Network Working Group C. Perkins Internet-Draft University of Glasgow Updates: RFC3550 T. Schierl (if approved) Fraunhofer HHI Intended status: Standards Track June 30, 2009 Expires: January 1, 2010 Rapid Synchronisation of RTP Flows draft-ietf-avt-rapid-rtp-sync-03.txt Status of this Memo This Internet-Draft is submitted to IETF in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. 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." The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt. The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. This Internet-Draft will expire on January 1, 2010. Copyright Notice Copyright (c) 2009 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents in effect on the date of publication of this document (http://trustee.ietf.org/license-info). Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Abstract This memo outlines how RTP sessions are synchronised, and discusses Perkins & Schierl Expires January 1, 2010 [Page 1] Internet-Draft RTP Synchronisation June 2009 how rapidly such synchronisation can occur. We show that most RTP sessions can be synchronised immediately, but that the use of video switching multipoint conference units (MCUs) or large source specific multicast (SSM) groups can greatly increase the synchronisation delay. This increase in delay can be unacceptable to some applications that use layered and/or multi-description codecs. This memo introduces three mechanisms to reduce the synchronisation delay for such sessions. First, it updates the RTP Control Protocol (RTCP) timing rules to reduce the initial synchronisation delay for SSM sessions. Second, a new feedback packet is defined for use with the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF), allowing video switching MCUs to rapidly request resynchronisation. Finally, new RTP header extensions are defined to allow rapid synchronisation of late joiners, and guarantee correct timestamp based decoding order recovery for layered codecs in the presence of clock skew. Perkins & Schierl Expires January 1, 2010 [Page 2] Internet-Draft RTP Synchronisation June 2009 Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Synchronisation of RTP Flows . . . . . . . . . . . . . . . . . 5 2.1. Initial Synchronisation Delay . . . . . . . . . . . . . . 6 2.1.1. Unicast Sessions . . . . . . . . . . . . . . . . . . . 6 2.1.2. Source Specific Multicast (SSM) Sessions . . . . . . . 7 2.1.3. Any Source Multicast (ASM) Sessions . . . . . . . . . 8 2.1.4. Discussion . . . . . . . . . . . . . . . . . . . . . . 9 2.2. Synchronisation for Late Joiners . . . . . . . . . . . . . 10 3. Reducing RTP Synchronisation Delays . . . . . . . . . . . . . 10 3.1. Reduced Initial RTCP Interval for SSM Senders . . . . . . 11 3.2. Rapid Resynchronisation Request . . . . . . . . . . . . . 11 3.3. In-band Delivery of Synchronisation Metadata . . . . . . . 12 4. Application to Decoding Order Recovery in Layered Codecs . . . 14 4.1. Problem description . . . . . . . . . . . . . . . . . . . 14 4.2. In-band Synchronisation for Decoding Order Recovery . . . 15 4.3. Timestamp based decoding order recovery . . . . . . . . . 16 5. Security Considerations . . . . . . . . . . . . . . . . . . . 19 6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 20 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 20 8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20 8.1. Normative References . . . . . . . . . . . . . . . . . . . 20 8.2. Informative References . . . . . . . . . . . . . . . . . . 21 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22 Perkins & Schierl Expires January 1, 2010 [Page 3] Internet-Draft RTP Synchronisation June 2009 1. Introduction When using RTP to deliver multimedia content it's often necessary to synchronise playout of audio and video components of a presentation. This is achieved using information contained in RTP Control Protocol (RTCP) Sender Report (SR) packets [1]. These are sent periodically, and the components of a multimedia session cannot be synchronised until sufficient RTCP SR packets have been received for each RTP flow to allow the receiver to establish mappings between the media clock used for each RTP flow, and the common (NTP-format) reference clock used to establish synchronisation. Recently, concern has been expressed that this synchronisation delay is problematic for some applications, for example those using layered or multi-description video coding. This memo reviews the operations of RTP synchronisation, and describes the synchronisation delay that can be expected. Three backwards compatible extensions to the basic RTP synchronisation mechanism are proposed: o The RTCP transmission timing rules are relaxed for SSM senders, to reduce the initial synchronisation latency for large SSM groups. See Section 3.1. o An enhancement to the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF) [2] is defined to allow receivers to request additional RTCP SR packets, providing the metadata needed to synchronise RTP flows. This can reduce the synchronisation delay when joining sessions with large RTCP reporting intervals, in the presence of packet loss, or when video switching MCUs are employed. See Section 3.2. o Two RTP header extensions are defined, to deliver synchronisation metadata in-band with RTP data packets. These extensions provide synchronisation metadata that is aligned with RTP data packets, and so eliminate the need to estimate clock-skew between flows before synchronisation. They can also reduce the need to receive RTCP SR packets before flows can be synchronising, although it does not eliminate the need for RTCP. See Section 3.3. The immediate use-case for these extensions is to reduce the delay due to synchronisation when joining a layered video session (e.g. an H.264/SVC session in NI-T mode [9]). The extensions are not specific to layered coding, however, and can be used in any environment when synchronisation latency is an issue. 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 RFC 2119 [3]. Perkins & Schierl Expires January 1, 2010 [Page 4] Internet-Draft RTP Synchronisation June 2009 2. Synchronisation of RTP Flows RTP flows are synchronised by receivers based on information that is contained in RTCP SR packets generated by senders (specifically, the NTP-format timestamp and the RTP timestamp). Synchronisation requires that a common reference clock MUST be used to generate the NTP-format timestamps in a set of flows that are to be synchronised. Furthermore, to achieve more rapid and accurate synchronisation, it is RECOMMENDED that senders and receivers use a common reference clock where possible (recognising that this is often not possible when RTP is used outside of controlled environments); the means by which that common reference clock is distributed are outside the scope of this memo. For multimedia sessions, each type of media (e.g. audio or video) is sent in a separate RTP session, and the receiver associates RTP flows to be synchronised by means of the canonical end-point identifier (CNAME) item included in the RTCP Source Description (SDES) packets generated by the sender or signalled out of band [10]. For layered media, different layers can be sent in different RTP sessions, or using different SSRC values within a single RTP session; in both cases, the CNAME is used to identify flows to be synchronised. To ensure synchronisation, an RTP sender MUST therefore send periodic compound RTCP packets following Section 6 of RFC 3550 [1]. The timing of these periodic compound RTCP packets will depend on the number of members in each RTP session, the fraction of those that are sending data, the session bandwidth, the configured RTCP bandwidth fraction, and whether the session is multicast or unicast (see RFC 3550 Section 6.2 for details). In summary, RTCP control traffic is allocated a small fraction, generally 5%, of the session bandwidth, and of that fraction, one quarter is allocated to active RTP senders, while receivers use the remaining three quarters (these fractions can be configured via SDP [11]). Each member of an RTP session derives an RTCP reporting interval based on these fractions, whether the session is multicast or unicast, the number of members it has observed, and whether it is actively sending data or not. It then sends a compound RTCP packet on average once per reporting interval (the actual packet transmission time is randomised in the range [0.5 ... 1.5] times the reporting interval to avoid synchronisation of reports). A minimum reporting interval of 5 seconds is RECOMMENDED, except that the delay before sending the initial report "MAY be set to half the minimum interval to allow quicker notification that the new participant is present" [1]. Also, for unicast sessions, "the delay before sending the initial compound RTCP packet MAY be zero" [1]. In addition, for unicast sessions, and for active senders in a multicast Perkins & Schierl Expires January 1, 2010 [Page 5] Internet-Draft RTP Synchronisation June 2009 session, the fixed minimum reporting interval MAY be scaled to "360 divided by the session bandwidth in kilobits/second. This minimum is smaller than 5 seconds for bandwidths greater than 72 kb/s." [1] 2.1. Initial Synchronisation Delay A multimedia session comprises a set of concurrent RTP sessions among a common group of participants, using one RTP session for each media type. For example, a videoconference (which is a multimedia session) might contain an audio RTP session and a video RTP session. To allow a receiver to synchronise the components of a multimedia session, a compound RTCP packet containing an RTCP SR packet and an RTCP SDES packet with a CNAME item MUST be sent to each of the RTP sessions in the multimedia session. A receiver cannot synchronise playout across the multimedia session until such RTCP packets have been received on all of the component RTP sessions. If there is no packet loss, this gives an expected initial synchronisation delay equal to the average time taken to receive the first RTCP packet in the RTP session with the longest RTCP reporting interval. This will vary between unicast and multicast RTP sessions. The initial synchronisation delay for layered sessions is similar to that for multimedia sessions. The layers cannot be synchronised until the RTCP SR and CNAME information has been received for each layer in the session. 2.1.1. Unicast Sessions For unicast multimedia or layered sessions, senders SHOULD transmit an initial compound RTCP packet (containing an RTCP SR packet and an RTCP SDES packet with a CNAME item) immediately on joining each RTP session in the multimedia session. The individual RTP sessions are considered to be joined once any in-band signalling for NAT traversal (e.g. [12]) and/or security keying (e.g. [13],[14]) has concluded, and the media path is open. This implies that the initial RTCP packet is sent in parallel with the first data packet following the guidance in RFC 3550 that "the delay before sending the initial compound RTCP packet MAY be zero" and, in the absence of any packet loss, flows can be synchronised immediately. Note that NAT pinholes, firewall holes, quality-of-service, and media security keys should have been negotiated as part of the signalling, whether in-band or out-of-band, before the first RTCP packet is sent. This should ensure that any middleboxes are ready to accept traffic, and reduce the likelihood that the initial RTCP packet will be lost. Perkins & Schierl Expires January 1, 2010 [Page 6] Internet-Draft RTP Synchronisation June 2009 2.1.2. Source Specific Multicast (SSM) Sessions For multicast sessions, the delay before sending the initial RTCP packet, and hence the synchronisation delay, varies with the session bandwidth and the number of members in the session. For a multicast multimedia or layered session, the average synchronisation delay will depend on the slowest of the component RTP sessions; this will generally be the session with the lowest bandwidth (assuming all the RTP sessions have the same number of members). When sending to a multicast group, the reduced minimum RTCP reporting interval of 360 seconds divided by the session bandwidth in kilobits per second [1] should be used when synchronisation latency is likely to be an issue. Also, as usual, the reporting interval is halved for the first RTCP packet. Depending on the session bandwidth and the number of members, this gives the average synchronisation delays shown in Figure 1. Session| Number of receivers: Bandwidth| 2 3 4 5 10 100 1000 10000 --+------------------------------------------------ 8 kbps| 2.73 4.10 5.47 5.47 5.47 5.47 5.47 5.47 16 kbps| 2.50 2.50 2.73 2.73 2.73 2.73 2.73 2.73 32 kbps| 2.50 2.50 2.50 2.50 2.50 2.50 2.50 2.50 64 kbps| 2.50 2.50 2.50 2.50 2.50 2.50 2.50 2.50 128 kbps| 1.41 1.41 1.41 1.41 1.41 1.41 1.41 1.41 256 kbps| 0.70 0.70 0.70 0.70 0.70 0.70 0.70 0.70 512 kbps| 0.35 0.35 0.35 0.35 0.35 0.35 0.35 0.35 1 Mbps| 0.18 0.18 0.18 0.18 0.18 0.18 0.18 0.18 2 Mbps| 0.09 0.09 0.09 0.09 0.09 0.09 0.09 0.09 4 Mbps| 0.04 0.04 0.04 0.04 0.04 0.04 0.04 0.04 Figure 1: Average RTCP reporting interval in seconds for an RTP Session with 1 sender. These numbers assume a source specific multicast channel with a single active sender, which the rules in RFC 3550 section 6.3 give a fixed fraction of the RTCP bandwidth irrespective of the number of receivers. It can be seen that they are sufficient for lip- synchronisation without excessive delay, but might be viewed as having too much latency for synchronising parts of a layered video stream. The RTCP interval is randomised in the usual manner, so the minimum synchronisation delay will be half these intervals, and the maximum delay will be 1.5 times these intervals. Note also that these RTCP intervals are calculated assuming perfect knowledge of the number of Perkins & Schierl Expires January 1, 2010 [Page 7] Internet-Draft RTP Synchronisation June 2009 members in the session. 2.1.3. Any Source Multicast (ASM) Sessions For ASM sessions, the fraction of members that are senders plays an important role, and causes more variation in average RTCP reporting interval. This is illustrated in Figure 2 and Figure 3, which show the RTCP reporting interval for the same session bandwidths and receiver populations as the SSM session described in Figure 1, but for sessions with 2 and 10 senders respectively. It can be seen that the initial synchronisation delay scales with the number of senders (this is to ensure that the total RTCP traffic from all group members does not grow without bound) and can be significantly larger than for single source groups. Despite this, the initial synchronisation time remains acceptable for lip-synchronisation in typical small-to-medium sized group conferencing scenarios. Session| Number of receivers: Bandwidth| 2 3 4 5 10 100 1000 10000 --+------------------------------------------------ 8 kbps| 2.73 4.10 5.47 6.84 10.94 10.94 10.94 10.94 16 kbps| 2.50 2.50 2.73 3.42 5.47 5.47 5.47 5.47 32 kbps| 2.50 2.50 2.50 2.50 2.73 2.73 2.73 2.73 64 kbps| 2.50 2.50 2.50 2.50 2.50 2.50 2.50 2.50 128 kbps| 1.41 1.41 1.41 1.41 1.41 1.41 1.41 1.41 256 kbps| 0.70 0.70 0.70 0.70 0.70 0.70 0.70 0.70 512 kbps| 0.35 0.35 0.35 0.35 0.35 0.35 0.35 0.35 1 Mbps| 0.18 0.18 0.18 0.18 0.18 0.18 0.18 0.18 2 Mbps| 0.09 0.09 0.09 0.09 0.09 0.09 0.09 0.09 4 Mbps| 0.04 0.04 0.04 0.04 0.04 0.04 0.04 0.04 Figure 2: Average RTCP reporting interval in seconds for an RTP Session with 2 senders. Perkins & Schierl Expires January 1, 2010 [Page 8] Internet-Draft RTP Synchronisation June 2009 Session| Number of receivers: Bandwidth| 2 3 4 5 10 100 1000 10000 --+------------------------------------------------ 8 kbps| 2.73 4.10 5.47 6.84 13.67 54.69 54.69 54.69 16 kbps| 2.50 2.50 2.73 3.42 6.84 27.34 27.34 27.34 32 kbps| 2.50 2.50 2.50 2.50 3.42 13.67 13.67 13.67 64 kbps| 2.50 2.50 2.50 2.50 2.50 6.84 6.84 6.84 128 kbps| 1.41 1.41 1.41 1.41 1.41 3.42 3.42 3.42 256 kbps| 0.70 0.70 0.70 0.70 0.70 1.71 1.71 1.71 512 kbps| 0.35 0.35 0.35 0.35 0.35 0.85 0.85 0.85 1 Mbps| 0.18 0.18 0.18 0.18 0.18 0.43 0.43 0.43 2 Mbps| 0.09 0.09 0.09 0.09 0.09 0.21 0.21 0.21 4 Mbps| 0.04 0.04 0.04 0.04 0.04 0.11 0.11 0.11 Figure 3: Average RTCP reporting interval in seconds for an RTP Session with 10 senders. Note that multi-sender groups implemented using multi-unicast with a central RTP translator (Topo-Translator in the terminology of [15]) or mixer (Topo-Mixer), or some forms of video switching MCU (Topo- Video-switch-MCU) distribute RTCP packets to all members of the group, and so scale in the same way as an ASM group with regards to initial synchronisation latency. 2.1.4. Discussion For unicast sessions, the existing RTCP SR-based mechanism allows for immediate synchronisation, provided the initial RTCP packet is not lost. For SSM sessions, the initial synchronisation delay is sufficient for lip-synchronisation, but may be larger than desired for some layered codecs. The rationale for not sending immediate RTCP packets for multicast groups is to avoid implosion of requests when large numbers of members simultaneously join the group ("flash crowd"). This is not an issue for SSM senders, since there can be at most one sender, so it is desirable to allow SSM senders to send an immediate RTCP SR on joining a session (as is currently allowed for unicast sessions, which also don't suffer from the implosion problem). SSM receivers using unicast feedback would not be allowed to send immediate RTCP. For ASM sessions, implosion of responses is a concern, so no change is proposed to the RTCP timing rules. In all cases, it is possible that the initial RTCP SR packet is lost. In this case, the receiver will not be able to synchronise the media until the reporting interval has passed, and the next RTCP SR packet is sent. This is undesirable. Section 3.2 defines a new RTP/AVPF transport layer feedback message to request an RTCP SR be generated, Perkins & Schierl Expires January 1, 2010 [Page 9] Internet-Draft RTP Synchronisation June 2009 allowing rapid resynchronisation in the case of packet loss. 2.2. Synchronisation for Late Joiners Synchronisation between RTP sessions is potentially slower for late joiners than for participants present at the start of the session. The reasons for this are two-fold: 1. Many of the optimisations that allow rapid transmission of RTCP SR packets apply only at the start of a session. This implies that a new participant may have to wait a complete RTCP reporting interval for each session before receiving the necessary data to synchronise media streams. This might potentially take several seconds, depending on the configured session bandwidth and the number of participants. 2. Additional synchronisation delay comes from the nature of the RTCP timing rules. Packets are generated on average once per reporting interval, but with the exact transmission times being randomised +/- 50% to avoid synchronisation of reports. This is important to avoid network congestion in multicast sessions, but does mean that the timing of RTCP SR reports for different RTP sessions isn't synchronised. Accordingly, a receiver must estimate the skew on the NTP-format clock in order to align RTP timestamps across sessions. This estimation is an essential part of an RTP synchronisation implementation, and can be done with high accuracy given sufficient reports. Collecting sufficient RTCP SR data to perform this estimation, however, may require reception of several RTCP reports, further increasing the synchronisation delay. 3. Many media codecs have the notion of periodic access points, such that a newly joined receiver often cannot start decoding a media stream until the packets corresponding to the access point have been received. These access points may be sent less often than RTCP SR packets, and so may be the limiting factor in starting synchronised media playout for late joiners. These delays are likely an issue for tuning in to an ongoing multicast RTP session, or for video switching MCUs. 3. Reducing RTP Synchronisation Delays Three backwards compatible RTP extensions are defined to reduce the possible synchronisation delay: a reduced initial RTCP interval for SSM senders, a rapid resynchronisation request message, and RTP header extensions that can convey synchronisation metadata in-band. Perkins & Schierl Expires January 1, 2010 [Page 10] Internet-Draft RTP Synchronisation June 2009 3.1. Reduced Initial RTCP Interval for SSM Senders In SSM sessions where the initial synchronisation delay is important, the RTP sender MAY set the delay before sending the initial compound RTCP packet to zero, and send its first RTCP packet immediately upon joining the SSM session. RTP receivers in an SSM session, sending unicast RTCP feedback, MUST NOT send RTCP packets with zero initial delay; the timing rules defined in [4] apply unchanged to receivers. 3.2. Rapid Resynchronisation Request The general format of an RTP/AVPF transport layer feedback message is shown in Figure 4 (see [2] for details). 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |V=2|P| FMT | PT=RTPFB=205 | length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of packet sender | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of media source | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ : Feedback Control Information (FCI) : : : Figure 4: RTP/AVP Transport Layer Feedback Message A new feedback message type, RTCP-SR-REQ, is defined with FMT = 5. The Feedback Control Information (FCI) part of the feedback message MUST be empty. The SSRC of packet sender indicates the member that is unable to synchronise media streams, while the SSRC of media source indicates the sender of the media it is unable to synchronise. The length MUST equal 2. This feedback message MAY be sent by a receiver to indicate that it's unable to synchronise some media streams, and desires that the media source transmit an RTCP SR packet as soon as possible (within the constraints of the RTCP timing rules for early feedback). When it receives such an indication, the media source SHOULD generate an RTCP SR packet as soon as possible within the RTCP early feedback rules. If the use of non-compound RTCP [5] was previously negotiated, both the feedback request and the RTCP SR response may be sent as non- compound RTCP packets. The RTCP-SR-REQ packet MAY be repeated once per RTCP reporting interval if no RTCP SR packet is forthcoming. When using SSM sessions with unicast feedback, is possible that the Perkins & Schierl Expires January 1, 2010 [Page 11] Internet-Draft RTP Synchronisation June 2009 feedback target and media source are not co-located. If a feedback target receives an RTCP-SR-REQ feedback message in such a case, the request should be forwarded to the media source. The mechanism to be used for forwarding such requests is not defined here. 3.3. In-band Delivery of Synchronisation Metadata The RTP header extension mechanism defined in [6] can be adopted to carry an OPTIONAL NTP format timestamp in RTP data packets. If such a timestamp is included, it MUST correspond to the same time instant as the RTP timestamp in the packet's header, and MUST be derived from the same clock used to generate the NTP format timestamps included in RTCP SR packets. Provided it has knowledge of the SSRC to CNAME mapping, either from prior receipt of an RTCP CNAME packet or via out-of-band signalling [10], the receiver can use the information provided as input to the synchronisation algorithm, in exactly the same way as if an additional RTCP SR packet was been received for the flow. Two variants are defined for this header extension. The first variant extends the RTP header with a 64 bit NTP timestamp format timestamp as defined in [7]. The second variant carries the lower 24 bit part of the Seconds of a NTP timestamp format timestamp and the 32 bit of the Fraction of a NTP timestamp format timestamp. The formats of the two variants are shown below. Variant A/64-bit NTP RTP header extension (length: 16 bytes): 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |V=2|P|1| CC |M| PT | sequence number | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+R | timestamp |T +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+P | synchronization source (SSRC) identifier | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | 0xBE | 0xDE | length=3 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+E | ID-A | L=7 | NTP timestamp format - Seconds (bit 0-23) |x +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+t |NTP Sec.(24-31)| NTP timestamp format - Fraction(bit 0-23) |n +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |NTP Frc.(24-31)| 0 (pad) | 0 (pad) | 0 (pad) | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | payload data | | .... | Perkins & Schierl Expires January 1, 2010 [Page 12] Internet-Draft RTP Synchronisation June 2009 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Variant B/56-bit NTP RTP header extension (length: 12 bytes): 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |V=2|P|1| CC |M| PT | sequence number | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+R | timestamp |T +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+P | synchronization source (SSRC) identifier | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | 0xBE | 0xDE | length=2 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+E | ID-B | L=6 | NTP timestamp format - Seconds (bit 8-31) |x +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+t | NTP timestamp format - Fraction (bit 0-31) |n +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | payload data | | .... | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ An NTP timestamp format timestamp MAY be included on any RTP packets the sender chooses, but it is RECOMMENDED when performing timestamp based decoding order recovery for layered codecs transported in multiple RTP flows, as further specified in Section 4.2. This header extension SHOULD be also sent on the RTP packets corresponding to a video random access point, and on the associated audio packets, to allow rapid synchronisation for late joiners in multimedia sessions, and in video switching scenarios. Note: The inclusion of an RTP header extension will reduce the efficiency of RTP header compression, if it is used. Furthermore, middle boxes which do not understand the header extensions may remove them or may not update the content according to this memo. In all cases, irrespective of whether in-band NTP timestamp format timestamps are included or not, regular RTCP SR packets MUST be sent to provide backwards compatibility with receivers that synchronize RTP flows according to [1], and robustness in the face of middleboxes (RTP translators) that might strip RTP header extensions. The sender reports are also required to receive the upper 8 bit of the Seconds of the NTP timestamp format timestamp not included in the Variant B/56-bit NTP RTP header extension (although this may generally be inferred from context). Perkins & Schierl Expires January 1, 2010 [Page 13] Internet-Draft RTP Synchronisation June 2009 When the SDP is used, the use of the RTP header extensions defined above MUST be indicated as specified in [6]. Therefore the following URIs MUST be used: o The URI used for signaling the use of Variant A/64-bit NTP RTP header extension in SDP is "urn:ietf:params:rtp-hdrext:ntp-64". o The URI used for signaling the use of Variant B/56-bit NTP RTP header extension in SDP is "urn:ietf:params:rtp-hdrext:ntp-56". 4. Application to Decoding Order Recovery in Layered Codecs Based on the timestamp contained in each RTP data packet, and the mapping to an NTP format timestamp, a decoding order recovery process may be applied if a media as result of a layered coding process is transported in multiple RTP flows. This recovers the decoding order of media frames or samples at the receiver. Especially when transporting layered video, the decoding order recovery process is not straight forward. In this section, we provide guidance on how to use RTP/NTP timing information for decoding order recovery. 4.1. Problem description One option for decoding order recovery in layered codecs is to use the NTP/sample presentation timestamps to reorder data of the same layered media transported in multiple RTP flows. For a timestamp- based decoding order recovery process, it is crucial to allow exact alignment of media frames respectively samples using the NTP timing information. In the presence of clock skew in NTP-format clock, it may not be possible to derive exact matching NTP timestamps using the NTP format clock in each RTP flow's RTCP sender reports. This is due to the fact that RTCP sender reports are not sent at the same point of time in the multiple RTP flows transporting data of the same layered media, while having a skew between those samples in the RTP flows RTCP sender reports. If the RTCP SR packets are not send synchronously in the multiple RTP flows, they therefore do not contain the same NTP-format timestamp. If there is a skew present in the clock used for NTP-format timestamp generation, using different NTP-format timestamps for the same sampling instance in the RTP flow inevitably leads to non-matching NTP timestamps generated from RTP timestamps and NTP-format timestamps in the multiple RTP flows. In order to allow a common and straight forward timestamp-based decoding order recovery process, it is important to guarantee exact matching of NTP timestamps. Thus in the presence of non-perfect clocks, which should be the normal case, an additional mechanism SHALL be used. An Perkins & Schierl Expires January 1, 2010 [Page 14] Internet-Draft RTP Synchronisation June 2009 exact inter-flow alignment of NTP timestamps can be guaranteed, if an RTP header extension containing an NTP timestamp is always inserted at the same timing position in all the RTP flows in question, and if those NTP header extensions are used to update the NTP-RTP relation in all RTP flows at the same point of time. This is called synchronous insertion of RTP header extensions in the following. 4.2. In-band Synchronisation for Decoding Order Recovery The RTP header extension to convey an NTP timestamp SHOULD be used with a layered, multi-description, or multi-view codec, to provide exact matching of NTP timestamps between layers, descriptions, or views transported in different RTP flows to allow timestamp-based decoding order recovery. If this header extension is inserted for RTP flows transporting samples or parts of samples of the same layered media, it SHALL be included at least once in each of the RTP flows of the same media for the sampling time instance of an insertion of an RTP header extension. Such synchronously inserted RTP header extensions SHALL contain the same NTP format timestamp. The frequency of inserting the header extensions in the RTP flows is up to the sender, but it should be noticed that higher insertion frequencies obviously lead to higher synchronization frequencies. For use cases where the same clock source has been used to generate the RTP timestamps in the multiple RTP flows, an application MAY rely on the RTP timestamps only for decoding order recovery starting from the point of synchronous insertion of the RTP header extensions containing NTP timestamps. Note: If the decoding order of RTP flows is given by any means (as e.g., for RTP session by mechanism defined in [8]), the NTP timestamp provided by the header extension allows to collect data of the same sample from the RTP flows, forming the sample decoding order. There may be future mechanism to allow indication of dependencies of RTP flows transported as RTP streams using SSRC multiplexing It is RECOMMENDED that the receiver uses for timestamp-based decoding order recovery the NTP timestamps provided in the RTP header extensions only, if such extensions are present for the RTP flows. Section 4.3 gives further details about the timestamp-based decoding order recovery. Note: Using the RTP header extensions described above allows the receiver to find the corresponding sample of the layered media, or parts thereof, in all RTP flows at the instant the RTP header extension is inserted into the flows. This guarantees that any clock skew present in the NTP timestamp generation process based on RTCP sender reports is avoided, and so allows direct comparison of NTP timestamps across multiple RTP flows. Furthermore, this approach Perkins & Schierl Expires January 1, 2010 [Page 15] Internet-Draft RTP Synchronisation June 2009 solves the possible problem of clock skews identified for the NI-T mode as defined in [9]. To ensure the absence of clock skew, a header extension containing the NTP timestamp MUST be inserted into the RTP flows comprising a layered media stream at the same instant in each RTP flow. This may require the insertion of extra packets in some of the RTP flows, since in layered video codecs not all sampling instances may be present in all the flows. If such a header extension is included in all flows at a sampling time instance, the NTP timestamps for samples following in decoding order the RTP header insertion point can be constructed using the RTP timestamps and identical reference NTP timestamps in the NTP header extension in all RTP flows. It should be noted that the frequency of inserting the RTP header extension containing the NTP timestamp is crucial in presence of clock skew, since the points of insertion may be the only points for a receiver to start the decoding order recovery. 4.3. Timestamp based decoding order recovery If parts or complete samples as result of a layered coding process are transported as different RTP flows, i.e. as different RTP streams, and/or as different RTP sessions, a decoding order recovery process is required to reorder the samples or parts of samples received. Such mechanism may be based on the NTP presentation timestamp which can be derived from the RTP timestamp using the NTP- format timestamp provided in the RTCP sender report packets. In order to guarantee the exact alignment of those derived NTP presentation timestamps, the RTP header extension as defined in this memo in Section 3.3 allows the receiver to start the decoding order recovery before the reception of a RTCP sender report if the RTP header extension is earlier provided in the RTP flow. Using the RTP header extensions may be the only way to allow correct decoding order recovery based on exact matching of NTP timestamps in the presence of clock skew in the clock used for generating the NTP format clock. Furthermore, some use cases may allow to use synchronously inserted RTP header extensions containing NTP timestamps to align the RTP timestamps of the multiple RTP flows, i.e. use cases where the RTP timestamps of the multiple RTP flows are generated from the same clock source. In such use cases, starting from a synchronous insertion of the RTP header extensions, the application may use the detected difference of RTP random offset values in the multiple sessions to align the media samples of parts of samples. Since typically for layered video codecs as, e.g. SVC [9], the decoding order is not equal to the presentation order of the media samples, media samples or parts of media samples cannot be simply ordered according to the presentation timestamp order. For this Perkins & Schierl Expires January 1, 2010 [Page 16] Internet-Draft RTP Synchronisation June 2009 reason, if transporting media samples or parts of media samples of a layered, multi-view or multi description codec in different RTP flows, the following rules SHOULD be kept for sending such flows: Note: The following rules are typically kept for layered audio codecs, which allows using the same algorithm for decoding order recovery of audio samples. Terminology: Following the decoding order of RTP flows as described above, an RTP flow containing sample data which is required to be accessed and/or decoded before decoding a second sample data of another RTP flow is called a lower RTP flow with respect to the second RTP flow. A second RTP flow, which requires for the decoding process accessing and/or decoding the sample data of the lower RTP flow is called the higher RTP flow. The lowest RTP flow is the flow, which does not require the presence of any other data. o The decoding order of media samples or part of the media samples transported in different RTP flows SHOULD be derivable by any means. This can be accomplished, e.g. by using the mechanisms defined in [8] if the sample data or parts of the sample data are transported in different RTP sessions or by any other means. o For each two RTP flows the following rules SHOULD be true in order to allow decoding order recovery based on matching NTP timestamps present in the different RTP flows: 1. The order of the RTP samples within an RTP flow is equal to the decoding order. 2. A higher RTP flow contains all the same sampling instances of the lower RTP flow. A higher RTP flow may contain additional sampling instances. Note: In some cases, it may be required to add packets in higher RTP flows in order to satisfy the second rule above. This may be achieved by placing empty RTP packets (containing padding data only) or by other payload means as, e.g. the Empty NAL unit packet as defined in [9]. If a packet must be inserted for satisfying the above rule, the NTP timestamp of such an inserted packet MUST be set equal to the NTP timestamp of a packet of the same sample present in any lower RTP flow and the lowest RTP flow. This is easy to accomplish if the packet can be inserted at the time of the RTP flow generation, since the NTP timestamp must be the same for the inserted packet and the packet of the corresponding sample. Perkins & Schierl Expires January 1, 2010 [Page 17] Internet-Draft RTP Synchronisation June 2009 The above rules allow the receiver to process the data of the RTP flows as follows: o Go through all received RTP flows starting with the highest RTP flow and aggregate the sample data or parts of the sample data with the same NTP timestamp in the order of RTP flows, starting from the lowest RTP flow up to the highest RTP flow received, to the sample with the NTP timestamp present in the highest RTP flow. The NTP timestamps MAY be derived using RTCP sender reports or MAY be directly taken from the NTP timestamp provided in an RTP header extension. The order of RTP flows may e.g. be indicated by mechanisms as defined in [8] or any other implicit or explicit means. Repeat the aforementioned process for each different NTP timestamp present in the highest RTP flow. Informative example: The example shown in Figure 3 refers to three RTP flows A, B and C containing a layered, a multi-view or a multi- description media stream. In the example, the dependency signalling as defined in [8] indicates that flow A is the lowest RTP flow, B is the first higher RTP flow and depends on A, and C is the second higher RTP flow corresponding to flow A and depends on A and B. A media coding structure is used that results in samples present in higher flows but not present in all lower flows. Flow A has the lowest frame rate and Flow B and C have the same but higher frame rate. The figure shows the full video samples with their corresponding RTP timestamps "(x)". The video samples are already re-ordered according to their RTP sequence number order. The figure indicates for the received sample in decoding order within each RTP flow, as well as the associated NTP media timestamps ("TS[..]"). These timestamps may be derived using the NTP format timestamp provided in the RTCP sender reports or as shown in the figure directly from the NTP timestamp contained in the RTP header extensions as indicate by the timestamp in "". Note that the timestamps are not in increasing order since, in this example, the decoding order is different from the output/presentation order. The process first proceeds to the sample parts associated with the first available synchronous insertion of NTP timestamp into RTP header extensions at NTP media timestamp TS=[8] and starts in the highest RTP flow C and removes/ignores all preceding sample parts (in decoding order) to sample parts with TS=[8] in each of the de- jittering buffers of RTP flows A, B, and C. Then, starting from flow C, the first media timestamp available in decoding order (TS=[8]) is selected and sample parts starting from RTP flow A, and flow B and C are placed in order of the RTP flow dependency as indicated by mechanisms defined in [8] (in the example for TS[8]: first flow B and then flow C into the video sample VS(TS[8]) associated with NTP media timestamp TS=[8]. Then the next media timestamp TS=[6] (RTP Perkins & Schierl Expires January 1, 2010 [Page 18] Internet-Draft RTP Synchronisation June 2009 timestamp=(4)) in order of appearance in the highest RTP flow C is processed and the process described above is repeated. Note that there may be video samples with no sample parts present, e.g., in the lowest RTP flow A (see, e.g., TS=[5]). The decoding order recovery process could be also started after receiving all RTP sender reports RTP timestamp to NTP-format timestamp mapping (indicated as timestamps "(x){y}") assuming that there is no clock skew in the source used for the NTP-format timestamp generation. C:-(0)----(2)----(7)<8>--(5)----(4)----(6)-----(11)----(9){10}- | | | | | | | | B:-(3)----(5)---(10)<8>--(8)----(7)----(9){7}--(14)----(12)---- | | | | A:---------------(3)<8>--(1)-------------------(7){12}-(5)----- ---------------------------------------decoding/transmission order-> TS:[1] [3] [8]=<8> [6] [5] [7] [12] [10] Key: A, B, C - RTP flows Integer values in "()"- video sample with its RTP timestamp as indicated in its RTP packet. "|" - indicates corresponding samples / parts of sample of the same video sample VS(TS[..]) in the RTP flows. Integer values in "[]"- NTP media timestamp TS, sampling time as derived from the NTP timestamp associated with the video sample AU(TS[..]), consisting of sample parts in the flows above. Integer values in "<>"- NTP media timestamp TS as directly taken from the NTP RTP header extensions. Integer values in "{}"- NTP media timestamp TS as provided in the RTCP sender reports. 5. Security Considerations The security considerations of the RTP specification [1], the Extended RTP profile for RTCP-Based Feedback [2], and the General Mechanism for RTP Header Extensions [6] apply. The extensions we define in this memo are not believed to introduce any additional security considerations. Perkins & Schierl Expires January 1, 2010 [Page 19] Internet-Draft RTP Synchronisation June 2009 6. IANA Considerations NOTE TO RFC EDITOR: Please replace "RFC XXXX" in the following with the RFC number assigned to this memo, and delete this note. The IANA is requested to register one new value in the table of FMT Values for RTPFB Payload Types [2] as follows: Name: RTCP-SR-REQ Long name: RTCP Rapid Resynchronisation Request Value: 5 Reference: RFC XXXX The IANA is also requested to register two new RTP Compact Header Extensions [6], according to the following: Extension URI: urn:ietf:params:rtp-hdrext:ntp-64 Description: Synchronisation metadata: 64-bit timestamp format Contact: Thomas Schierl IETF Audio/Video Transport Working Group Reference: RFC XXXX Extension URI: urn:ietf:params:rtp-hdrext:ntp-56 Description: Synchronisation metadata: 56-bit timestamp format Contact: Thomas Schierl IETF Audio/Video Transport Working Group Reference: RFC XXXX 7. Acknowledgements This memo has benefitted from discussions with numerous members of the IETF AVT working group, including Jonathan Lennox, Magnus Westerlund, Randell Jesup, Gerard Babonneau, Ingemar Johansson, Ali C. Begen, Ye-Kui Wang, Roni Even, Michael Dolan, and Art Allison. The header extension format of Variant A in Section 3.3 was suggested by Dave Singer, matching a similar mechanism specified by ISMA. 8. References 8.1. Normative References [1] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. Perkins & Schierl Expires January 1, 2010 [Page 20] Internet-Draft RTP Synchronisation June 2009 [2] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 2006. [3] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [4] Schooler, E., Ott, J., and J. Chesterfield, "RTCP Extensions for Single-Source Multicast Sessions with Unicast Feedback", draft-ietf-avt-rtcpssm-18 (work in progress), March 2009. [5] Johansson, I. and M. Westerlund, "Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences", RFC 5506, April 2009. [6] Singer, D. and H. Desineni, "A General Mechanism for RTP Header Extensions", RFC 5285, July 2008. [7] Mills, D., "Network Time Protocol (Version 3) Specification, Implementation", RFC 1305, March 1992. [8] Schierl, T. and S. Wenger, "Signaling media decoding dependency in Session Description Protocol (SDP)", draft-ietf-mmusic-decoding-dependency-08 (work in progress), April 2009. 8.2. Informative References [9] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, "RTP Payload Format for SVC Video", draft-ietf-avt-rtp-svc-18 (work in progress), March 2009. [10] Lennox, J., Ott, J., and T. Schierl, "Source-Specific Media Attributes in the Session Description Protocol (SDP)", draft-ietf-mmusic-sdp-source-attributes-02 (work in progress), October 2008. [11] Casner, S., "Session Description Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 3556, July 2003. [12] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols", draft-ietf-mmusic-ice-19 (work in progress), October 2007. [13] McGrew, D. and E. Rescorla, "Datagram Transport Layer Security (DTLS) Extension to Establish Keys for Secure Real-time Perkins & Schierl Expires January 1, 2010 [Page 21] Internet-Draft RTP Synchronisation June 2009 Transport Protocol (SRTP)", draft-ietf-avt-dtls-srtp-05 (work in progress), September 2008. [14] Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media Path Key Agreement for Secure RTP", draft-zimmermann-avt-zrtp-13 (work in progress), January 2009. [15] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, January 2008. Authors' Addresses Colin Perkins University of Glasgow Department of Computing Science Sir Alwyn Williams Building Lilybank Gardens Glasgow G12 8QQ UK Email: csp@csperkins.org Thomas Schierl Fraunhofer HHI Einsteinufer 37 D-10587 Berlin Germany Phone: +49-30-31002-227 Email: Thomas.Schierl@hhi.fraunhofer.de Perkins & Schierl Expires January 1, 2010 [Page 22]