--- 1/draft-ietf-rtcweb-rtp-usage-24.txt 2015-06-12 05:15:07.374471393 -0700 +++ 2/draft-ietf-rtcweb-rtp-usage-25.txt 2015-06-12 05:15:07.474473811 -0700 @@ -1,21 +1,21 @@ RTCWEB Working Group C. Perkins Internet-Draft University of Glasgow Intended status: Standards Track M. Westerlund -Expires: November 30, 2015 Ericsson +Expires: December 14, 2015 Ericsson J. Ott Aalto University - May 29, 2015 + June 12, 2015 Web Real-Time Communication (WebRTC): Media Transport and Use of RTP - draft-ietf-rtcweb-rtp-usage-24 + draft-ietf-rtcweb-rtp-usage-25 Abstract 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. @@ -28,21 +28,21 @@ Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." - This Internet-Draft will expire on November 30, 2015. + This Internet-Draft will expire on December 14, 2015. Copyright Notice Copyright (c) 2015 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents @@ -54,69 +54,69 @@ Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 5 4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . 5 4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 8 - 4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 9 + 4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 10 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 10 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 11 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 11 - 4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 11 + 4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 12 4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 12 4.10. Handling of Leap Seconds . . . . . . . . . . . . . . . . 13 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 13 - 5.1. Conferencing Extensions and Topologies . . . . . . . . . 13 + 5.1. Conferencing Extensions and Topologies . . . . . . . . . 14 5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 15 5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 15 - 5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 15 + 5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 16 5.1.4. Reference Picture Selection Indication (RPSI) . . . . 16 5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 16 5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 16 - 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 16 + 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 17 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 17 5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 17 5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 18 5.2.4. Media Stream Identification . . . . . . . . . . . . . 18 5.2.5. Coordination of Video Orientation . . . . . . . . . . 18 - 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 18 + 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 19 6.1. Negative Acknowledgements and RTP Retransmission . . . . 19 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 20 7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 20 7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 21 - 7.2. Congestion Control Interoperability and Legacy Systems . 21 + 7.2. Congestion Control Interoperability and Legacy Systems . 22 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 22 - 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 22 + 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 23 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 23 - 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 24 + 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 25 12. RTP Implementation Considerations . . . . . . . . . . . . . . 27 12.1. Configuration and Use of RTP Sessions . . . . . . . . . 27 12.1.1. Use of Multiple Media Sources Within an RTP Session 27 12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 28 12.1.3. Differentiated Treatment of RTP Packet Streams . . . 33 12.2. Media Source, RTP Packet Streams, and Participant Identification . . . . . . . . . . . . . . . . . . . . . 35 12.2.1. Media Source Identification . . . . . . . . . . . . 35 12.2.2. SSRC Collision Detection . . . . . . . . . . . . . . 36 12.2.3. Media Synchronisation Context . . . . . . . . . . . 37 13. Security Considerations . . . . . . . . . . . . . . . . . . . 37 14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 39 15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 39 16. References . . . . . . . . . . . . . . . . . . . . . . . . . 39 16.1. Normative References . . . . . . . . . . . . . . . . . . 39 - 16.2. Informative References . . . . . . . . . . . . . . . . . 42 - Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 44 + 16.2. Informative References . . . . . . . . . . . . . . . . . 43 + Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 45 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 signalling, these form the basis for many teleconferencing systems. @@ -153,21 +153,21 @@ 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 RTP feature set for all WebRTC Endpoints. This builds on - the past 20 years development of RTP to mandate the use of extensions + 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 @@ -182,40 +182,47 @@ 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: WebRTC MediaStream: The MediaStream concept defined by the W3C in - the WebRTC API [W3C.WD-mediacapture-streams-20130903]. + the WebRTC API [W3C.WD-mediacapture-streams-20130903]. 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-20130903]. 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. Transport-layer Flow: A uni-directional flow of transport packets that are identified by having a particular 5-tuple of source IP address, source port, destination IP address, destination port, and transport protocol used. Bi-directional Transport-layer Flow: A bi-directional 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 [I-D.ietf-avtext-rtp-grouping-taxonomy] and [I-D.ietf-rtcweb-overview]. Other terms are used according to their definitions from the RTP Specification [RFC3550]. Especially note the following frequently used terms: RTP Packet Stream, RTP Session, - and End-point. + 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 @@ -224,21 +231,21 @@ 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 are REQUIRED in all WebRTC Endpoints: o Support for use of multiple simultaneous SSRC values in a single - RTP session, including support for RTP end-points that send many + RTP session, including support for RTP endpoints that send many SSRC values simultaneously, following [RFC3550] and [I-D.ietf-avtcore-rtp-multi-stream]. The RTCP optimisations for multi-SSRC sessions defined in [I-D.ietf-avtcore-rtp-multi-stream-optimisation] MAY be supported; if supported the usage MUST be signalled. o Random choice of SSRC on joining a session; collision detection and resolution for SSRC values (see also Section 4.8). o Support for reception of RTP data packets containing CSRC lists, @@ -250,39 +257,41 @@ extensions. o Support for multiple synchronisation contexts. Participants that send multiple simultaneous RTP packet streams SHOULD do so as part of a single synchronisation context, using a single RTCP CNAME for all streams and allowing receivers to play the streams out in a synchronised manner. For compatibility with potential future versions of this specification, or for interoperability with non- WebRTC devices through a gateway, receivers MUST support multiple synchronisation contexts, indicated by the use of multiple RTCP - CNAMEs in an RTP session. This specification requires the usage + CNAMEs in an RTP session. This specification mandates the usage of a single CNAME when sending RTP Packet Streams in some circumstances, see Section 4.9. o Support for sending and receiving RTCP SR, RR, SDES, and BYE - packet types, with OPTIONAL support for other RTCP packet types - 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 SDP bundle negotiation extension will - use the SDP grouping framework 'mid' attribute to identify media - streams. Such endpoints MUST implement the RTCP SDES MID item - described in [I-D.ietf-mmusic-sdp-bundle-negotiation]. + 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 SDP bundle negotiation + extension will use the SDP grouping framework 'mid' attribute to + identify media streams. Such endpoints MUST implement the RTCP + SDES MID item described in + [I-D.ietf-mmusic-sdp-bundle-negotiation]. - o Support for multiple end-points in a single RTP session, and for + o 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 randomised RTCP transmission intervals to avoid synchronisation of RTCP reports; support for RTCP timer reconsideration (Section 6.3.6 of + [RFC3550]) and reverse reconsideration (Section 6.3.4 of [RFC3550]). o 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]. o Support for the reduced minimum RTCP reporting interval described in Section 6.2 of [RFC3550]. When using the reduced minimum RTCP @@ -295,21 +304,21 @@ o 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 is not audio specific, and can be used with any RTP payload format. o Ignore unknown RTCP packet types and RTP header extensions. This - to ensure robust handling of future extensions, middlebox + is to ensure robust handling of future extensions, middlebox behaviours, etc., that can result in not signalled RTCP packet types or RTP header extensions being received. If a compound RTCP packet is received that contains a mixture of known and unknown RTCP packet types, the known packets types need to be processed as usual, with only the unknown packet types being discarded. It is known that a significant number of legacy RTP implementations, especially those targeted at 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 @@ -325,33 +334,32 @@ 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, 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 - end-points via a gateway is to set the trr-int parameter to a - value representing 4 seconds, see Section 6.1 in + endpoints via a gateway is to set the trr-int parameter to a value + representing 4 seconds, see Section 6.1 in [I-D.ietf-avtcore-rtp-multi-stream]). 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 (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 @@ -368,21 +376,21 @@ 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. - End-points can signal support for multiple RTP payload formats, or + 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 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. This leads to the following considerations: @@ -399,21 +407,21 @@ 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 "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 + An endpoint that has signalled 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 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. @@ -421,22 +429,22 @@ 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 end-points communicating using RTP is - known as an RTP session [RFC3550]. An end-point can be involved in + 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 @@ -494,35 +502,34 @@ 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. Implementations MUST support sending and receiving non-compound RTCP feedback packets [RFC5506]. Use of non-compound RTCP packets MUST be negotiated using the signalling channel. If SDP is used for signalling, 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 end-point - does not agree to the use of non-compound RTCP in the signalling - exchange. + the use of compound RTCP feedback packets if the remote endpoint does + not agree to the use of non-compound RTCP in the signalling 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 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 + specifically ports at the endpoints, 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. 4.8. Choice of RTP Synchronisation Source (SSRC) Implementations are REQUIRED to support signalled RTP synchronisation source (SSRC) identifiers. If SDP is used, this MUST be done using the "a=ssrc:" SDP attribute defined in Section 4.1 and Section 5 of [RFC5576] and the "previous-ssrc" source attribute defined in Section 6.2 of [RFC5576]; other per-SSRC attributes defined in @@ -547,44 +554,44 @@ 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). 4.9. Generation of the RTCP Canonical Name (CNAME) 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 + identifier for an RTP endpoint. While the Synchronisation Source + (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 a RTCPeerConnection - [W3C.WD-webrtc-20130910], so that RTP end-points can be uniquely + [W3C.WD-webrtc-20130910], so that RTP endpoints 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 + 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 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 + If an endpoint 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 Endpoint 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. + avoid having to resynchronize media from multiple different endpoints + 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 (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 as an existing RTCPeerConnection within their common same-origin context. @@ -662,21 +669,21 @@ 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 [I-D.ietf-avtcore-rtp-topologies-update]). 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 + 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 and, 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 the memo on Codec Control Messages (CCM) in RTP/ AVPF [RFC5104]; they are fully usable by the Secure variant of this @@ -921,37 +928,37 @@ 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 [I-D.ietf-rtcweb-fec]. 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 set of link technologies, including both wired and wireless + 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 bit-rates, load levels, and traffic - mixtures. Individual end-points can send one or more RTP packet + 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, 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 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. - End-points have to be designed so they do not transmit significantly + 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 @@ -965,23 +972,25 @@ 7.1. Boundary Conditions and Circuit Breakers WebRTC Endpoints 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. + 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 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 packetisation choices that exist. In addition, the signalling channel can establish maximum media bit-rate boundaries using, for example, 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). Signalled bandwidth limitations, such as SDP "b=AS:" or @@ -993,30 +1002,30 @@ 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 mechanisms. 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 + interwork with such endpoints 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 + 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 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 + parameter, and the possibility that the endpoint supports some useful + feedback format for congestion control purpose such as TMMBR + [RFC5104]. Implementations that have to interwork with such + endpoints 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 @@ -1033,32 +1042,34 @@ 8. WebRTC Use of RTP: Performance Monitoring 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 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 such packets SHOULD be signalled in advance. + RTCP Extended Reports (XR) framework, see [RFC3611][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 + signalled; 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 MUST be made + of WebRTC. In this case, future updates to this memo have to 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 @@ -1107,46 +1118,47 @@ RTCP packet types, including any necessary parameters, MUST be signalled. This signalling is to ensure that a WebRTC Endpoint's behaviour, especially when sending, of any extensions is predictable and consistent. For robustness, and for 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 to the - end-points will be necessary. This SHALL be done as described in + 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 - end-points have a common view of the RTCP bandwidth. A common - RTCP bandwidth is important as a too different view of the - bandwidths can lead to failure to interoperate. + 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 on 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 signalling model and API how these parameters need to be configured but they will be need to either be set in the API or explicitly signalled between the peers. 11. WebRTC API Considerations 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. + are possible. The MediaStreamTracks within a MediaStream might need + to be synchronized during play back. 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 (Section 6.2) or RTP retransmission (Section 6.1) to @@ -1165,31 +1177,31 @@ choose to use only one encoded stream to create the different RTP packet streams. Note that such optimisations 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 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 synchronisation base. To ensure that this works in - all cases, and does not force an end-point to to disrupt the media by + all cases, and does not force an endpoint to disrupt the media by changing synchronisation base and CNAME during delivery of any ongoing packet streams, all MediaStreamTracks and their associated - SSRCs originating from the same end-point need to be sent using the - same CNAME within one RTCPeerConnection. This is motivating the - discussion in Section 4.9 to only use a single CNAME. + 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 on using the same CNAME for all SSRCs that - originate from the same end-point, does not require a middlebox - that forwards traffic from multiple end-points to only use a - single CNAME. + 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 [I-D.ietf-rtcweb-security] for details). A web application can request that the CNAMEs used in different RTCPeerConnections (within a same-orign context) be the same, this allows for synchronization of the endpoint's RTP packet streams across the different RTCPeerConnections. @@ -1206,122 +1218,123 @@ synchronisation. 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 synchronisation. 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 etc., followed by encoding of the media again into the outgoing packet stream. Optimisations of - this method is clearly possible and implementation specific. + this method are clearly possible and implementation specific. A WebRTC Endpoint MUST support receiving multiple MediaStreamTracks, - where each of different MediaStreamTracks (and their sets of + where each of the different MediaStreamTracks (and their sets of associated packet streams) uses different CNAMEs. However, MediaStreamTracks that are received with different CNAMEs have no defined synchronisation. 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 end-points relay/ - forward streams. It also ensures that end-points can interoperate - with certain types of multi-stream middleboxes or end-points that - are not WebRTC. + enable more efficient stream handling when endpoints relay/forward + streams. It also ensures that endpoints can interoperate with + certain types of multi-stream middleboxes or endpoints that are + not WebRTC. - 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. This later is relevant to handle some cases of legacy - interop. 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. + Javascript Session Establishment Protocol [I-D.ietf-rtcweb-jsep] + specifies that 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]. The MSID document [I-D.ietf-mmusic-msid] + also defines, in section 4.1, how to map unknown source packet stream + 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) and 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 behaviour 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 can include media data from multiple end-points. In the + and can include media data from multiple endpoints. In the following, some ways in which WebRTC Endpoints can configure and use - RTP sessions is outlined. + RTP sessions are outlined. 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 sources (e.g., one RTP session for + 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 bi- directional transport-layer flow, but will contain multiple RTP packet streams, each containing a different type of media. A - common example might be an end-point with a camera and microphone + 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 session. Multiple Capture Devices: A WebRTC Endpoint might have multiple cameras, microphones, or other media capture devices, and so 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 end-point sending multiple RTP + 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 end-point might send a RTP packet stream + Associated Repair Data: An endpoint might send a 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: An end-point can use a + Layered or Multiple Description Coding: An endpoint can use a layered media codec, for example H.264 SVC, or a multiple description codec, that generates multiple RTP packet streams, each with a distinct RTP SSRC, within a single RTP session. RTP Mixers, Translators, and Other Middleboxes: An RTP session, in the WebRTC context, is a point-to-point association between an - end-point and some other peer device, where those devices share a + 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, that 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 end-points. WebRTC is a group communication - environment and end-points need to be capable of receiving, + 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: @@ -1336,42 +1349,41 @@ To provide enhanced quality of service: Some network-based quality of service mechanisms operate on the granularity of transport layer flows. If it is desired to use these mechanisms to provide differentiated quality of service for some RTP packet streams, 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 end-point 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 centralised multiparty - conferencing systems display the active speaker in high - resolution, but show low resolution "thumbnails" of other - participants. Such systems might configure the end-points 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 bit-rate, and 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, but it can be easier to distinguish the RTP packet - streams if they arrive on separate RTP sessions on separate - transport-layer flows. + 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 centralised 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 bit-rate, and + 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, + but 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 multi-party 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 @@ -1387,39 +1399,38 @@ 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 between the - relation between RTP sessions and RTCPeerConnection objects it is - recommend that this is implemented as several individual RTP - sessions. The only downside is that end-point 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 it 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 showed - 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 bit-rates and - RTP session configurations between the different peers, thus not - forcing B to endure the same quality reductions if there are - limitations in the transport from A to C as C will. It is - believed that these advantages outweigh the limitations in - debugging power. + pair of peers. To maintain a coherent mapping of the relationship + between RTP sessions and RTCPeerConnection objects it is recommend + that this is 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 showed 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 bit-rates and RTP session + configurations between the different peers, thus not forcing B to + endure the same quality reductions if there are limitations in the + transport from A to C as C will. It is believed that these + advantages outweigh the limitations in debugging power. To indirectly connect with multiple peers: A common scenario in multi-party 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 centralised conference. The middlebox acts to optimise 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. @@ -1431,110 +1442,111 @@ | 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 - end-points in one multi-party session. Other types of middlebox - might use separate RTP sessions between each end-point and the + endpoints in one multi-party session. Other types of middlebox + 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 have the encoder produce a so called Intra frame. Another is limiting the bit-rate of a given stream to better suit the mixer view of the multiple down-streams. Others are controlling the most suitable frame-rate, picture resolution, the trade-off between frame-rate and spatial quality. The middlebox has the responsibility to correctly perform congestion control, source identification, manage synchronisation while providing the application with suitable media optimisations. 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 end-point, before sending it on - towards the end-point(s), thus they need to be able to decrypt and + itself (or both) received from one endpoint, before sending it 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. - RTP Mixers can create a situation where an end-point experiences a - situation in-between a session with only two end-points and + RTP Mixers can create a situation where an endpoint experiences a + situation in-between a session with only two endpoints and multiple RTP sessions. Mixers are expected to not forward RTCP reports regarding RTP packet streams across themselves. This is due to the difference in the RTP packet streams provided to the - different end-points. The original media source lacks information + different endpoints. The original media source lacks information about a mixer's manipulations prior to sending it the different - receivers. This scenario also results in that an end-point's - feedback or requests goes to the mixer. When the mixer can't act - on this by itself, it is forced to go to the original media source - to fulfil the receivers request. This will not necessarily be - explicitly visible any RTP and RTCP traffic, but the interactions - and the time to complete them will indicate such dependencies. + receivers. This scenario also results in that an endpoint's + feedback or requests go to the mixer. When the mixer can't act on + this by itself, it is forced to go to the original media source to + fulfil the receivers 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 + dependencies. Providing source authentication in multi-party scenarios is a - challenge. In the mixer-based topologies, end-points source + 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 end- - point. In RTP sessions where multiple end-points are directly - visible to an end-point, all end-points will have knowledge about + 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 claimed to - come from another end-point in the session. Any node performing + come from another endpoint in the session. Any node performing relay can perform non-cryptographic mitigation by preventing forwarding of packets that have SSRC fields that came from other - end-points before. For cryptographic verification of the source, + endpoints before. For cryptographic verification of the source, SRTP would require additional security mechanisms, for example 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 end-point that receives an RTP packet stream to be able to + 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 media source that is re-encoding 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 end-point that is performing the forwarding is responsible for + 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 to the RTP session on which the media was received. This will require media transcoding for congestion control purpose to produce a suitable bit-rate for the outgoing RTP session, reducing media quality and forcing the forwarding - end-point to spend the resource on the transcoding. The media + 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 end- - point to optimise its media transcoding operation. The cost is + removing almost all dependencies, and allowing the forwarding + endpoint to optimise 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 device. 12.1.3. Differentiated Treatment of RTP Packet 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 end-point + system. First of all is the prioritization within the endpoint 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 @@ -1546,21 +1558,21 @@ 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 consisting of the actual mechanism to prioritize packets. Three common methods for classifying IP packets are: - DiffServ: The end-point marks a packet with a DiffServ code point to + DiffServ: The endpoint marks a packet with a DiffServ code point to indicate to the network that the packet belongs to a particular class. 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. @@ -1571,23 +1583,23 @@ 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 end-point or a classifier can mark + DiffServ assumes that either the endpoint 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 + 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 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]. Deep Packet Inspectors will, despite the SRTP media encryption, still be fairly capable at classifying the RTP streams. The reason is that @@ -1598,25 +1610,25 @@ 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 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 a 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. + see [I-D.ietf-tsvwg-rtcweb-qos] and [I-D.ietf-dart-dscp-rtp] for + further discussion. As a 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 @@ -1658,76 +1670,75 @@ 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., resolve the conflict - when two different end-points 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: + 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: o In a point-to-point session where each SSRC is associated with - either of the two end-points and where the main media carrying - SSRC identifier will be announced in the signalling channel, a + either of the two endpoints and where the main media carrying SSRC + identifier will be announced in the signalling 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 end-points start using a new SSRC identifier prior - to having signalled it to the peer and received acknowledgement on + occur if both endpoints start using a new SSRC identifier prior to + having signalled it to the peer and received acknowledgement on the signalling message. The Source-Specific SDP Attributes - [RFC5576] contains a mechanism to signal how the end-point - resolved the SSRC collision. + [RFC5576] contains a mechanism to signal how the endpoint resolved + the SSRC collision. o SSRC values that have not been signalled 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 to the SSRC of the source RTP packet stream [RFC4588]. In those cases, an - end-point can create a new SSRC that strictly doesn't need to be + endpoint can create a new SSRC that strictly doesn't need to be announced over the signalling channel to function correctly on both RTP and RTCPeerConnection level. - o Multiple end-points in a multiparty conference can create new + o 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 end- - points from the RTP middlebox collisions can occur. + where the SSRC/CSRC are propagated between the different endpoints + from the RTP middlebox collisions can occur. - o An RTP middlebox could connect an end-point's RTCPeerConnection to - another RTCPeerConnection from the same end-point, thus forming a - loop where the end-point will receive its own traffic. While it - is clearly considered a bug, it is important that the end-point is + o 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 is able to recognise and handle the case when it occurs. This case becomes even more problematic when media mixers, and so on, 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 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, identification of the media source(s) part of a MediaStreamTrack being propagated across multiple interconnected RTCPeerConnection needs to be preserved across these interconnections. 12.2.3. Media Synchronisation Context - When an end-point sends media from more than one media source, it + 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, 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 + synchronisation 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 @@ -1742,32 +1753,33 @@ 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 not believed 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 + and partial source authentication. A mandatory to implement and use + media security solution is created by combining 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. + this memo mandates generation of short-term persistent RTCP CNAMES, + as specified in RFC7022, 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: @@ -1809,35 +1821,44 @@ 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 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. + In multi-party communication scenarios using RTP Middleboxes, a lot + of trust is placed on these middleboxes to preserve the sessions + security. The middlebox needs to maintain the confidentiality, + integrity and perform source authentication. As discussed in + Section 12.1.1 the middlebox can perform checks that prevents any + endpoint participating in a conference to impersonate another. Some + additional security considerations regarding multi-party topologies + can be found in [I-D.ietf-avtcore-rtp-topologies-update]. + 14. IANA Considerations This memo makes no request of IANA. Note to RFC Editor: this section is to be removed on publication as an RFC. 15. Acknowledgements The authors would like to thank Bernard Aboba, Harald Alvestrand, - Cary Bran, Ben Campbell, Alissa Cooper, Charles Eckel, Alex - Eleftheriadis, Christian Groves, 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. + 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. 16. References 16.1. Normative References [I-D.ietf-avtcore-multi-media-rtp-session] Westerlund, M., Perkins, C., and J. Lennox, "Sending Multiple Types of Media in a Single RTP Session", draft- ietf-avtcore-multi-media-rtp-session-07 (work in progress), March 2015. @@ -1854,36 +1875,46 @@ draft-ietf-avtcore-rtp-multi-stream-07 (work in progress), March 2015. [I-D.ietf-avtcore-rtp-multi-stream-optimisation] Lennox, J., Westerlund, M., Wu, W., and C. Perkins, "Sending Multiple Media Streams in a Single RTP Session: Grouping RTCP Reception Statistics and Other Feedback", draft-ietf-avtcore-rtp-multi-stream-optimisation-05 (work in progress), February 2015. + [I-D.ietf-avtcore-rtp-topologies-update] + Westerlund, M. and S. Wenger, "RTP Topologies", draft- + ietf-avtcore-rtp-topologies-update-07 (work in progress), + April 2015. + [I-D.ietf-mmusic-sdp-bundle-negotiation] Holmberg, C., Alvestrand, H., and C. Jennings, "Negotiating Media Multiplexing Using the Session Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- negotiation-19 (work in progress), March 2015. [I-D.ietf-rtcweb-audio] Valin, J. and C. Bran, "WebRTC Audio Codec and Processing Requirements", draft-ietf-rtcweb-audio-08 (work in progress), April 2015. [I-D.ietf-rtcweb-fec] Uberti, J., "WebRTC Forward Error Correction Requirements", draft-ietf-rtcweb-fec-01 (work in progress), March 2015. + [I-D.ietf-rtcweb-overview] + Alvestrand, H., "Overview: Real Time Protocols for + Browser-based Applications", draft-ietf-rtcweb-overview-13 + (work in progress), November 2014. + [I-D.ietf-rtcweb-security] Rescorla, E., "Security Considerations for WebRTC", draft- ietf-rtcweb-security-08 (work in progress), February 2015. [I-D.ietf-rtcweb-security-arch] Rescorla, E., "WebRTC Security Architecture", draft-ietf- rtcweb-security-arch-11 (work in progress), March 2015. [I-D.ietf-rtcweb-video] Roach, A., "WebRTC Video Processing and Codec @@ -1977,59 +2008,73 @@ [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)", RFC 7022, September 2013. [RFC7160] Petit-Huguenin, M. and G. Zorn, "Support for Multiple Clock Rates in an RTP Session", RFC 7160, April 2014. [RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds", RFC 7164, March 2014. + [W3C.WD-mediacapture-streams-20130903] + Burnett, D., Bergkvist, A., Jennings, C., and A. + Narayanan, "Media Capture and Streams", World Wide Web + Consortium WD WD-mediacapture-streams-20130903, September + 2013, . + + [W3C.WD-webrtc-20130910] + Bergkvist, A., Burnett, D., Jennings, C., and A. + Narayanan, "WebRTC 1.0: Real-time Communication Between + Browsers", World Wide Web Consortium WD WD-webrtc- + 20130910, September 2013, + . + 16.2. Informative References [I-D.ietf-avtcore-multiplex-guidelines] Westerlund, M., Perkins, C., and H. Alvestrand, "Guidelines for using the Multiplexing Features of RTP to Support Multiple Media Streams", draft-ietf-avtcore- multiplex-guidelines-03 (work in progress), October 2014. - [I-D.ietf-avtcore-rtp-topologies-update] - Westerlund, M. and S. Wenger, "RTP Topologies", draft- - ietf-avtcore-rtp-topologies-update-07 (work in progress), - April 2015. - [I-D.ietf-avtext-rtp-grouping-taxonomy] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and B. Burman, "A Taxonomy of Grouping Semantics and Mechanisms for Real-Time Transport Protocol (RTP) Sources", draft-ietf-avtext-rtp-grouping-taxonomy-06 (work in progress), March 2015. + [I-D.ietf-dart-dscp-rtp] + Black, D. and P. Jones, "Differentiated Services + (DiffServ) and Real-time Communication", draft-ietf-dart- + dscp-rtp-10 (work in progress), November 2014. + [I-D.ietf-mmusic-msid] Alvestrand, H., "WebRTC MediaStream Identification in the Session Description Protocol", draft-ietf-mmusic-msid-10 (work in progress), April 2015. [I-D.ietf-payload-rtp-howto] Westerlund, M., "How to Write an RTP Payload Format", draft-ietf-payload-rtp-howto-14 (work in progress), May 2015. [I-D.ietf-rmcat-cc-requirements] Jesup, R. and Z. Sarker, "Congestion Control Requirements for Interactive Real-Time Media", draft-ietf-rmcat-cc- requirements-09 (work in progress), December 2014. - [I-D.ietf-rtcweb-overview] - Alvestrand, H., "Overview: Real Time Protocols for - Browser-based Applications", draft-ietf-rtcweb-overview-13 - (work in progress), November 2014. + [I-D.ietf-rtcweb-jsep] + Uberti, J., Jennings, C., and E. Rescorla, "Javascript + Session Establishment Protocol", draft-ietf-rtcweb-jsep-09 + (work in progress), March 2015. [I-D.ietf-tsvwg-rtcweb-qos] Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J. Polk, "DSCP and other packet markings for RTCWeb QoS", draft-ietf-tsvwg-rtcweb-qos-03 (work in progress), November 2014. [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol Extended Reports (RTCP XR)", RFC 3611, November 2003. @@ -2055,44 +2100,31 @@ Keeping Alive the NAT Mappings Associated with RTP / RTP Control Protocol (RTCP) Flows", RFC 6263, June 2011. [RFC6792] Wu, Q., Hunt, G., and P. Arden, "Guidelines for Use of the RTP Monitoring Framework", RFC 6792, November 2012. [RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- Time Communication Use Cases and Requirements", RFC 7478, March 2015. - [W3C.WD-mediacapture-streams-20130903] - Burnett, D., Bergkvist, A., Jennings, C., and A. - Narayanan, "Media Capture and Streams", World Wide Web - Consortium WD WD-mediacapture-streams-20130903, September - 2013, . - - [W3C.WD-webrtc-20130910] - Bergkvist, A., Burnett, D., Jennings, C., and A. - Narayanan, "WebRTC 1.0: Real-time Communication Between - Browsers", World Wide Web Consortium WD WD-webrtc- - 20130910, September 2013, - . - Authors' Addresses Colin Perkins University of Glasgow School of Computing Science Glasgow G12 8QQ United Kingdom Email: csp@csperkins.org - URI: http://csperkins.org/ + URI: https://csperkins.org/ + Magnus Westerlund Ericsson Farogatan 6 SE-164 80 Kista Sweden Phone: +46 10 714 82 87 Email: magnus.westerlund@ericsson.com Joerg Ott