RTP Retransmission Payload Format
draft-ietf-avt-rtp-retransmission-12
The information below is for an old version of the document that is already published as an RFC.
| Document | Type |
This is an older version of an Internet-Draft that was ultimately published as RFC 4588.
|
|
|---|---|---|---|
| Authors | David Leon, Rolf Hakenberg , Jose Rey , Akihiro Miyazaki , Victor Varsa | ||
| Last updated | 2020-07-29 (Latest revision 2005-09-15) | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Intended RFC status | Proposed Standard | ||
| Formats | |||
| Additional resources | Mailing list discussion | ||
| Stream | WG state | (None) | |
| Document shepherd | (None) | ||
| IESG | IESG state | Became RFC 4588 (Proposed Standard) | |
| Action Holders |
(None)
|
||
| Consensus boilerplate | Unknown | ||
| Telechat date | (None) | ||
| Responsible AD | Allison J. Mankin | ||
| Send notices to | <csp@csperkins.org>, <magnus.westerlund@ericsson.com> |
draft-ietf-avt-rtp-retransmission-12
Internet Draft
draft-ietf-avt-rtp-retransmission- J. Rey/Panasonic
12.txt D. Leon/Nokia
A. Miyazaki/Panasonic
V. Varsa/Nokia
R. Hakenberg/Panasonic
Expires: March 15, 2006 September 15, 2005
RTP Retransmission Payload Format
Status of this Memo
By submitting this Internet-Draft, each author represents that any
applicable patent or other IPR claims of which he or she is aware
have been or will be disclosed, and any of which he or she becomes
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[Note to RFC Editor: This paragraph shall be deleted upon
publication as an RFC. References in this draft to RFC XXXX
should be replaced with the RFC number assigned to this document.]
Abstract
RTP retransmission is an effective packet loss recovery technique
for real-time applications with relaxed delay bounds. This
document describes an RTP payload format for performing
retransmissions. Retransmitted RTP packets are sent in a separate
stream from the original RTP stream. It is assumed that feedback
from receivers to senders is available. In particular, it is
assumed that RTCP feedback as defined in the extended RTP profile
for RTCP-based feedback (denoted RTP/AVPF), is available in this
memo.
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Table of Contents
1. Introduction..................................................3
2. Terminology...................................................3
3. Requirements and design rationale for a retransmission scheme.4
3.1 Multiplexing scheme choice..................................6
4. Retransmission payload format.................................7
5. Association of retransmission and original streams............9
5.1 Retransmission session sharing..............................9
5.2 CNAME use...................................................9
5.3 Association at the receiver.................................9
6. Use with the extended RTP profile for RTCP-based feedback....10
6.1 RTCP at the sender.........................................11
6.2 RTCP Receiver Reports......................................11
6.3 Retransmission requests....................................11
6.4 Timing rules...............................................12
7. Congestion control...........................................13
8. Retransmission Payload Format MIME type registration.........14
8.1 Introduction...............................................14
8.2 Registration of audio/rtx..................................15
8.3 Registration of video/rtx..................................16
8.4 Registration of text/rtx...................................17
8.5 Registration of application/rtx............................17
8.6 Mapping to SDP.............................................18
8.7 SDP description with session-multiplexing..................19
8.8 SDP description with SSRC-multiplexing.....................20
9. RTSP considerations..........................................20
9.1 RTSP control with SSRC-multiplexing........................21
9.2 RTSP control with session-multiplexing.....................21
9.3 RTSP control of the retransmission stream..................22
9.4 Cache control..............................................22
10. Implementation examples.....................................22
10.1 A minimal receiver implementation example.................22
10.2 Retransmission of Layered Encoded Media in Multicast......23
11. IANA considerations.........................................24
12. Security considerations.....................................24
13. Acknowledgements............................................25
14. References..................................................25
14.1 Normative References......................................25
14.2 Informative References....................................26
15. Author's Addresses..........................................26
Appendix A. How to control the number of rtxs. per packet.......27
IPR Notices.....................................................31
Full Copyright Statement........................................32
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1. Introduction
Packet losses between an RTP sender and receiver may significantly
degrade the quality of the received media. Several techniques,
such as forward error correction (FEC), retransmissions or
interleaving may be considered to increase packet loss resiliency.
RFC 2354 [8] discusses the different options.
When choosing a repair technique for a particular application, the
tolerable latency of the application has to be taken into account.
In the case of multimedia conferencing, the end-to-end delay has
to be at most a few hundred milliseconds in order to guarantee
interactivity, which usually excludes the use of retransmission.
With sufficient latency, the efficiency of the repair scheme can
be increased. The sender may use the receiver feedback in
order to react to losses before their playout time at the
receiver.
In the case of multimedia streaming, the user can tolerate an
initial latency as part of the session set-up and thus an end-to-
end delay of several seconds may be acceptable. RTP
retransmission as defined in this document is targeted at such
applications.
Furthermore, the RTP retransmission method defined herein is
applicable to unicast and (small) multicast groups. The present
document defines a payload format for retransmitted RTP packets
and provides protocol rules for the sender and the receiver
involved in retransmissions.
This retransmission payload format was designed for use with the
extended RTP profile for RTCP-based feedback, AVPF [1]. It may
also be used with other RTP profiles defined in the future.
The AVPF profile allows for more frequent feedback and for early
feedback. It defines a general-purpose feedback message, i.e.
NACK, as well as codec and application-specific feedback messages.
See [1] for details.
2. Terminology
The following terms are used in this document:
Original packet: refers to an RTP packet which carries user data
sent for the first time by an RTP sender.
Original stream: refers to the RTP stream of original packets.
Retransmission packet: refers to an RTP packet which is to be used
by the receiver instead of a lost original packet. Such a
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retransmission packet is said to be associated with the original
RTP packet.
Retransmission request: a means by which an RTP receiver is able
to request that the RTP sender should send a retransmission packet
for a given original packet. Usually, an RTCP NACK packet as
specified in [1] is used as retransmission request for lost
packets.
Retransmission stream: the stream of retransmission packets
associated with an original stream.
Session-multiplexing: scheme by which the original stream and the
associated retransmission stream are sent into two different RTP
sessions.
SSRC-multiplexing: scheme by which the original stream and the
retransmission stream are sent in the same RTP session with
different SSRC values.
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
[2].
3. Requirements and design rationale for a retransmission scheme
The use of retransmissions in RTP as a repair method for streaming
media is appropriate in those scenarios with relaxed delay bounds
and where full reliability is not a requirement. More
specifically, RTP retransmission allows to trade-off reliability
vs. delay, i.e. the endpoints may give up retransmitting a lost
packet after a given buffering time has elapsed. Unlike TCP there
is thus no head-of-line blocking caused by RTP retransmissions.
The implementer should be aware that in cases where full
reliability is required or higher delay and jitter can be
tolerated, TCP or other transport options should be considered.
The RTP retransmission scheme defined in this document is designed
to fulfil the following set of requirements:
1. It must not break general RTP and RTCP mechanisms.
2. It must be suitable for unicast and small multicast groups.
3. It must work with mixers and translators.
4. It must work with all known payload types.
5. It must not prevent the use of multiple payload types in a
session.
6. In order to support the largest variety of payload formats, the
RTP receiver must be able to derive how many and which RTP
packets were lost as a result of a gap in received RTP sequence
numbers. This requirement is referred to as sequence number
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preservation. Without such a requirement, it would be
impossible to use retransmission with payload formats, such as
conversational text [9] or most audio/video streaming
applications, that use the RTP sequence number to detect lost
packets.
When designing a solution for RTP retransmission, several
approaches may be considered for the multiplexing of the original
RTP packets and the retransmitted RTP packets.
One approach may be to retransmit the RTP packet with its original
sequence number and send original and retransmission packets in
the same RTP stream. The retransmission packet would then be
identical to the original RTP packet, i.e. the same header (and
thus same sequence number) and the same payload. However, such an
approach is not acceptable because it would corrupt the RTCP
statistics. As a consequence, requirement 1 would not be met.
Correct RTCP statistics require that for every RTP packet within
the RTP stream, the sequence number be increased by one.
Another approach may be to multiplex original RTP packets and
retransmission packets in the same RTP stream using different
payload type values. With such an approach, the original packets
and the retransmission packets would share the same sequence
number space. As a result, the RTP receiver would not be able to
infer how many and which original packets (which sequence numbers)
were lost.
In other words, this approach does not satisfy the sequence number
preservation requirement (requirement 6). This in turn implies
that requirement 4 would not be met. Interoperability with mixers
and translators would also be more difficult if they did not
understand this new retransmission payload type in a sender RTP
stream. For these reasons, a solution based on payload type
multiplexing of original packets and retransmission packets in the
same RTP stream is excluded.
Finally, the original and retransmission packets may be sent in
two separate streams. These two streams may be multiplexed either
by sending them in two different sessions , i.e., session-
multiplexing, or in the same session using different SSRC values,
i.e. SSRC-multiplexing. Since original and retransmission packets
carry media of the same type, the objections in Section 5.2 of RTP
[3] to RTP multiplexing do not apply in this case.
Mixers and translators may process the original stream and simply
discard the retransmission stream if they are unable to utilise
it.
On the other hand, sending the original and retransmission packets
in two separate streams does not alone satisfy requirements 1 and
6. For this purpose, this document includes the original sequence
number in the retransmitted packets.
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In this manner, using two separate streams satisfies all the
requirements listed in this section.
3.1 Multiplexing scheme choice
Session-multiplexing and SSRC-multiplexing have different pros and
cons:
Session-multiplexing is based on sending the retransmission stream
in a different RTP session (as defined in RTP [3]) from that of
the original stream, i.e. the original and retransmission streams
are sent to different network addresses and/or port numbers.
Having a separate session allows more flexibility. In multicast,
using two separate sessions for the original and the
retransmission streams allows a receiver to choose whether or not
to subscribe to the RTP session carrying the retransmission
stream. The original session may also be single-source multicast
while separate unicast sessions are used to convey retransmissions
to each of the receivers, which as a result will receive only the
retransmission packets they request.
The use of separate sessions also facilitates differential
treatment by the network and may simplify processing in mixers,
translators and packet caches.
With SSRC-multiplexing, a single session is needed for the
original and the retransmission stream. This allows streaming
servers and middleware which are involved in a high number of
concurrent sessions to minimise their port usage.
This retransmission payload format allows both session-
multiplexing and SSRC-multiplexing for unicast sessions. From an
implementation point of view, there is little difference between
the two approaches. Hence, in order to maximise interoperability,
both multiplexing approaches SHOULD be supported by senders and
receivers. For multicast sessions, session-multiplexing MUST be
used because the association of the original stream and the
retransmission stream is problematic if SSRC-multiplexing is used
with multicast sessions(see Section 5.3 for motivation).
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4. Retransmission payload format
The format of a retransmission packet is shown below:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTP Header |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| OSN | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| Original RTP Packet Payload |
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The RTP header usage is as follows:
In the case of session-multiplexing, the same SSRC value MUST be
used for the original stream and the retransmission stream. In
the case of an SSRC collision in either the original session or
the retransmission session, the RTP specification requires that an
RTCP BYE packet MUST be sent in the session where the collision
happened. In addition, an RTCP BYE packet MUST also be sent for
the associated stream in its own session. After a new SSRC
identifier is obtained, the SSRC of both streams MUST be set to
this value.
In the case of SSRC-multiplexing, two different SSRC values MUST
be used for the original stream and the retransmission stream as
required by RTP. If an SSRC collision is detected for either the
original stream or the retransmission stream, the RTP
specification requires that an RTCP BYE packet MUST be sent for
this stream. An RTCP BYE packet MUST NOT be sent for the
associated stream. Therefore, only the stream that experienced
SSRC collision MUST choose a new SSRC value. Refer to Section 5.3
for the implications on the original and retransmission stream
SSRC association at the receiver.
For either multiplexing scheme, the sequence number has the
standard definition, i.e. it MUST be one higher than the sequence
number of the preceding packet sent in the retransmission stream.
The retransmission packet timestamp MUST be set to the original
timestamp, i.e. to the timestamp of the original packet. As a
consequence, the initial RTP timestamp for the first packet of the
retransmission stream is not random but equal to the original
timestamp of the first packet that is retransmitted. See the
security considerations section in this document for security
implications.
Implementers have to be aware that the RTCP jitter value for the
retransmission stream does not reflect the actual network jitter
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since there could be little correlation between the time a packet
is retransmitted and its original timestamp.
The payload type is dynamic. If multiple payload types using
retransmission are present in the original stream, then for each
of these, a dynamic payload type MUST be mapped to the
retransmission payload format. See Section 8.1 for the
specification of how the mapping between original and
retransmission payload types is done with SDP.
As the retransmission packet timestamp carries the original media
timestamp, the timestamp clockrate used by the retransmission
payload type MUST be the same as the one used by the associated
original payload type. Therefore, if an RTP stream carries
payload types of different clockrates, this will also be the case
for the associated retransmission stream. Note that an RTP stream
does not usually carry payload types of different clockrates.
The payload of the RTP retransmission packet comprises the
retransmission payload header followed by the payload of the
original RTP packet. The length of the retransmission payload
header is 2 octets. This payload header contains only one field,
OSN (original sequence number), which MUST be set to the sequence
number of the associated original RTP packet. The original RTP
packet payload, including any possible payload headers specific to
the original payload type, MUST be placed right after the
retransmission payload header.
For payload formats that support encoding at multiple rates,
instead of retransmitting the same payload as the original RTP
packet the sender MAY retransmit the same data encoded at a lower
rate. This aims at limiting the bandwidth usage of the
retransmission stream. When doing so, the sender MUST ensure that
the receiver will still be able to decode the payload of the
already sent original packets that might have been encoded based
on the payload of the lost original packet. In addition, if the
sender chooses to retransmit at a lower rate, the values in the
payload header of the original RTP packet may not longer apply to
the retransmission packet and may need to be modified in the
retransmission packet to reflect the change in rate. The sender
SHOULD trade-off the decrease in bandwidth usage with the decrease
in quality caused by resending at a lower rate.
If the original RTP header carried any profile-specific
extensions, the retransmission packet SHOULD include the same
extensions immediately following the fixed RTP header as expected
by applications running under this profile. In this case, the
retransmission payload header MUST be placed after the profile-
specific extensions.
If the original RTP header carried an RTP header extension, the
retransmission packet SHOULD carry the same header extension.
This header extension MUST be placed right after the fixed RTP
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header, as specified in RTP [3]. In this case, the retransmission
payload header MUST be placed after the header extension.
If the original RTP packet contained RTP padding, that padding
MUST be removed before constructing the retransmission packet. If
padding of the retransmission packet is needed, padding MUST be
performed as with any RTP packets and the padding bit MUST be set.
The marker bit (M), the CSRC count (CC) and the CSRC list of the
original RTP header MUST be copied "as is" into the RTP header of
the retransmission packet.
5. Association of retransmission and original streams
5.1 Retransmission session sharing
In the case of session-multiplexing, a retransmission session MUST
map to exactly one original session, i.e. the same retransmission
session cannot be used for different original sessions.
If retransmission session sharing were allowed, it would be a
problem for receivers, since they would receive retransmissions
for original sessions they might not have joined. For example, a
receiver wishing to receive only audio would receive also
retransmitted video packets if an audio and video session shared
the same retransmission session.
5.2 CNAME use
In both the session-multiplexing and the SSRC-multiplexing cases,
a sender MUST use the same CNAME [3] for an original stream and
its associated retransmission stream.
5.3 Association at the receiver
A receiver receiving multiple original and retransmission streams
needs to associate each retransmission stream with its original
stream. The association is done differently depending on whether
session-multiplexing or SSRC-multiplexing is used.
If session-multiplexing is used, the receiver associates the two
streams having the same SSRC in the two sessions. Note that the
payload type field cannot be used to perform the association as
several media streams may have the same payload type value. The
two sessions are themselves associated out-of-band. See Section 8
for how the grouping of the two sessions is done with SDP.
If SSRC-multiplexing is used, the receiver should first of all
look for two streams that have the same CNAME in the session. In
some cases, the CNAME may not be enough to determine the
association as multiple original streams in the same session may
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share the same CNAME. For example, there can be in the same video
session multiple video streams mapping to different SSRCs and
still using the same CNAME and possibly the same PT values. Each
(or some) of these streams may have an associated retransmission
stream.
In this case, in order to find out the association between
original and retransmission streams having the same CNAME, the
receiver SHOULD behave as follows.
The association can generally be resolved when the receiver
receives a retransmission packet matching a retransmission request
which had been sent earlier. Upon reception of a retransmission
packet whose original sequence number has been previously
requested, the receiver can derive that the SSRC of the
retransmission packet is associated to the sender SSRC from which
the packet was requested.
However, this mechanism might fail if there are two outstanding
requests for the same packet sequence number in two different
original streams of the session. Note that since the initial
packet sequence numbers are random, the probability of having two
outstanding requests for the same packet sequence number would be
very small. Nevertheless, in order to avoid ambiguity in the
unicast case, the receiver MUST NOT have two outstanding requests
for the same packet sequence number in two different original
streams before the association is resolved. In multicast, this
ambiguity cannot be completely avoided, because another receiver
may have requested the same sequence number from another stream.
Therefore, SSRC-multiplexing MUST NOT be used in multicast
sessions.
If the receiver discovers that two senders are using the same SSRC
or if it receives an RTCP BYE packet, it MUST stop requesting
retransmissions for that SSRC. Upon reception of original RTP
packets with a new SSRC, the receiver MUST perform the SSRC
association again as described in this section.
6. Use with the extended RTP profile for RTCP-based feedback
This section gives general hints for the usage of this payload
format with the extended RTP profile for RTCP-based feedback,
denoted AVPF [1]. Note that the general RTCP send and receive
rules and the RTCP packet format as specified in RTP apply, except
for the changes that the AVPF profile introduces. In short, the
AVPF profile relaxes the RTCP timing rules and specifies
additional general-purpose RTCP feedback messages. See [1] for
details.
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6.1 RTCP at the sender
In the case of session-multiplexing, Sender Report (SR) packets
for the original stream are sent in the original session and SR
packets for the retransmission stream are sent in the
retransmission session according to the rules of RTP.
In the case of SSRC-multiplexing, SR packets for both original and
retransmission streams are sent in the same session according to
the rules of RTP. The original and retransmission streams are
seen, as far the RTCP bandwidth calculation is concerned, as
independent senders belonging to the same RTP session and are thus
equally sharing the RTCP bandwidth assigned to senders.
Note that in both cases, session- and SSRC-multiplexing, BYE
packets MUST still be sent for both streams as specified in RTP.
In other words, it is not enough to send BYE packets for the
original stream only.
6.2 RTCP Receiver Reports
In the case of session-multiplexing, the receiver will send report
blocks for the original stream and the retransmission stream in
separate Receiver Report (RR) packets belonging to separate RTP
sessions. RR packets reporting on the original stream are sent in
the original RTP session while RR packets reporting on the
retransmission stream are sent in the retransmission session. The
RTCP bandwidth for these two sessions may be chosen independently
(for example through RTCP bandwidth modifiers [4]).
In the case of SSRC-multiplexing, the receiver sends report blocks
for the original and the retransmission streams in the same RR
packet since there is a single session.
6.3 Retransmission requests
The NACK feedback message format defined in the AVPF profile
SHOULD be used by receivers to send retransmission requests.
Whether a receiver chooses to request a packet or not is an
implementation issue. An actual receiver implementation should
take into account such factors as the tolerable application delay,
the network environment and the media type.
The receiver should generally assess whether the retransmitted
packet would still be useful at the time it is received. The
timestamp of the missing packet can be estimated from the
timestamps of packets preceding and/or following the sequence
number gap caused by the missing packet in the original stream.
In most cases, some form of linear estimate of the timestamp is
good enough.
Furthermore, a receiver should compute an estimate of the round-
trip time (RTT) to the sender. This can be done, for example, by
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measuring the retransmission delay to receive a retransmission
packet after a NACK has been sent for that packet. This estimate
may also be obtained from past observations, RTCP report round-
trip time if available or any other means. A standard mechanism
for the receiver to estimate the RTT is specified in RTP Extended
Reports [11].
The receiver should not send a retransmission request as soon as
it detects a missing sequence number but should add some extra
delay to compensate for packet reordering. This extra delay may,
for example, be based on past observations of the experienced
packet reordering. It should be noted that, in environments where
packet reordering is rare or does not take place, e.g., if the
underlying datalink layer affords ordered delivery, the delay may
be extremely low or even take the value zero. In such cases, an
appropriate "reorder delay" algorithm may not actually be timer-
based, but packet-based. E.g., if n number of packets are
received after a gap is detected, then it may be assumed that the
packet was truly lost rather than out of order. This may turn out
to be far easier to code on some platforms as a very short fixed
FIFO packet buffer as opposed to the timer-based mechanism.
To increase the robustness to the loss of a NACK or of a
retransmission packet, a receiver may send a new NACK for the same
packet. This is referred to as multiple retransmissions. Before
sending a new NACK for a missing packet, the receiver should rely
on a timer to be reasonably sure that the previous retransmission
attempt has failed and so avoid unnecessary retransmissions. The
timer value shall be based on the observed round-trip time. Both,
a static or an adaptive value MAY be used. E.g.: an adaptive timer
could be one that changes its value with every new request for the
same packet. This document does not provide any guidelines as to
how this adaptive value should be calculated because no
experiments have been done to find this out.
NACKs MUST be sent only for the original RTP stream. Otherwise,
if a receiver wanted to perform multiple retransmissions by
sending a NACK in the retransmission stream, it would not be able
to know the original sequence number and a timestamp estimation of
the packet it requests.
Appendix A gives some guidelines as to how to control the number
of retransmissions.
6.4 Timing rules
The NACK feedback message may be sent in a regular full compound
RTCP packet or in an early RTCP packet, as per AVPF [1]. Sending
a NACK in an early packet allows to react more quickly to a given
packet loss. However, in that case if a new packet loss occurs
right after the early RTCP packet was sent, the receiver will then
have to wait for the next regular RTCP compound packet after the
early packet. Sending NACKs only in regular RTCP compound
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decreases the maximum delay between detecting an original packet
loss and being able to send a NACK for that packet. Implementers
should consider the possible implications of this fact for the
application being used.
Furthermore, receivers may make use of the minimum interval
between regular RTCP compound packets. This interval can be used
to keep regular receiver reporting down to a minimum, while still
allowing receivers to send early RTCP packets during periods
requiring more frequent feedback, e.g. times of higher packet loss
rate. Note that although RTCP packets may be suppressed because
they do not contain NACKs, the same RTCP bandwidth as if they were
sent needs to be available. See AVPF [1] for details on the use
of the minimum interval.
7. Congestion control
RTP retransmission poses a risk of increasing network congestion.
In a best-effort environment, packet loss is caused by congestion.
Reacting to loss by retransmission of older data without
decreasing the rate of the original stream would thus further
increase congestion. Implementations SHOULD follow the
recommendations below in order to use retransmission.
The RTP profile under which the retransmission scheme is used
defines an appropriate congestion control mechanism in different
environments. Following the rules under the profile, an RTP
application can determine its acceptable bitrate and packet rate
in order to be fair to other TCP or RTP flows.
If an RTP application uses retransmission, the acceptable packet
rate and bitrate includes both the original and retransmitted
data. This guarantees that an application using retransmission
achieves the same fairness as one that does not. Such a rule
would translate in practice into the following actions:
If enhanced service is used, it should be made sure that the total
bitrate and packet rate do not exceed that of the requested
service. It should be further monitored that the requested
services are actually delivered. In a best-effort environment,
the sender SHOULD NOT send retransmission packets without reducing
the packet rate and bitrate of the original stream (for example by
encoding the data at a lower rate).
In addition, the sender MAY selectively retransmit only the
packets that it deems important and ignore NACK messages for other
packets in order to limit the bitrate.
These congestion control mechanisms should keep the packet loss
rate within acceptable parameters. In the context of congestion
control, packet loss is considered acceptable if a TCP flow across
the same network path and experiencing the same network conditions
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Internet Draft RTP Retransmission Payload Format September 2005
would achieve, on a reasonable timescale, an average throughput,
that is not less than the one the RTP flow achieves. If congestion
is not kept under control, then retransmission SHOULD NOT be used.
Retransmissions MAY still be sent in some cases, e. g., in
wireless links where packet losses are not caused by congestion,
if the server (or the client that makes the retransmission
request) estimates that a particular packet or frame is important
to continue play out, or if an RTSP PAUSE has been issued to allow
the buffer to fill up (RTSP PAUSE does not affect the sending of
retransmissions.)
Finally, it may further be necessary to adapt the transmission
rate (or the number of layers subscribed for a layered multicast
session), or to arrange for the receiver to leave the session.
8. Retransmission Payload Format MIME type registration
8.1 Introduction
The following MIME subtype name and parameters are introduced in
this document: "rtx", "rtx-time" and "apt".
The binding used for the retransmission stream to the payload type
number is indicated by an rtpmap attribute. The MIME subtype name
used in the binding is "rtx".
The "apt" (associated payload type) parameter MUST be used to map
the retransmission payload type to the associated original stream
payload type. If multiple original payload types are used, then
multiple "apt" parameters MUST be included to map each original
payload type to a different retransmission payload type.
An OPTIONAL payload-format-specific parameter, "rtx-time",
indicates the maximum time a sender will keep an original RTP
packet in its buffers available for retransmission. This time
starts with the first transmission of the packet.
The syntax is as follows:
a=fmtp:<number> apt=<apt-value>;rtx-time=<rtx-time-val>
where,
<number>: indicates the dynamic payload type number assigned
to the retransmission payload format in an rtpmap attribute.
<apt-value>: the value of the original stream payload type to
which this retransmission stream payload type is associated.
<rtx-time-val>: specifies the time in milliseconds (measured
from the time a packet was first sent) that a sender keeps an
RTP packet in its buffers available for retransmission. The
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Internet Draft RTP Retransmission Payload Format September 2005
absence of the rtx-time parameter for a retransmission stream
means that the maximum retransmission time is not defined,
but MAY be negotiated by other means.
8.2 Registration of audio/rtx
MIME type: audio
MIME subtype: rtx
Required parameters:
rate: the RTP timestamp clockrate is equal to the RTP
timestamp clockrate of the media that is retransmitted.
apt: associated payload type. The value of this parameter is
the payload type of the associated original stream.
Optional parameters:
rtx-time: indicates the time in milliseconds (measured from
the time a packet was first sent) that the sender keeps an
RTP packet in its buffers available for retransmission.
Encoding considerations: this type is only defined for transfer
via RTP.
Security considerations: see Section 12 of RFC XXXX
Interoperability considerations: none
Published specification: RFC XXXX
Applications which use this media type: multimedia streaming
applications
Additional information: none
Person & email address to contact for further information:
jose.rey@eu.panasonic.com
david.leon@nokia.com
avt@ietf.org
Intended usage: COMMON
Authors:
Jose Rey
David Leon
Change controller:
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Internet Draft RTP Retransmission Payload Format September 2005
IETF AVT WG delegated from the IESG
8.3 Registration of video/rtx
MIME type: video
MIME subtype: rtx
Required parameters:
rate: the RTP timestamp clockrate is equal to the RTP
timestamp clockrate of the media that is retransmitted.
apt: associated payload type. The value of this parameter is
the payload type of the associated original stream.
Optional parameters:
rtx-time: indicates the time in milliseconds (measured from
the time a packet was first sent) that the sender keeps an
RTP packet in its buffers available for retransmission.
Encoding considerations: this type is only defined for transfer
via RTP.
Security considerations: see Section 12 of RFC XXXX
Interoperability considerations: none
Published specification: RFC XXXX
Applications which use this media type: multimedia streaming
applications
Additional information: none
Person & email address to contact for further information:
jose.rey@eu.panasonic.com
david.leon@nokia.com
avt@ietf.org
Intended usage: COMMON
Authors:
Jose Rey
David Leon
Change controller:
IETF AVT WG delegated from the IESG
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Internet Draft RTP Retransmission Payload Format September 2005
8.4 Registration of text/rtx
MIME type: text
MIME subtype: rtx
Required parameters:
rate: the RTP timestamp clockrate is equal to the RTP
timestamp clockrate of the media that is retransmitted.
apt: associated payload type. The value of this parameter is
the payload type of the associated original stream.
Optional parameters:
rtx-time: indicates the time in milliseconds (measured from
the time a packet was first sent) that the sender keeps an
RTP packet in its buffers available for retransmission.
Encoding considerations: this type is only defined for transfer
via RTP.
Security considerations: see Section 12 of RFC XXXX
Interoperability considerations: none
Published specification: RFC XXXX
Applications which use this media type: multimedia streaming
applications
Additional information: none
Person & email address to contact for further information:
jose.rey@eu.panasonic.com
david.leon@nokia.com
avt@ietf.org
Intended usage: COMMON
Authors:
Jose Rey
David Leon
Change controller:
IETF AVT WG delegated from the IESG
8.5 Registration of application/rtx
MIME type: application
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Internet Draft RTP Retransmission Payload Format September 2005
MIME subtype: rtx
Required parameters:
rate: the RTP timestamp clockrate is equal to the RTP
timestamp clockrate of the media that is retransmitted.
apt: associated payload type. The value of this parameter is
the payload type of the associated original stream.
Optional parameters:
rtx-time: indicates the time in milliseconds (measured from
the time a packet was first sent) that the sender keeps an
RTP packet in its buffers available for retransmission.
Encoding considerations: this type is only defined for transfer
via RTP.
Security considerations: see Section 12 of RFC XXXX
Interoperability considerations: none
Published specification: RFC XXXX
Applications which use this media type: multimedia streaming
applications
Additional information: none
Person & email address to contact for further information:
jose.rey@eu.panasonic.com
david.leon@nokia.com
avt@ietf.org
Intended usage: COMMON
Authors:
Jose Rey
David Leon
Change controller:
IETF AVT WG delegated from the IESG
8.6 Mapping to SDP
The information carried in the MIME media type specification has a
specific mapping to fields in SDP [5], which is commonly used to
describe RTP sessions. When SDP is used to specify
retransmissions for an RTP stream, the mapping is done as
follows:
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Internet Draft RTP Retransmission Payload Format September 2005
- The MIME types ("video"), ("audio"), ("text") and
("application") go in the SDP "m=" as the media name.
- The MIME subtype ("rtx") goes in SDP "a=rtpmap" as the encoding
name. The RTP clock rate in "a=rtpmap" MUST be that of the
retransmission payload type. See Section 4 for details on this.
- The AVPF profile-specific parameters "ack" and "nack" go in SDP
"a=rtcp-fb". Several SDP "a=rtcp-fb" are used for several types
of feedback. See the AVPF profile [1] for details.
- The retransmission payload-format-specific parameters "apt" and
"rtx-time" go in the SDP "a=fmtp" as a semicolon separated list of
parameter=value pairs.
- Any remaining parameters go in the SDP "a=fmtp" attribute by
copying them directly from the MIME media type string as a
semicolon separated list of parameter=value pairs.
In the following sections some example SDP descriptions are
presented. In some of these examples, long lines are folded to
meet the column width constraints of this document; the backslash
("\") at the end of a line and the carriage return that follows it
should be ignored.
8.7 SDP description with session-multiplexing
In the case of session-multiplexing, the SDP description contains
one media specification "m" line per RTP session. The SDP MUST
provide the grouping of the original and associated retransmission
sessions' "m" lines, using the Flow Identification (FID) semantics
defined in RFC 3388 [6].
The following example specifies two original, AMR and MPEG-4,
streams on ports 49170 and 49174 and their corresponding
retransmission streams on ports 49172 and 49176, respectively:
v=0
o=mascha 2980675221 2980675778 IN IP4 host.example.net
c=IN IP4 192.0.2.0
a=group:FID 1 2
a=group:FID 3 4
m=audio 49170 RTP/AVPF 96
a=rtpmap:96 AMR/8000
a=fmtp:96 octet-align=1
a=rtcp-fb:96 nack
a=mid:1
m=audio 49172 RTP/AVPF 97
a=rtpmap:97 rtx/8000
a=fmtp:97 apt=96;rtx-time=3000
a=mid:2
m=video 49174 RTP/AVPF 98
a=rtpmap:98 MP4V-ES/90000
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Internet Draft RTP Retransmission Payload Format September 2005
a=rtcp-fb:98 nack
a=fmtp:98 profile-level-id=8;config=01010000012000884006682C209\
0A21F
a=mid:3
m=video 49176 RTP/AVPF 99
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98;rtx-time=3000
a=mid:4
A special case of the SDP description is a description that
contains only one original session "m" line and one retransmission
session "m" line, the grouping is then obvious and FID semantics
MAY be omitted in this special case only.
This is illustrated in the following example, which is an SDP
description for a single original MPEG-4 stream and its
corresponding retransmission session:
v=0
o=mascha 2980675221 2980675778 IN IP4 host.example.net
c=IN IP4 192.0.2.0
m=video 49170 RTP/AVPF 96
a=rtpmap:96 MP4V-ES/90000
a=rtcp-fb:96 nack
a=fmtp:96 profile-level-id=8;config=01010000012000884006682C209\
0A21F
m=video 49172 RTP/AVPF 97
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96;rtx-time=3000
8.8 SDP description with SSRC-multiplexing
The following is an example of an SDP description for an RTP video
session using SSRC-multiplexing with similar parameters as in the
single-session example above:
v=0
o=mascha 2980675221 2980675778 IN IP4 host.example.net
c=IN IP4 192.0.2.0
m=video 49170 RTP/AVPF 96 97
a=rtpmap:96 MP4V-ES/90000
a=rtcp-fb:96 nack
a=fmtp:96 profile-level-id=8;config=01010000012000884006682C209\
0A21F
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96;rtx-time=3000
9. RTSP considerations
The Real-time Streaming Protocol (RTSP), RFC 2326 [7] is an
application-level protocol for control over the delivery of data
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Internet Draft RTP Retransmission Payload Format September 2005
with real-time properties. This section looks at the issues
involved in controlling RTP sessions that use retransmissions.
9.1 RTSP control with SSRC-multiplexing
In the case of SSRC-multiplexing, the "m" line includes both
original and retransmission payload types and has a single RTSP
"control" attribute. The receiver uses the "m" line to request
SETUP and TEARDOWN of the whole media session. The RTP profile
contained in the Transport header MUST be the AVPF profile or
another suitable profile allowing extended feedback. If the SSRC
value is included in the SETUP response's Transport header, it
MUST be that of the original stream.
In order to control the sending of the session original media
stream, the receiver sends as usual PLAY and PAUSE requests to the
sender for the session. The RTP-info header that is used to set
RTP-specific parameters in the PLAY response MUST be set according
to the RTP information of the original stream.
When the receiver starts receiving the original stream, it can
then request retransmission through RTCP NACKs without additional
RTSP signalling.
9.2 RTSP control with session-multiplexing
In the case of session-multiplexing, each SDP "m" line has an RTSP
"control" attribute. Hence, when retransmission is used, both the
original session and the retransmission have their own "control"
attributes. The receiver can associate the original session and
the retransmission session through the FID semantics as specified
in Section 8.
The original and the retransmission streams are set up and torn
down separately through their respective media "control"
attribute. The RTP profile contained in the Transport header MUST
be the AVPF profile or another suitable profile allowing extended
feedback for both the original and the retransmission session.
The RTSP presentation SHOULD support aggregate control and SHOULD
contain a session level RTSP URL. The receiver SHOULD use
aggregate control for an original session and its associated
retransmission session. Otherwise, there would need to be two
different 'session-id' values, i.e. different values for the
original and retransmission sessions, and the sender would not
know how to associate them.
The session-level "control" attribute is then used as usual to
control the playing of the original stream. When the receiver
starts receiving the original stream, it can then request
retransmissions through RTCP without additional RTSP signalling.
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Internet Draft RTP Retransmission Payload Format September 2005
9.3 RTSP control of the retransmission stream
Because of the nature of retransmissions, the sending of
retransmission packets SHOULD NOT be controlled through RTSP PLAY
and PAUSE requests. The PLAY and PAUSE requests SHOULD NOT affect
the retransmission stream. Retransmission packets are sent upon
receiver requests in the original RTCP stream, regardless of the
state.
9.4 Cache control
Retransmission streams SHOULD NOT be cached.
In the case of session-multiplexing, the "Cache-Control" header
SHOULD be set to "no-cache" for the retransmission stream.
In the case of SSRC-multiplexing, RTSP cannot specify independent
caching for the retransmission stream, because there is a single
"m" line in SDP. Therefore, the implementer should take this fact
into account when deciding whether to cache an SSRC-multiplexed
session or not.
10. Implementation examples
This document mandates only the sender and receiver behaviours
that are necessary for interoperability. In addition, certain
algorithms, such as rate control or buffer management when
targeted at specific environments, may enhance the retransmission
efficiency.
This section gives an overview of different implementation options
allowed within this specification.
The first example describes a minimal receiver implementation.
With this implementation, it is possible to retransmit lost RTP
packets, detect efficiently the loss of retransmissions and
perform multiple retransmissions, if needed. Most of the
necessary processing is done at the server.
The second example shows how retransmissions may be used in
(small) multicast groups in conjunction with layered encoding. It
illustrates that retransmissions and layered encoding may be
complementary techniques.
10.1 A minimal receiver implementation example
This section gives an example of an implementation supporting
multiple retransmissions. The sender transmits the original data
in RTP packets using the MPEG-4 video RTP payload format.
It is assumed that NACK feedback messages are used, as per
[1]. An SDP description example with SSRC-multiplexing is given
below:
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Internet Draft RTP Retransmission Payload Format September 2005
v=0
o=mascha 2980675221 2980675778 IN IP4 host.example.net
c=IN IP4 192.0.2.0
m=video 49170 RTP/AVPF 96 97
a=rtpmap:96 MP4V-ES/90000
a=rtcp-fb:96 nack
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96;rtx-time=3000
The format-specific parameter "rtx-time" indicates that the server
will buffer the sent packets in a retransmission buffer for 3.0
seconds, after which the packets are deleted from the
retransmission buffer and will never be sent again.
In this implementation example, the required RTP receiver
processing to handle retransmission is kept to a minimum. The
receiver detects packet loss from the gaps observed in the
received sequence numbers. It signals lost packets to the sender
through NACKs as defined in the AVPF profile [1]. The receiver
should take into account the signalled sender retransmission
buffer length in order to dimension its own reception buffer. It
should also derive from the buffer length the maximum number of
times the retransmission of a packet can be requested.
The sender should retransmit the packets selectively, i.e. it
should choose whether to retransmit a requested packet depending
on the packet importance, the observed QoS and congestion state of
the network connection to the receiver. Obviously, the sender
processing increases with the number of receivers as state
information and processing load must be allocated to each
receiver.
10.2 Retransmission of Layered Encoded Media in Multicast
This section shows how to combine retransmissions with layered
encoding in multicast sessions. Note that the retransmission
framework is not intended as a complete solution to reliable
multicast. Refer to RFC 2887 [10], for an overview of the
problems related with reliable multicast transmission.
Packets of different importance are sent in different RTP
sessions. The retransmission streams corresponding to the
different layers can themselves be seen as different
retransmission layers. The relative importance of the different
retransmission streams should reflect the relative importance of
the different original streams.
In multicast, SSRC-multiplexing of the original and retransmission
streams is not allowed as per Section 5.3 of this document. For
this reason, the retransmission stream(s) MUST be sent in
different RTP session(s) using session-multiplexing.
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Internet Draft RTP Retransmission Payload Format September 2005
An SDP description example of multicast retransmissions for
layered encoded media is given below:
m=video 8000 RTP/AVPF 98
c=IN IP4 224.2.1.0/127/3
a=rtpmap:98 MP4V-ES/90000
a=rtcp-fb:98 nack
m=video 8000 RTP/AVPF 99
c=IN IP4 224.2.1.3/127/3
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98;rtx-time=3000
The server and the receiver may implement the retransmission
methods illustrated in the previous examples. In addition, they
may choose to request and retransmit a lost packet depending on
the layer it belongs to.
11. IANA considerations
A new MIME subtype name, "rtx", has been registered for four
different media types, as follows: "video", "audio", "text" and
"application". An additional REQUIRED parameter, "apt", and an
OPTIONAL parameter, "rtx-time", are defined. See Section 8 for
details.
12. Security considerations
RTP packets using the payload format defined in this specification
are subject to the general security considerations discussed in
RTP, Section 9.
In common streaming scenarios message authentication, data
integrity, replay protection and confidentiality are desired.
The absence of authentication may enable man-in-the-middle and
replay attacks, which can be very harmful for RTP retransmission.
For example: tampered RTCP packets may trigger inappropriate
retransmissions that effectively reduce the actual bitrate share
allocated to the original data stream, tampered RTP retransmission
packets could cause the client's decoder to crash, tampered
retransmission requests may invalidate the SSRC association
mechanism described in Section 5 of this document. On the other
hand, replayed packets could lead to false re-ordering and RTT
measurements (required for the retransmission request strategy)
and may cause the receiver buffer to overflow.
Further, in order to ensure confidentiality of the data, the
original payload data needs to be encrypted. There is actually no
need to encrypt the 2-byte retransmission payload header since it
does not provide any hints about the data content.
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Internet Draft RTP Retransmission Payload Format September 2005
Furthermore, it is RECOMMENDED that the cryptography mechanisms
used for this payload format provide protection against known
plaintext attacks. RTP recommends that the initial RTP timestamp
SHOULD be random to secure the stream against known plaintext
attacks. This payload format does not follow this recommendation
as the initial timestamp will be the media timestamp of the first
retransmitted packet. However, since the initial timestamp of the
original stream is itself random, if the original stream is
encrypted, the first retransmitted packet timestamp would also be
random to an attacker. Therefore, confidentiality would not be
compromised.
If cryptography is used to provide security services on the
original stream, then the same services, with equivalent
cryptographic strength, MUST be provided on the retransmission
stream. The use of the same key for the retransmitted stream and
the original stream may lead to security problems, e. g., two-time
pads. Refer to Section 9.1 of the Secure Real-Time Transport
Protocol (SRTP)[12] for a discussion the implications of two-time
pads and how to avoid them.
At the time of writing this document, SRTP does not provide all
the security services mentioned. There are, at least, two reasons
for this: 1) the occurrence of two-time pads and 2) the fact that
this payload format typically works under the RTP/AVPF profile
while SRTP only supports RTP/AVP. An adapted variant of SRTP
shall solve these shortcomings in the future.
Congestion control considerations with the use of retransmission
are dealt with in Section 7 of this document.
13. Acknowledgements
We would like to express our gratitude to Carsten Burmeister for
his participation in the development of this document. Our thanks
also go to Koichi Hata, Colin Perkins, Stephen Casner, Magnus
Westerlund, Go Hori and Rahul Agarwal for their helpful comments.
14. References
14.1 Normative References
1 J. Ott, S. Wenger, N. Sato, C. Burmeister, J. Rey, "Extended RTP
profile for RTCP-based feedback", draft-ietf-avt-rtcp-feedback-
11.txt, August 2004.
2 S. Bradner, "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997
3 H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson, "RTP: A
Transport Protocol for Real-Time Applications", RFC 3550, July
2003.
Rey, et al. [Page 25]
Internet Draft RTP Retransmission Payload Format September 2005
4 S. Casner, "SDP bandwidth modifiers for RTCP bandwidth", RFC
3556, July 2003.
5 M. Handley, V. Jacobson, "SDP: Session Description Protocol",
RFC 2327, April 1998.
6 G. Camarillo, J. Holler, G. AP. Eriksson, "Grouping of media
lines in the Session Description Protocol (SDP)", RFC 3388,
December 2002.
7 H. Schulzrinne, A. Rao, R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, April 1998.
14.2 Informative References
8 C. Perkins, O. Hodson, "Options for Repair of Streaming Media",
RFC 2354, June 1998.
9 G. Hellstrom, "RTP for conversational text", RFC 2793, May 2000
10 M. Handley, et al., "The Reliable Multicast Design Space for
Bulk Data Transfer", RFC 2887, August 2000.
11 Friedman, et. al., "RTP Extended Reports", Work in Progress.
12 M. Baugher, D. A. McGrew, D. Oran, R. Blom, E. Carrara, M.
Naslund, K. Norrman, "The Secure Real-Time Transport Protocol",
RFC 3711, March 2004.
13 R. Hovey and S. Bradner, "The Organizations Involved in the IETF
Standards Process," BCP 11, RFC 2028, IETF, October 1996.
15. Author's Addresses
Jose Rey jose.rey@eu.panasonic.com
Panasonic R&D Center Germany GmbH
Monzastr. 4c
D-63225 Langen, Germany
Phone: +49-6103-766-134
Fax: +49-6103-766-166
David Leon david.leon@nokia.com
Nokia Research Center
6000 Connection Drive
Irving, TX. USA
Phone: 1-972-374-1860
Akihiro Miyazaki miyazaki.akihiro@jp.panasonic.com
CE Architecture Development Center
Matsushita Electric Industrial Co., Ltd.
1006 Kadoma, Kadoma City, Osaka 571-8501, Japan
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Internet Draft RTP Retransmission Payload Format September 2005
Phone: +81-6-6900-9172
Fax: +81-6-6900-9173
Viktor Varsa viktor.varsa@nokia.com
Nokia Research Center
6000 Connection Drive
Irving, TX. USA
Phone: 1-972-374-1861
Rolf Hakenberg rolf.hakenberg@eu.panasonic.com
Panasonic R&D Center Germany GmbH
Monzastr. 4c
D-63225 Langen, Germany
Phone: +49-6103-766-162
Fax: +49-6103-766-166
Appendix A. How to control the number of rtxs. per packet
Finding out the number of retransmissions (rtxs.) per packet for
achieving the best possible transmission is a difficult task. Of
course, the absolute minimum should be one (1) - otherwise, do not
use this payload format. Moreover, as of date of publication, the
authors were not aware of any studies on the number of
retransmissions per packet that should be used for best
performance. To help implementers and researchers on this item,
this section describes an estimate of the buffering time required
for achieving a given number of retransmissions. Once this time
has been calculated, it can be communicated to the client via SDP
parameter "rtx-time", as defined in this document.
Scenario and Assumptions
========================
* Streaming scenario with relaxed delay bounds. Client and server
are provided with buffering space as indicated by the parameter
"rtx-time" in SDP.
* RTP AVPF profile used with SSRC-multiplexing retransmission
scheme: 1 SSRC for original packets, 1 for retransmission packets.
* Default RTCP bandwidth share for SRs and RRs, i.e., SR+RR=0.05.
We have senders (2) and receivers (1). Receivers and senders get
equally 1/3 of the RTCP bandwidth share because the proportion of
senders is greater than 1/4 of session members.
* avg-rtcp-size is approximated by 120 bytes. This is a rounded-
up average of 2 SRs, one for each SSRC, containing 40/8/28/32
bytes for IPv6/UDP/SR/SDES with CNAME, thus making 105 bytes each;
and a RR with 40/8/64/32 bytes for IPv6/UDP/2*RR/SDES, making 157
bytes. Since senders and receivers share the RTCP bandwidth
equally, then avg-rtcp-size=(157+105+105)/3=117,3~=120 bytes. The
important characteristic of this value is that it is something
over 100 bytes, which seems to be a representative figure for
typical configurations.
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Internet Draft RTP Retransmission Payload Format September 2005
* The profile used is AVPF [1] and Generic NACKs are used for
requesting retransmissions. This adds 16 bytes of overhead for 1
NACK and 4 bytes more for every additional NACK FCI field.
* We assume a worst-case scenario in which each packet exhausts
its corresponding number of available retransmissions, N, before
it is received. This means that if a packet may be requested for
retransmission a maximum of 2 times, the corresponding generic
NACK report block requesting that particular packet is sent in two
consecutive RTCP compounds; likewise, if it is requested for
retransmission 10 times, then the generic NACK is sent 10 times.
This assumption makes the RTCP packet size approx. constant after
N*RTCP intervals (seconds), namely to avg-rtcp-size= 120 +
(receiver-RTCP-bw-share)*(12 + 4*N). In our case, the receiver
RTCP bandwidth share is 1/3, thus avg-rtcp-size = 124 + 4*N/3.
* Two delay parameters are difficult to approximate and may be
implementation-dependent. Therefore, we list them here explicitly
without assigning them a particular value: one is the packet loss
detection time (T2) and the other feedback processing and queuing
time for retransmissions (T5). Implementers shall assign
appropriate values to these two parameters .
Graphically, we have:
Sender
+-+---------------------------------^-----\-----------------
\ \ / \
\ \ | |
SN=0 \ \ SN=1 / \ RTX(SN=0)
\ \ / \
X \ / \
`. / \
\ / \
\ | |
\ / \ ......
\ / \
-------------V----D--------/-----------------------V--------
T1 T2 T3 T4 T5 T1 ........
Receiver
Legend:
=======
DL : downlink (client->server)
UL : uplink (server->client)
Time unit is seconds, s.
Bitrate unit is bit per second, bps.
DL transmission time : T1= physical-delay-DL +
tx-delay-DL(=avg-pkt-size/DL-bitrate) + interarrival-delay-jitter
Time to detect packet loss : T2= pkt-loss-detect-time
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Internet Draft RTP Retransmission Payload Format September 2005
Time to report packet loss : T3= time-to-next-rtcp-report
UL transmission time : T4= physical-delay-UL +
transmission-delay-UL + interarrival-delay-jitter
Retransmissions processing time : T5= feedback-processing-time +
rtx-queuing-time
Goal
====
To find an estimate of the buffering time, T(), that a streaming
server shall use in order to enable a given number of
retransmissions for each packet, N. This time is approximately
equal at the server and at the client, if one considers that the
client starts buffering T1 seconds later.
Solution
========
First we find the value of the estimate for 1 retransmission,
T(1)=T:
T = T1 + T2 + T3 + T4 + T5
Since T1 + T4 ~= RTT,
T = RTT + T2 + T3 + T5
The worst case for T3 would be that we assume that reporting has
to wait a whole RTCP interval and that the maximum randomization
factor of 1.5 is applied. Therefore, after applying the
subsequent compensation to avoid traffic bursts (see RTP Section
A.7 [3]), we have that T3 = 1.5/1.21828*RTCP-Interval. Thus,
T = RTT + 1.2312*RTCP-Interval + T2 + T5
On the other hand, RTCP-Interval = avg-rtcp-size*8*(senders +
receivers)/(RR+RS). In this scenario: sender + receivers = 3;
RR+RS is the receiver report plus sender report bandwidth share,
in this case, equal to the default 5% of session bandwidth, bw.
We assume an average RTCP packet size, avg-rtcp-size=120 bytes.
This includes Thus:
T = RTT + 1.2312*avg-rtcp-size*8*3/(0.05*bw) + T2 + T5
for 1 retransmission.
For enabling N retransmissions, the available buffering time in a
streaming server or client is
approximately:
T(N) = N*(RTT+1.2312*avg-rtcp-size*8*3/(0.05*bw) + T2 + T5)
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Internet Draft RTP Retransmission Payload Format September 2005
where, as per above,
avg-rtcp-size = 120 + (receiver-RTCP-bw-share=1/3)*(12 + 4*N) =
= 124 + 4*N/3.
Numbers
========
If we ignore the effect of T2 and T5, i.e., assume all losses are
detected immediately and that there is no additional delay due to
feedback processing or retransmission queuing, we have the
following buffering times for different values of N:
RTCP w/ several Generic NACKs; variable packet size= 124 + 4*N/3
bytes
|============|=====|======================================|
| RTP BW | RTT | N value |
|============|=====|======================================|
1,00 2,00 5,00 7,00 10,00
64000 0,05 1,21 2,44 6,28 8,97 13,18
128000 0,05 0,63 1,27 3,27 4,66 6,84
256000 0,05 0,34 0,68 1,76 2,50 3,67
512000 0,05 0,19 0,39 1,00 1,43 2,09
1024000 0,05 0,12 0,25 0,63 0,89 1,29
5000000 0,05 0,06 0,13 0,33 0,46 0,66
10000000 0,05 0,06 0,11 0,29 0,41 0,58
64000 0,2 1,36 2,74 7,03 10,02 14,68
128000 0,2 0,78 1,57 4,02 5,71 8,34
256000 0,2 0,49 0,98 2,51 3,55 5,17
512000 0,2 0,34 0,69 1,75 2,48 3,59
1024000 0,2 0,27 0,55 1,38 1,94 2,79
5000000 0,2 0,21 0,43 1,08 1,51 2,16
10000000 0,2 0,21 0,41 1,04 1,46 2,08
64000 1 2,16 4,34 11,03 15,62 22,68
128000 1 1,58 3,17 8,02 11,31 16,34
256000 1 1,29 2,58 6,51 9,15 13,17
512000 1 1,14 2,29 5,75 8,08 11,59
1024000 1 1,07 2,15 5,38 7,54 10,79
5000000 1 1,01 2,03 5,08 7,11 10,16
10000000 1 1,01 2,01 5,04 7,06 10,08
To quantify the error of not taking the Generic NACKS into
account, we can do the same numbers, but ignoring the Generic NACK
contribution, avg-rtcp-size ~= 120 bytes. As we see from below,
this may result in a buffer estimation error of 1-1.5 seconds (5-
10%) for lower bandwidth values and higher number of
retransmissions. This effect is low in this case. Nevertheless,
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Internet Draft RTP Retransmission Payload Format September 2005
it should be carefully evaluated for the particular scenario; that
is why the formula includes it.
RTCP w/o Generic NACK, fixed packet size ~= 120 bytes
|============|=====|======================================|
| RTP BW | RTT | N value |
|============|=====|======================================|
1,00 2,00 5,00 7,00 10,00
64000 0,05 1,16 2,32 5,79 8,11 11,58
128000 0,05 0,60 1,21 3,02 4,23 6,04
256000 0,05 0,33 0,65 1,64 2,29 3,27
512000 0,05 0,19 0,38 0,94 1,32 1,89
1024000 0,05 0,12 0,24 0,60 0,83 1,19
5000000 0,05 0,06 0,13 0,32 0,45 0,64
10000000 0,05 0,06 0,11 0,29 0,40 0,57
64000 0,2 1,31 2,62 6,54 9,16 13,08
128000 0,2 0,75 1,51 3,77 5,28 7,54
256000 0,2 0,48 0,95 2,39 3,34 4,77
512000 0,2 0,34 0,68 1,69 2,37 3,39
1024000 0,2 0,27 0,54 1,35 1,88 2,69
5000000 0,2 0,21 0,43 1,07 1,50 2,14
10000000 0,2 0,21 0,41 1,04 1,45 2,07
64000 1 2,11 4,22 10,54 14,76 21,08
128000 1 1,55 3,11 7,77 10,88 15,54
256000 1 1,28 2,55 6,39 8,94 12,77
512000 1 1,14 2,28 5,69 7,97 11,39
1024000 1 1,07 2,14 5,35 7,48 10,69
5000000 1 1,01 2,03 5,07 7,10 10,14
10000000 1 1,01 2,01 5,04 7,05 10,07
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Rey, et al. [Page 31]
Internet Draft RTP Retransmission Payload Format September 2005
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Rey, et al. [Page 32]