| RTCWEB Working Group | C. S. Perkins |
| Internet-Draft | University of Glasgow |
| Intended status: Standards Track | M. Westerlund |
| Expires: October 25, 2014 | Ericsson |
| J. Ott | |
| Aalto University | |
| April 23, 2014 |
Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
draft-ietf-rtcweb-rtp-usage-13
The Web Real-Time Communication (WebRTC) framework 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.
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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 signalling, 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-aware end-points, 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 [I-D.ietf-rtcweb-overview].
The structure of this memo is as follows. Section 2 outlines our rationale in 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 that can be used in the WebRTC context. 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 signalling 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 [sec-security] and IANA considerations [sec-iana].
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 to support many new media encodings, but 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 to define a common baseline feature set for all WebRTC implementations of RTP. This builds on the past 20 years development of RTP 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 end-points if they are beneficial for new use cases. However, they are not necessary to address the WebRTC use cases and requirements identified in [I-D.ietf-rtcweb-use-cases-and-requirements].
While the baseline set of RTP features and extensions defined in this memo is targeted at the requirements of the WebRTC framework, it is expected to be broadly useful for other conferencing-related uses of RTP. In particular, it is likely that this set of RTP features and extensions will be appropriate for other desktop or mobile video conferencing systems, or for room-based high-quality telepresence applications.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in [RFC2119]. The RFC 2119 interpretation of these key words applies only when written in ALL CAPS. 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:
This document uses the terminology from [I-D.ietf-avtext-rtp-grouping-taxonomy]. Other terms are used according to their definitions from the RTP Specification [RFC3550]. We especially note the following frequently used terms: RTP Packet Stream, RTP Session, and End-point.
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 implementations need to implement to function effectively on today's networks.
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 in all WebRTC applications.
The following RTP and RTCP features are sometimes omitted in limited functionality implementations of RTP, but are REQUIRED in all WebRTC implementations:
It is known that a significant number of legacy RTP implementations, especially those targeted at VoIP-only systems, do not support all of the above features, and in some cases do not support RTCP at all. Implementers are advised to consider the requirements for graceful degradation when interoperating with legacy implementations.
Other implementation considerations are discussed in Section 12.
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 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 end-point 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.
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 implementations 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 (i.e., implementations MUST use SRTP and SRTCP). The RTP/SAVPF profile MUST be configured using the cipher suites, DTLS-SRTP protection profiles, keying mechanisms, and other parameters described in [I-D.ietf-rtcweb-security-arch].
The set of mandatory to implement codecs and RTP payload formats for WebRTC is not specified in this memo, instead they are defined in separate specifications, such as [I-D.ietf-rtcweb-audio]. Implementations can support any codec for which an RTP payload format and associated signalling is defined. Implementation 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.
End-points 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 signalling 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 signalled in the "a=rtpmap:" lines in the media sections of the SDP. If RTP packet streams are being associated with signalling contexts based on the RTP payload type, then the assignment of RTP payload type numbers MUST be unique across signalling 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 signalling 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 different "m=" lines in an SDP bundle group [I-D.ietf-mmusic-sdp-bundle-negotiation] form a single RTP session).
An end-point that has signalled support for multiple RTP payload formats SHOULD be able to accept data in any of those payload formats at any time, unless it has previously signalled 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 signalled for the type of media that is being sent by that source; see Section 4.4. To support rapid rate adaptation by changing codec, RTP does not require advance signalling for changes between RTP payload formats used by a single SSRC that were signalled during session set-up.
An RTP sender that changes between two RTP payload types that use different RTP clock rates 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).
An association amongst a set of end-points communicating using RTP is known as an RTP session [RFC3550]. An end-point 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 implementations of RTP are REQUIRED to implement support for multimedia sessions in this way, separating each session using different transport-layer flows for compatibility with legacy systems.
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 [I-D.ietf-avtcore-multi-media-rtp-session]. 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, [I-D.ietf-mmusic-sdp-bundle-negotiation] 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 are suitable can be found in [I-D.ietf-avtcore-multiplex-guidelines].
Historically, RTP and RTCP have been run on separate transport layer flows (e.g., two UDP ports for each RTP session, one port for RTP and one port 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 set-up times, support for multiplexing RTP data packets and RTCP control packets on a single transport-layer flow for each RTP session is REQUIRED, provided it is negotiated in the signalling channel before use as specified in [RFC5761]. For backwards compatibility, implementations are also REQUIRED to support RTP and RTCP sent on separate transport-layer flows.
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, and is the recommend method for application level keep-alives of RTP sessions [RFC6263].
RTCP packets are usually sent as compound RTCP packets, and [RFC3550] requires that those compound packets start with an Sender Report (SR) or Receiver Report (RR) packet. When using frequent RTCP feedback messages under the RTP/AVPF Profile [RFC4585] these statistics are not needed in every packet, and unnecessarily increase the mean RTCP packet size. This can limit the frequency at which RTCP packets can be sent within the RTCP bandwidth share.
To avoid this problem, [RFC5506] specifies how to reduce the mean RTCP message size and allow for more frequent feedback. Frequent feedback, in turn, is essential to make real-time applications quickly aware of changing network conditions, and to allow them to adapt their transmission and encoding behaviour. Support for non-compound RTCP feedback packets [RFC5506] is REQUIRED, but MUST be negotiated using the signalling channel before use. For backwards compatibility, implementations are also REQUIRED to support the use of compound RTCP feedback packets if the remote end-point does not agree to the use of non-compound RTCP in the signalling exchange.
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 bi-directional 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 end-points, but also in the network as NAT mappings or firewall state is not unnecessary bloated. The amount of per flow QoS state kept in the network is also reduced.
Implementations are REQUIRED to support signalled RTP synchronisation source (SSRC) identifiers, using the "a=ssrc:" SDP attribute defined in Section 4.1 and Section 5 of [RFC5576]. Implementations MUST also support the "previous-ssrc" source attribute defined in Section 6.2 of [RFC5576]. Other per-SSRC attributes defined in [RFC5576] MAY be supported.
Use of the "a=ssrc:" attribute to signal SSRC identifiers in an RTP session is OPTIONAL. Implementations MUST be prepared to accept RTP and RTCP packets using SSRCs that have not been explicitly signalled ahead of time. Implementations MUST support random SSRC assignment, and MUST support SSRC collision detection and resolution, according to [RFC3550]. When using signalled 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 are 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 signalled 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 signalled, the RTP payload type numbers used in an RTP packet stream are often sufficient to associate that packet stream with a signalling context (e.g., 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).
The RTCP Canonical Name (CNAME) provides a persistent transport-level identifier for an RTP end-point. While the Synchronisation Source (SSRC) identifier for an RTP end-point 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 a RTCPeerConnection [W3C.WD-webrtc-20130910], so that RTP end-points can be uniquely identified and associated with their RTP packet streams within a set of related RTP sessions.
Each RTP end-point MUST have at least one RTCP CNAME, and that RTCP CNAME MUST be unique within the RTCPeerConnection. RTCP CNAMEs identify a particular synchronisation context, i.e., all SSRCs associated with a single RTCP CNAME share a common reference clock. If an end-point has SSRCs that are associated with several unsynchronised reference clocks, and hence different synchronisation contexts, it will need to use multiple RTCP CNAMEs, one for each synchronisation context.
Taking the discussion in Section 11 into account, a WebRTC end-point MUST NOT use more than one RTCP CNAME in the RTP sessions belonging to single RTCPeerConnection (that is, an RTCPeerConnection forms a synchronisation 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 end-points part of a multi-party 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 [sec-security]. 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 as an existing RTCPeerConnection within their common same-origin context.
An WebRTC end-point 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.
The guidelines regarding handling of leap seconds to limit their impact on RTP media playout and synchronization given in [RFC7164] SHOULD be followed.
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 application 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 the WebRTC context.
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 [I-D.ietf-avtcore-rtp-topologies-update].
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 implementations are not expected to support topologies based on IP multicast groups or to support mesh-based topologies, such as a point-to-multipoint mesh configured as a single RTP session (Topo-Mesh in the terminology of [I-D.ietf-avtcore-rtp-topologies-update]). However, a point-to-multipoint mesh constructed using several RTP sessions, implemented in the WebRTC context using independent RTCPeerConnections, can be expected to be utilised by WebRTC applications and needs to be supported.
WebRTC implementations of RTP endpoints implemented according to this memo are expected to support all the topologies described in [I-D.ietf-avtcore-rtp-topologies-update] 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 the WebRTC environment. Specifically:
The following topology can be used, however it has some issues worth noting:
The RTP extensions described in Section 5.1.1 to Section 5.1.6 are designed to be used with centralised 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 end-point 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 and, to provide a reasonable baseline quality, some of these extensions are mandatory to be supported by WebRTC end-points.
The RTCP conferencing extensions are defined in Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF) [RFC4585] and the memo on Codec Control Messages (CCM) in RTP/AVPF [RFC5104]; they are fully usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
The Full Intra Request message is defined in Sections 3.5.1 and 4.3.1 of the 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 senders MUST understand and react to FIR feedback messages they receiver, since this greatly improves the user experience when using centralised mixer-based conferencing. Support for sending FIR messages is OPTIONAL.
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 fulfil the request. WebRTC senders MUST understand and react to PLI feedback messages as a loss tolerance mechanism. Receivers MAY send PLI messages.
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).
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 learnt that encoder-decoder synchronisation 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 signalled. Receivers that detect that encoder-decoder synchronisation has been lost SHOULD generate an RPSI feedback message if codec being used supports reference picture selection. A 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.
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.
The TMMBR feedback message is defined in Sections 3.5.4 and 4.2.1 of the Codec Control Messages [RFC5104]. This request and its notification message 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. This 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 senders 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 requests is OPTIONAL.
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 the WebRTC context, but if they are used, they MUST be formatted and signalled following the general mechanism for RTP header extensions defined in [RFC5285], since this gives well-defined semantics to RTP header extensions.
As noted in [RFC5285], 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 signalled (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.
Many RTP sessions require synchronisation between audio, video, and other content. This synchronisation 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 synchronisation extensions described in [RFC6051] be implemented in addition to RTCP SR-based synchronisation. The rapid synchronisation extensions use the general RTP header extension mechanism [RFC5285], which requires signalling, but are otherwise backwards compatible.
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 mixer. 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 [RFC6464] header extension is RECOMMENDED to be implemented. If this header extension is implemented, it is REQUIRED that implementations are 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 is used, unless the encryption has been explicitly disabled through API or signalling.
The Mixer to Client Audio Level header extension [RFC6465] provides an endpoint with the audio level of the different sources mixed into a common mix by a 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 optimisations of non critical functions, and is hence OPTIONAL to implement. If this header extension is implemented, it is REQUIRED that implementations are 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 is used, unless the encryption has been explicitly disabled through API or signalling.
There are tools that can make RTP packet streams robust against packet loss and reduce the impact of loss on media quality. However, they all add overhead compared to a non-robust stream. The overhead needs to be considered, and the aggregate bit-rate 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 sub-sections can be used to improve tolerance to packet loss.
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 optimise 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 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 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 caused increased packet loss if the original packet loss was caused by network congestion. We 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 needs to be considered before RTP retransmission is used.
Receivers are REQUIRED to implement support for RTP retransmission packets [RFC4588]. Senders MAY send RTP retransmission packets in response to NACKs if the RTP retransmission payload format has been negotiated for the session, and if the sender believes it is useful to send a retransmission of the packet(s) referenced in the NACK. An RTP sender does not need to retransmit every NACKed packet.
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, others operate across RTP packets and can be used with any payload format. It needs to be noted that using redundant encoding or FEC will lead to increased play out delay, which needs to be considered when choosing the redundancy or FEC formats and their respective parameters.
If an RTP payload format negotiated for use in a RTCPeerConnection supports redundant transmission or FEC as a standard feature of that payload format, then that support MAY be used in the RTCPeerConnection, subject to any appropriate signalling.
There are several block-based FEC schemes that are designed for use with RTP independent of the chosen RTP payload format. At the time of this writing there is no consensus on which, if any, of these FEC schemes is appropriate for use in the WebRTC context. Accordingly, this memo makes no recommendation on the choice of block-based FEC for WebRTC use.
WebRTC will be used in heterogeneous network environments using a variety set 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 bit-rates, load levels, and traffic mixtures. Individual end-points can send one or more RTP packet streams to each participant in a WebRTC conference, and there can be several participants. Each of these RTP packet streams can contain different types of media, and the type of media, bit rate, 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 end-points 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 implementation is very dependent on effective adaptation of the media to the limitations of the network. End-points 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 [I-D.ietf-rmcat-cc-requirements]. It is expected that a future version of this memo will mandate the use of a congestion control algorithm that satisfies these requirements.
In the absence of a concrete congestion control algorithm, all WebRTC implementations MUST implement the RTP circuit breaker algorithm that is described in [I-D.ietf-avtcore-rtp-circuit-breakers]. The RTP circuit breaker is designed to enable applications to recognise 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. Any future RTP congestion control algorithms are expected to operate within the envelope allowed by the circuit breaker.
The session establishment signalling will also necessarily establish boundaries to which the media bit-rate will conform. The choice of media codecs provides upper- and lower-bounds on the supported bit-rates that the application can utilise to provide useful quality, and the packetization choices that exist. In addition, the signalling channel can establish maximum media bit-rate boundaries using the SDP "b=AS:" or "b=CT:" lines, and the RTP/AVPF Temporary Maximum Media Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of this memo). The combination of media codec choice and signalled bandwidth limits SHOULD be used to limit traffic based on known bandwidth limitations, for example the capacity of the edge links, to the extent possible.
Experience with the congestion control algorithms of TCP [RFC5681], TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown that feedback on packet arrivals needs to be sent frequently (roughly once per round trip time is common). We note that the real-time media traffic might not be able to adapt to changing path conditions as rapidly as elastic applications using TCP, but frequent feedback, perhaps on the order of once per video frame, is still needed to allow the congestion control algorithm to track the path dynamics.
As an example of the type of RTCP congestion control feedback that is possible, consider one of the simplest scenarios for WebRTC: a point to point video call between two end systems. There will be four RTP flows in this scenario, two audio and two video, with all four flows being active for essentially all the time (the audio flows will likely use voice activity detection and comfort noise to reduce the packet rate during silent periods, but doesn't cause transmissions to stop). Assume all four flows are sent in a single RTP session, each using a separate SSRC. Further, assume each SSRC sends RTCP reports for all other SSRCs in the session (i.e., the optimisations in [I-D.ietf-avtcore-rtp-multi-stream-optimisation] are not used, giving the worst case for the RTCP overhead). When all members are senders like this, the RTCP timing rules in Sections 6.2 and 6.3 of [RFC3550] and [RFC4585] reduce to:
rtcp_interval = avg_rtcp_size * n / rtcp_bw
where avg_rtcp_size is measured in octets, and the rtcp_bw is the bandwidth available for RTCP. The average RTCP size will depend on the amount of feedback that is sent in each RTCP packet, on the number of members in the session, and on the size of source description (RTCP SDES) information sent. As a baseline, each RTCP packet will be a compound RTCP packet that contains an RTCP SR and an RTCP SDES packet. In the scenario above, each RTCP SR packet will contain three report blocks, once for each of the other RTP SSRCs sending data, for a total of 100 octets (this is 8 octets header, 20 octets sender info, and 3 * 24 octets report blocks). The RTCP SDES packet will comprise a header (4 octets), an originating SSRC (4 octets), a CNAME chunk, and padding. If the CNAME follows [RFC7022] and it will be 19 octets in size, and require 1 octet of padding. The resulting compound RTCP packet will be 128 octets in size. If sent in UDP/IPv4 with no IP options and using Secure RTP, which adds 20 (IPv4) + 8 (UDP) + 14 (SRTP with 80 bit Authentication tag), the avg_rtcp_size will therefore be 170 octets, including the header overhead. The value n is this scenario is 4, and the rtcp_bw is assumed to be 5% of the session bandwidth.
If it is desired to send RTCP feedback packets on average 30 times per second, to correspond to one RTCP report every frame for 30fps video, we can invert the above rtcp_interval calculation to get an rtcp_bw that gives an interval of 1/30th of a second or lower. This corresponds to an rtcp_bw of 20400 octets per second (since 1/30 = 170 * 4 / 20400). This is 163200 bits per second, which if 5% of the session bandwidth, gives a session bandwidth of approximately 3.3Mbps (i.e., 3.3Mbps media rate, plus an additional 5% for RTCP, to give a total data rate of approximately 3.4Mbps). That is, RTCP can report on every frame of video provided the session bandwidth is 3.3Mbps or larger, when every SSRC sends a report for every video frame. Please note that the actual RTCP transmission intervals will be within the interval [0.0135, 0.0406]s, but maintaining an average RTCP transmission interval of 0.033s.
If additional feedback beyond the standard report block is needed, the session bandwidth needed will increase. For example, with an additional 20 octets data being reported in each RTCP packet, the session bandwidth needed increases to 3.5Mbps for every SSRC to be able to report on every frame. However, the above baseline might not be the most appropriate usage of the RTCP bandwidth. Depending on needs, a less frequent usage of regular RTCP compound packets, controlled by T_rr_interval combined with using the reduced size RTCP packets, can achieve more frequent and useful reporting. Also the reporting requirements defined in [I-D.ietf-avtcore-rtp-multi-stream-optimisation] will reduced the amount of bandwidth consumed for reporting when each endpoint has multiple SSRCs.
Calculations such as these show that RTCP cannot be used to send per-packet congestion feedback. RTCP can, however, be used to send congestion feedback on each frame of video sent in an interactive video conferencing scenario, provided the RTCP parameters are correctly configured and the overall session bandwidth exceeds a couple of megabits per second (the exact rate depending on the number of session participants, the RTCP bandwidth fraction, and whether audio and video are sent in one or two RTP sessions). Using similar calculations, it can be shown that RTCP can likely also be used to send feedback on a per-RTT basis, provided the RTT is not too low.
Interactive communication might not be able to afford to wait for packet losses to occur to indicate congestion, because an increase in play out delay due to queuing (most prominent in wireless networks) can easily lead to packets being dropped due to late arrival at the receiver. Therefore, more sophisticated cues might need to be reported -- to be defined in a suitable congestion control framework as noted above -- which, in turn, increase the report size again. For example, different RTCP XR report blocks (jointly) provide the necessary details to implement a variety of congestion control algorithms, but the (compound) report size grows quickly.
There are legacy RTP implementations that do not implement RTCP, and hence do not provide any congestion feedback. Congestion control cannot be performed with these end-points. WebRTC implementations that need to interwork with such end-points MUST limit their transmission to a low rate, equivalent to a VoIP call using a low bandwidth codec, that is unlikely to cause any significant congestion.
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 end-points MUST ensure that they keep within the RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the congestion they can cause.
If a legacy end-point supports RTP/AVPF, this enables negotiation of important parameters for frequent reporting, such as the "trr-int" parameter, and the possibility that the end-point supports some useful feedback format for congestion control purpose such as TMMBR [RFC5104]. Implementations that have to interwork with such end-points MUST ensure that they stay within the RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the congestion they can cause, but 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 situation where one direction is dual controlled, when the other direction is not controlled. This memo cannot mandate behaviour for proprietary congestion control algorithms, but implementations that use such algorithms ought to be aware of this issue, and try to ensure that both effective congestion control is negotiated for media flowing in both directions. If the IETF were to standardise 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.
As described in Section 4.1, implementations are REQUIRED to generate RTCP Sender Report (SR) and Reception 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 [RFC3611][RFC6792]. At the time of this writing, it is not clear what extended metrics are suitable for use in the WebRTC context, 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 such packets SHOULD be signalled in advance.
All WebRTC implementations MUST be prepared to receive RTP XR report packets, whether or not they were signalled. There is no requirement that the data contained in such reports be used, or exposed to the Javascript application, however.
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 applications. In this case, future updates to this memo MUST be made following the Guidelines for Writers of RTP Payload Format Specifications [RFC2736], How to Write an RTP Payload Format [I-D.ietf-payload-rtp-howto] and Guidelines for Extending the RTP Control Protocol [RFC5968], and SHOULD take into account any future guidelines for extending RTP and related protocols that have been developed.
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.
RTP is built with the assumption that an external signalling channel exists, and can be used to configure RTP sessions and their features. The basic configuration of an RTP session consists of the following parameters:
These parameters are often expressed in SDP messages conveyed within an offer/answer exchange. RTP does not depend on SDP or on the offer/answer model, but does require all the necessary parameters to be agreed upon, and provided to the RTP implementation. We note that in the WebRTC context it will depend on the signalling model and API how these parameters need to be configured but they will be need to either set in the API or explicitly signalled between the peers.
The WebRTC API [W3C.WD-webrtc-20130910] and the Media Capture and Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses 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 like a microphone or a camera, but also conceptual sources, like a audio mix or a video composition, are possible. The MediaStreamTracks within a MediaStream need to be possible to play out synchronised.
A MediaStreamTrack's realisation in RTP in the context of an RTCPeerConnection consists of a source packet stream identified with an SSRC within an RTP session 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 [sec-FEC] or RTP retransmission [sec-rtx] 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 a 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. Thus it is possible for multiple source packet streams to share encoded streams (but not packet streams), but this is an implementation choice to try to utilise such optimisations. Note that such optimizations would need to take into account that the constraints for one of the MediaStreamTracks can at any moment change, meaning that the encoding configurations might no longer be identical.
The same MediaStreamTrack can also be included in multiple MediaStreams, thus multiple sets of MediaStreams can implicitly need to use the same synchronisation base. To ensure that this works in all cases, and don't forces a end-point to change synchronisation base and CNAME in the middle of a ongoing delivery of any packet streams, which would cause media disruption; all MediaStreamTracks and their associated SSRCs originating from the same end-point needs to be sent using the same CNAME within one RTCPeerConnection. This is motivating the strong recommendation in Section 4.9 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 [I-D.ietf-rtcweb-security] for details). A web application can request that the CNAMEs used in different RTCPeerConnection within a same-orign context to be the same, this allow for synchronization of the endpoint's RTP packet streams across the different RTCPeerConnections.
The above will currently force a WebRTC end-point that receives an MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing on any RTCPeerConnection to perform resynchronisation of the stream. This, as the sending party needs to change the CNAME, which implies that it has to use a locally available system clock as timebase for the synchronisation. Thus, the relative relation between the timebase of the incoming stream and the system sending out needs to defined. This relation also needs monitoring for clock drift and likely adjustments of the synchronisation. The sending entity is also responsible for congestion control for its the 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 etc followed by encoding of the media again into the outgoing packet stream. Optimisations of this method is clearly possible and implementation specific.
A WebRTC end-point MUST support receiving multiple MediaStreamTracks, where each of different MediaStreamTracks (and their sets of associated packet streams) uses different CNAMEs. However, MediaStreamTracks that are received with different CNAMEs have no defined synchronisation.
The binding between the WebRTC MediaStreams, MediaStreamTracks and the SSRC is done as specified in "Cross Session Stream Identification in the Session Description Protocol" [I-D.ietf-mmusic-msid]. This document [I-D.ietf-mmusic-msid] also defines, in section 4.1, how to map unknown source packet stream SSRCs to MediaStreamTracks and MediaStreams. 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 [sec-rtp-rtcp] as well as the Mixer-to-Client audio levels [sec-mixer-to-client] (when supported) and the basic requirements for this is further discussed in Section 12.2.1.
The following discussion provides some guidance on the implementation of the RTP features described in this memo. The focus is on a WebRTC end-point implementation perspective, and while some mention is made of the behaviour of middleboxes, that is not the focus of this memo.
A WebRTC end-point will be a simultaneous participant in one or more RTP sessions. Each RTP session can convey multiple media sources, and can include media data from multiple end-points. In the following, we outline some ways in which WebRTC end-points can configure and use RTP sessions.
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:
In addition to sending and receiving multiple RTP packet streams within a single RTP session, a WebRTC end-point might participate in multiple RTP sessions. There are several reasons why a WebRTC end-point might choose to do this:
+---+ +---+
| A |<--->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 1: Multi-unicast using several RTP sessions
+---+ +-------------+ +---+
| A |<---->| |<---->| B |
+---+ | RTP mixer, | +---+
| translator, |
| or other |
+---+ | middlebox | +---+
| C |<---->| |<---->| D |
+---+ +-------------+ +---+
Figure 2: RTP mixer with only unicast paths
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 end-point sending the media, which controls, both which RTP packet streams that will be sent, and their allocation of bit-rate out of the current available aggregate as determined by the congestion control.
It is expected that the WebRTC API [W3C.WD-webrtc-20130910] 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 congestion control response in how to divide the available bandwidth between the RTP packet streams. 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, in regards to how robust that codec is to packet loss. However, a default policy might to be to use the same priority for redundant RTP packet stream as for the source RTP packet stream.
Secondly, the network can prioritize transport-layer flows and sub-flows, 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 the actual mechanism to prioritize packets. This is done according to three methods:
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 in a WebRTC application. When flow-based differentiation is available the WebRTC application needs to know about it so that it can provide the separation of the RTP packet streams onto different UDP flows to enable a more granular usage of flow based differentiation. That way at least providing different prioritization of audio and video if desired by application.
DiffServ assumes that either the end-point or a classifier can mark the packets with an appropriate DSCP so that the packets are treated according to that marking. If the end-point is to mark the traffic two requirements arise in the WebRTC context: 1) The WebRTC application or browser has to know which DSCP to use and 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 "DSCP and other packet markings for RTCWeb QoS" [I-D.ietf-tsvwg-rtcweb-qos].
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 prioritise any payloads carrying only B frames less, as these are less damaging to loose. However, depending on the QoS mechanism and what markings that are applied, this can result in not only different packet drop probabilities but also packet reordering, see [I-D.ietf-tsvwg-rtcweb-qos] for further discussion. As default policy all RTP packets related to a 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 (SR), 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 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 feedback.
Each RTP packet stream is identified by a unique synchronisation source (SSRC) identifier. The SSRC identifier is carried in each of the RTP packets comprising a 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 end-point 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 needs 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 name space to JavaScript 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. This information can usefully be exposed in the user interface.
The RTP standard [RFC3550] requires any RTP implementation to have support for detecting and handling SSRC collisions, i.e., resolve the conflict when two different end-points use the same SSRC value. This requirement also applies to WebRTC end-points. There are several scenarios where SSRC collisions can occur:
These SSRC/CSRC collisions can only be handled on RTP level as long as the same RTP session is extended across multiple RTCPeerConnections by a RTP middlebox. To resolve the more generic case where multiple RTCPeerConnections are interconnected, then identification of the media source(s) part of a MediaStreamTrack being propagated across multiple interconnected RTCPeerConnection needs to be preserved across these interconnections.
When an end-point 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, synchronisation is provided by having a set of RTP packet streams be indicated as coming from the same synchronisation context and logical end-point 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; it is up to the receiver to use it or not.
The overall security architecture for WebRTC is described in [I-D.ietf-rtcweb-security-arch], and security considerations for the WebRTC framework are described in [I-D.ietf-rtcweb-security]. These considerations also apply to this memo.
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. We do not believe there are any new security considerations resulting from the combination of these various protocol extensions.
The Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides handling of fundamental issues by offering confidentiality, integrity and partial source authentication. A mandatory to implement media security solution is created by combing this secured RTP profile and DTLS-SRTP keying [RFC5764] as defined by Section 5.5 of [I-D.ietf-rtcweb-security-arch].
RTCP packets convey a Canonical Name (CNAME) identifier that is used to associate RTP packet streams that need to be synchronised 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 provides guidelines for generation of untraceable CNAME values that alleviate this risk.
The guidelines in [RFC6562] apply when using variable bit rate (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 or third party monitoring that will greatly benefit from the information, and this has been expressed using API or signalling. If further evidence are produced to show that information leakage is significant from audio level indications, then use of encryption needs to be mandated at that time.
This memo makes no request of IANA.
Note to RFC Editor: this section is to be removed on publication as an RFC.
The authors would like to thank Bernard Aboba, Harald Alvestrand, Cary Bran, Charles Eckel, Christian Groves, Cullen Jennings, Dan Romascanu, Martin Thomson, and the other members of the IETF RTCWEB working group for their valuable feedback.