draft-ietf-avt-rtp-retransmission-04.txt   draft-ietf-avt-rtp-retransmission-05.txt 
Internet Draft Internet Draft
draft-ietf-avt-rtp-retransmission- Jose Rey/Matsushita draft-ietf-avt-rtp-retransmission-05.txt J. Rey/Matsushita
04.txt David Leon/Nokia D. Leon/Nokia
Akihiro Miyazaki/Matsushita A. Miyazaki/Matsushita
Viktor Varsa/Nokia V. Varsa/Nokia
Rolf Hakenberg/Matsushita R. Hakenberg/Matsushita
Expires: May 2003 December 2002 Expires: August 2003 February 2003
RTP retransmission payload format RTP Retransmission Payload Format
Status of this Memo Status of this Memo
This document is an Internet-Draft and is in full conformance This document is an Internet-Draft and is in full conformance
with all provisions of Section 10 of RFC2026. with all provisions of Section 10 of RFC2026.
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skipping to change at line 33 skipping to change at page 1, line 34
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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 [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 draft is published as an RFC. References in this draft to RFC XXXX
should be replaced with the RFC number assigned to this document. should be replaced with the RFC number assigned to this document.]
References in this draft to RFC YYYY should be replaced with the RFC
number assigned the draft-ietf-mmusic-fid when published as RFC.
References in this draft to RFC ZZZZ should be replaced with the RFC
number assigned the draft-ietf-avt-rtcp-bw when published as RFC.
References in this draft to RFC UUUU should be replaced with the
RFC number assigned the draft-ietf-avt-srtp when published as RFC.
References in this draft to RFC VVVV should be replaced with the RFC
number assigned the draft-ietf-avt-rtcp-feedback when published as
RFC. References in this draft to RFC WWWW should be replaced with
the RFC number of the revision of RFC 1889 being drafted as draft-
ietf-avt-rtp-new. Main changes since draft-ietf-avt-rtp-
retransmission-02.txt: this document is the result of the merging of
draft-ietf-avt-selret-05.txt and draft-ietf-avt-rtp-retransmission-
02.txt. Main changes since draft-ietf-avt-rtp-retransmission-03.txt:
RTSP section new drafted.]
IETF draft - Expires May 2003 1
Abstract Abstract
RTP retransmission is an effective packet loss recovery technique RTP retransmission is an effective packet loss recovery technique
for real-time applications with relaxed delay bounds. This document for real-time applications with relaxed delay bounds. This document
describes an RTP payload format for performing retransmissions. describes an RTP payload format for performing retransmissions.
Retransmitted RTP packets are sent in a separate stream from the Retransmitted RTP packets are sent in a separate stream from the
original RTP stream. It is assumed that feedback from receivers to original RTP stream. It is assumed that feedback from receivers to
senders is available. In particular, it is assumed that RTCP senders is available. In particular, it is assumed that RTCP
feedback as defined in the extended RTP profile for RTCP-based feedback as defined in the extended RTP profile for RTCP-based
feedback [1] ( denoted AVPF ), is available in this memo. feedback (denoted RTP/AVPF), is available in this memo.
Table of Contents Table of Contents
1. Introduction....................................................3 1. Introduction....................................................3
2. Terminology.....................................................3 2. Terminology.....................................................3
3. Requirements and design rationale for a retransmission scheme...4 3. Requirements and design rationale for a retransmission scheme...4
4. Retransmission payload format...................................6 4. Retransmission payload format...................................6
5. Association of a retransmission stream with its original stream.7 5. Association of a retransmission stream with its original stream.8
6. Use with the extended RTP profile for RTCP-based feedback.......8 6. Use with the extended RTP profile for RTCP-based feedback......10
7. Congestion control.............................................10 7. Congestion control.............................................12
8. SDP usage......................................................11 8. Retransmission Payload Format MIME type registration...........13
9. RTSP considerations............................................14 9. RTSP considerations............................................19
10. Implementation examples.......................................15 10. Implementation examples.......................................20
11. IANA considerations...........................................18 11. IANA considerations...........................................23
12. Security considerations.......................................22 12. Security considerations.......................................23
13. Acknowledgements..............................................22 13. Acknowledgements..............................................24
14. References....................................................22 14. References....................................................24
Author's Addresses................................................23 Author's Addresses................................................25
Rey/Leon/Miyazaki/Varsa/Hakenberg 2
1. Introduction 1. Introduction
Packet losses between an RTP sender and receiver may significantly Packet losses between an RTP sender and receiver may significantly
degrade the quality of the received media. Several techniques, such degrade the quality of the received media. Several techniques, such
as forward error correction (FEC), retransmissions or interleaving as forward error correction (FEC), retransmissions or interleaving
may be considered to increase packet loss resiliency. RFC 2354 [9] may be considered to increase packet loss resiliency. RFC 2354 [8]
discusses the different options. discusses the different options.
When choosing a repair technique for a particular application, the When choosing a repair technique for a particular application, the
tolerable latency of the application has to be taken into account. tolerable latency of the application has to be taken into account.
In the case of multimedia conferencing, the end-to-end delay has to In the case of multimedia conferencing, the end-to-end delay has to
be at most a few hundred milliseconds in order to guarantee be at most a few hundred milliseconds in order to guarantee
interactivity, which usually excludes the use of retransmission. interactivity, which usually excludes the use of retransmission.
However, in the case of multimedia streaming, the user can tolerate 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- an initial latency as part of the session set-up and thus an end-to-
skipping to change at line 126 skipping to change at page 3, line 47
2. Terminology 2. Terminology
The following terms are used in this document: The following terms are used in this document:
Original packet: refers to an RTP packet which carries user data Original packet: refers to an RTP packet which carries user data
sent for the first time by an RTP sender. sent for the first time by an RTP sender.
Original stream: refers to the RTP stream of original packets. Original stream: refers to the RTP stream of original packets.
Retransmission packet: refers to an RTP packet whose payload Retransmission packet: refers to an RTP packet which is to be used
includes the payload of an already sent original packet. Such a by the receiver instead of a lost original packet. Such a
retransmission packet is said to be associated with the original RTP retransmission packet is said to be associated with the original RTP
packet. packet.
Retransmission request: a means by which an RTP receiver is able to Retransmission request: a means by which an RTP receiver is able to
request that the RTP sender should send a retransmission packet for request that the RTP sender should send a retransmission packet for
a given original packet. Usually, an RTCP NACK message as specified a given original packet. Usually, an RTCP NACK packet as specified
in [1] is used as retransmission request for lost packets. in [1] is used as retransmission request for lost packets.
Rey/Leon/Miyazaki/Varsa/Hakenberg 3
Retransmission stream: the stream of retransmission packets Retransmission stream: the stream of retransmission packets
associated with an original stream. associated with an original stream.
Session-multiplexing: scheme by which the original stream and the Session-multiplexing: scheme by which the original stream and the
associated retransmission stream are sent into two different RTP associated retransmission stream are sent into two different RTP
sessions. sessions.
SSRC-multiplexing: scheme by which the original stream and the SSRC-multiplexing: scheme by which the original stream and the
retransmission stream are sent in the same RTP session with retransmission stream are sent in the same RTP session with
different SSRC values. different SSRC values.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [2]. document are to be interpreted as described in RFC 2119 [2].
3. Requirements and design rationale for a retransmission scheme 3. Requirements and design rationale for a retransmission scheme
The retransmission scheme is designed to fulfil the following set of The use of retransmissions in RTP as a repair method for streaming
requirements: 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.
1. It must not break general RTP and RTCP mechanisms 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. 2. It must be suitable for unicast and small multicast groups.
3. It must work with mixers and translators. 3. It must work with mixers and translators.
4. It must work with all known payload types. 4. It must work with all known payload types.
5. It must not prevent the use multiple payload types in a session. 5. It must not prevent the use of multiple payload types in a
6. In order to support the largest variety of payload formats the session.
RTP receiver must be able to indicate how many and which RTP 6. In order to support the largest variety of payload formats, the
packets were lost. This requirement is referred to as sequence RTP receiver must be able to derive how many and which RTP
number preservation. Without such a requirement, it would be packets were lost as a result of a gap in received RTP sequence
impossible to use retransmission with payload formats, such as numbers. This requirement is referred to as sequence number
conversational text [10] or most audio/video streaming 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 applications, that use the RTP sequence number to detect lost
packets. packets.
When designing a solution for RTP retransmission, several approaches When designing a solution for RTP retransmission, several approaches
may be considered for the multiplexing of the original RTP packets may be considered for the multiplexing of the original RTP packets
and the retransmitted RTP packets. and the retransmitted RTP packets.
One approach may be to retransmit the RTP packet with its original One approach may be to retransmit the RTP packet with its original
sequence number and send original and retransmission packets in the sequence number and send original and retransmission packets in the
same stream. The retransmission packet would then be identical to same RTP stream. The retransmission packet would then be identical
the original RTP packet, i.e. the same header (and thus same to the original RTP packet, i.e. the same header (and thus same
sequence number) and the same payload. However, such an approach is sequence number) and the same payload. However, such an approach is
not acceptable because it would corrupt the RTCP statistics. As a not acceptable because it would corrupt the RTCP statistics. As a
consequence, requirement 1 would not be met. Correct RTCP statistics consequence, requirement 1 would not be met. Correct RTCP
require that for every RTP packet within the RTP stream, the statistics require that for every RTP packet within the RTP stream,
sequence number be increased by one. the sequence number be increased by one.
Another approach may be to multiplex original RTP packets and Another approach may be to multiplex original RTP packets and
retransmission packets in the same stream using the payload type retransmission packets in the same RTP stream using different
field. With such an approach the original stream and the payload type values. With such an approach, the original packets
and the retransmission packets would share the same sequence number
Rey/Leon/Miyazaki/Varsa/Hakenberg 4 space. As a result, the RTP receiver would not be able to infer how
retransmission stream would share the same sequence number space. As many and which original packets (which sequence numbers) were lost.
a result, the RTP receiver would not be able to infer how many and
which original packets (i.e. with which sequence number) were lost.
In other words, this feature does not satisfy the sequence number In other words, this approach does not satisfy the sequence number
preservation requirement (requirement 6). This in turn implies that preservation requirement (requirement 6). This in turn implies that
requirement 4 would not be met. Interoperability with mixers and requirement 4 would not be met. Interoperability with mixers and
translators would also be more difficult if they do not understand translators would also be more difficult if they did not understand
this new payload type in a sender RTP stream. For these reasons, a this new retransmission payload type in a sender RTP stream. For
solution based on payload type multiplexing of original packets and these reasons, a solution based on payload type multiplexing of
retransmission packets in the same RTP stream is excluded. original packets and retransmission packets in the same RTP stream
is excluded.
Finally, the original and retransmission packets may be sent in two Finally, the original and retransmission packets may be sent in two
separate streams. These two streams may be multiplexed either by separate streams. These two streams may be multiplexed either by
sending them in two different sessions , i.e. session-multiplexing, sending them in two different sessions , i.e. session-multiplexing,
or in the same session using different SSRCs, i.e. SSRC-multi- or in the same session using different SSRC values, i.e. SSRC-
plexing. Since original and retransmission packets carry media of multiplexing. Since original and retransmission packets carry media
the same type, the objections in Section 5.2 of RTP, RFC WWWW [3] to of the same type, the objections in Section 5.2 of RTP [3] to RTP
RTP multiplexing do not apply. multiplexing do not apply in this case.
Using two separate streams satisfies all the requirements listed in Mixers and translators may process the original stream and simply
this section. Mixers and translators may process the original stream discard the retransmission stream if they are unable to utilise it.
and simply discard the retransmission stream if they are unable to Using two separate streams thus satisfies all the requirements
utilise it. listed in this section.
3.1 Multiplexing scheme choice 3.1 Multiplexing scheme choice
Session-multiplexing and SSRC-multiplexing have different pros and Session-multiplexing and SSRC-multiplexing have different pros and
cons: cons:
Session-multiplexing is based on sending the retransmission stream Session-multiplexing is based on sending the retransmission stream
in a different RTP session (as defined in RTP [3]) from that of the in a different RTP session (as defined in RTP [3]) from that of the
original stream, i.e. the original and retransmission streams are original stream, i.e. the original and retransmission streams are
sent to different network addresses and/or port numbers. Having a sent to different network addresses and/or port numbers. Having a
separate session allows more flexibility. In multicast, using two separate session allows more flexibility. In multicast, using two
sessions for retransmission allows a receiver to choose whether to separate sessions for the original and the retransmission streams
subscribe or not to the RTP session carrying the retransmission allows a receiver to choose whether or not to subscribe to the RTP
stream. It is also possible for the original session to be single- session carrying the retransmission stream. The original session
source multicast and have separate unicast sessions to convey may also be single-source multicast while separate unicast sessions
retransmissions to each of the receivers, which will then receive are used to convey retransmissions to each of the receivers, which
only the retransmission packets they requested. as a result will receive only the retransmission packets they
request.
The use of separate sessions also allows differential treatment by The use of separate sessions also facilitates differential treatment
the network and may simplify processing in mixers, translators and by the network and may simplify processing in mixers, translators
packet caches. and packet caches.
With SSRC-multiplexing, a single session is needed for the original With SSRC-multiplexing, a single session is needed for the original
and the retransmission stream. This allows streaming servers and and the retransmission stream. This allows streaming servers and
middleware which are involved in a high number of concurrent middleware which are involved in a high number of concurrent
sessions to minimise their port usage. sessions to minimise their port usage.
This retransmission payload format allows both session-multiplexing This retransmission payload format allows both session-multiplexing
and SSRC-multiplexing. From an implementation point of view, there and SSRC-multiplexing for unicast sessions. From an implementation
point of view, there is little difference between the two
Rey/Leon/Miyazaki/Varsa/Hakenberg 5 approaches. Hence, in order to maximise interoperability, both
is little difference between the two approaches. Hence, in order to multiplexing approaches SHOULD be supported by senders and
maximise interoperability, both multiplexing approaches SHOULD be receivers. For multicast sessions, session-multiplexing MUST be
supported. 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 4. Retransmission payload format
The format of a retransmission packet is shown below: The format of a retransmission packet is shown below:
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTP Header | | RTP Header |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| OSN | | | OSN | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| Original RTP Packet Payload | | Original RTP Packet Payload |
| | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The RTP header usage is as follows: The RTP header usage is as follows:
If the original and the retransmission streams are sent in separate In the case of session-multiplexing, the same SSRC value MUST be
RTP sessions, the same SSRC value MUST be used for the original used for the original stream and the retransmission stream. In the
stream and the retransmission stream. In case of an SSRC collision, case of an SSRC collision in either the original session or the
an RTCP BYE packet MUST be sent for the original RTP session. After retransmission session, the RTP specification requires that an RTCP
a new SSRC identifier is obtained, the SSRC of the retransmission BYE packet MUST be sent in the session where the collision happened.
session MUST be set to this value. 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.
If the original stream and the retransmission stream are sent in the In the case of SSRC-multiplexing, two different SSRC values MUST be
same RTP session, two different SSRC values MUST be used for the used for the original stream and the retransmission stream as
original stream and the retransmission stream as required by RTP. 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 For either multiplexing scheme, the sequence number has the standard
definition, i.e. it MUST be one higher than the sequence number of definition, i.e. it MUST be one higher than the sequence number of
the preceding packet sent in the retransmission stream. the preceding packet sent in the retransmission stream.
The retransmission packet timestamp is set to the original The retransmission packet timestamp is set to the original
timestamp, i.e. to the timestamp of the original packet. As a timestamp, i.e. to the timestamp of the original packet. As a
consequence, the initial RTP timestamp for the first packet of the consequence, the initial RTP timestamp for the first packet of the
retransmission stream is not random but equal to the original retransmission stream is not random but equal to the original
timestamp of the first packet requested for retransmission. See the timestamp of the first packet that is retransmitted. See the
security considerations section in this document for security security considerations section in this document for security
implications. implications.
Implementers have to be aware that the RTCP jitter value for the Implementers have to be aware that the RTCP jitter value for the
retransmission stream does not reflect the actual network jitter retransmission stream does not reflect the actual network jitter
since there could be little correlation between the time a packet is since there could be little correlation between the time a packet is
retransmitted and its original timestamp. retransmitted and its original timestamp.
The payload type is dynamic. Each payload type of the original The payload type is dynamic. Each payload type of the original
stream MUST map to a different payload type value in the stream MUST map to a different payload type value in the
retransmission stream. Therefore, when multiple payload types are retransmission stream. Therefore, when multiple payload types are
Rey/Leon/Miyazaki/Varsa/Hakenberg 6
used in the original stream, multiple dynamic payload types will be used in the original stream, multiple dynamic payload types will be
mapped to this retransmission payload format. See Section 8 for the mapped to the retransmission payload format. See Section 8.1 for
specification of how the mapping between original and retransmission the specification of how the mapping between original and
payload types is done. retransmission payload types is done with SDP.
As the retransmission packet timestamp carries the original media As the retransmission packet timestamp carries the original media
timestamp, the timestamp clockrate used by the retransmission timestamp, the timestamp clockrate used by the retransmission
payload type is the same as the one used by the associated original payload type is the same as the one used by the associated original
payload type. It is thus possible to retransmit RTP packets whose payload type. It is thus possible to send retransmission packets
payload types have different timestamp clockrates in the same whose original payload types have different timestamp clockrates in
retransmission stream if the original payload types have different the same retransmission stream. Note that an RTP stream does not
clock rates, but this is usually not the case. usually carry payload types of different clockrates.
If the original RTP header carried any profile-specific payload The payload of the RTP retransmission packet comprises the
header, the retransmission packet MUST include this payload header. 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.
If the original RTP header carried an RTP header extension, the For payload types that support encoding at multiple rates, instead
retransmission packet SHOULD carry the same header extension. 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.
The retransmission payload carries a payload header followed by the When doing so, the sender MUST ensure that the receiver will still
original RTP packet payload. The length of payload header is 2 be able to decode the payload of the already sent original packets
octets. The payload header contains only one field, OSN, which MUST that might have been encoded based on the payload of the lost
be set to the sequence number of the associated original RTP packet. 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 packet contained RTP padding, that padding must If the original RTP header carried any profile-specific extensions,
be removed before constructing the retransmission packet. If padding the retransmission packet SHOULD include the same extensions
of the retransmission packet is needed, padding is performed as with immediately following the fixed RTP header as expected by
any RTP packets and the padding bit is set. applications running under this profile. In this case, the
retransmission payload header is thus placed after the profile-
specific extensions.
All other fields of the RTP header MUST have the same value as in If the original RTP header carried an RTP header extension, the
the associated original RTP packet 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. Association of a retransmission stream with its original stream
5.1 Retransmission session sharing 5.1 Retransmission session sharing
In the case of session-multiplexing, a retransmission session MUST In the case of session-multiplexing, a retransmission session MUST
map to exactly one original session, i.e. the same retransmission map to exactly one original session, i.e. the same retransmission
session cannot be used for different original sessions. session cannot be used for different original sessions.
If retransmission session sharing were allowed, a receiver joining If retransmission session sharing were allowed, it would be a
the retransmission session would also receive the retransmissions problem for receivers, since they would receive retransmissions for
belonging to all other original sessions which the receiver may have original sessions they might not have joined. For example, a
not joined. For example, a receiver wishing to receive only audio receiver wishing to receive only audio would receive also
would receive retransmitted video packets if an audio and video retransmitted video packets if an audio and video session shared the
session would share the same retransmission session. same retransmission session.
5.2 CNAME use 5.2 CNAME use
A sender MUST use the same CNAME for an original stream and its 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. associated retransmission stream.
Rey/Leon/Miyazaki/Varsa/Hakenberg 7
5.3 Association at the receiver 5.3 Association at the receiver
A receiver receiving multiple original and retransmission streams A receiver receiving multiple original and retransmission streams
needs to associate each retransmission stream with its original needs to associate each retransmission stream with its original
stream. The association is done differently depending on whether stream. The association is done differently depending on whether
session-multiplexing or SSRC-multiplexing is used. session-multiplexing or SSRC-multiplexing is used.
If session-multiplexing is used, the receiver associates the two If session-multiplexing is used, the receiver associates the two
streams having the same SSRC in the two sessions. Note that the streams having the same SSRC in the two sessions. Note that the
payload type field cannot be used to do this coupling as several payload type field cannot be used to perform the association as
media streams may have the same payload type value. The two sessions several media streams may have the same payload type value. The two
are themselves associated out-of-band. See the SDP section to see sessions are themselves associated out-of-band. See Section 8 for
how the grouping of the two sessions is done with SDP. how the grouping of the two sessions is done with SDP.
If SSRC-multiplexing is used, the receiver should first of all look 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 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 cases, the CNAME may not be enough to determine the association as
multiple original streams in the same session may share the same multiple original streams in the same session may share the same
CNAME. For example, there can be in the same video session multiple CNAME. For example, there can be in the same video session multiple
video streams mapping to different SSRCs and still using the same video streams mapping to different SSRCs and still using the same
CNAME and possibly the same PT values. Each (or some of) these CNAME and possibly the same PT values. Each (or some) of these
streams may have an associated retransmission stream. streams may have an associated retransmission stream.
In order to find out the association between original and In this case, in order to find out the association between original
retransmission streams having the same CNAME, the receiver SHOULD and retransmission streams having the same CNAME, the receiver
behave as follows. SHOULD behave as follows.
The association can generally be resolved when the receiver receives The association can generally be resolved when the receiver receives
a retransmission packet matching a retransmission request which had a retransmission packet matching a retransmission request which had
been sent earlier. Upon reception of a retransmission whose original been sent earlier. Upon reception of a retransmission packet whose
sequence number had been previously requested, the receiver can original sequence number has been previously requested, the receiver
derive that the SSRC of the retransmission packet is associated to can derive that the SSRC of the retransmission packet is associated
the sender SSRC from which the packet was requested. In order to to the sender SSRC from which the packet was requested.
avoid ambiguity, the receiver MUST NOT have two outstanding requests
for the same packet sequence number in two different original However, this mechanism might fail if there are two outstanding
streams before the association is resolved. Note that since the requests for the same packet sequence number in two different
initial packet timestamps are random, the probability of having two 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 outstanding requests for the same packet sequence number would be
very small. 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 If the receiver discovers that two senders are using the same SSRC
or if it receives an RTCP BYE packet, it MUST stop requesting or if it receives an RTCP BYE packet, it MUST stop requesting
retransmissions for that SSRC. Upon reception of original RTP retransmissions for that SSRC. Upon reception of original RTP
packets with a new SSRC, the receiver MUST perform the SSRC packets with a new SSRC, the receiver MUST perform the SSRC
association again as described in this section. association again as described in this section.
6. Use with the extended RTP profile for RTCP-based feedback 6. Use with the extended RTP profile for RTCP-based feedback
This section gives general hints for the usage of this payload This section gives general hints for the usage of this payload
format with the extended RTP profile for RTCP-based feedback [1], format with the extended RTP profile for RTCP-based feedback,
denoted AVPF. 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.
Rey/Leon/Miyazaki/Varsa/Hakenberg 8 6.1 RTCP at the sender
6.1 RTCP Receiver reports 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.
If the original RTP stream and the retransmission stream are sent to In the case of SSRC-multiplexing, SR packets for both original and
separate RTP sessions, the receiver will then send report blocks for retransmission streams are sent in the same session according to the
the original stream and the retransmission stream in separate RTCP rules of RTP. The original and retransmission streams are seen, as
receiver reports (RR) packets belonging to separate RTP sessions. far the RTCP bandwidth calculation is concerned, as independent
RTCP packets reporting on the original stream are sent in the senders belonging to the same RTP session and are thus equally
original RTP session while RTCP packets reporting on the 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 retransmission stream are sent in the retransmission session. The
RTCP bandwidth for these two sessions may be chosen independently RTCP bandwidth for these two sessions may be chosen independently
(for example through RTCP bandwidth modifiers RFC ZZZZ [4]). (for example through RTCP bandwidth modifiers [4]).
If the original RTP stream and the retransmission stream are sent in In the case of SSRC-multiplexing, the receiver sends report blocks
the same session (SSRC multiplexing), the receiver sends report for the original and the retransmission streams in the same RR
blocks for the original and the retransmission streams in the same packet since there is a single session.
RTCP RR packet.
6.2 Retransmission requests 6.3 Retransmission requests
The NACK message format defined in the AVPF profile SHOULD be used The NACK feedback message format defined in the AVPF profile SHOULD
by receivers to send retransmission requests. be used by receivers to send retransmission requests. Whether a
Whether a receiver chooses to request a packet or not is an receiver chooses to request a packet or not is an implementation
implementation issue. An actual receiver implementation should take issue. An actual receiver implementation should take into account
into account such factors as the tolerable application delay, the such factors as the tolerable application delay, the network
network environment and the media type. environment and the media type.
The receiver should generally assess whether the retransmitted The receiver should generally assess whether the retransmitted
packet would still be useful at the time it is received. The packet would still be useful at the time it is received. The
timestamp of the missing packet can be estimated from the timestamps timestamp of the missing packet can be estimated from the timestamps
of packets preceding and/or following the sequence number gap caused of packets preceding and/or following the sequence number gap caused
by the missing packet in the original stream. In most cases, some by the missing packet in the original stream. In most cases, some
form of linear estimate of the timestamp is good enough. form of linear estimate of the timestamp is good enough.
Furthermore, a receiver should compute an estimate of the round-trip Furthermore, a receiver should compute an estimate of the round-trip
time (RTT) to the sender. This can be done, for example, by time (RTT) to the sender. This can be done, for example, by
measuring the retransmission delay to receive a retransmission measuring the retransmission delay to receive a retransmission
packet after a NACK message has been sent for that packet. This packet after a NACK has been sent for that packet. This estimate
estimate may also be obtained from past observations, RTCP report may also be obtained from past observations, RTCP report round-trip
round-trip time if available or any other means. 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 The receiver should not send a retransmission request as soon as it
detects a missing sequence number but should add some extra delay to detects a missing sequence number but should add some extra delay to
compensate for packet reordering. This extra delay may, for example, compensate for packet reordering. This extra delay may, for
be based on past observations of the experienced packet reordering. example, be based on past observations of the experienced packet
reordering.
To increase the robustness to the loss of a NACK message or of a To increase the robustness to the loss of a NACK or of a
retransmission packet, a receiver may send a new NACK message. This retransmission packet, a receiver may send a new NACK for the same
is referred to as multiple retransmissions. Before sending a new packet. This is referred to as multiple retransmissions. Before
NACK message for a missing packet, the receiver should rely on a sending a new NACK for a missing packet, the receiver should rely on
timer to be reasonably sure that the previous retransmission attempt a timer to be reasonably sure that the previous retransmission
has failed in order not to cause unnecessary retransmissions. attempt has failed in order to avoid unnecessary retransmissions.
Rey/Leon/Miyazaki/Varsa/Hakenberg 9 NACKs MUST be sent only for the original RTP stream. Otherwise, if
NACK packets MUST be sent only for the original RTP stream. If a a receiver wanted to perform multiple retransmissions by sending a
receiver wanted to perform multiple-retransmissions by sending a
NACK in the retransmission stream, it would not be able to know the NACK in the retransmission stream, it would not be able to know the
original sequence number and a timestamp estimation of the packet it original sequence number and a timestamp estimation of the packet it
requests. requests.
6.3 Timing rules 6.4 Timing rules
The RTCP NACK packet may be sent in a regular full compound RTCP The NACK feedback message may be sent in a regular full compound
packet or in an early RTCP packet, as per AVPF [1]. Sending a NACK RTCP packet or in an early RTCP packet, as per AVPF [1]. Sending a
in an early packet allows to react more quickly to a given packet NACK in an early packet allows to react more quickly to a given
loss. However, in that case if a new packet loss occurs right after packet loss. However, in that case if a new packet loss occurs
the early RTCP packet was sent, the receiver will then have to wait right after the early RTCP packet was sent, the receiver will then
for the next regular RTCP compound packet after the early packet. have to wait for the next regular RTCP compound packet after the
Sending NACK packets only in regular RTCP compound decreases the early packet. Sending NACKs only in regular RTCP compound decreases
maximum delay between detecting an original packet loss and being the maximum delay between detecting an original packet loss and
able to send a NACK message for that packet. Implementers should being able to send a NACK for that packet. Implementers should
consider the possible implications of this fact for the application consider the possible implications of this fact for the application
being used. being used.
Furthermore, receivers may make use of the minimum interval between Furthermore, receivers may make use of the minimum interval between
regular RTCP compound packets. This can be used, for example, to regular RTCP compound packets. This interval can be used to keep
keep reception reporting down to a given minimum, while still regular receiver reporting down to a minimum, while still allowing
allowing receivers to react to periods requiring more frequent receivers to send early RTCP packets during periods requiring more
feedback, e.g. times of higher packet loss rate. In this way, frequent feedback, e.g. times of higher packet loss rate.. Note
receivers will try to keep the amount of sent RTCP packets as low as that although RTCP packets may be suppressed because they do not
specified by the minimum interval, but are still able to report contain NACKs, the same RTCP bandwidth as if they were sent needs to
packet losses quickly enough. Note that although RTCP packets may be be available. See AVPF [1] for details on the use of the minimum
suppressed because they do not contain NACK packets, the reserved interval.
RTCP bandwidth is the same as if they were sent. See AVPF [1] for
details.
7. Congestion control 7. Congestion control
RTP retransmission poses a risk of increasing network congestion. In RTP retransmission poses a risk of increasing network congestion.
a best-effort environment, packet loss is caused by congestion. In a best-effort environment, packet loss is caused by congestion.
Reacting to loss by retransmission of older data without decreasing Reacting to loss by retransmission of older data without decreasing
the rate of the original stream would thus further increase the rate of the original stream would thus further increase
congestion. Implementations SHOULD follow the recommendations below congestion. Implementations SHOULD follow the recommendations below
in order to use retransmission. in order to use retransmission.
The RTP profile under which the retransmission scheme is used The RTP profile under which the retransmission scheme is used
defines an appropriate congestion control mechanism in different defines an appropriate congestion control mechanism in different
environments. Following the rules under the profile, an RTP environments. Following the rules under the profile, an RTP
application can determine its acceptable bitrate and packet rate in application can determine its acceptable bitrate and packet rate in
order to be fair to other TCP or RTP flows. order to be fair to other TCP or RTP flows.
If an RTP application uses retransmission, the acceptable packet If an RTP application uses retransmission, the acceptable packet
rate and bitrate includes both the original and retransmitted data. rate and bitrate includes both the original and retransmitted data.
This guarantees that an application using retransmission achieves This guarantees that an application using retransmission achieves
the same fairness as one that does not. Such a rule would translate the same fairness as one that does not. Such a rule would translate
in practice into the following actions: in practice into the following actions:
Rey/Leon/Miyazaki/Varsa/Hakenberg 10
If enhanced service is used, it should be made sure that the total 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. bitrate and packet rate do not exceed that of the requested service.
It should be further monitored that the requested services are It should be further monitored that the requested services are
actually delivered. In a best-effort environment, the sender SHOULD actually delivered. In a best-effort environment, the sender SHOULD
NOT send retransmission packets without reducing the packet rate and NOT send retransmission packets without reducing the packet rate and
bitrate of the original stream (for example by encoding the data at bitrate of the original stream (for example by encoding the data at
a lower rate). a lower rate).
In addition, the sender MAY selectively retransmit only the packets In addition, the sender MAY selectively retransmit only the packets
that it deems important and ignore NACK messages for other packets that it deems important and ignore NACK messages for other packets
in order to limit the bitrate. in order to limit the bitrate.
These congestion control mechanisms should keep the packet loss rate These congestion control mechanisms should keep the packet loss rate
within acceptable parameters. Packet loss is considered acceptable within acceptable parameters. Packet loss is considered acceptable
if a TCP flow across the same network path and experiencing the same if a TCP flow across the same network path and experiencing the same
network conditions would achieve an average throughput, measured on network conditions would achieve, on a reasonable timescale, an
a reasonable timescale, that is not less than the RTP flow is average throughput, that is not less than the one the RTP flow
achieving. If the packet loss rate exceed acceptable parameters, achieves. If the packet loss rate exceeds an acceptable level, it
this would mean that congestion is not kept under control and should be concluded that congestion is not kept under control and
retransmission should then not be used. It may further be necessary retransmission should then not be used. It may further be necessary
to adapt the transmission rate (or the number of layers subscribed to adapt the transmission rate (or the number of layers subscribed
for a layered multicast session), or to arrange for the receiver to for a layered multicast session), or to arrange for the receiver to
leave the session if the loss rate is unacceptably high. leave the session.
8. SDP usage 8. Retransmission Payload Format MIME type registration
8.1 Introduction 8.1 Introduction
This section specifies how to describe the use of retransmission The following MIME subtype name and parameters are introduced in
with the Session Description Protocol (SDP), RFC 2327 [5]. As this document: "rtx", "rtx-time" and "apt".
specified in this document, the retransmission stream may be
conveyed in a separate RTP session, i.e. through session-
multiplexing, or in the same RTP session as the original stream
through SSRC-multiplexing.
The following attributes and parameters are introduced in this
document: "rtx", "rtx-time" and "apt".
The binding used for the retransmission stream to the payload type The binding used for the retransmission stream to the payload type
number is indicated by an rtpmap attribute. The MIME subtype name number is indicated by an rtpmap attribute. The MIME subtype name
used in the binding is "rtx", as specified in Section 11. 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.
An OPTIONAL payload format-specific parameter indicates the maximum
time a server will try to retransmit a packet.
The syntax is as follows: The syntax is as follows:
a=fmtp <number>: rtx-time=<rtx-time-val> a=fmtp:<number> apt=<apt-value>;rtx-time=<rtx-time-val>
where, where,
<number> indicates the dynamic payload type number assigned to
<number>: indicates the dynamic payload type number assigned to
the retransmission payload format in an rtpmap attribute. the retransmission payload format in an rtpmap attribute.
Rey/Leon/Miyazaki/Varsa/Hakenberg 11 <apt-value>: the value of the original stream payload type to
<rtx-time-val> indicates the time in milliseconds, measured 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 from the time a packet was first sent until the time the server
will stop trying to retransmit the packet. The absence of the will stop trying to retransmit the packet. The absence of the
rtx-time parameter for a retransmission stream means that the rtx-time parameter for a retransmission stream means that the
maximum retransmission time is not defined, but MAY be maximum retransmission time is not defined, but MAY be
negotiated by other means. negotiated by other means.
Additionally, a new SDP payload-format-specific parameter "apt" MUST 8.2 Registration of audio/rtx
be used to map the RTX payload type to the associated original
stream payload type as seen in the SDP description examples below.
If multiple payload types are used in the original stream, then
multiple "apt" parameters MUST be included to map each original
stream payload type to a different RTX payload type. The syntax of
this parameter is as follows:
a=fmtp <number>: apt=<apt-value> MIME type: audio
where,
<number> indicates the dynamic payload type number assigned to
the retransmission payload format.
<apt-value> indicates the original stream payload type to which
this retransmission stream payload type is associated.
Some SDP description examples are presented in the following MIME subtype: rtx
subsections.
8.2 Mapping MIME Parameters into SDP 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
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.
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.
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
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 The information carried in the MIME media type specification has a
specific mapping to fields in SDP [5], which is commonly used to specific mapping to fields in SDP [5], which is commonly used to
describe RTP sessions. When SDP is used to specify retransmissions describe RTP sessions. When SDP is used to specify retransmissions
for an RTP stream, the mapping is done as follows: for an RTP stream, the mapping is done as follows:
- The MIME types ("video"), ("audio") and ("text") go in the SDP - The MIME types ("video"), ("audio") and ("text") go in the SDP
"m=" as the media name. "m=" as the media name.
- The MIME subtype ("rtx") goes in SDP "a=rtpmap" as the encoding - 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 name. The RTP clock rate in "a=rtpmap" MUST be that of the
retransmission payload type. See Section 4 for details on this. retransmission payload type. See Section 4 for details on this.
- The AVPF profile-specific parameters "ack" and "nack" go in SDP - 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 "a=rtcp-fb". Several SDP "a=rtcp-fb" are used for several types of
feedback. See the AVPF profile [1] for details. feedback. See the AVPF profile [1] for details.
- The retransmission payload format-specific parameters "apt" and - The retransmission payload-format-specific parameters "apt" and
"rtx-time" go in the SDP "a=fmtp" as a semicolon separated list of "rtx-time" go in the SDP "a=fmtp" as a semicolon separated list of
parameter=value pairs. parameter=value pairs.
- Any remaining parameters go in the SDP "a=fmtp" attribute by - Any remaining parameters go in the SDP "a=fmtp" attribute by
copying them directly from the MIME media type string as a semicolon copying them directly from the MIME media type string as a semicolon
separated list of parameter=value pairs. separated list of parameter=value pairs.
Rey/Leon/Miyazaki/Varsa/Hakenberg 12
In the following sections some example SDP descriptions are In the following sections some example SDP descriptions are
presented. presented.
8.3 SDP description with session-multiplexing 8.7 SDP description with session-multiplexing
In the case of session-multiplexing, the SDP description contains In the case of session-multiplexing, the SDP description contains
one media specification "m" line per RTP session. The SDP MUST one media specification "m" line per RTP session. The SDP MUST
provide the grouping of the original and associated retransmission provide the grouping of the original and associated retransmission
sessions' "m" lines, using the Flow Identification (FID) semantics sessions' "m" lines, using the Flow Identification (FID) semantics
defined in RFC YYYY [6]. defined in RFC 3388 [6].
The following example specifies two original, AMR and MPEG-4, The following example specifies two original, AMR and MPEG-4,
streams on ports 49170 and 49174 and their corresponding streams on ports 49170 and 49174 and their corresponding
retransmission streams on ports 49172 and 49176, respectively: retransmission streams on ports 49172 and 49176, respectively:
v=0 v=0
o=mascha 2980675221 2980675778 IN IP4 at.home.ru o=mascha 2980675221 2980675778 IN IP4 at.home.ru
c=IN IP4 125.25.5.1 c=IN IP4 125.25.5.1
a=group:FID 1 2 a=group:FID 1 2
a=group:FID 3 4 a=group:FID 3 4
skipping to change at line 668 skipping to change at page 18, line 49
omitted in this special case only. omitted in this special case only.
This is illustrated in the following example, which is an SDP This is illustrated in the following example, which is an SDP
description for a single original MPEG-4 stream and its description for a single original MPEG-4 stream and its
corresponding retransmission session: corresponding retransmission session:
v=0 v=0
o=mascha 2980675221 2980675778 IN IP4 at.home.ru o=mascha 2980675221 2980675778 IN IP4 at.home.ru
c=IN IP4 125.25.5.1 c=IN IP4 125.25.5.1
m=video 49170 RTP/AVPF 96 m=video 49170 RTP/AVPF 96
Rey/Leon/Miyazaki/Varsa/Hakenberg 13
a=rtpmap:96 MP4V-ES/90000 a=rtpmap:96 MP4V-ES/90000
a=rtcp-fb:96 nack a=rtcp-fb:96 nack
a=fmtp:96 profile-level-id=8;config=01010000012000884006682C2090A21F a=fmtp:96 profile-level-id=8;config=01010000012000884006682C2090A21F
m=video 49172 RTP/AVPF 97 m=video 49172 RTP/AVPF 97
a=rtpmap:97 rtx/90000 a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96;rtx-time=3000 a=fmtp:97 apt=96;rtx-time=3000
8.4 SDP description with SSRC-multiplexing 8.8 SDP description with SSRC-multiplexing
The following is an example of an SDP description for an RTP video The following is an example of an SDP description for an RTP video
session using SSRC-multiplexing with similar parameters as in the session using SSRC-multiplexing with similar parameters as in the
single-session example above: single-session example above:
v=0 v=0
o=mascha 2980675221 2980675778 IN IP4 at.home.ru o=mascha 2980675221 2980675778 IN IP4 at.home.ru
c=IN IP4 125.25.5.1 c=IN IP4 125.25.5.1
m=video 49170 RTP/AVPF 96 97 m=video 49170 RTP/AVPF 96 97
a=rtpmap:96 MP4V-ES/90000 a=rtpmap:96 MP4V-ES/90000
a=rtcp-fb:96 nack a=rtcp-fb:96 nack
a=fmtp:96 profile-level-id=8;config=01010000012000884006682C2090A21F a=fmtp:96 profile-level-id=8;config=01010000012000884006682C2090A21F
a=rtpmap:97 rtx/90000 a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96;rtx-time=3000 a=fmtp:97 apt=96;rtx-time=3000
9. RTSP considerations 9. RTSP considerations
The Real-time Streaming Protocol (RTSP), RFC 2326 [7] is an The Real-time Streaming Protocol (RTSP), RFC 2326 [7] is an
application-level protocol for control over the delivery of data application-level protocol for control over the delivery of data
with real-time properties. This section looks at the issues involved with real-time properties. This section looks at the issues
in controlling RTP sessions that use retransmissions. involved in controlling RTP sessions that use retransmissions.
9.1 RTSP control with SSRC-multiplexing 9.1 RTSP control with SSRC-multiplexing
In the case of SSRC-multiplexing, the "m" line includes both In the case of SSRC-multiplexing, the "m" line includes both
original and retransmission payload types and has a single RTSP original and retransmission payload types and has a single RTSP
"control" attribute. The receiver uses the "m" line to request SETUP "control" attribute. The receiver uses the "m" line to request
and TEARDOWN of the whole media session. The RTP profile contained SETUP and TEARDOWN of the whole media session. The RTP profile
in the transport header MUST be the AVPF profile or another suitable contained in the transport header MUST be the AVPF profile or
profile allowing extended feedback. another suitable profile allowing extended feedback.
In order to control the sending of the session original media In order to control the sending of the session original media
stream, the receiver sends as usual PLAY and PAUSE requests to the 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- sender for the session. The RTP-info header that is used to set
specific parameters in the PLAY response MUST be set according to RTP-specific parameters in the PLAY response MUST be set according
the RTP information of the original stream. to the RTP information of the original stream.
When the receiver starts receiving the original stream, it can then When the receiver starts receiving the original stream, it can then
request retransmission through RTCP NACKs without additional RTSP request retransmission through RTCP NACKs without additional RTSP
signalling. signalling.
9.2 RTSP control with session-multiplexing 9.2 RTSP control with session-multiplexing
Rey/Leon/Miyazaki/Varsa/Hakenberg 14
In the case of session-multiplexing, each SDP "m" line has an RTSP In the case of session-multiplexing, each SDP "m" line has an RTSP
"control" attribute. Hence, when retransmission is used, both the "control" attribute. Hence, when retransmission is used, both the
original session and the retransmission have their own "control" original session and the retransmission have their own "control"
attributes. The receiver can associate the original session and the attributes. The receiver can associate the original session and the
retransmission session through the FID semantics as specified in retransmission session through the FID semantics as specified in
Section 8. Section 8.
The original and the retransmission streams are set up and torn down The original and the retransmission streams are set up and torn down
separately through their respective media "control" attribute. The separately through their respective media "control" attribute. The
RTP profile contained in the transport header MUST be the AVPF RTP profile contained in the transport header MUST be the AVPF
skipping to change at line 744 skipping to change at page 20, line 20
The RTSP presentation SHOULD support aggregate control and SHOULD The RTSP presentation SHOULD support aggregate control and SHOULD
contain a session level RTSP URL. The receiver SHOULD use aggregate contain a session level RTSP URL. The receiver SHOULD use aggregate
control for an original session and its associated retransmission control for an original session and its associated retransmission
session. Otherwise, there would need to be two different 'session- session. Otherwise, there would need to be two different 'session-
id' values, i.e. different values for the original and id' values, i.e. different values for the original and
retransmission sessions, and the sender would not know how to retransmission sessions, and the sender would not know how to
associate them. associate them.
The session-level "control" attribute is then used as usual to The session-level "control" attribute is then used as usual to
control the playing of the original stream. When the receiver starts control the playing of the original stream. When the receiver
receiving the original stream, it can then request retransmissions starts receiving the original stream, it can then request
through RTCP without additional RTSP signalling. retransmissions through RTCP without additional RTSP signalling.
9.3 Retransmission in pause state 9.3 RTSP control of the retransmission stream
Because of the nature of retransmissions, the sending of Because of the nature of retransmissions, the sending of
retransmission packets SHOULD NOT be controlled through RTSP PLAY retransmission packets SHOULD NOT be controlled through RTSP PLAY
and PAUSE requests. The PLAY and PAUSE requests should not affect and PAUSE requests. The PLAY and PAUSE requests should not affect
the retransmission stream. Retransmission packets are sent upon the retransmission stream. Retransmission packets are sent upon
receiver requests in the original RTCP stream, regardless of the receiver requests in the original RTCP stream, regardless of the
state. state.
9.4 Cache control 9.4 Cache control
skipping to change at line 774 skipping to change at page 20, line 50
In the case of SSRC-multiplexing, RTSP cannot specify independent In the case of SSRC-multiplexing, RTSP cannot specify independent
caching for the retransmission stream, because there is a single "m" caching for the retransmission stream, because there is a single "m"
line in SDP. Therefore, the implementer should take this fact into line in SDP. Therefore, the implementer should take this fact into
account when deciding whether to cache an SSRC-multiplexed session account when deciding whether to cache an SSRC-multiplexed session
or not. or not.
10. Implementation examples 10. Implementation examples
This document mandates only the sender and receiver behaviours that This document mandates only the sender and receiver behaviours that
are necessary for interoperability. In addition, certain algorithms, are necessary for interoperability. In addition, certain algorithms,
Rey/Leon/Miyazaki/Varsa/Hakenberg 15
such as rate control or buffer management when targeted at specific such as rate control or buffer management when targeted at specific
environments, may enhance the retransmission efficiency. environments, may enhance the retransmission efficiency.
This section gives an overview of different implementation options This section gives an overview of different implementation options
allowed within this specification. allowed within this specification.
The first example is a server-driven retransmission implementation. The first example describes a minimal receiver implementation. With
With this implementation, it is possible to retransmit lost RTP this implementation, it is possible to retransmit lost RTP packets,
packets, detect efficiently the loss of retransmissions and perform detect efficiently the loss of retransmissions and perform multiple
multiple retransmissions, if needed. Most of the necessary processing retransmissions, if needed. Most of the necessary processing is done
is done at the server. at the server.
The second example shows a receiver-driven implementation. It The second example shows how a receiver may implement additional
illustrates how a receiver may increase the retransmission enhancements that might help reduce sender buffer requirements and
efficiency. This implementation also increases the sender scalability optimise the retransmission efficiency
by reducing the load required at the sender.
The third example shows how retransmissions may be used in (small) The third example shows how retransmissions may be used in (small)
multicast groups in conjunction with layered encoding. It illustrates multicast groups in conjunction with layered encoding. It
that retransmissions and layered encoding may be complementary illustrates that retransmissions and layered encoding may be
techniques. complementary techniques.
10.1 A sender-driven retransmission example 10.1 A minimal receiver implementation example
This section gives an implementation example of multiple This section gives an example of an implementation supporting
retransmissions. The sender transmits the original data in RTP multiple retransmissions. The sender transmits the original data in
packets using the MPEG-4 video RTP payload format. RTP packets using the MPEG-4 video RTP payload format.
It is assumed that Generic NACK feedback messages are used, as per It is assumed that NACK feedback messages are used, as per
[1]. An SDP description example with SSRC-multiplexing is given [1]. An SDP description example with SSRC-multiplexing is given
below: below:
v=0 v=0
o=mascha 2980675221 2980675778 IN IP4 at.home.ru o=mascha 2980675221 2980675778 IN IP4 at.home.ru
c=IN IP4 125.25.5.1 c=IN IP4 125.25.5.1
m=video 49170 RTP/AVPF 96 97 m=video 49170 RTP/AVPF 96 97
a=rtpmap:96 MP4V-ES/90000 a=rtpmap:96 MP4V-ES/90000
a=rtcp-fb:96 nack a=rtcp-fb:96 nack
a=rtpmap:97 rtx/90000 a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96;rtx-time=3000 a=fmtp:97 apt=96;rtx-time=3000
The format-specific parameter "rtx-time" would indicate that the The format-specific parameter "rtx-time" indicates that the server
server will buffer the sent packets in a retransmission will buffer the sent packets in a retransmission buffer for 3.0
buffer for 3.0 seconds, after which the packets are deleted from seconds, after which the packets are deleted from the retransmission
the retransmission buffer and will never be sent again. buffer and will never be sent again.
In this implementation example, the required RTP receiver processing In this implementation example, the required RTP receiver processing
to handle retransmission is very limited. The receiver detects packet to handle retransmission is kept to a minimum. The receiver detects
loss from the gaps observed in the received sequence numbers. It packet loss from the gaps observed in the received sequence numbers.
signals lost packets to the sender through RTCP NACK messages as It signals lost packets to the sender through NACKs as defined in the
defined in the AVPF profile [1]. The receiver should take into AVPF profile [1]. The receiver should take into account the
account the signalled sender retransmission buffer length in order to signalled sender retransmission buffer length in order to dimension
dimension its own reception buffer. It should also derive from the its own reception buffer. It should also derive from the buffer
length the maximum number of times the retransmission of a packet can
Rey/Leon/Miyazaki/Varsa/Hakenberg 16 be requested.
buffer length the maximum number of times retransmission of a packet
can be requested.
The sender should retransmit the packets selectively, i.e. it should The sender should retransmit the packets selectively, i.e. it should
choose whether to retransmit a requested packet depending on the choose whether to retransmit a requested packet depending on the
packet importance, the observed QoS and congestion state of the packet importance, the observed QoS and congestion state of the
network connection to the receiver. Obviously, the sender processing network connection to the receiver. Obviously, the sender processing
increases with the number of receivers as state information and increases with the number of receivers as state information and
processing load must be allocated to each receiver. processing load must be allocated to each receiver.
10.2 A receiver-driven retransmission example 10.2 An enhanced receiver implementation example
The receiver may have more accurate information than the sender about The receiver may have more accurate information than the sender about
the current network QoS such as available bandwidth, packet loss the current network QoS such as available bandwidth, packet loss
rate, delay and jitter. In addition, other receiver-specific rate, delay and jitter. In addition, other receiver-specific
parameters like buffer level, estimated importance of the lost packet parameters such as buffer level, estimated importance of the lost
and application level QoS may be used by the receiver to make a more packet and application level QoS may be used by the receiver to make
efficient use of RTP retransmission through selective requests. 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 Furthermore, a receiver may acknowledge the received packets. This
can be done by sending ACK messages, as per [1]. Upon receiving an can be done by sending ACKs, as per [1]. Upon receiving an ACK, the
ACK, the sender may delete all the acknowledged packets from its sender may delete all the acknowledged packets from its
retransmission buffer. Note that this would also require only limited retransmission buffer. Note that this would also require only
increase in the required RTCP bandwidth as long as ACK packets are limited increase in the required RTCP bandwidth as long as ACK
sent seldom enough. With the receiver-driven retransmission packets are sent seldom enough.
implementation, processing load and buffer requirements at the sender
are decreased, allowing greater sender scalability.
Note that choosing between the sender-driven implementation and the This implementation may help reduce buffer requirements at the sender
receiver-driven implementation does not imply any changes in the SDP and optimise the performance of the implementation by using selective
description, except for the need to signal the use of ACK RTCP requests.
packets, by means of an additional SDP "a=rtcp-fb" line, as follows:
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 v=0
o=mascha 2980675221 2980675778 IN IP4 at.home.ru o=mascha 2980675221 2980675778 IN IP4 at.home.ru
c=IN IP4 125.25.5.1 c=IN IP4 125.25.5.1
m=video 49170 RTP/AVPF 96 97 m=video 49170 RTP/AVPF 96 97
a=rtpmap:96 MP4V-ES/90000 a=rtpmap:96 MP4V-ES/90000
a=rtcp-fb:96 nack a=rtcp-fb:96 nack
a=rtcp-fb:96 ack a=rtcp-fb:96 ack
a=rtpmap:97 rtx/90000 a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96;rtx-time=3000 a=fmtp:97 apt=96;rtx-time=3000
10.3 Retransmissions with Layered Transmissions 10.3 Retransmission of Layered Encoded Media in Multicast
This section shows how to combine retransmissions with layered This section shows how to combine retransmissions with layered
encoding. Note that the retransmission framework is not intended as a encoding in multicast sessions. Note that the retransmission
complete solution to reliable multicast. Refer to RFC 2887 [11], for framework is not intended as a complete solution to reliable
an overview of the problems related with reliable multicast multicast. Refer to RFC 2887 [10], for an overview of the problems
transmission. related with reliable multicast transmission.
Packets of different importance are sent in different RTP sessions. Packets of different importance are sent in different RTP sessions.
The retransmission streams corresponding to the different layers can The retransmission streams corresponding to the different layers can
Rey/Leon/Miyazaki/Varsa/Hakenberg 17
themselves be seen as different retransmission layers. The relative themselves be seen as different retransmission layers. The relative
importance of the different retransmission streams should reflect the importance of the different retransmission streams should reflect the
relative importance of the different original streams. relative importance of the different original streams.
A retransmission stream may be sent in the same RTP session as its In multicast, SSRC-multiplexing of the original and retransmission
corresponding original layer through SSRC multiplexing or in a streams is not allowed as per Section 5.3 of this document. For this
different RTP session through session multiplexing. reason, the retransmission stream(s) MUST be sent in different RTP
session(s) using session-multiplexing.
An SDP description example for SSRC-multiplexing is given below: An SDP description example of multicast retransmissions for layered
encoded media is given below:
c=IN IP4 224.2.1.1/127/3 c=IN IP4 224.2.1.1/127/3
m=video 8000 RTP/AVPF 98 99 m=video 8000 RTP/AVPF 98
a=rtpmap:98 MP4V-ES/90000 a=rtpmap:98 MP4V-ES/90000
a=rtcp-fb:98 nack 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=rtpmap:99 rtx/90000
a=fmtp:99 apt=98;rtx-time=3000 a=fmtp:99 apt=98;rtx-time=3000
The server and the receiver may implement the retransmission methods The server and the receiver may implement the retransmission methods
illustrated in the previous examples. In addition, they may choose to illustrated in the previous examples. In addition, they may choose
request and retransmit a lost packet depending on the layer it to request and retransmit a lost packet depending on the layer it
belongs to. belongs to.
11. IANA considerations 11. IANA considerations
11.1 Registration of audio/rtx A new MIME subtype name, "rtx", has been registered. An additional
REQUIRED parameter, "apt", and an OPTIONAL parameter, "rtx-time",
MIME type: audio are defined. See Section 8 for details.
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/Leon/Miyazaki/Varsa/Hakenberg 18
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
11.2 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 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
Rey/Leon/Miyazaki/Varsa/Hakenberg 19
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
11.3 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
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
Rey/Leon/Miyazaki/Varsa/Hakenberg 20
Author/Change controller:
Jose Rey
David Leon
IETF AVT WG
11.4 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
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
Rey/Leon/Miyazaki/Varsa/Hakenberg 21
12. Security considerations 12. Security considerations
Applications utilising encryption SHOULD encrypt both the original Applications utilising encryption SHOULD encrypt both the original
and the retransmission stream. Old keys will likely need to be and the retransmission stream. Old keys will likely need to be
cached so that when the keys change for the original stream, the old 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 key is used until it is determined that the key has changed on the
retransmission packets as well. retransmission packets as well.
The use of the same key for the retransmitted stream and the The use of the same key for the retransmitted stream and the
original stream may lead to security problems, e.g. two-time pads. original stream may lead to security problems, e.g. two-time pads.
This sharing has to be evaluated towards the chosen security This sharing has to be evaluated towards the chosen security
protocol and security algorithms, e.g. the Secure Real-Time protocol and security algorithms.
Transport Protocol (SRTP) RFC UUUU [8] establishes requirements for
avoiding the two-time pad.
RTP recommends that the initial RTP timestamp SHOULD be random to RTP recommends that the initial RTP timestamp SHOULD be random to
secure the stream against known plain text attacks. This payload secure the stream against known plain text attacks. This payload
format does not follow this recommendation as the initial timestamp format does not follow this recommendation as the initial timestamp
will be the media timestamp of the first retransmitted packet. will be the media timestamp of the first retransmitted packet.
However, since the initial timestamp of the original stream is However, since the initial timestamp of the original stream is
itself random, if the original stream is encrypted, the first itself random, if the original stream is encrypted, the first
retransmitted packet timestamp would also be random to an attacker. retransmitted packet timestamp would also be random to an attacker.
Therefore, security would not be compromised. Therefore, confidentiality would not be compromised.
Congestion control considerations with the use of retransmission are Congestion control considerations with the use of retransmission are
dealt with in Section 7 of this document. dealt with in Section 7 of this document.
Any other security considerations of the profile under which the Any other security considerations of the profile under which the
retransmission scheme is used should be applied. 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 13. Acknowledgements
We would like to express our gratitude to Carsten Burmeister for his We would like to express our gratitude to Carsten Burmeister for his
participation in the development of this document. Our thanks also participation in the development of this document. Our thanks also
go to Koichi Hata, Colin Perkins, Stephen Casner, Magnus Westerlund, go to Koichi Hata, Colin Perkins, Stephen Casner, Magnus Westerlund,
Go Hori and Rahul Agarwal for their helpful comments. Go Hori and Rahul Agarwal for their helpful comments.
14. References 14. References
14.1 Normative References 14.1 Normative References
1 J. Ott, S. Wenger, N. Sato, C. Burmeister, J. Rey, "Extended RTP 1 J. Ott, S. Wenger, N. Sato, C. Burmeister, J. Rey, "Extended RTP
profile for RTCP-based feedback", RFC VVVV, September 2002. 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 2 S. Bradner, "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997 Levels", BCP 14, RFC 2119, March 1997
3 H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson, "RTP: A 3 H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson, "RTP: A
Transport Protocol for Real-Time Applications", RFC WWWW, May Transport Protocol for Real-Time Applications", draft-ietf-avt-
2002. rtp-new-11.txt, May 2002.
Rey/Leon/Miyazaki/Varsa/Hakenberg 22 4 S. Casner, "SDP bandwidth modifiers for RTCP bandwidth", draft-
4 S. Casner, "SDP bandwidth modifiers for RTCP bandwidth", RFC ZZZZ, ietf-avt-rtcp-bw-05.txt, May 2002.
May 2002.
5 M. Handley, V. Jacobson, "SDP: Session Description Protocol", RFC 5 M. Handley, V. Jacobson, "SDP: Session Description Protocol", RFC
2327, April 1998. 2327, April 1998.
6 G. Camarillo,J. Holler, G. AP. Eriksson, "Grouping of media lines 6 G. Camarillo,J. Holler, G. AP. Eriksson, "Grouping of media lines
in SDP", RFC YYYY, February 2002. in the Session Description Protocol (SDP)", RFC 3388, December
2002.
7 H. Schulzrinne, A. Rao, R. Lanphier, "Real Time Streaming Protocol 7 H. Schulzrinne, A. Rao, R. Lanphier, "Real Time Streaming Protocol
(RTSP)", RFC 2326, April 1998. (RTSP)", RFC 2326, April 1998.
8 M. Baugher, D. A. McGrew, D. Oran, R. Blom, E. Carrara, M. 14.2 Informative References
Naslund, K. Norrman, "The Secure Real-Time Transport Protocol",
RFC UUUU, June 2002.
14.2 Non-normative References
9 C. Perkins, O. Hodson, "Options for Repair of Streaming Media", 8 C. Perkins, O. Hodson, "Options for Repair of Streaming Media",
RFC 2354, June 1998. RFC 2354, June 1998.
10 J. Hellstrom, "RTP for conversational text", RFC 2793, May 2000 9 G. Hellstrom, "RTP for conversational text", RFC 2793, May 2000
11 M. Handley, et al., "The Reliable Multicast Design Space for Bulk 10 M. Handley, et al., "The Reliable Multicast Design Space for Bulk
Data Transfer", RFC 2887, August 2000. 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 Author's Addresses
Jose Rey rey@panasonic.de Jose Rey rey@panasonic.de
Panasonic European Laboratories GmbH Panasonic European Laboratories GmbH
Monzastr. 4c Monzastr. 4c
D-63225 Langen, Germany D-63225 Langen, Germany
Phone: +49-6103-766-134 Phone: +49-6103-766-134
Fax: +49-6103-766-166 Fax: +49-6103-766-166
David Leon david.leon@nokia.com David Leon david.leon@nokia.com
skipping to change at line 1202 skipping to change at page 25, line 46
Akihiro Miyazaki akihiro@isl.mei.co.jp Akihiro Miyazaki akihiro@isl.mei.co.jp
Core Software Development Center Core Software Development Center
Corporate Software Development Division Corporate Software Development Division
Matsushita Electric Industrial Co., Ltd. Matsushita Electric Industrial Co., Ltd.
1006 Kadoma, Kadoma City, Osaka 571-8501, Japan 1006 Kadoma, Kadoma City, Osaka 571-8501, Japan
Phone: +81-6-6900-9192 Phone: +81-6-6900-9192
Fax: +81-6-6900-9193 Fax: +81-6-6900-9193
Viktor Varsa viktor.varsa@nokia.com Viktor Varsa viktor.varsa@nokia.com
Nokia Research Center Nokia Research Center
Rey/Leon/Miyazaki/Varsa/Hakenberg 23
6000 Connection Drive 6000 Connection Drive
Irving, TX. USA Irving, TX. USA
Phone: 1-972-374-1861 Phone: 1-972-374-1861
Rolf Hakenberg hakenberg@panasonic.de Rolf Hakenberg hakenberg@panasonic.de
Panasonic European Laboratories GmbH Panasonic European Laboratories GmbH
Monzastr. 4c Monzastr. 4c
D-63225 Langen, Germany D-63225 Langen, Germany
Phone: +49-6103-766-162 Phone: +49-6103-766-162
Fax: +49-6103-766-166 Fax: +49-6103-766-166
Full Copyright Statement
Rey/Leon/Miyazaki/Varsa/Hakenberg 24 "Copyright (C) The Internet Society (date). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.
The limited permissions granted above are perpetual and will
not be revoked by the Internet Society or its successors or
assigns.
This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
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HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
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