Network Working Group C. Perkins
Internet-Draft University of Glasgow
Updates: RFC3550 T. Schierl
(if approved) Fraunhofer HHI
Intended status: Standards Track November 9, 2009
Expires: May 13, 2010
Rapid Synchronisation of RTP Flows
draft-ietf-avt-rapid-rtp-sync-07.txt
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Abstract
This memo outlines how RTP sessions are synchronised, and discusses
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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.
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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 . . . . . . . . . . . . . 9
3. Reducing RTP Synchronisation Delays . . . . . . . . . . . . . 10
3.1. Reduced Initial RTCP Interval for SSM Senders . . . . . . 10
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. In-band Synchronisation for Decoding Order Recovery . . . 14
4.2. Timestamp based decoding order recovery . . . . . . . . . 15
4.3. Example . . . . . . . . . . . . . . . . . . . . . . . . . 16
5. Security Considerations . . . . . . . . . . . . . . . . . . . 18
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 18
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 19
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 19
8.1. Normative References . . . . . . . . . . . . . . . . . . . 19
8.2. Informative References . . . . . . . . . . . . . . . . . . 20
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 20
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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 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 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 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].
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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 [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
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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]
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. [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.
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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 [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 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
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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 [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.
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 initial synchronisation delay in seconds for an RTP
Session with 2 senders.
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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 initial synchronisation delay in seconds for an RTP
Session with 10 senders.
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,
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 three-fold:
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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.
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
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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
One 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.
If the RTP/AVPF profile [2] is in use, 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
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.
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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 in Figure 5 and Figure 6.
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
| synchronisation 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 |
| .... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 5: Variant A/64-bit NTP RTP header extension
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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|1| CC |M| PT | sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+R
| timestamp |T
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+P
| synchronisation 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 |
| .... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 6: Variant B/56-bit NTP RTP header extension
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 4.1. 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 synchronise
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).
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:
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o The URI used for signalling 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 signalling 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
Packets in RTP flows are often predictively coded, with a receiver
having to arrange the packets into a particular order before it can
decode the media data. Depending on the payload format, the decoding
order might be explicitly specified as a field in the RTP payload
header, or the receiver might decode the packets in order of their
RTP timestamps. If a layered encoding is used, where the media data
is split across several RTP flows, then it is often necessary to
exactly synchronise the RTP flows comprising the different layers
before layers other than the base layer can be decoded. Examples of
such layered encodings are H.264 SVC in NI-T mode [9] and MPEG
surround multi-channel audio [16]. As described in Section 2, such
synchronisation is possible in RTP, but can be difficult to perform
rapidly. In the following, we describe how the extensions defined in
Section 3.3 can be used to synchronise layered flows, and provide a
common timestamp-based decoding order.
4.1. In-band Synchronisation for Decoding Order Recovery
When a layered, multi-description, or multi-view codec is used, with
the different components of the media being transferred on separate
RTP flows, the RTP sender SHOULD use periodic synchronous in-band
delivery of synchronisation metadata to allow receivers to rapidly
and accurately synchronise the separate components of the layered
media flow. There are three parts to this:
o The sender must negotiate the use of the RTP header extensions
described in Section 3.3, and must periodically and synchronously
insert such header extensions into all the RTP flows forming the
separate components of the layered, multi-description, or multi-
view flow.
o Synchronous insertion requires the sender insert these RTP header
extensions into packets corresponding to the exact same sampling
instant in all the flows. Since the header extensions for each
flow are inserted at exactly the same sampling instant, they will
have identical NTP-format timestamps, hence allowing receivers to
exactly align the RTP timestamps for the component flows. This
may require the insertion of extra data packets into some of the
component RTP flows, if some component flows contain packets for
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sampling instants that do not exist in other flows (for example, a
layered video codec, where the layers have differing frame rates).
o The frequency with which the sender inserts the header extensions
will directly correspond to the synchronisation latency, with more
frequent insertion leading to higher per-flows overheads, but
lower synchronisation latency. It is RECOMMENDED that the sender
insert the header extensions synchronously into all component RTP
flows at least once per random access point of the media, but they
MAY be inserted more often.
The sender MUST continue to send periodic RTCP reports including SR
packets, and MUST ensure the RTP timestamp to NTP-format timestamp
mapping in the RTCP SR packets is consistent with that used in the
RTP header extensions. Receivers should use both the information
contained in RTCP SR packets and the in-band mapping of RTP and NTP-
format timestamps as input to the synchronisation process, but it is
RECOMMENDED that receivers sanity check the mappings received and
discard outliers, to provide robustness against invalid data (one
might think it more likely that the RTCP SR mappings are invalid,
since they are sent at irregular times and subject to skew, but the
presence of broken RTP translators could also corrupt the timestamps
in the RTP header extension; receivers need to cope with both types
of failure).
4.2. Timestamp based decoding order recovery
Once a receiver has synchronised the components of a layered, multi-
description, or multi-view flow using the RTP header extensions as
described in Section 4.1, it may then derive a decoding order based
on the synchronised timestamps as follows (or it may use information
in the RTP payload header to derive the decoding order, if present
and desired).
There may be explicit dependencies between the component flows of a
layered, multi-description, or multi-view flow. For example, it is
common for layered flows to be arranged in a hierarchy, where flows
from "higher" layers cannot be decoded until the corresponding data
in "lower" layer flows has been received and decoded. If such a
decoding hierarchy exists, it MUST be signalled out of band, for
example using [8] when SDP signalling is used.
Each component RTP flow MUST contain packets corresponding to all the
sampling instants of the RTP flows on which it depends. If such
packets are not naturally present in the RTP flow, the sender MUST
generate additional packets as necessary in order to satisfy this
rule. The format of these packets depends on the payload format
used. For H.264 SVC, the Empty NAL unit packet [9] should be used.
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Flows may also include packets corresponding to additional sampling
instants that are not present in the flows on which they depend.
The receiver should decode the packets in all the component RTP flows
as follows:
o For each RTP packet in each flow, use the mapping contained in the
RTP header extensions and RTCP SR packets to derive the NTP-format
timestamp corresponding to its RTP timestamp.
o Group together RTP data packets from all component flows that have
identical calculated NTP-format timestamps.
o Processing groups in order of ascending NTP-format timestamp,
decode the RTP packets in each group according to the signalled
RTP flow decoding hierarchy. That is, pass the RTP packet data
from the flow on which all other flows depend to the decoder
first, then that from the next dependent flow, and so on. The
decoding order of the RTP flow hierarchy may be indicated by
mechanisms defined in [8] or by some other means.
Note that the decoding order will not necessarily match the packet
transmission order. The receiver will need to buffer packets for a
codec-dependent amount of time in order for all necessary packets to
arrive to allow decoding.
4.3. 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 "<x>". Note
that the timestamps are not in increasing order since, in this
example, the decoding order is different from the output/presentation
order.
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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
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.
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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 RTP header extensions defined in Section 3.3 include an NTP-
format timestamp. When an RTP session using this header extension is
protected by the Secure RTP framework [17], that header extension is
not part of the encrypted portion of the RTP data packets or RTCP
control packets; however these NTP-format timestamps are encrypted
when using SRTP without this header extension. This is a minor
information leak, but one that is not believed to be significant.
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 <Thomas.Schierl@hhi.fraunhofer.de>
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 <ts@thomas-schierl.de>
IETF Audio/Video Transport Working Group
Reference: RFC XXXX
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7. Acknowledgements
This memo has benefited 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, Art Allison, and Stefan Doehla.
The RTP 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.
[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.
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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
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.
[16] Bont, F., Doehla, S., Schmidt, M., and R. Sperschneider, "RTP
Payload Format for Elementary Streams with MPEG Surround multi-
channel audio", draft-ietf-avt-rtp-mps-02 (work in progress),
January 2009.
[17] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
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Authors' Addresses
Colin Perkins
University of Glasgow
Department of Computing Science
Glasgow G12 8QQ
UK
Email: csp@csperkins.org
Thomas Schierl
Fraunhofer HHI
Einsteinufer 37
D-10587 Berlin
Germany
Phone: +49-30-31002-227
Email: ts@thomas-schierl.de
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