Internet Draft
draft-ietf-avt-rtp-retransmission-05.txt J. Rey/Matsushita
D. Leon/Nokia
A. Miyazaki/Matsushita
V. Varsa/Nokia
R. Hakenberg/Matsushita
Expires: August 2003 February 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
Task Force (IETF), its areas, and its working groups. Note that
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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 from receivers to
senders is available. In particular, it is assumed that RTCP
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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. Association of a retransmission stream with 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.......................................20
11. IANA considerations...........................................23
12. Security considerations.......................................23
13. Acknowledgements..............................................24
14. References....................................................24
Author's Addresses................................................25
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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 specified
in [1] is used as retransmission request for lost packets.
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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.
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.
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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. 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
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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:
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 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
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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,
OSN, 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.
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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 M, CC and CSRC bit of the original RTP header MUST remain
unchanged in the retransmission packet.
5. Association of a retransmission stream with 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.
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.
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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
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.
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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
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
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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.
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.
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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. 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
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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 payload types are used for the original
streams, then multiple "apt" parameters MUST be included to map each
original stream payload type to a different retransmission payload
type.
An OPTIONAL payload-format-specific parameter, "rtx-time," indicates
the maximum time a server will try to retransmit a 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>: indicates the time in milliseconds, measured
from the time a packet was first sent until the time the server
will stop trying to retransmit the packet. 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:
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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 until the time the server will
stop trying to retransmit the packet.
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
8.3 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.
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Optional parameters:
rtx-time: indicates the time in milliseconds, measured from the
time a packet was first sent until the time the server will
stop trying to retransmit the packet.
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
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 until the time the server will
stop trying to retransmit the packet.
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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.
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 until the time the server will
stop trying to retransmit the packet.
Encoding considerations: this type is only defined for transfer via
RTP.
Security considerations: see Section 12 of RFC XXXX
Interoperability considerations: none
Rey, et al. [Page 16]
Internet Draft RTP Retransmission Payload Format February 2003
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") and ("text") 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.
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
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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 at.home.ru
c=IN IP4 125.25.5.1
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=01010000012000884006682C2090A21F
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 at.home.ru
c=IN IP4 125.25.5.1
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=01010000012000884006682C2090A21F
m=video 49172 RTP/AVPF 97
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96;rtx-time=3000
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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 at.home.ru
c=IN IP4 125.25.5.1
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=01010000012000884006682C2090A21F
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
"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.
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.
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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.
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.
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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 at.home.ru
c=IN IP4 125.25.5.1
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
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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.
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 at.home.ru
c=IN IP4 125.25.5.1
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
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Internet Draft RTP Retransmission Payload Format February 2003
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:
c=IN IP4 224.2.1.1/127/3
m=video 8000 RTP/AVPF 98
a=rtpmap:98 MP4V-ES/90000
a=rtcp-fb:98 nack
c=IN IP4 224.2.1.4/127/3
m=video 8000 RTP/AVPF 99
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. An additional
REQUIRED parameter, "apt", and an OPTIONAL parameter, "rtx-time",
are defined. See Section 8 for details.
12. Security considerations
Applications utilising encryption SHOULD encrypt both the original
and 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.
RTP recommends that the initial RTP timestamp SHOULD be random to
secure the stream against known plain text attacks. This payload
format does not follow this recommendation as the initial timestamp
will be the media timestamp of the first retransmitted packet.
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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 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-11.txt, May 2002.
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.
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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.
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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Rey, et al. [Page 26]