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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 [RFC3550]. 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:

  • The RTCP transmission timing rules are relaxed for source-specific multicast (SSM) senders, to reduce the initial synchronisation latency for large SSM groups. See Section 3.1.

  • An enhancement to the extended RTP profile for RTCP-based feedback (RTP/AVPF) [RFC4585] 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.

  • 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 synchronised, 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 (Scalable Video Coding) session in Non-Interleaved Timestamp-based (NI-T) mode [AVT-RTP-SVC]). 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 [RFC2119].

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 (i.e., when synchronising several RTP flows, the RTP timestamps for each flow are derived from separate, and media specific, clocks, but the NTP-format timestamps in the RTCP SR packets of all flows to be synchronised MUST be sampled from the same clock). To achieve faster and more accurate synchronisation, it is further RECOMMENDED that senders and receivers use a synchronised common NTP-format reference clock with common properties, especially timebase, 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 and its properties are signalled and distributed is 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 [RFC5576]. For layered media, different layers can be sent in different RTP sessions, or using different synchronisation source (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 [RFC3550].

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 the Session Description Protocol (SDP) [RFC3556]). 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" [RFC3550]. Also, for unicast sessions, "the delay before sending the initial compound RTCP packet MAY be zero" [RFC3550]. In addition, for unicast sessions, and for active senders in a multicast 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" [RFC3550].

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 by each sender. 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., [RFC5245]) and/or security keying (e.g., [RFC5764], [ZRTP]) 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.

It is expected that NAT pinholes, firewall holes, quality-of-service, and media security keys will 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.

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 [RFC3550] 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 Initial Synchronisation Delay in Seconds
                     for an RTP Session with 1 Sender

These numbers assume a source-specific multicast channel with a single active sender, assuming an average RTCP packet size of 70 octets. These intervals 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 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 source-specific groups. Despite this, the initial synchronisation time remains acceptable for lip-synchronisation in typical small-to-medium sized group video conferencing scenarios.

Note that multi-sender groups implemented using multi-unicast with a central RTP translator (Topo-Translator in the terminology of [RFC5117]) have similar synchronisation performance to ASM groups (Topo-ASM), since the translator effectively forms an ASM group.

(Figures 2 and 3 omitted in this chunk but discussed in next sections if available).

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