.  Congestion Control Interoperability and Legacy Systems

   All endpoints that wish to interwork with WebRTC MUST implement RTCP
   and provide congestion feedback via the defined RTCP reporting

   When interworking with legacy implementations that support RTCP using
   the RTP/AVP profile [RFC3551], congestion feedback is provided in
   RTCP RR packets every few seconds.  Implementations that have to
   interwork with such endpoints MUST ensure that they keep within the
   RTP circuit breaker [RFC8083] constraints to limit the congestion
   they can cause.

   If a legacy endpoint supports RTP/AVPF, this enables negotiation of
   important parameters for frequent reporting, such as the "trr-int"
   parameter, and the possibility that the endpoint supports some useful
   feedback format for congestion control purposes such as TMMBR
   [RFC5104].  Implementations that have to interwork with such
   endpoints MUST ensure that they stay within the RTP circuit breaker
   [RFC8083] constraints to limit the congestion they can cause, but
   they might find that they can achieve better congestion response
   depending on the amount of feedback that is available.

   With proprietary congestion control algorithms, issues can arise when
   different algorithms and implementations interact in a communication
   session.  If the different implementations have made different
   choices in regards to the type of adaptation, for example one sender
   based, and one receiver based, then one could end up in a situation
   where one direction is dual controlled when the other direction is
   not controlled.  This memo cannot mandate behavior for proprietary
   congestion control algorithms, but implementations that use such
   algorithms ought to be aware of this issue and try to ensure that
   effective congestion control is negotiated for media flowing in both
   directions.  If the IETF were to standardize both sender- and
   receiver-based congestion control algorithms for WebRTC traffic in
   the future, the issues of interoperability, control, and ensuring
   that both directions of media flow are congestion controlled would
   also need to be considered.

8.  WebRTC Use of RTP: Performance Monitoring

   As described in Section 4.1, implementations are REQUIRED to generate
   RTCP Sender Report (SR) and Receiver Report (RR) packets relating to
   the RTP packet streams they send and receive.  These RTCP reports can
   be used for performance monitoring purposes, since they include basic
   packet-loss and jitter statistics.

   A large number of additional performance metrics are supported by the
   RTCP Extended Reports (XR) framework; see [RFC3611] and [RFC6792].
   At the time of this writing, it is not clear what extended metrics
   are suitable for use in WebRTC, so there is no requirement that
   implementations generate RTCP XR packets.  However, implementations
   that can use detailed performance monitoring data MAY generate RTCP
   XR packets as appropriate.  The use of RTCP XR packets SHOULD be
   signaled; implementations MUST ignore RTCP XR packets that are
   unexpected or not understood.

9.  WebRTC Use of RTP: Future Extensions

   It is possible that the core set of RTP protocols and RTP extensions
   specified in this memo will prove insufficient for the future needs
   of WebRTC.  In this case, future updates to this memo have to be made
   following "Guidelines for Writers of RTP Payload Format
   Specifications" [RFC2736], "How to Write an RTP Payload Format"
   [RFC8088], and "Guidelines for Extending the RTP Control Protocol
   (RTCP)" [RFC5968].  They also SHOULD take into account any future
   guidelines for extending RTP and related protocols that have been

   Authors of future extensions are urged to consider the wide range of
   environments in which RTP is used when recommending extensions, since
   extensions that are applicable in some scenarios can be problematic
   in others.  Where possible, the WebRTC framework will adopt RTP
   extensions that are of general utility, to enable easy implementation
   of a gateway to other applications using RTP, rather than adopt
   mechanisms that are narrowly targeted at specific WebRTC use cases.

10.  Signaling Considerations

   RTP is built with the assumption that an external signaling channel
   exists and can be used to configure RTP sessions and their features.
   The basic configuration of an RTP session consists of the following

   RTP profile:  The name of the RTP profile to be used in the session.
      The RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can
      interoperate on a basic level, as can their secure variants, RTP/
      SAVP [RFC3711] and RTP/SAVPF [RFC5124].  The secure variants of
      the profiles do not directly interoperate with the nonsecure
      variants, due to the presence of additional header fields for
      authentication in SRTP packets and cryptographic transformation of
      the payload.  WebRTC requires the use of the RTP/SAVPF profile,
      and this MUST be signaled.  Interworking functions might transform
      this into the RTP/SAVP profile for a legacy use case by indicating
      to the WebRTC endpoint that the RTP/SAVPF is used and configuring
      a "trr-int" value of 4 seconds.

   Transport information:  Source and destination IP address(es) and
      ports for RTP and RTCP MUST be signaled for each RTP session.  In
      WebRTC, these transport addresses will be provided by Interactive
      Connectivity Establishment (ICE) [RFC8445] that signals candidates
      and arrives at nominated candidate address pairs.  If RTP and RTCP
      multiplexing [RFC5761] is to be used such that a single port --
      i.e., transport-layer flow -- is used for RTP and RTCP flows, this
      MUST be signaled (see Section 4.5).

   RTP payload types, media formats, and format parameters:  The mapping
      between media type names (and hence the RTP payload formats to be
      used) and the RTP payload type numbers MUST be signaled.  Each
      media type MAY also have a number of media type parameters that
      MUST also be signaled to configure the codec and RTP payload
      format (the "a=fmtp:" line from SDP).  Section 4.3 of this memo
      discusses requirements for uniqueness of payload types.

   RTP extensions:  The use of any additional RTP header extensions and
      RTCP packet types, including any necessary parameters, MUST be
      signaled.  This signaling ensures that a WebRTC endpoint's
      behavior, especially when sending, is predictable and consistent.
      For robustness and compatibility with non-WebRTC systems that
      might be connected to a WebRTC session via a gateway,
      implementations are REQUIRED to ignore unknown RTCP packets and
      RTP header extensions (see also Section 4.1).

   RTCP bandwidth:  Support for exchanging RTCP bandwidth values with
      the endpoints will be necessary.  This SHALL be done as described
      in "Session Description Protocol (SDP) Bandwidth Modifiers for RTP
      Control Protocol (RTCP) Bandwidth" [RFC3556] if using SDP, or
      something semantically equivalent.  This also ensures that the
      endpoints have a common view of the RTCP bandwidth.  A common view
      of the RTCP bandwidth among different endpoints is important to
      prevent differences in RTCP packet timing and timeout intervals
      causing interoperability problems.

   These parameters are often expressed in SDP messages conveyed within
   an offer/answer exchange.  RTP does not depend on SDP or the offer/
   answer model but does require all the necessary parameters to be
   agreed upon and provided to the RTP implementation.  Note that in
   WebRTC, it will depend on the signaling model and API how these
   parameters need to be configured, but they will need to either be set
   in the API or explicitly signaled between the peers.

11.  WebRTC API Considerations

   The WebRTC API [W3C.WebRTC] and the Media Capture and Streams API
   [W3C.WD-mediacapture-streams] define and use the concept of a
   MediaStream that consists of zero or more MediaStreamTracks.  A
   MediaStreamTrack is an individual stream of media from any type of
   media source, such as a microphone or a camera, but conceptual
   sources, like an audio mix or a video composition, are also possible.
   The MediaStreamTracks within a MediaStream might need to be
   synchronized during playback.

   A MediaStreamTrack's realization in RTP, in the context of an
   RTCPeerConnection, consists of a source packet stream, identified by
   an SSRC, sent within an RTP session that is part of the
   RTCPeerConnection.  The MediaStreamTrack can also result in
   additional packet streams, and thus SSRCs, in the same RTP session.
   These can be dependent packet streams from scalable encoding of the
   source stream associated with the MediaStreamTrack, if such a media
   encoder is used.  They can also be redundancy packet streams; these
   are created when applying Forward Error Correction (Section 6.2) or
   RTP retransmission (Section 6.1) to the source packet stream.

   It is important to note that the same media source can be feeding
   multiple MediaStreamTracks.  As different sets of constraints or
   other parameters can be applied to the MediaStreamTrack, each
   MediaStreamTrack instance added to an RTCPeerConnection SHALL result
   in an independent source packet stream with its own set of associated
   packet streams and thus different SSRC(s).  It will depend on applied
   constraints and parameters if the source stream and the encoding
   configuration will be identical between different MediaStreamTracks
   sharing the same media source.  If the encoding parameters and
   constraints are the same, an implementation could choose to use only
   one encoded stream to create the different RTP packet streams.  Note
   that such optimizations would need to take into account that the
   constraints for one of the MediaStreamTracks can change at any
   moment, meaning that the encoding configurations might no longer be
   identical, and two different encoder instances would then be needed.

   The same MediaStreamTrack can also be included in multiple
   MediaStreams; thus, multiple sets of MediaStreams can implicitly need
   to use the same synchronization base.  To ensure that this works in
   all cases and does not force an endpoint to disrupt the media by
   changing synchronization base and CNAME during delivery of any
   ongoing packet streams, all MediaStreamTracks and their associated
   SSRCs originating from the same endpoint need to be sent using the
   same CNAME within one RTCPeerConnection.  This is motivating the use
   of a single CNAME in Section 4.9.

      |  The requirement to use the same CNAME for all SSRCs that
      |  originate from the same endpoint does not require a middlebox
      |  that forwards traffic from multiple endpoints to only use a
      |  single CNAME.

   Different CNAMEs normally need to be used for different
   RTCPeerConnection instances, as specified in Section 4.9.  Having two
   communication sessions with the same CNAME could enable tracking of a
   user or device across different services (see Section 4.4.1 of
   [RFC8826] for details).  A web application can request that the
   CNAMEs used in different RTCPeerConnections (within a same-origin
   context) be the same; this allows for synchronization of the
   endpoint's RTP packet streams across the different

      |  Note: This doesn't result in a tracking issue, since the
      |  creation of matching CNAMEs depends on existing tracking within
      |  a single origin.

   The above will currently force a WebRTC endpoint that receives a
   MediaStreamTrack on one RTCPeerConnection and adds it as outgoing one
   on any RTCPeerConnection to perform resynchronization of the stream.
   Since the sending party needs to change the CNAME to the one it uses,
   this implies it has to use a local system clock as the timebase for
   the synchronization.  Thus, the relative relation between the
   timebase of the incoming stream and the system sending out needs to
   be defined.  This relation also needs monitoring for clock drift and
   likely adjustments of the synchronization.  The sending entity is
   also responsible for congestion control for its sent streams.  In
   cases of packet loss, the loss of incoming data also needs to be
   handled.  This leads to the observation that the method that is least
   likely to cause issues or interruptions in the outgoing source packet
   stream is a model of full decoding, including repair, followed by
   encoding of the media again into the outgoing packet stream.
   Optimizations of this method are clearly possible and implementation

   A WebRTC endpoint MUST support receiving multiple MediaStreamTracks,
   where each of the different MediaStreamTracks (and its sets of
   associated packet streams) uses different CNAMEs.  However,
   MediaStreamTracks that are received with different CNAMEs have no
   defined synchronization.

      |  Note: The motivation for supporting reception of multiple
      |  CNAMEs is to allow for forward compatibility with any future
      |  changes that enable more efficient stream handling when
      |  endpoints relay/forward streams.  It also ensures that
      |  endpoints can interoperate with certain types of multistream
      |  middleboxes or endpoints that are not WebRTC.

   "JavaScript Session Establishment Protocol (JSEP)" [RFC8829]
   specifies that the binding between the WebRTC MediaStreams,
   MediaStreamTracks, and the SSRC is done as specified in "WebRTC
   MediaStream Identification in the Session Description Protocol"
   [RFC8830].  Section 4.1 of the MediaStream Identification (MSID)
   document [RFC8830] also defines how to map source packet streams with
   unknown SSRCs to MediaStreamTracks and MediaStreams.  This later is
   relevant to handle some cases of legacy interoperability.  Commonly,
   the RTP payload type of any incoming packets will reveal if the
   packet stream is a source stream or a redundancy or dependent packet
   stream.  The association to the correct source packet stream depends
   on the payload format in use for the packet stream.

   Finally, this specification puts a requirement on the WebRTC API to
   realize a method for determining the CSRC list (Section 4.1) as well
   as the mixer-to-client audio levels (Section 5.2.3) (when supported);
   the basic requirements for this is further discussed in
   Section 12.2.1.

12.  RTP Implementation Considerations

   The following discussion provides some guidance on the implementation
   of the RTP features described in this memo.  The focus is on a WebRTC
   endpoint implementation perspective, and while some mention is made
   of the behavior of middleboxes, that is not the focus of this memo.

12.1.  Configuration and Use of RTP Sessions

   A WebRTC endpoint will be a simultaneous participant in one or more
   RTP sessions.  Each RTP session can convey multiple media sources and
   include media data from multiple endpoints.  In the following, some
   ways in which WebRTC endpoints can configure and use RTP sessions are

12.1.1.  Use of Multiple Media Sources within an RTP Session

   RTP is a group communication protocol, and every RTP session can
   potentially contain multiple RTP packet streams.  There are several
   reasons why this might be desirable:

   *  Multiple media types:

      Outside of WebRTC, it is common to use one RTP session for each
      type of media source (e.g., one RTP session for audio sources and
      one for video sources, each sent over different transport-layer
      flows).  However, to reduce the number of UDP ports used, the
      default in WebRTC is to send all types of media in a single RTP
      session, as described in Section 4.4, using RTP and RTCP
      multiplexing (Section 4.5) to further reduce the number of UDP
      ports needed.  This RTP session then uses only one bidirectional
      transport-layer flow but will contain multiple RTP packet streams,
      each containing a different type of media.  A common example might
      be an endpoint with a camera and microphone that sends two RTP
      packet streams, one video and one audio, into a single RTP

   *  Multiple capture devices:

      A WebRTC endpoint might have multiple cameras, microphones, or
      other media capture devices, and so it might want to generate
      several RTP packet streams of the same media type.  Alternatively,
      it might want to send media from a single capture device in
      several different formats or quality settings at once.  Both can
      result in a single endpoint sending multiple RTP packet streams of
      the same media type into a single RTP session at the same time.

   *  Associated repair data:

      An endpoint might send an RTP packet stream that is somehow
      associated with another stream.  For example, it might send an RTP
      packet stream that contains FEC or retransmission data relating to
      another stream.  Some RTP payload formats send this sort of
      associated repair data as part of the source packet stream, while
      others send it as a separate packet stream.

   *  Layered or multiple-description coding:

      Within a single RTP session, an endpoint can use a layered media
      codec -- for example, H.264 Scalable Video Coding (SVC) -- or a
      multiple-description codec that generates multiple RTP packet
      streams, each with a distinct RTP SSRC.

   *  RTP mixers, translators, and other middleboxes:

      An RTP session, in the WebRTC context, is a point-to-point
      association between an endpoint and some other peer device, where
      those devices share a common SSRC space.  The peer device might be
      another WebRTC endpoint, or it might be an RTP mixer, translator,
      or some other form of media-processing middlebox.  In the latter
      cases, the middlebox might send mixed or relayed RTP streams from
      several participants, which the WebRTC endpoint will need to
      render.  Thus, even though a WebRTC endpoint might only be a
      member of a single RTP session, the peer device might be extending
      that RTP session to incorporate other endpoints.  WebRTC is a
      group communication environment, and endpoints need to be capable
      of receiving, decoding, and playing out multiple RTP packet
      streams at once, even in a single RTP session.

12.1.2.  Use of Multiple RTP Sessions

   In addition to sending and receiving multiple RTP packet streams
   within a single RTP session, a WebRTC endpoint might participate in
   multiple RTP sessions.  There are several reasons why a WebRTC
   endpoint might choose to do this:

   *  To interoperate with legacy devices:

      The common practice in the non-WebRTC world is to send different
      types of media in separate RTP sessions -- for example, using one
      RTP session for audio and another RTP session, on a separate
      transport-layer flow, for video.  All WebRTC endpoints need to
      support the option of sending different types of media on
      different RTP sessions so they can interwork with such legacy
      devices.  This is discussed further in Section 4.4.

   *  To provide enhanced quality of service:

      Some network-based quality-of-service mechanisms operate on the
      granularity of transport-layer flows.  If use of these mechanisms
      to provide differentiated quality of service for some RTP packet
      streams is desired, then those RTP packet streams need to be sent
      in a separate RTP session using a different transport-layer flow,
      and with appropriate quality-of-service marking.  This is
      discussed further in Section 12.1.3.

   *  To separate media with different purposes:

      An endpoint might want to send RTP packet streams that have
      different purposes on different RTP sessions, to make it easy for
      the peer device to distinguish them.  For example, some
      centralized multiparty conferencing systems display the active
      speaker in high resolution but show low-resolution "thumbnails" of
      other participants.  Such systems might configure the endpoints to
      send simulcast high- and low-resolution versions of their video
      using separate RTP sessions to simplify the operation of the RTP
      middlebox.  In the WebRTC context, this is currently possible by
      establishing multiple WebRTC MediaStreamTracks that have the same
      media source in one (or more) RTCPeerConnection.  Each
      MediaStreamTrack is then configured to deliver a particular media
      quality and thus media bitrate, and it will produce an
      independently encoded version with the codec parameters agreed
      specifically in the context of that RTCPeerConnection.  The RTP
      middlebox can distinguish packets corresponding to the low- and
      high-resolution streams by inspecting their SSRC, RTP payload
      type, or some other information contained in RTP payload, RTP
      header extension, or RTCP packets.  However, it can be easier to
      distinguish the RTP packet streams if they arrive on separate RTP
      sessions on separate transport-layer flows.

   *  To directly connect with multiple peers:

      A multiparty conference does not need to use an RTP middlebox.
      Rather, a multi-unicast mesh can be created, comprising several
      distinct RTP sessions, with each participant sending RTP traffic
      over a separate RTP session (that is, using an independent
      RTCPeerConnection object) to every other participant, as shown in
      Figure 1.  This topology has the benefit of not requiring an RTP
      middlebox node that is trusted to access and manipulate the media
      data.  The downside is that it increases the used bandwidth at
      each sender by requiring one copy of the RTP packet streams for
      each participant that is part of the same session beyond the
      sender itself.

      +---+     +---+
      | A |<--->| B |
      +---+     +---+
        ^         ^
         \       /
          \     /
           v   v
           | C |

              Figure 1: Multi-unicast Using Several RTP Sessions

      The multi-unicast topology could also be implemented as a single
      RTP session, spanning multiple peer-to-peer transport-layer
      connections, or as several pairwise RTP sessions, one between each
      pair of peers.  To maintain a coherent mapping of the relationship
      between RTP sessions and RTCPeerConnection objects, it is
      RECOMMENDED that this be implemented as several individual RTP
      sessions.  The only downside is that endpoint A will not learn of
      the quality of any transmission happening between B and C, since
      it will not see RTCP reports for the RTP session between B and C,
      whereas it would if all three participants were part of a single
      RTP session.  Experience with the Mbone tools (experimental RTP-
      based multicast conferencing tools from the late 1990s) has shown
      that RTCP reception quality reports for third parties can be
      presented to users in a way that helps them understand asymmetric
      network problems, and the approach of using separate RTP sessions
      prevents this.  However, an advantage of using separate RTP
      sessions is that it enables using different media bitrates and RTP
      session configurations between the different peers, thus not
      forcing B to endure the same quality reductions as C will if there
      are limitations in the transport from A to C.  It is believed that
      these advantages outweigh the limitations in debugging power.

   *  To indirectly connect with multiple peers:

      A common scenario in multiparty conferencing is to create indirect
      connections to multiple peers, using an RTP mixer, translator, or
      some other type of RTP middlebox.  Figure 2 outlines a simple
      topology that might be used in a four-person centralized
      conference.  The middlebox acts to optimize the transmission of
      RTP packet streams from certain perspectives, either by only
      sending some of the received RTP packet stream to any given
      receiver, or by providing a combined RTP packet stream out of a
      set of contributing streams.

      +---+      +-------------+      +---+
      | A |<---->|             |<---->| B |
      +---+      | RTP mixer,  |      +---+
                 | translator, |
                 | or other    |
      +---+      | middlebox   |      +---+
      | C |<---->|             |<---->| D |
      +---+      +-------------+      +---+

                 Figure 2: RTP Mixer with Only Unicast Paths

      There are various methods of implementation for the middlebox.  If
      implemented as a standard RTP mixer or translator, a single RTP
      session will extend across the middlebox and encompass all the
      endpoints in one multiparty session.  Other types of middleboxes
      might use separate RTP sessions between each endpoint and the
      middlebox.  A common aspect is that these RTP middleboxes can use
      a number of tools to control the media encoding provided by a
      WebRTC endpoint.  This includes functions like requesting the
      breaking of the encoding chain and having the encoder produce a
      so-called Intra frame.  Another common aspect is limiting the
      bitrate of a stream to better match the mixed output.  Other
      aspects are controlling the most suitable frame rate, picture
      resolution, and the trade-off between frame rate and spatial
      quality.  The middlebox has the responsibility to correctly
      perform congestion control, identify sources, and manage
      synchronization while providing the application with suitable
      media optimizations.  The middlebox also has to be a trusted node
      when it comes to security, since it manipulates either the RTP
      header or the media itself (or both) received from one endpoint
      before sending them on towards the endpoint(s); thus they need to
      be able to decrypt and then re-encrypt the RTP packet stream
      before sending it out.

      Mixers are expected to not forward RTCP reports regarding RTP
      packet streams across themselves.  This is due to the difference
      between the RTP packet streams provided to the different
      endpoints.  The original media source lacks information about a
      mixer's manipulations prior to being sent to the different
      receivers.  This scenario also results in an endpoint's feedback
      or requests going to the mixer.  When the mixer can't act on this
      by itself, it is forced to go to the original media source to
      fulfill the receiver's request.  This will not necessarily be
      explicitly visible to any RTP and RTCP traffic, but the
      interactions and the time to complete them will indicate such

      Providing source authentication in multiparty scenarios is a
      challenge.  In the mixer-based topologies, endpoints source
      authentication is based on, firstly, verifying that media comes
      from the mixer by cryptographic verification and, secondly, trust
      in the mixer to correctly identify any source towards the
      endpoint.  In RTP sessions where multiple endpoints are directly
      visible to an endpoint, all endpoints will have knowledge about
      each others' master keys and can thus inject packets claiming to
      come from another endpoint in the session.  Any node performing
      relay can perform noncryptographic mitigation by preventing
      forwarding of packets that have SSRC fields that came from other
      endpoints before.  For cryptographic verification of the source,
      SRTP would require additional security mechanisms -- for example,
      Timed Efficient Stream Loss-Tolerant Authentication (TESLA) for
      SRTP [RFC4383] -- that are not part of the base WebRTC standards.

   *  To forward media between multiple peers:

      It is sometimes desirable for an endpoint that receives an RTP
      packet stream to be able to forward that RTP packet stream to a
      third party.  The are some obvious security and privacy
      implications in supporting this, but also potential uses.  This is
      supported in the W3C API by taking the received and decoded media
      and using it as a media source that is re-encoded and transmitted
      as a new stream.

      At the RTP layer, media forwarding acts as a back-to-back RTP
      receiver and RTP sender.  The receiving side terminates the RTP
      session and decodes the media, while the sender side re-encodes
      and transmits the media using an entirely separate RTP session.
      The original sender will only see a single receiver of the media,
      and will not be able to tell that forwarding is happening based on
      RTP-layer information, since the RTP session that is used to send
      the forwarded media is not connected to the RTP session on which
      the media was received by the node doing the forwarding.

      The endpoint that is performing the forwarding is responsible for
      producing an RTP packet stream suitable for onwards transmission.
      The outgoing RTP session that is used to send the forwarded media
      is entirely separate from the RTP session on which the media was
      received.  This will require media transcoding for congestion
      control purposes to produce a suitable bitrate for the outgoing
      RTP session, reducing media quality and forcing the forwarding
      endpoint to spend the resource on the transcoding.  The media
      transcoding does result in a separation of the two different legs,
      removing almost all dependencies, and allowing the forwarding
      endpoint to optimize its media transcoding operation.  The cost is
      greatly increased computational complexity on the forwarding node.
      Receivers of the forwarded stream will see the forwarding device
      as the sender of the stream and will not be able to tell from the
      RTP layer that they are receiving a forwarded stream rather than
      an entirely new RTP packet stream generated by the forwarding

12.1.3.  Differentiated Treatment of RTP Streams

   There are use cases for differentiated treatment of RTP packet
   streams.  Such differentiation can happen at several places in the
   system.  First of all is the prioritization within the endpoint
   sending the media, which controls both which RTP packet streams will
   be sent and their allocation of bitrate out of the current available
   aggregate, as determined by the congestion control.

   It is expected that the WebRTC API [W3C.WebRTC] will allow the
   application to indicate relative priorities for different
   MediaStreamTracks.  These priorities can then be used to influence
   the local RTP processing, especially when it comes to determining how
   to divide the available bandwidth between the RTP packet streams for
   the sake of congestion control.  Any changes in relative priority
   will also need to be considered for RTP packet streams that are
   associated with the main RTP packet streams, such as redundant
   streams for RTP retransmission and FEC.  The importance of such
   redundant RTP packet streams is dependent on the media type and codec
   used, with regard to how robust that codec is against packet loss.
   However, a default policy might be to use the same priority for a
   redundant RTP packet stream as for the source RTP packet stream.

   Secondly, the network can prioritize transport-layer flows and
   subflows, including RTP packet streams.  Typically, differential
   treatment includes two steps, the first being identifying whether an
   IP packet belongs to a class that has to be treated differently, the
   second consisting of the actual mechanism for prioritizing packets.
   Three common methods for classifying IP packets are:

   DiffServ:  The endpoint marks a packet with a DiffServ code point to
      indicate to the network that the packet belongs to a particular

   Flow based:  Packets that need to be given a particular treatment are
      identified using a combination of IP and port address.

   Deep packet inspection:  A network classifier (DPI) inspects the
      packet and tries to determine if the packet represents a
      particular application and type that is to be prioritized.

   Flow-based differentiation will provide the same treatment to all
   packets within a transport-layer flow, i.e., relative prioritization
   is not possible.  Moreover, if the resources are limited, it might
   not be possible to provide differential treatment compared to best
   effort for all the RTP packet streams used in a WebRTC session.  The
   use of flow-based differentiation needs to be coordinated between the
   WebRTC system and the network(s).  The WebRTC endpoint needs to know
   that flow-based differentiation might be used to provide the
   separation of the RTP packet streams onto different UDP flows to
   enable a more granular usage of flow-based differentiation.  The used
   flows, their 5-tuples, and prioritization will need to be
   communicated to the network so that it can identify the flows
   correctly to enable prioritization.  No specific protocol support for
   this is specified.

   DiffServ assumes that either the endpoint or a classifier can mark
   the packets with an appropriate Differentiated Services Code Point
   (DSCP) so that the packets are treated according to that marking.  If
   the endpoint is to mark the traffic, two requirements arise in the
   WebRTC context: 1) The WebRTC endpoint has to know which DSCPs to use
   and know that it can use them on some set of RTP packet streams. 2)
   The information needs to be propagated to the operating system when
   transmitting the packet.  Details of this process are outside the
   scope of this memo and are further discussed in "Differentiated
   Services Code Point (DSCP) Packet Markings for WebRTC QoS" [RFC8837].

   Despite the SRTP media encryption, deep packet inspectors will still
   be fairly capable of classifying the RTP streams.  The reason is that
   SRTP leaves the first 12 bytes of the RTP header unencrypted.  This
   enables easy RTP stream identification using the SSRC and provides
   the classifier with useful information that can be correlated to
   determine, for example, the stream's media type.  Using packet sizes,
   reception times, packet inter-spacing, RTP timestamp increments, and
   sequence numbers, fairly reliable classifications are achieved.

   For packet-based marking schemes, it might be possible to mark
   individual RTP packets differently based on the relative priority of
   the RTP payload.  For example, video codecs that have I, P, and B
   pictures could prioritize any payloads carrying only B frames less,
   as these are less damaging to lose.  However, depending on the QoS
   mechanism and what markings are applied, this can result in not only
   different packet-drop probabilities but also packet reordering; see
   [RFC8837] and [RFC7657] for further discussion.  As a default policy,
   all RTP packets related to an RTP packet stream ought to be provided
   with the same prioritization; per-packet prioritization is outside
   the scope of this memo but might be specified elsewhere in future.

   It is also important to consider how RTCP packets associated with a
   particular RTP packet stream need to be marked.  RTCP compound
   packets with Sender Reports (SRs) ought to be marked with the same
   priority as the RTP packet stream itself, so the RTCP-based round-
   trip time (RTT) measurements are done using the same transport-layer
   flow priority as the RTP packet stream experiences.  RTCP compound
   packets containing an RR packet ought to be sent with the priority
   used by the majority of the RTP packet streams reported on.  RTCP
   packets containing time-critical feedback packets can use higher
   priority to improve the timeliness and likelihood of delivery of such

12.2.  Media Source, RTP Streams, and Participant Identification

12.2.1.  Media Source Identification

   Each RTP packet stream is identified by a unique synchronization
   source (SSRC) identifier.  The SSRC identifier is carried in each of
   the RTP packets comprising an RTP packet stream, and is also used to
   identify that stream in the corresponding RTCP reports.  The SSRC is
   chosen as discussed in Section 4.8.  The first stage in
   demultiplexing RTP and RTCP packets received on a single transport-
   layer flow at a WebRTC endpoint is to separate the RTP packet streams
   based on their SSRC value; once that is done, additional
   demultiplexing steps can determine how and where to render the media.

   RTP allows a mixer, or other RTP-layer middlebox, to combine encoded
   streams from multiple media sources to form a new encoded stream from
   a new media source (the mixer).  The RTP packets in that new RTP
   packet stream can include a contributing source (CSRC) list,
   indicating which original SSRCs contributed to the combined source
   stream.  As described in Section 4.1, implementations need to support
   reception of RTP data packets containing a CSRC list and RTCP packets
   that relate to sources present in the CSRC list.  The CSRC list can
   change on a packet-by-packet basis, depending on the mixing operation
   being performed.  Knowledge of what media sources contributed to a
   particular RTP packet can be important if the user interface
   indicates which participants are active in the session.  Changes in
   the CSRC list included in packets need to be exposed to the WebRTC
   application using some API if the application is to be able to track
   changes in session participation.  It is desirable to map CSRC values
   back into WebRTC MediaStream identities as they cross this API, to
   avoid exposing the SSRC/CSRC namespace to WebRTC applications.

   If the mixer-to-client audio level extension [RFC6465] is being used
   in the session (see Section 5.2.3), the information in the CSRC list
   is augmented by audio-level information for each contributing source.
   It is desirable to expose this information to the WebRTC application
   using some API, after mapping the CSRC values to WebRTC MediaStream
   identities, so it can be exposed in the user interface.

12.2.2.  SSRC Collision Detection

   The RTP standard requires RTP implementations to have support for
   detecting and handling SSRC collisions -- i.e., be able to resolve
   the conflict when two different endpoints use the same SSRC value
   (see Section 8.2 of [RFC3550]).  This requirement also applies to
   WebRTC endpoints.  There are several scenarios where SSRC collisions
   can occur:

   *  In a point-to-point session where each SSRC is associated with
      either of the two endpoints and the main media-carrying SSRC
      identifier will be announced in the signaling channel, a collision
      is less likely to occur due to the information about used SSRCs.
      If SDP is used, this information is provided by source-specific
      SDP attributes [RFC5576].  Still, collisions can occur if both
      endpoints start using a new SSRC identifier prior to having
      signaled it to the peer and received acknowledgement on the
      signaling message.  "Source-Specific Media Attributes in the
      Session Description Protocol (SDP)" [RFC5576] contains a mechanism
      to signal how the endpoint resolved the SSRC collision.

   *  SSRC values that have not been signaled could also appear in an
      RTP session.  This is more likely than it appears, since some RTP
      functions use extra SSRCs to provide their functionality.  For
      example, retransmission data might be transmitted using a separate
      RTP packet stream that requires its own SSRC, separate from the
      SSRC of the source RTP packet stream [RFC4588].  In those cases,
      an endpoint can create a new SSRC that strictly doesn't need to be
      announced over the signaling channel to function correctly on both
      RTP and RTCPeerConnection level.

   *  Multiple endpoints in a multiparty conference can create new
      sources and signal those towards the RTP middlebox.  In cases
      where the SSRC/CSRC are propagated between the different endpoints
      from the RTP middlebox, collisions can occur.

   *  An RTP middlebox could connect an endpoint's RTCPeerConnection to
      another RTCPeerConnection from the same endpoint, thus forming a
      loop where the endpoint will receive its own traffic.  While it is
      clearly considered a bug, it is important that the endpoint be
      able to recognize and handle the case when it occurs.  This case
      becomes even more problematic when media mixers and such are
      involved, where the stream received is a different stream but
      still contains this client's input.

   These SSRC/CSRC collisions can only be handled on the RTP level when
   the same RTP session is extended across multiple RTCPeerConnections
   by an RTP middlebox.  To resolve the more generic case where multiple
   RTCPeerConnections are interconnected, identification of the media
   source or sources that are part of a MediaStreamTrack being
   propagated across multiple interconnected RTCPeerConnection needs to
   be preserved across these interconnections.

12.2.3.  Media Synchronization Context

   When an endpoint sends media from more than one media source, it
   needs to consider if (and which of) these media sources are to be
   synchronized.  In RTP/RTCP, synchronization is provided by having a
   set of RTP packet streams be indicated as coming from the same
   synchronization context and logical endpoint by using the same RTCP
   CNAME identifier.

   The next provision is that the internal clocks of all media sources
   -- i.e., what drives the RTP timestamp -- can be correlated to a
   system clock that is provided in RTCP Sender Reports encoded in an
   NTP format.  By correlating all RTP timestamps to a common system
   clock for all sources, the timing relation of the different RTP
   packet streams, also across multiple RTP sessions, can be derived at
   the receiver and, if desired, the streams can be synchronized.  The
   requirement is for the media sender to provide the correlation
   information; whether or not the information is used is up to the

13.  Security Considerations

   The overall security architecture for WebRTC is described in
   [RFC8827], and security considerations for the WebRTC framework are
   described in [RFC8826].  These considerations also apply to this

   The security considerations of the RTP specification, the RTP/SAVPF
   profile, and the various RTP/RTCP extensions and RTP payload formats
   that form the complete protocol suite described in this memo apply.
   It is believed that there are no new security considerations
   resulting from the combination of these various protocol extensions.

   "Extended Secure RTP Profile for Real-time Transport Control Protocol
   (RTCP)-Based Feedback (RTP/SAVPF)" [RFC5124] provides handling of
   fundamental issues by offering confidentiality, integrity, and
   partial source authentication.  A media-security solution that is
   mandatory to implement and use is created by combining this secured
   RTP profile and DTLS-SRTP keying [RFC5764], as defined by Section 5.5
   of [RFC8827].

   RTCP packets convey a Canonical Name (CNAME) identifier that is used
   to associate RTP packet streams that need to be synchronized across
   related RTP sessions.  Inappropriate choice of CNAME values can be a
   privacy concern, since long-term persistent CNAME identifiers can be
   used to track users across multiple WebRTC calls.  Section 4.9 of
   this memo mandates generation of short-term persistent RTCP CNAMES,
   as specified in RFC 7022, resulting in untraceable CNAME values that
   alleviate this risk.

   Some potential denial-of-service attacks exist if the RTCP reporting
   interval is configured to an inappropriate value.  This could be done
   by configuring the RTCP bandwidth fraction to an excessively large or
   small value using the SDP "b=RR:" or "b=RS:" lines [RFC3556] or some
   similar mechanism, or by choosing an excessively large or small value
   for the RTP/AVPF minimal receiver report interval (if using SDP, this
   is the "a=rtcp-fb:... trr-int" parameter) [RFC4585].  The risks are
   as follows:

   1.  the RTCP bandwidth could be configured to make the regular
       reporting interval so large that effective congestion control
       cannot be maintained, potentially leading to denial of service
       due to congestion caused by the media traffic;

   2.  the RTCP interval could be configured to a very small value,
       causing endpoints to generate high-rate RTCP traffic, potentially
       leading to denial of service due to the RTCP traffic not being
       congestion controlled; and

   3.  RTCP parameters could be configured differently for each
       endpoint, with some of the endpoints using a large reporting
       interval and some using a smaller interval, leading to denial of
       service due to premature participant timeouts due to mismatched
       timeout periods that are based on the reporting interval.  This
       is a particular concern if endpoints use a small but nonzero
       value for the RTP/AVPF minimal receiver report interval (trr-int)
       [RFC4585], as discussed in Section 6.1 of [RFC8108].

   Premature participant timeout can be avoided by using the fixed
   (nonreduced) minimum interval when calculating the participant
   timeout (see Section 4.1 of this memo and Section 7.1.2 of
   [RFC8108]).  To address the other concerns, endpoints SHOULD ignore
   parameters that configure the RTCP reporting interval to be
   significantly longer than the default five-second interval specified
   in [RFC3550] (unless the media data rate is so low that the longer
   reporting interval roughly corresponds to 5% of the media data rate),
   or that configure the RTCP reporting interval small enough that the
   RTCP bandwidth would exceed the media bandwidth.

   The guidelines in [RFC6562] apply when using variable bitrate (VBR)
   audio codecs such as Opus (see Section 4.3 for discussion of mandated
   audio codecs).  The guidelines in [RFC6562] also apply, but are of
   lesser importance, when using the client-to-mixer audio level header
   extensions (Section 5.2.2) or the mixer-to-client audio level header
   extensions (Section 5.2.3).  The use of the encryption of the header
   extensions are RECOMMENDED, unless there are known reasons, like RTP
   middleboxes performing voice-activity-based source selection or
   third-party monitoring that will greatly benefit from the
   information, and this has been expressed using API or signaling.  If
   further evidence is produced to show that information leakage is
   significant from audio-level indications, then use of encryption
   needs to be mandated at that time.

   In multiparty communication scenarios using RTP middleboxes, a lot of
   trust is placed on these middleboxes to preserve the session's
   security.  The middlebox needs to maintain confidentiality and
   integrity and perform source authentication.  As discussed in
   Section 12.1.1, the middlebox can perform checks that prevent any
   endpoint participating in a conference from impersonating another.
   Some additional security considerations regarding multiparty
   topologies can be found in [RFC7667].

14.  IANA Considerations

   This document has no IANA actions.

15.  References

15.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

   [RFC2736]  Handley, M. and C. Perkins, "Guidelines for Writers of RTP
              Payload Format Specifications", BCP 36, RFC 2736,
              DOI 10.17487/RFC2736, December 1999,

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003, <>.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              DOI 10.17487/RFC3551, July 2003,

   [RFC3556]  Casner, S., "Session Description Protocol (SDP) Bandwidth
              Modifiers for RTP Control Protocol (RTCP) Bandwidth",
              RFC 3556, DOI 10.17487/RFC3556, July 2003,

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, DOI 10.17487/RFC3711, March 2004,

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <>.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
              DOI 10.17487/RFC4585, July 2006,

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              DOI 10.17487/RFC4588, July 2006,

   [RFC4961]  Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
              BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007,

   [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
              "Codec Control Messages in the RTP Audio-Visual Profile
              with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
              February 2008, <>.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
              2008, <>.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
              2009, <>.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761,
              DOI 10.17487/RFC5761, April 2010,

   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764,
              DOI 10.17487/RFC5764, May 2010,

   [RFC6051]  Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
              Flows", RFC 6051, DOI 10.17487/RFC6051, November 2010,

   [RFC6464]  Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time
              Transport Protocol (RTP) Header Extension for Client-to-
              Mixer Audio Level Indication", RFC 6464,
              DOI 10.17487/RFC6464, December 2011,

   [RFC6465]  Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real-
              time Transport Protocol (RTP) Header Extension for Mixer-
              to-Client Audio Level Indication", RFC 6465,
              DOI 10.17487/RFC6465, December 2011,

   [RFC6562]  Perkins, C. and JM. Valin, "Guidelines for the Use of
              Variable Bit Rate Audio with Secure RTP", RFC 6562,
              DOI 10.17487/RFC6562, March 2012,

   [RFC6904]  Lennox, J., "Encryption of Header Extensions in the Secure
              Real-time Transport Protocol (SRTP)", RFC 6904,
              DOI 10.17487/RFC6904, April 2013,

   [RFC7007]  Terriberry, T., "Update to Remove DVI4 from the
              Recommended Codecs for the RTP Profile for Audio and Video
              Conferences with Minimal Control (RTP/AVP)", RFC 7007,
              DOI 10.17487/RFC7007, August 2013,

   [RFC7022]  Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              "Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
              September 2013, <>.

   [RFC7160]  Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple
              Clock Rates in an RTP Session", RFC 7160,
              DOI 10.17487/RFC7160, April 2014,

   [RFC7164]  Gross, K. and R. Brandenburg, "RTP and Leap Seconds",
              RFC 7164, DOI 10.17487/RFC7164, March 2014,

   [RFC7742]  Roach, A.B., "WebRTC Video Processing and Codec
              Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,

   [RFC7874]  Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing
              Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016,

   [RFC8083]  Perkins, C. and V. Singh, "Multimedia Congestion Control:
              Circuit Breakers for Unicast RTP Sessions", RFC 8083,
              DOI 10.17487/RFC8083, March 2017,

   [RFC8108]  Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
              "Sending Multiple RTP Streams in a Single RTP Session",
              RFC 8108, DOI 10.17487/RFC8108, March 2017,

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <>.

   [RFC8285]  Singer, D., Desineni, H., and R. Even, Ed., "A General
              Mechanism for RTP Header Extensions", RFC 8285,
              DOI 10.17487/RFC8285, October 2017,

   [RFC8825]  Alvestrand, H., "Overview: Real-Time Protocols for
              Browser-Based Applications", RFC 8825,
              DOI 10.17487/RFC8825, January 2021,

   [RFC8826]  Rescorla, E., "Security Considerations for WebRTC",
              RFC 8826, DOI 10.17487/RFC8826, January 2021,

   [RFC8827]  Rescorla, E., "WebRTC Security Architecture", RFC 8827,
              DOI 10.17487/RFC8827, January 2021,

   [RFC8843]  Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", RFC 8843,
              DOI 10.17487/RFC8843, January 2021,

   [RFC8854]  Uberti, J., "WebRTC Forward Error Correction
              Requirements", RFC 8854, DOI 10.17487/RFC8854, January
              2021, <>.

   [RFC8858]  Holmberg, C., "Indicating Exclusive Support of RTP and RTP
              Control Protocol (RTCP) Multiplexing Using the Session
              Description Protocol (SDP)", RFC 8858,
              DOI 10.17487/RFC8858, January 2021,

   [RFC8860]  Westerlund, M., Perkins, C., and J. Lennox, "Sending
              Multiple Types of Media in a Single RTP Session",
              RFC 8860, DOI 10.17487/RFC8860, January 2021,

   [RFC8861]  Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
              "Sending Multiple RTP Streams in a Single RTP Session:
              Grouping RTP Control Protocol (RTCP) Reception Statistics
              and Other Feedback", RFC 8861, DOI 10.17487/RFC8861,
              January 2021, <>.

              Jennings, C., Aboba, B., Bruaroey, J-I., and H. Boström,
              "Media Capture and Streams", W3C Candidate Recommendation,

              Jennings, C., Boström, H., and J-I. Bruaroey, "WebRTC 1.0:
              Real-time Communication Between Browsers", W3C Proposed
              Recommendation, <>.

15.2.  Informative References

   [RFC3611]  Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
              "RTP Control Protocol Extended Reports (RTCP XR)",
              RFC 3611, DOI 10.17487/RFC3611, November 2003,

   [RFC4383]  Baugher, M. and E. Carrara, "The Use of Timed Efficient
              Stream Loss-Tolerant Authentication (TESLA) in the Secure
              Real-time Transport Protocol (SRTP)", RFC 4383,
              DOI 10.17487/RFC4383, February 2006,

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,

   [RFC5968]  Ott, J. and C. Perkins, "Guidelines for Extending the RTP
              Control Protocol (RTCP)", RFC 5968, DOI 10.17487/RFC5968,
              September 2010, <>.

   [RFC6263]  Marjou, X. and A. Sollaud, "Application Mechanism for
              Keeping Alive the NAT Mappings Associated with RTP / RTP
              Control Protocol (RTCP) Flows", RFC 6263,
              DOI 10.17487/RFC6263, June 2011,

   [RFC6792]  Wu, Q., Ed., Hunt, G., and P. Arden, "Guidelines for Use
              of the RTP Monitoring Framework", RFC 6792,
              DOI 10.17487/RFC6792, November 2012,

   [RFC7478]  Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use Cases and Requirements", RFC 7478,
              DOI 10.17487/RFC7478, March 2015,

   [RFC7656]  Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
              B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
              for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
              DOI 10.17487/RFC7656, November 2015,

   [RFC7657]  Black, D., Ed. and P. Jones, "Differentiated Services
              (Diffserv) and Real-Time Communication", RFC 7657,
              DOI 10.17487/RFC7657, November 2015,

   [RFC7667]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
              DOI 10.17487/RFC7667, November 2015,

   [RFC8088]  Westerlund, M., "How to Write an RTP Payload Format",
              RFC 8088, DOI 10.17487/RFC8088, May 2017,

   [RFC8445]  Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
              Connectivity Establishment (ICE): A Protocol for Network
              Address Translator (NAT) Traversal", RFC 8445,
              DOI 10.17487/RFC8445, July 2018,

   [RFC8829]  Uberti, J., Jennings, C., and E. Rescorla, Ed.,
              "JavaScript Session Establishment Protocol (JSEP)",
              RFC 8829, DOI 10.17487/RFC8829, January 2021,

   [RFC8830]  Alvestrand, H., "WebRTC MediaStream Identification in the
              Session Description Protocol", RFC 8830,
              DOI 10.17487/RFC8830, January 2021,

   [RFC8836]  Jesup, R. and Z. Sarker, Ed., "Congestion Control
              Requirements for Interactive Real-Time Media", RFC 8836,
              DOI 10.17487/RFC8836, January 2021,

   [RFC8837]  Jones, P., Dhesikan, S., Jennings, C., and D. Druta,
              "Differentiated Services Code Point (DSCP) Packet Markings
              for WebRTC QoS", RFC 8837, DOI 10.17487/RFC8837, January
              2021, <>.

   [RFC8872]  Westerlund, M., Burman, B., Perkins, C., Alvestrand, H.,
              and R. Even, "Guidelines for Using the Multiplexing
              Features of RTP to Support Multiple Media Streams",
              RFC 8872, DOI 10.17487/RFC8872, January 2021,


   The authors would like to thank Bernard Aboba, Harald Alvestrand,
   Cary Bran, Ben Campbell, Alissa Cooper, Spencer Dawkins, Charles
   Eckel, Alex Eleftheriadis, Christian Groves, Chris Inacio, Cullen
   Jennings, Olle Johansson, Suhas Nandakumar, Dan Romascanu, Jim
   Spring, Martin Thomson, and the other members of the IETF RTCWEB
   working group for their valuable feedback.

Authors' Addresses

   Colin Perkins
   University of Glasgow
   School of Computing Science
   G12 8QQ
   United Kingdom


   Magnus Westerlund
   Torshamnsgatan 23
   SE-164 80 Kista


   Jörg Ott
   Technical University Munich
   Department of Informatics
   Chair of Connected Mobility
   Boltzmannstrasse 3
   85748 Garching