Internet Draft
draft-ietf-avt-rtp-retransmission- J. Rey/Matsushita
07.txt D. Leon/Nokia
A. Miyazaki/Matsushita
V. Varsa/Nokia
R. Hakenberg/Matsushita
Expires: November 2003 April 2003
RTP Retransmission Payload Format
Status of this Memo
This document is an Internet-Draft and is in full conformance
with all provisions of Section 10 of RFC 2026.
Internet-Drafts are working documents of the Internet Engineering
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Copyright Notice
Copyright (C) The Internet Society (2003). All Rights Reserved.
[Note to RFC Editor: This paragraph is to be deleted when this
draft is published 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
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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.
Table of Contents
1. Introduction..................................................3
2. Terminology...................................................3
3. Requirements and design rationale for a retransmission scheme.4
4. Retransmission payload format.................................6
5. Asocciation of a retransmission stream to its original stream.8
6. Use with the extended RTP profile for RTCP-based feedback....10
7. Congestion control...........................................12
8. Retransmission Payload Format MIME type registration.........13
9. RTSP considerations..........................................19
10. Implementation examples.....................................21
11. IANA considerations.........................................24
12. Security considerations.....................................24
13. Acknowledgements............................................24
14. References..................................................25
15. Author's Addresses..........................................26
IPR Notices.....................................................26
Full Copyright Statement........................................27
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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.
However, 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.
Retransmission may thus be considered for such applications.
This document specifies a retransmission method for RTP applicable
to unicast and (small) multicast groups: it defines a payload
format for retransmitted RTP packets and provides protocol rules
for the sender and the receiver involved in retransmissions.
Furthermore, 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 small number of general-purpose feedback
messages, e.g. ACKs and NACKs, 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
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
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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
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.
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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. Using two separate streams thus 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.
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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).
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
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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. No RTCP BYE packet MUST be sent for the associated
stream. Therefore, only the stream that experienced SSRC
collision will 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 is 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
since there could be little correlation between the time a packet
is retransmitted and its original timestamp.
The payload type is dynamic. Each payload type of the original
stream MUST map to a different payload type value in the
retransmission stream. Therefore, when multiple payload types are
used in the original stream, multiple dynamic payload types will
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 is the same as the one used by the associated
original payload type. It is thus possible to send retransmission
packets whose original payload types have different timestamp
clockrates in the same 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,
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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, is placed right after the
retransmission payload header.
For payload types 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 is thus 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
header, as specified in RTP [3]. In this case, the retransmission
payload header is thus 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 is
performed as with any RTP packets and the padding bit is 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 a retransmission stream to its original stream
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.
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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 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
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
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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.
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
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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
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.
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 in order to avoid unnecessary retransmissions.
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.
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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
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
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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. Packet loss is considered
acceptable if a TCP flow across the same network path and
experiencing the same network conditions would achieve, on a
reasonable timescale, an average throughput, that is not less than
the one the RTP flow achieves. If the packet loss rate exceeds an
acceptable level, it should be concluded that congestion is not
kept under control and retransmission should then not be used. 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.
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<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
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:
rey@panasonic.de
david.leon@nokia.com
avt@ietf.org
Intended usage: COMMON
Author/Change controller:
Jose Rey
David Leon
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IETF AVT WG
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:
rey@panasonic.de
david.leon@nokia.com
avt@ietf.org
Intended usage: COMMON
Author/Change controller:
Jose Rey
David Leon
IETF AVT WG
8.4 Registration of text/rtx
MIME type: text
MIME subtype: rtx
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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:
rey@panasonic.de
david.leon@nokia.com
avt@ietf.org
Intended usage: COMMON
Author/Change controller:
Jose Rey
David Leon
IETF AVT WG
8.5 Registration of application/rtx
MIME type: application
MIME subtype: rtx
Required parameters:
rate: the RTP timestamp clockrate is equal to the RTP
timestamp clockrate of the media that is retransmitted.
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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:
rey@panasonic.de
david.leon@nokia.com
avt@ietf.org
Intended usage: COMMON
Author/Change controller:
Jose Rey
David Leon
IETF AVT WG
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:
- 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.
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- 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
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
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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
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
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"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.
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.
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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 a receiver may implement additional
enhancements that might help reduce sender buffer requirements and
optimise the retransmission efficiency
The third 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:
v=0
o=mascha 2980675221 2980675778 IN IP4 host.example.net
c=IN IP4 192.0.2.0
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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 An enhanced receiver implementation example
The receiver may have more accurate information than the sender
about the current network QoS such as available bandwidth, packet
loss rate, delay and jitter. In addition, other receiver-specific
parameters such as buffer level, estimated importance of the lost
packet and application level QoS may be used by the receiver to
make a more efficient use of RTP retransmission by selectively
sending NACKs for important lost packets and not for others. For
example, a receiver may decide to suppress a request for a packet
loss that could be concealed locally, or for a retransmission that
would arrive late.
Furthermore, a receiver may acknowledge the received packets.
This can be done by sending ACKs, as per [1]. Upon receiving an
ACK, the sender may delete all the acknowledged packets from its
retransmission buffer. Note that this would also require only
limited increase in the required RTCP bandwidth as long as ACK
packets are sent seldom enough.
This implementation may help reduce buffer requirements at the
sender and optimise the performance of the implementation by using
selective requests.
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Note that these receiver enhancements do not need to be negotiated
as they do not affect the sender implementation. However, in
order to allow the receiver to acknowledge packets, it is needed
to allow the use of ACKs in the SDP description, by means of an
additional SDP "a=rtcp-fb" line, as follows:
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=rtcp-fb:96 ack
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96;rtx-time=3000
10.3 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.
An SDP description example of multicast retransmissions for
layered encoded media is given below:
m=video 8000 RTP/AVPF 98
c=IN IP4 192.0.2.0/127/3
a=rtpmap:98 MP4V-ES/90000
a=rtcp-fb:98 nack
m=video 8000 RTP/AVPF 99
c=IN IP4 192.0.2.4/127/3
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.
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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
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. Old keys will likely need to be cached so that when the
keys change for the original stream, the old key is used until it
is determined that the key has changed on the retransmission
packets as well.
The use of the same key for the retransmitted stream and the
original stream may lead to security problems, e.g. two-time pads.
This sharing has to be evaluated towards the chosen security
protocol and security algorithms.
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.
Congestion control considerations with the use of retransmission
are dealt with in Section 7 of this document.
Any other security considerations of the profile under which the
retransmission scheme is used should be applied. The
retransmission payload format MUST NOT be used under the SAVP
profile defined by the Secure Real-Time Transport Protocol
(SRTP)[12] but instead an extension of SRTP should be defined to
secure the AVPF profile. The definition of such a profile is out
of the scope 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
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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-
04.txt, September 2002.
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", draft-ietf-avt-
rtp-new-12.txt, March 2003.
4 S. Casner, "SDP bandwidth modifiers for RTCP bandwidth", draft-
ietf-avt-rtcp-bw-05.txt, May 2002.
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",
draft-ietf-avt-srtp-05.txt, June 2002.
13 R. Hovey and S. Bradner, "The Organizations Involved in the IETF
Standards Process," BCP 11, RFC 2028, IETF, October 1996.
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15. Author's Addresses
Jose Rey rey@panasonic.de
Panasonic European Laboratories 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 akihiro@isl.mei.co.jp
Core Software Development Center
Corporate Software Development Division
Matsushita Electric Industrial Co., Ltd.
1006 Kadoma, Kadoma City, Osaka 571-8501, Japan
Phone: +81-6-6900-9192
Fax: +81-6-6900-9193
Viktor Varsa viktor.varsa@nokia.com
Nokia Research Center
6000 Connection Drive
Irving, TX. USA
Phone: 1-972-374-1861
Rolf Hakenberg hakenberg@panasonic.de
Panasonic European Laboratories GmbH
Monzastr. 4c
D-63225 Langen, Germany
Phone: +49-6103-766-162
Fax: +49-6103-766-166
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Information on the IETF's procedures with respect to rights in
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or the result of an attempt made to obtain a general license or
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or users of this specification can be obtained from the IETF
Secretariat.
Rey, et al. [Page 26]
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Rey, et al. [Page 27]
