One document matched: draft-perkins-avt-rapid-rtp-sync-00.txt
Network Working Group C. Perkins
Internet-Draft University of Glasgow
Intended status: Standards Track October 25, 2008
Expires: April 28, 2009
Rapid Synchronisation of RTP Flows
draft-perkins-avt-rapid-rtp-sync-00.txt
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Copyright (C) The IETF Trust (2008).
Abstract
This memo outlines how RTP multimedia sessions are synchronised, and
discusses how rapidly such synchronisation can occur. It is shown
that unicast sessions can be synchronised immediately in most cases.
Multicast groups have longer synchronisation delay. A modification
to the RTCP timing rules is suggested to improve synchronisation time
for SSM senders. A new RTP/AVPF feedback packet is defined to
improve general synchronisation times.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Synchronisation of RTP Flows . . . . . . . . . . . . . . . . . 3
3. Initial Synchronisation Delay . . . . . . . . . . . . . . . . . 4
3.1. Unicast Sessions . . . . . . . . . . . . . . . . . . . . . 4
3.2. Source Specific Multicast (SSM) Sessions . . . . . . . . . 5
3.3. Any Source Multicast (ASM) Sessions . . . . . . . . . . . . 6
4. Discussion . . . . . . . . . . . . . . . . . . . . . . . . . . 6
5. Rapid Resynchronisation . . . . . . . . . . . . . . . . . . . . 6
6. Security Considerations . . . . . . . . . . . . . . . . . . . . 7
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 7
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 7
8.1. Normative References . . . . . . . . . . . . . . . . . . . 7
8.2. Informative References . . . . . . . . . . . . . . . . . . 8
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 8
Intellectual Property and Copyright Statements . . . . . . . . . . 9
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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 [1]. These are sent periodically
and the components of a multimedia session cannot be synchronised
until an RTCP SR packet has been received for each flow. Recently,
concern has been expressed that this initial synchronisation delay is
problematic for some applications, for example those using layered or
multiple description video coding. This memo reviews the operation
of RTP synchronisation, describes the initial synchronisation delay
that can be expected, and defines an enhancement to the Extended RTP
Profile for RTCP-based Feedback (RTP/AVPF) [2] to provide faster
synchronisation in some circumstances.
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].
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 and RTP timestamps). Each media type 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.
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.
Of that RTCP bandwidth fraction, one quarter is allocated to active
RTP senders, while receivers use the remaining three quarters (these
fractions can be configured via SDP [4]). 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 transmission delay for each packet is randomised
in the range [0.5 ... 1.5] times the reporting interval to avoid
synchronisation of reports).
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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
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]
3. 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.
3.1. Unicast Sessions
For unicast multimedia 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. [5])
and/or security keying (e.g. [6],[7]) 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.
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3.2. 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 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 following average synchronisation
delays:
Session| Number of receivers (single sender assumed):
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 0.07 0.07 0.07 0.07 0.07 0.07 0.07
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 (seconds)
These numbers assume a single-source multicast channal 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
members in the session. In practice, an implementation will have
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only limited knowledge of the size of the session when joining, and
will likely send its initial report early compared to these values,
following the RTCP reconsideration rules.
3.3. Any Source Multicast (ASM) Sessions
(tbd)
For ASM sessions, the fraction of members that are senders plays an
important role, and imply more varation in average RTCP reporting
interval.
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 might be desirable to allow SSM senders to send an immediate
RTCP SR on joining a session (as is currenly 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. This would be a change to RFC 3550, if accepted.
For ASM session... (tbd)
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. The following section defines a new
RTP/AVPF transport layer feedback message to request an RTCP SR be
generated, allowing rapid resynchronisation.
5. Rapid Resynchronisation
The general format of an RTP/AVPF transport layer feedback message is
shown below.
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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) :
: :
A new feedback message type, RTCP-SR-REQ, is defined with FMT = XXX.
(the next available FMT is 5?) This MAY be sent to indicate that a
receiver is unable to synchronise media streams, and desires that the
media source send an RTCP SR packet as soon as possible (within the
constraints of RTCP the early feedback rules). On receipt of this,
the media source SHOULD generate an RTCP SR packet as soon as
possible within the RTCP early feedback rules. That RTCP SR packet
MAY be sent as a non-compound RTCP packet, if this has been
negotiated.
The Feedback Control Information (FCI) part of the packet is emtpy.
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 lenght MUST
equal 2.
6. Security Considerations
The security considerations of the RTP specification [1] and RTP/AVPF
profile [2] apply. No addtional security considerations apply due to
the RTP/AVPF rapid resynchronisation mechanism defined in Section 5.
7. IANA Considerations
(tbd - this needs to register the new RTP/AVPF transport layer
feedback packet type)
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,
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RFC 3550, July 2003.
[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.
8.2. Informative References
[4] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 3556,
July 2003.
[5] 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.
[6] McGrew, D. and E. Rescorla, "Datagram Transport Layer Security
(DTLS) Extension to Establish Keys for Secure Real-time
Transport Protocol (SRTP)", draft-ietf-avt-dtls-srtp-05 (work in
progress), September 2008.
[7] Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media Path
Key Agreement for Secure RTP", draft-zimmermann-avt-zrtp-09
(work in progress), September 2008.
Author's Address
Colin Perkins
University of Glasgow
Department of Computing Science
Sir Alwyn Williams Building
Lilybank Gardens
Glasgow G12 8QQ
UK
Email: csp@csperkins.org
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