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rfc:rfc4588

Network Working Group J. Rey Request for Comments: 4588 Panasonic Category: Standards Track D. Leon

                                                            Consultant
                                                           A. Miyazaki
                                                             Panasonic
                                                              V. Varsa
                                                                 Nokia
                                                          R. Hakenberg
                                                             Panasonic
                                                             July 2006
                 RTP Retransmission Payload Format

Status of This Memo

 This document specifies an Internet standards track protocol for the
 Internet community, and requests discussion and suggestions for
 improvements.  Please refer to the current edition of the "Internet
 Official Protocol Standards" (STD 1) for the standardization state
 and status of this protocol.  Distribution of this memo is unlimited.

Copyright Notice

 Copyright (C) The Internet Society (2006).

Abstract

 RTP retransmission is an effective packet loss recovery technique for
 real-time applications with relaxed delay bounds.  This document
 describes an RTP payload format for performing retransmissions.
 Retransmitted RTP packets are sent in a separate stream from the
 original RTP stream.  It is assumed that feedback from receivers to
 senders is available.  In particular, it is assumed that Real-time
 Transport Control Protocol (RTCP) feedback as defined in the extended
 RTP profile for RTCP-based feedback (denoted RTP/AVPF) is available
 in this memo.

Rey, et al. Standards Track [Page 1] RFC 4588 RTP Retransmission Payload Format July 2006

Table of Contents

 1. Introduction ....................................................3
 2. Terminology .....................................................3
 3. Requirements and Design Rationale for a Retransmission Scheme ...4
    3.1. Multiplexing Scheme Choice .................................6
 4. Retransmission Payload Format ...................................7
 5. Association of Retransmission and Original Streams ..............9
    5.1. Retransmission Session Sharing .............................9
    5.2. CNAME Use ..................................................9
    5.3. Association at the Receiver ................................9
 6. Use with the Extended RTP Profile for RTCP-based Feedback ......11
    6.1. RTCP at the Sender ........................................11
    6.2. RTCP Receiver Reports .....................................11
    6.3. Retransmission Requests ...................................12
    6.4. Timing Rules ..............................................13
 7. Congestion Control .............................................13
 8. Retransmission Payload Format MIME Type Registration ...........15
    8.1. Introduction ..............................................15
    8.2. Registration of audio/rtx .................................16
    8.3. Registration of video/rtx .................................17
    8.4. Registration of text/rtx ..................................18
    8.5. Registration of application/rtx ...........................19
    8.6. Mapping to SDP ............................................20
    8.7. SDP Description with Session-Multiplexing .................20
    8.8. SDP Description with SSRC-Multiplexing ....................21
 9. RTSP Considerations ............................................22
    9.1. RTSP Control with SSRC-Multiplexing .......................22
    9.2. RTSP Control with Session-Multiplexing ....................22
    9.3. RTSP Control of the Retransmission Stream .................23
    9.4. Cache Control .............................................23
 10. Implementation Examples .......................................23
    10.1. A Minimal Receiver Implementation Example ................24
    10.2. Retransmission of Layered Encoded Media in Multicast .....25
 11. IANA Considerations ...........................................26
 12. Security Considerations .......................................26
 13. Acknowledgements ..............................................27
 14. References ....................................................27
    14.1. Normative References .....................................27
    14.2. Informative References ...................................28
 Appendix A. How to Control the Number of Rtxs. per Packet .........29

Rey, et al. Standards Track [Page 2] RFC 4588 RTP Retransmission Payload Format July 2006

1. Introduction

 Packet losses between an RTP sender and receiver may significantly
 degrade the quality of the received media.  Several techniques, such
 as forward error correction (FEC), retransmissions, or interleaving,
 may be considered to increase packet loss resiliency.  RFC 2354 [8]
 discusses the different options.
 When choosing a repair technique for a particular application, the
 tolerable latency of the application has to be taken into account.
 In the case of multimedia conferencing, the end-to-end delay has to
 be at most a few hundred milliseconds in order to guarantee
 interactivity, which usually excludes the use of retransmission.
 With sufficient latency, the efficiency of the repair scheme can be
 increased.  The sender may use the receiver feedback in order to
 react to losses before their playout time at the receiver.
 In the case of multimedia streaming, the user can tolerate an initial
 latency as part of the session set-up and thus an end-to-end delay of
 several seconds may be acceptable.  RTP retransmission as defined in
 this document is targeted at such applications.
 Furthermore, the RTP retransmission method defined herein is
 applicable to unicast and (small) multicast groups.  The present
 document defines a payload format for retransmitted RTP packets and
 provides protocol rules for the sender and the receiver involved in
 retransmissions.
 This retransmission payload format was designed for use with the
 extended RTP profile for RTCP-based feedback, AVPF [1].  It may also
 be used with other RTP profiles defined in the future.
 The AVPF profile allows for more frequent feedback and for early
 feedback.  It defines a general-purpose feedback message, i.e., NACK,
 as well as codec and application-specific feedback messages.  See [1]
 for details.

2. Terminology

 The following terms are used in this document:
 CSRC: contributing source.  See [3].
 Original packet: an RTP packet that carries user data sent for the
 first time by an RTP sender.
 Original stream: the RTP stream of original packets.

Rey, et al. Standards Track [Page 3] RFC 4588 RTP Retransmission Payload Format July 2006

 Retransmission packet: an RTP packet that is to be used by the
 receiver instead of a lost original packet.  Such a retransmission
 packet is said to be associated with the original RTP packet.
 Retransmission request: a means by which an RTP receiver is able to
 request that the RTP sender should send a retransmission packet for a
 given original packet.  Usually, an RTCP NACK packet as specified in
 [1] is used as retransmission request for lost packets.
 Retransmission stream: the stream of retransmission packets
 associated with an original stream.
 Session-multiplexing: scheme by which the original stream and the
 associated retransmission stream are sent into two different RTP
 sessions.
 SSRC: synchronization source.  See [3].
 SSRC-multiplexing: scheme by which the original stream and the
 retransmission stream are sent in the same RTP session with different
 SSRC values.
 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
 document are to be interpreted as described in RFC 2119 [2].

3. Requirements and Design Rationale for a Retransmission Scheme

 The use of retransmissions in RTP as a repair method for streaming
 media is appropriate in those scenarios with relaxed delay bounds and
 where full reliability is not a requirement.  More specifically, RTP
 retransmission allows one to trade off reliability vs. delay; i.e.,
 the endpoints may give up retransmitting a lost packet after a given
 buffering time has elapsed.  Unlike TCP, there is thus no head-of-
 line blocking caused by RTP retransmissions.  The implementer should
 be aware that in cases where full reliability is required or higher
 delay and jitter can be tolerated, TCP or other transport options
 should be considered.
 The RTP retransmission scheme defined in this document is designed to
 fulfill the following set of requirements:
 1. It must not break general RTP and RTCP mechanisms.
 2. It must be suitable for unicast and small multicast groups.
 3. It must work with mixers and translators.
 4. It must work with all known payload types.
 5. It must not prevent the use of multiple payload types in a
    session.

Rey, et al. Standards Track [Page 4] RFC 4588 RTP Retransmission Payload Format July 2006

 6. In order to support the largest variety of payload formats, the
    RTP receiver must be able to derive how many and which RTP packets
    were lost as a result of a gap in received RTP sequence numbers.
    This requirement is referred to as sequence number preservation.
    Without such a requirement, it would be impossible to use
    retransmission with payload formats, such as conversational text
    [9] or most audio/video streaming applications, that use the RTP
    sequence number to detect lost packets.
 When designing a solution for RTP retransmission, several approaches
 may be considered for the multiplexing of the original RTP packets
 and the retransmitted RTP packets.
 One approach may be to retransmit the RTP packet with its original
 sequence number and send original and retransmission packets in the
 same RTP stream.  The retransmission packet would then be identical
 to the original RTP packet, i.e., the same header (and thus same
 sequence number) and the same payload.  However, such an approach is
 not acceptable because it would corrupt the RTCP statistics.  As a
 consequence, requirement 1 would not be met.  Correct RTCP statistics
 require that for every RTP packet within the RTP stream, the sequence
 number be increased by one.
 Another approach may be to multiplex original RTP packets and
 retransmission packets in the same RTP stream using different payload
 type values.  With such an approach, the original packets and the
 retransmission packets would share the same sequence number space.
 As a result, the RTP receiver would not be able to infer how many and
 which original packets (which sequence numbers) were lost.
 In other words, this approach does not satisfy the sequence number
 preservation requirement (requirement 6).  This in turn implies that
 requirement 4 would not be met.  Interoperability with mixers and
 translators would also be more difficult if they did not understand
 this new retransmission payload type in a sender RTP stream.  For
 these reasons, a solution based on payload type multiplexing of
 original packets and retransmission packets in the same RTP stream is
 excluded.
 Finally, the original and retransmission packets may be sent in two
 separate streams.  These two streams may be multiplexed either by
 sending them in two different sessions , i.e., session-multiplexing,
 or in the same session using different SSRC values, i.e., SSRC-
 multiplexing.  Since original and retransmission packets carry media
 of the same type, the objections in Section 5.2 of RTP [3] to RTP
 multiplexing do not apply in this case.

Rey, et al. Standards Track [Page 5] RFC 4588 RTP Retransmission Payload Format July 2006

 Mixers and translators may process the original stream and simply
 discard the retransmission stream if they are unable to utilise it.
 On the other hand, sending the original and retransmission packets in
 two separate streams does not alone satisfy requirements 1 and 6.
 For this purpose, this document includes the original sequence number
 in the retransmitted packets.
 In this manner, using two separate streams satisfies all the
 requirements listed in this section.

3.1. Multiplexing Scheme Choice

 Session-multiplexing and SSRC-multiplexing have different pros and
 cons:
 Session-multiplexing is based on sending the retransmission stream in
 a different RTP session (as defined in RTP [3]) from that of the
 original stream; i.e., the original and retransmission streams are
 sent to different network addresses and/or port numbers.  Having a
 separate session allows more flexibility.  In multicast, using two
 separate sessions for the original and the retransmission streams
 allows a receiver to choose whether or not to subscribe to the RTP
 session carrying the retransmission stream.  The original session may
 also be single-source multicast while separate unicast sessions are
 used to convey retransmissions to each of the receivers, which as a
 result will receive only the retransmission packets they request.
 The use of separate sessions also facilitates differential treatment
 by the network and may simplify processing in mixers, translators,
 and packet caches.
 With SSRC-multiplexing, a single session is needed for the original
 and the retransmission streams.  This allows streaming servers and
 middleware that are involved in a high number of concurrent sessions
 to minimise their port usage.
 This retransmission payload format allows both session-multiplexing
 and SSRC-multiplexing for unicast sessions.  From an implementation
 point of view, there is little difference between the two approaches.
 Hence, in order to maximise interoperability, both multiplexing
 approaches SHOULD be supported by senders and receivers.  For
 multicast sessions, session-multiplexing MUST be used because the
 association of the original stream and the retransmission stream is
 problematic if SSRC-multiplexing is used with multicast sessions(see
 Section 5.3 for motivation).

Rey, et al. Standards Track [Page 6] RFC 4588 RTP Retransmission Payload Format July 2006

4. Retransmission Payload Format

 The format of a retransmission packet is shown below:
  0                   1                   2                   3
  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |                         RTP Header                            |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |            OSN                |                               |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               |
 |                  Original RTP Packet Payload                  |
 |                                                               |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 The RTP header usage is as follows:
 In the case of session-multiplexing, the same SSRC value MUST be used
 for the original stream and the retransmission stream.  In the case
 of an SSRC collision in either the original session or the
 retransmission session, the RTP specification requires that an RTCP
 BYE packet MUST be sent in the session where the collision happened.
 In addition, an RTCP BYE packet MUST also be sent for the associated
 stream in its own session.  After a new SSRC identifier is obtained,
 the SSRC of both streams MUST be set to this value.
 In the case of SSRC-multiplexing, two different SSRC values MUST be
 used for the original stream and the retransmission stream as
 required by RTP.  If an SSRC collision is detected for either the
 original stream or the retransmission stream, the RTP specification
 requires that an RTCP BYE packet MUST be sent for this stream.  An
 RTCP BYE packet MUST NOT be sent for the associated stream.
 Therefore, only the stream that experienced SSRC collision MUST
 choose a new SSRC value.  Refer to Section 5.3 for the implications
 on the original stream and retransmission stream SSRC association at
 the receiver.
 For either multiplexing scheme, the sequence number has the standard
 definition; i.e., it MUST be one higher than the sequence number of
 the preceding packet sent in the retransmission stream.
 The retransmission packet timestamp MUST be set to the original
 timestamp, i.e., to the timestamp of the original packet.  As a
 consequence, the initial RTP timestamp for the first packet of the
 retransmission stream is not random but equal to the original
 timestamp of the first packet that is retransmitted.  See the
 Security Considerations section in this document for security
 implications.

Rey, et al. Standards Track [Page 7] RFC 4588 RTP Retransmission Payload Format July 2006

 Implementers have to be aware that the RTCP jitter value for the
 retransmission stream does not reflect the actual network jitter
 since there could be little correlation between the time a packet is
 retransmitted and its original timestamp.
 The payload type is dynamic.  If multiple payload types using
 retransmission are present in the original stream, then for each of
 these, a dynamic payload type MUST be mapped to the retransmission
 payload format.  See Section 8.1 for the specification of how the
 mapping between original and retransmission payload types is done
 with Session Description Protocol (SDP).
 As the retransmission packet timestamp carries the original media
 timestamp, the timestamp clockrate used by the retransmission payload
 type MUST be the same as the one used by the associated original
 payload type.  Therefore, if an RTP stream carries payload types of
 different clockrates, this will also be the case for the associated
 retransmission stream.  Note that an RTP stream does not usually
 carry payload types of different clockrates.
 The payload of the RTP retransmission packet comprises the
 retransmission payload header followed by the payload of the original
 RTP packet.  The length of the retransmission payload header is 2
 octets.  This payload header contains only one field, OSN (original
 sequence number), which MUST be set to the sequence number of the
 associated original RTP packet.  The original RTP packet payload,
 including any possible payload headers specific to the original
 payload type, MUST be placed right after the retransmission payload
 header.
 For payload formats that support encoding at multiple rates, instead
 of retransmitting the same payload as the original RTP packet the
 sender MAY retransmit the same data encoded at a lower rate.  This
 aims at limiting the bandwidth usage of the retransmission stream.
 When doing so, the sender MUST ensure that the receiver will still be
 able to decode the payload of the already sent original packets that
 might have been encoded based on the payload of the lost original
 packet.  In addition, if the sender chooses to retransmit at a lower
 rate, the values in the payload header of the original RTP packet may
 no longer apply to the retransmission packet and may need to be
 modified in the retransmission packet to reflect the change in rate.
 The sender SHOULD trade off the decrease in bandwidth usage with the
 decrease in quality caused by resending at a lower rate.
 If the original RTP header carried any profile-specific extensions,
 the retransmission packet SHOULD include the same extensions
 immediately following the fixed RTP header as expected by
 applications running under this profile.  In this case, the

Rey, et al. Standards Track [Page 8] RFC 4588 RTP Retransmission Payload Format July 2006

 retransmission payload header MUST be placed after the profile-
 specific extensions.
 If the original RTP header carried an RTP header extension, the
 retransmission packet SHOULD carry the same header extension.  This
 header extension MUST be placed right after the fixed RTP header, as
 specified in RTP [3].  In this case, the retransmission payload
 header MUST be placed after the header extension.
 If the original RTP packet contained RTP padding, that padding MUST
 be removed before constructing the retransmission packet.  If padding
 of the retransmission packet is needed, padding MUST be performed as
 with any RTP packets and the padding bit MUST be set.
 The marker bit (M), the CSRC count (CC), and the CSRC list of the
 original RTP header MUST be copied "as is" into the RTP header of the
 retransmission packet.

5. Association of Retransmission and Original Streams

5.1. Retransmission Session Sharing

 In the case of session-multiplexing, a retransmission session MUST
 map to exactly one original session; i.e., the same retransmission
 session cannot be used for different original sessions.
 If retransmission session sharing were allowed, it would be a problem
 for receivers, since they would receive retransmissions for original
 sessions they might not have joined.  For example, a receiver wishing
 to receive only audio would receive also retransmitted video packets
 if an audio and video session shared the same retransmission session.

5.2. CNAME Use

 In both the session-multiplexing and the SSRC-multiplexing cases, a
 sender MUST use the same RTCP CNAME [3] for an original stream and
 its associated retransmission stream.

5.3. Association at the Receiver

 A receiver receiving multiple original and retransmission streams
 needs to associate each retransmission stream with its original
 stream.  The association is done differently depending on whether
 session-multiplexing or SSRC-multiplexing is used.
 If session-multiplexing is used, the receiver associates the two
 streams having the same SSRC in the two sessions.  Note that the
 payload type field cannot be used to perform the association as

Rey, et al. Standards Track [Page 9] RFC 4588 RTP Retransmission Payload Format July 2006

 several media streams may have the same payload type value.  The two
 sessions are themselves associated out-of-band.  See Section 8 for
 how the grouping of the two sessions is done with SDP.
 If SSRC-multiplexing is used, the receiver should first of all look
 for two streams that have the same CNAME in the session.  In some
 cases, the CNAME may not be enough to determine the association as
 multiple original streams in the same session may share the same
 CNAME.  For example, there can be in the same video session multiple
 video streams mapping to different SSRCs and still using the same
 CNAME and possibly the same payload type (PT) values.  Each (or some)
 of these streams may have an associated retransmission stream.
 In this case, in order to find out the association between original
 and retransmission streams having the same CNAME, the receiver SHOULD
 behave as follows.
 The association can generally be resolved when the receiver receives
 a retransmission packet matching a retransmission request that had
 been sent earlier.  Upon reception of a retransmission packet whose
 original sequence number has been previously requested, the receiver
 can derive that the SSRC of the retransmission packet is associated
 to the sender SSRC from which the packet was requested.
 However, this mechanism might fail if there are two outstanding
 requests for the same packet sequence number in two different
 original streams of the session.  Note that since the initial packet
 sequence numbers are random, the probability of having two
 outstanding requests for the same packet sequence number would be
 very small.  Nevertheless, in order to avoid ambiguity in the unicast
 case, the receiver MUST NOT have two outstanding requests for the
 same packet sequence number in two different original streams before
 the association is resolved.  In multicast, this ambiguity cannot be
 completely avoided, because another receiver may have requested the
 same sequence number from another stream.  Therefore, SSRC-
 multiplexing MUST NOT be used in multicast sessions.
 If the receiver discovers that two senders are using the same SSRC or
 if it receives an RTCP BYE packet, it MUST stop requesting
 retransmissions for that SSRC.  Upon reception of original RTP
 packets with a new SSRC, the receiver MUST perform the SSRC
 association again as described in this section.

Rey, et al. Standards Track [Page 10] RFC 4588 RTP Retransmission Payload Format July 2006

6. Use with the Extended RTP Profile for RTCP-based Feedback

 This section gives general hints for the usage of this payload format
 with the extended RTP profile for RTCP-based feedback, denoted AVPF
 [1].  Note that the general RTCP send and receive rules and the RTCP
 packet format as specified in RTP apply, except for the changes that
 the AVPF profile introduces.  In short, the AVPF profile relaxes the
 RTCP timing rules and specifies additional general-purpose RTCP
 feedback messages.  See [1] for details.

6.1. RTCP at the Sender

 In the case of session-multiplexing, Sender Report (SR) packets for
 the original stream are sent in the original session and SR packets
 for the retransmission stream are sent in the retransmission session
 according to the rules of RTP.
 In the case of SSRC-multiplexing, SR packets for both original and
 retransmission streams are sent in the same session according to the
 rules of RTP.  The original and retransmission streams are seen, as
 far as the RTCP bandwidth calculation is concerned, as independent
 senders belonging to the same RTP session and are thus equally
 sharing the RTCP bandwidth assigned to senders.
 Note that in both cases, session- and SSRC-multiplexing, BYE packets
 MUST still be sent for both streams as specified in RTP.  In other
 words, it is not enough to send BYE packets for the original stream
 only.

6.2. RTCP Receiver Reports

 In the case of session-multiplexing, the receiver will send report
 blocks for the original stream and the retransmission stream in
 separate Receiver Report (RR) packets belonging to separate RTP
 sessions.  RR packets reporting on the original stream are sent in
 the original RTP session while RR packets reporting on the
 retransmission stream are sent in the retransmission session.  The
 RTCP bandwidth for these two sessions may be chosen independently
 (e.g., through RTCP bandwidth modifiers [4]).
 In the case of SSRC-multiplexing, the receiver sends report blocks
 for the original and the retransmission streams in the same RR packet
 since there is a single session.

Rey, et al. Standards Track [Page 11] RFC 4588 RTP Retransmission Payload Format July 2006

6.3. Retransmission Requests

 The NACK feedback message format defined in the AVPF profile SHOULD
 be used by receivers to send retransmission requests.  Whether or not
 a receiver chooses to request a packet is an implementation issue.
 An actual receiver implementation should take into account such
 factors as the tolerable application delay, the network environment,
 and the media type.
 The receiver should generally assess whether the retransmitted packet
 would still be useful at the time it is received.  The timestamp of
 the missing packet can be estimated from the timestamps of packets
 preceding and/or following the sequence number gap caused by the
 missing packet in the original stream.  In most cases, some form of
 linear estimate of the timestamp is good enough.
 Furthermore, a receiver should compute an estimate of the round-trip
 time (RTT) to the sender.  This can be done, for example, by
 measuring the retransmission delay to receive a retransmission packet
 after a NACK has been sent for that packet.  This estimate may also
 be obtained from past observations, RTCP report round-trip time if
 available, or any other means.  A standard mechanism for the receiver
 to estimate the RTT is specified in "RTP Control Protocol Extended
 Reports (RTCP XR)" [11].
 The receiver should not send a retransmission request as soon as it
 detects a missing sequence number but should add some extra delay to
 compensate for packet reordering.  This extra delay may, for example,
 be based on past observations of the experienced packet reordering.
 It should be noted that, in environments where packet reordering is
 rare or does not take place, e.g., if the underlying datalink layer
 affords ordered delivery, the delay may be extremely low or even take
 the value zero.  In such cases, an appropriate "reorder delay"
 algorithm may not actually be timer based, but packet based.  For
 example, if n number of packets are received after a gap is detected,
 then it may be assumed that the packet was truly lost rather than out
 of order.  This may turn out to be far easier to code on some
 platforms as a very short fixed FIFO packet buffer as opposed to the
 timer-based mechanism.
 To increase the robustness to the loss of a NACK or of a
 retransmission packet, a receiver may send a new NACK for the same
 packet.  This is referred to as multiple retransmissions.  Before
 sending a new NACK for a missing packet, the receiver should rely on
 a timer to be reasonably sure that the previous retransmission
 attempt has failed and so avoid unnecessary retransmissions.  The
 timer value shall be based on the observed round-trip time.  A static
 or an adaptive value MAY be used.  For example, an adaptive timer

Rey, et al. Standards Track [Page 12] RFC 4588 RTP Retransmission Payload Format July 2006

 could be one that changes its value with every new request for the
 same packet.  This document does not provide any guidelines as to how
 this adaptive value should be calculated because no experiments have
 been done to find this out.
 NACKs MUST be sent only for the original RTP stream.  Otherwise, if a
 receiver wanted to perform multiple retransmissions by sending a NACK
 in the retransmission stream, it would not be able to know the
 original sequence number and a timestamp estimation of the packet it
 requests.
 Appendix A gives some guidelines as to how to control the number of
 retransmissions.

6.4. Timing Rules

 The NACK feedback message may be sent in a regular full compound RTCP
 packet or in an early RTCP packet, as per AVPF [1].  Sending a NACK
 in an early packet allows reacting more quickly to a given packet
 loss.  However, in that case if a new packet loss occurs right after
 the early RTCP packet was sent, the receiver will then have to wait
 for the next regular RTCP compound packet after the early packet.
 Sending NACKs only in regular RTCP compound decreases the maximum
 delay between detecting an original packet loss and being able to
 send a NACK for that packet.  Implementers should consider the
 possible implications of this fact for the application being used.
 Furthermore, receivers may make use of the minimum interval between
 regular RTCP compound packets.  This interval can be used to keep
 regular receiver reporting down to a minimum, while still allowing
 receivers to send early RTCP packets during periods requiring more
 frequent feedback, e.g., times of higher packet loss rate.  Note that
 although RTCP packets may be suppressed because they do not contain
 NACKs, the same RTCP bandwidth as if they were sent needs to be
 available.  See AVPF [1] for details on the use of the minimum
 interval.

7. Congestion Control

 RTP retransmission poses a risk of increasing network congestion.  In
 a best-effort environment, packet loss is caused by congestion.
 Reacting to loss by retransmission of older data without decreasing
 the rate of the original stream would thus further increase
 congestion.  Implementations SHOULD follow the recommendations below
 in order to use retransmission.

Rey, et al. Standards Track [Page 13] RFC 4588 RTP Retransmission Payload Format July 2006

 The RTP profile under which the retransmission scheme is used defines
 an appropriate congestion control mechanism in different
 environments.  Following the rules under the profile, an RTP
 application can determine its acceptable bitrate and packet rate in
 order to be fair to other TCP or RTP flows.
 If an RTP application uses retransmission, the acceptable packet rate
 and bitrate include both the original and retransmitted data.  This
 guarantees that an application using retransmission achieves the same
 fairness as one that does not.  Such a rule would translate in
 practice into the following actions:
 If enhanced service is used, it should be made sure that the total
 bitrate and packet rate do not exceed that of the requested service.
 It should be further monitored that the requested services are
 actually delivered.  In a best-effort environment, the sender SHOULD
 NOT send retransmission packets without reducing the packet rate and
 bitrate of the original stream (for example, by encoding the data at
 a lower rate).
 In addition, the sender MAY selectively retransmit only the packets
 that it deems important and ignore NACK messages for other packets in
 order to limit the bitrate.
 These congestion control mechanisms should keep the packet loss rate
 within acceptable parameters.  In the context of congestion control,
 packet loss is considered acceptable if a TCP flow across the same
 network path and experiencing the same network conditions would
 achieve, on a reasonable timescale, an average throughput that is not
 less than the one the RTP flow achieves.  If congestion is not kept
 under control, then retransmission SHOULD NOT be used.
 Retransmissions MAY still be sent in some cases, e.g., in wireless
 links where packet losses are not caused by congestion, if the server
 (or the client that makes the retransmission request) estimates that
 a particular packet or frame is important to continue play out, or if
 an RTSP PAUSE has been issued to allow the buffer to fill up (RTSP
 PAUSE does not affect the sending of retransmissions).
 Finally, it may further be necessary to adapt the transmission rate
 (or the number of layers subscribed for a layered multicast session),
 or to arrange for the receiver to leave the session.

Rey, et al. Standards Track [Page 14] RFC 4588 RTP Retransmission Payload Format July 2006

8. Retransmission Payload Format MIME Type Registration

8.1. Introduction

 The following MIME subtype name and parameters are introduced in this
 document: "rtx", "rtx-time", and "apt".
 The binding used for the retransmission stream to the payload type
 number is indicated by an rtpmap attribute.  The MIME subtype name
 used in the binding is "rtx".
 The "apt" (associated payload type) parameter MUST be used to map the
 retransmission payload type to the associated original stream payload
 type.  If multiple original payload types are used, then multiple
 "apt" parameters MUST be included to map each original payload type
 to a different retransmission payload type.
 An OPTIONAL payload-format-specific parameter, "rtx-time", indicates
 the maximum time a sender will keep an original RTP packet in its
 buffers available for retransmission.  This time starts with the
 first transmission of the packet.
 The syntax is as follows:
    a=fmtp:<number> apt=<apt-value>;rtx-time=<rtx-time-val>
 where
    <number>: indicates the dynamic payload type number assigned to
    the retransmission payload format in an rtpmap attribute.
    <apt-value>: is the value of the original stream payload type to
    which this retransmission stream payload type is associated.
    <rtx-time-val>: specifies the time in milliseconds (measured from
    the time a packet was first sent) that a sender keeps an RTP
    packet in its buffers available for retransmission.  The absence
    of the rtx-time parameter for a retransmission stream means that
    the maximum retransmission time is not defined, but MAY be
    negotiated by other means.

Rey, et al. Standards Track [Page 15] RFC 4588 RTP Retransmission Payload Format July 2006

8.2. Registration of audio/rtx

 MIME type: audio
 MIME subtype: rtx
 Required parameters:
    rate: the RTP timestamp clockrate is equal to the RTP timestamp
    clockrate of the media that is retransmitted.
    apt: associated payload type.  The value of this parameter is the
    payload type of the associated original stream.
 Optional parameters:
    rtx-time: indicates the time in milliseconds (measured from the
    time a packet was first sent) that the sender keeps an RTP packet
    in its buffers available for retransmission.
 Encoding considerations: this type is only defined for transfer via
 RTP.
 Security considerations: see Section 12 of RFC 4588
 Interoperability considerations: none
 Published specification: RFC 4588
 Applications which use this media type: multimedia streaming
 applications
 Additional information: none
 Person & email address to contact for further information:
 jose.rey@eu.panasonic.com
 davidleon123@yahoo.com
 avt@ietf.org
 Intended usage: COMMON
 Authors:
 Jose Rey
 David Leon
 Change controller:
 IETF AVT WG delegated from the IESG

Rey, et al. Standards Track [Page 16] RFC 4588 RTP Retransmission Payload Format July 2006

8.3. Registration of video/rtx

 MIME type: video
 MIME subtype: rtx
 Required parameters:
    rate: the RTP timestamp clockrate is equal to the RTP timestamp
    clockrate of the media that is retransmitted.
    apt: associated payload type.  The value of this parameter is the
    payload type of the associated original stream.
 Optional parameters:
    rtx-time: indicates the time in milliseconds (measured from the
    time a packet was first sent) that the sender keeps an RTP packet
    in its buffers available for retransmission.
 Encoding considerations: this type is only defined for transfer via
 RTP.
 Security considerations: see Section 12 of RFC 4588
 Interoperability considerations: none
 Published specification: RFC 4588
 Applications which use this media type: multimedia streaming
 applications
 Additional information: none
 Person & email address to contact for further information:
 jose.rey@eu.panasonic.com
 davidleon123@yahoo.com
 avt@ietf.org
 Intended usage: COMMON
 Authors:
 Jose Rey
 David Leon
 Change controller:
 IETF AVT WG delegated from the IESG

Rey, et al. Standards Track [Page 17] RFC 4588 RTP Retransmission Payload Format July 2006

8.4. Registration of text/rtx

 MIME type: text
 MIME subtype: rtx
 Required parameters:
    rate: the RTP timestamp clockrate is equal to the RTP timestamp
    clockrate of the media that is retransmitted.
    apt: associated payload type.  The value of this parameter is the
    payload type of the associated original stream.
 Optional parameters:
    rtx-time: indicates the time in milliseconds (measured from the
    time a packet was first sent) that the sender keeps an RTP packet
    in its buffers available for retransmission.
 Encoding considerations: this type is only defined for transfer via
 RTP.
 Security considerations: see Section 12 of RFC 4588
 Interoperability considerations: none
 Published specification: RFC 4588
 Applications which use this media type: multimedia streaming
 applications
 Additional information: none
 Person & email address to contact for further information:
 jose.rey@eu.panasonic.com
 davidleon123@yahoo.com
 avt@ietf.org
 Intended usage: COMMON
 Authors:
 Jose Rey
 David Leon
 Change controller:
 IETF AVT WG delegated from the IESG

Rey, et al. Standards Track [Page 18] RFC 4588 RTP Retransmission Payload Format July 2006

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) that the sender keeps an RTP packet
    in its buffers available for retransmission.
 Encoding considerations: this type is only defined for transfer via
 RTP.
 Security considerations: see Section 12 of RFC 4588
 Interoperability considerations: none
 Published specification: RFC 4588
 Applications which use this media type: multimedia streaming
 applications
 Additional information: none
 Person & email address to contact for further information:
 jose.rey@eu.panasonic.com
 davidleon123@yahoo.com
 avt@ietf.org
 Intended usage: COMMON
 Authors:
 Jose Rey
 David Leon
 Change controller:
 IETF AVT WG delegated from the IESG

Rey, et al. Standards Track [Page 19] RFC 4588 RTP Retransmission Payload Format July 2006

8.6. Mapping to SDP

 The information carried in the MIME media type specification has a
 specific mapping to fields in SDP [5], which is commonly used to
 describe RTP sessions.  When SDP is used to specify retransmissions
 for an RTP stream, the mapping is done as follows:
  1. The MIME types ("video"), ("audio"), ("text"), and ("application")

go in the SDP "m=" as the media name.

  1. The MIME subtype ("rtx") goes in SDP "a=rtpmap" as the encoding

name. The RTP clockrate in "a=rtpmap" MUST be that of the

    retransmission payload type.  See Section 4 for details on this.
  1. The AVPF profile-specific parameters "ack" and "nack" go in SDP

"a=rtcp-fb". Several SDP "a=rtcp-fb" are used for several types

    of feedback.  See the AVPF profile [1] for details.
  1. The retransmission payload-format-specific parameters "apt" and

"rtx-time" go in the SDP "a=fmtp" as a semicolon-separated list of

    parameter=value pairs.
  1. Any remaining parameters go in the SDP "a=fmtp" attribute by

copying them directly from the MIME media type string as a

    semicolon-separated list of parameter=value pairs.
 In the following sections, some example SDP descriptions are
 presented.  In some of these examples, long lines are folded to meet
 the column width constraints of this document; the backslash ("\") at
 the end of a line and the carriage return that follows it should be
 ignored.

8.7. SDP Description with Session-Multiplexing

 In the case of session-multiplexing, the SDP description contains one
 media specification "m" line per RTP session.  The SDP MUST provide
 the grouping of the original and associated retransmission sessions'
 "m" lines, using the Flow Identification (FID) semantics defined in
 RFC 3388 [6].
 The following example specifies two original, AMR and MPEG-4, streams
 on ports 49170 and 49174 and their corresponding retransmission
 streams on ports 49172 and 49176, respectively:
 v=0
 o=mascha 2980675221 2980675778 IN IP4 host.example.net
 c=IN IP4 192.0.2.0
 a=group:FID 1 2

Rey, et al. Standards Track [Page 20] RFC 4588 RTP Retransmission Payload Format July 2006

 a=group:FID 3 4
 m=audio 49170 RTP/AVPF 96
 a=rtpmap:96 AMR/8000
 a=fmtp:96 octet-align=1
 a=rtcp-fb:96 nack
 a=mid:1
 m=audio 49172 RTP/AVPF 97
 a=rtpmap:97 rtx/8000
 a=fmtp:97 apt=96;rtx-time=3000
 a=mid:2
 m=video 49174 RTP/AVPF 98
 a=rtpmap:98 MP4V-ES/90000
 a=rtcp-fb:98 nack
 a=fmtp:98 profile-level-id=8;config=01010000012000884006682C209\
 0A21F
 a=mid:3
 m=video 49176 RTP/AVPF 99
 a=rtpmap:99 rtx/90000
 a=fmtp:99 apt=98;rtx-time=3000
 a=mid:4
 A special case of the SDP description is a description that contains
 only one original session "m" line and one retransmission session "m"
 line, the grouping is then obvious and FID semantics MAY be omitted
 in this special case only.
 This is illustrated in the following example, which is an SDP
 description for a single original MPEG-4 stream and its corresponding
 retransmission session:
 v=0
 o=mascha 2980675221 2980675778 IN IP4 host.example.net
 c=IN IP4 192.0.2.0
 m=video 49170 RTP/AVPF 96
 a=rtpmap:96 MP4V-ES/90000
 a=rtcp-fb:96 nack
 a=fmtp:96 profile-level-id=8;config=01010000012000884006682C209\
 0A21F
 m=video 49172 RTP/AVPF 97
 a=rtpmap:97 rtx/90000
 a=fmtp:97 apt=96;rtx-time=3000

8.8. SDP Description with SSRC-Multiplexing

 The following is an example of an SDP description for an RTP video
 session using SSRC-multiplexing with similar parameters as in the
 single-session example above:

Rey, et al. Standards Track [Page 21] RFC 4588 RTP Retransmission Payload Format July 2006

 v=0
 o=mascha 2980675221 2980675778 IN IP4 host.example.net
 c=IN IP4 192.0.2.0
 m=video 49170 RTP/AVPF 96 97
 a=rtpmap:96 MP4V-ES/90000
 a=rtcp-fb:96 nack
 a=fmtp:96 profile-level-id=8;config=01010000012000884006682C209\
 0A21F
 a=rtpmap:97 rtx/90000
 a=fmtp:97 apt=96;rtx-time=3000

9. RTSP Considerations

 The Real Time Streaming Protocol (RTSP), RFC 2326 [7], is an
 application-level protocol for control over the delivery of data with
 real-time properties.  This section looks at the issues involved in
 controlling RTP sessions that use retransmissions.

9.1. RTSP Control with SSRC-Multiplexing

 In the case of SSRC-multiplexing, the "m" line includes both original
 and retransmission payload types and has a single RTSP "control"
 attribute.  The receiver uses the "m" line to request SETUP and
 TEARDOWN of the whole media session.  The RTP profile contained in
 the Transport header MUST be the AVPF profile or another suitable
 profile allowing extended feedback.  If the SSRC value is included in
 the SETUP response's Transport header, it MUST be that of the
 original stream.
 In order to control the sending of the session original media stream,
 the receiver sends as usual PLAY and PAUSE requests to the sender for
 the session.  The RTP-info header that is used to set RTP-specific
 parameters in the PLAY response MUST be set according to the RTP
 information of the original stream.
 When the receiver starts receiving the original stream, it can then
 request retransmission through RTCP NACKs without additional RTSP
 signalling.

9.2. RTSP Control with Session-Multiplexing

 In the case of session-multiplexing, each SDP "m" line has an RTSP
 "control" attribute.  Hence, when retransmission is used, both the
 original session and the retransmission have their own "control"
 attributes.  The receiver can associate the original session and the
 retransmission session through the FID semantics as specified in
 Section 8.

Rey, et al. Standards Track [Page 22] RFC 4588 RTP Retransmission Payload Format July 2006

 The original and the retransmission streams are set up and torn down
 separately through their respective media "control" attribute.  The
 RTP profile contained in the Transport header MUST be the AVPF
 profile or another suitable profile allowing extended feedback for
 both the original and the retransmission sessions.
 The RTSP presentation SHOULD support aggregate control and SHOULD
 contain a session-level RTSP URL.  The receiver SHOULD use aggregate
 control for an original session and its associated retransmission
 session.  Otherwise, there would need to be two different 'session-
 id' values, i.e., different values for the original and
 retransmission sessions, and the sender would not know how to
 associate them.
 The session-level "control" attribute is then used as usual to
 control the playing of the original stream.  When the receiver starts
 receiving the original stream, it can then request retransmissions
 through RTCP without additional RTSP signalling.

9.3. RTSP Control of the Retransmission Stream

 Because of the nature of retransmissions, the sending of
 retransmission packets SHOULD NOT be controlled through RTSP PLAY and
 PAUSE requests.  The PLAY and PAUSE requests SHOULD NOT affect the
 retransmission stream.  Retransmission packets are sent upon receiver
 requests in the original RTCP stream, regardless of the state.

9.4. Cache Control

 Retransmission streams SHOULD NOT be cached.
 In the case of session-multiplexing, the "Cache-Control" header
 SHOULD be set to "no-cache" for the retransmission stream.
 In the case of SSRC-multiplexing, RTSP cannot specify independent
 caching for the retransmission stream, because there is a single "m"
 line in SDP.  Therefore, the implementer should take this fact into
 account when deciding whether or not to cache an SSRC-multiplexed
 session.

10. Implementation Examples

 This document mandates only the sender and receiver behaviours that
 are necessary for interoperability.  In addition, certain algorithms,
 such as rate control or buffer management when targeted at specific
 environments, may enhance the retransmission efficiency.

Rey, et al. Standards Track [Page 23] RFC 4588 RTP Retransmission Payload Format July 2006

 This section gives an overview of different implementation options
 allowed within this specification.
 The first example describes a minimal receiver implementation.  With
 this implementation, it is possible to retransmit lost RTP packets,
 detect efficiently the loss of retransmissions, and perform multiple
 retransmissions, if needed.  Most of the necessary processing is done
 at the server.
 The second example shows how retransmissions may be used in (small)
 multicast groups in conjunction with layered encoding.  It
 illustrates that retransmissions and layered encoding may be
 complementary techniques.

10.1. A Minimal Receiver Implementation Example

 This section gives an example of an implementation supporting
 multiple retransmissions.  The sender transmits the original data in
 RTP packets using the MPEG-4 video RTP payload format.  It is assumed
 that NACK feedback messages are used, as per [1].  An SDP description
 example with SSRC-multiplexing is given below:
 v=0
 o=mascha 2980675221 2980675778 IN IP4 host.example.net
 c=IN IP4 192.0.2.0
 m=video 49170 RTP/AVPF 96 97
 a=rtpmap:96 MP4V-ES/90000
 a=rtcp-fb:96 nack
 a=rtpmap:97 rtx/90000
 a=fmtp:97 apt=96;rtx-time=3000
 The format-specific parameter "rtx-time" indicates that the server
 will buffer the sent packets in a retransmission buffer for 3.0
 seconds, after which the packets are deleted from the retransmission
 buffer and will never be sent again.
 In this implementation example, the required RTP receiver processing
 to handle retransmission is kept to a minimum.  The receiver detects
 packet loss from the gaps observed in the received sequence numbers.
 It signals lost packets to the sender through NACKs as defined in the
 AVPF profile [1].  The receiver should take into account the
 signalled sender retransmission buffer length in order to dimension
 its own reception buffer.  It should also derive from the buffer
 length the maximum number of times the retransmission of a packet can
 be requested.

Rey, et al. Standards Track [Page 24] RFC 4588 RTP Retransmission Payload Format July 2006

 The sender should retransmit the packets selectively; i.e., it should
 choose whether to retransmit a requested packet depending on the
 packet importance, the observed Quality of Service (QoS), and
 congestion state of the network connection to the receiver.
 Obviously, the sender processing increases with the number of
 receivers as state information and processing load must be allocated
 to each receiver.

10.2. Retransmission of Layered Encoded Media in Multicast

 This section shows how to combine retransmissions with layered
 encoding in multicast sessions.  Note that the retransmission
 framework is offered only for small multicast applications.  Refer to
 RFC 2887 [10] for a discussion of the problems of NACK implosion,
 severe congestion caused by feedback traffic, in large-group reliable
 multicast applications.
 Packets of different importance are sent in different RTP sessions.
 The retransmission streams corresponding to the different layers can
 themselves be seen as different retransmission layers.  The relative
 importance of the different retransmission streams should reflect the
 relative importance of the different original streams.
 In multicast, SSRC-multiplexing of the original and retransmission
 streams is not allowed as per Section 5.3 of this document.  For this
 reason, the retransmission stream(s) MUST be sent in different RTP
 session(s) using session-multiplexing.
 An SDP description example of multicast retransmissions for layered
 encoded media is given below:
 m=video 8000 RTP/AVPF 98
 c=IN IP4 224.2.1.0/127/3
 a=rtpmap:98 MP4V-ES/90000
 a=rtcp-fb:98 nack
 m=video 8000 RTP/AVPF 99
 c=IN IP4 224.2.1.3/127/3
 a=rtpmap:99 rtx/90000
 a=fmtp:99 apt=98;rtx-time=3000
 The server and the receiver may implement the retransmission methods
 illustrated in the previous examples.  In addition, they may choose
 to request and retransmit a lost packet depending on the layer it
 belongs to.

Rey, et al. Standards Track [Page 25] RFC 4588 RTP Retransmission Payload Format July 2006

11. IANA Considerations

 A new MIME subtype name, "rtx", has been registered for four
 different media types, as follows: "video", "audio", "text" and
 "application".  An additional REQUIRED parameter, "apt", and an
 OPTIONAL parameter, "rtx-time", are defined.  See Section 8 for
 details.

12. Security Considerations

 RTP packets using the payload format defined in this specification
 are subject to the general security considerations discussed in RTP
 [3], Section 9.
 In common streaming scenarios message authentication, data integrity,
 replay protection, and confidentiality are desired.
 The absence of authentication may enable man-in-the-middle and replay
 attacks, which can be very harmful for RTP retransmission.  For
 example: tampered RTCP packets may trigger inappropriate
 retransmissions that effectively reduce the actual bitrate share
 allocated to the original data stream, tampered RTP retransmission
 packets could cause the client's decoder to crash, and tampered
 retransmission requests may invalidate the SSRC association mechanism
 described in Section 5 of this document.  On the other hand, replayed
 packets could lead to false reordering and RTT measurements (required
 for the retransmission request strategy) and may cause the receiver
 buffer to overflow.
 Furthermore, in order to ensure confidentiality of the data, the
 original payload data needs to be encrypted.  There is actually no
 need to encrypt the 2-byte retransmission payload header since it
 does not provide any hints about the data content.
 Furthermore, it is RECOMMENDED that the cryptography mechanisms used
 for this payload format provide protection against known plaintext
 attacks.  RTP recommends that the initial RTP timestamp SHOULD be
 random to secure the stream against known plaintext attacks.  This
 payload format does not follow this recommendation as the initial
 timestamp will be the media timestamp of the first retransmitted
 packet.  However, since the initial timestamp of the original stream
 is itself random, if the original stream is encrypted, the first
 retransmitted packet timestamp would also be random to an attacker.
 Therefore, confidentiality would not be compromised.
 If cryptography is used to provide security services on the original
 stream, then the same services, with equivalent cryptographic
 strength, MUST be provided on the retransmission stream.  The use of

Rey, et al. Standards Track [Page 26] RFC 4588 RTP Retransmission Payload Format July 2006

 the same key for the retransmitted stream and the original stream may
 lead to security problems, e.g., two-time pads.  Refer to Section 9.1
 of the Secure Real-Time Transport Protocol (SRTP) [12] for a
 discussion the implications of two-time pads and how to avoid them.
 At the time of writing this document, SRTP does not provide all the
 security services mentioned.  There are, at least, two reasons for
 this: 1) the occurrence of two-time pads and 2) the fact that this
 payload format typically works under the RTP/AVPF profile whereas
 SRTP only supports RTP/AVP.  An adapted variant of SRTP shall solve
 these shortcomings in the future.
 Congestion control considerations with the use of retransmission are
 dealt with in Section 7 of this document.

13. Acknowledgements

 We would like to express our gratitude to Carsten Burmeister for his
 participation in the development of this document.  Our thanks also
 go to Koichi Hata, Colin Perkins, Stephen Casner, Magnus Westerlund,
 Go Hori, and Rahul Agarwal for their helpful comments.

14. References

14.1. Normative References

 [1]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
      "Extended RTP profile for Real-time Transport Control Protocol
      (RTCP)-Based feedback", RFC 4585, July 2006.
 [2]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
      Levels", BCP 14, RFC 2119, March 1997.
 [3]  Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
      "RTP: A Transport Protocol for Real-Time Applications", STD 64,
      RFC 3550, July 2003.
 [4]  Casner, S., "Session Description Protocol (SDP) Bandwidth
      Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 3556,
      July 2003.
 [5]  Handley, M. and V. Jacobson, "SDP: Session Description
      Protocol", RFC 2327, April 1998.
 [6]  Camarillo, G., Eriksson, G., Holler, J., and H. Schulzrinne,
      "Grouping of Media Lines in the Session Description Protocol
      (SDP)", RFC 3388, December 2002.

Rey, et al. Standards Track [Page 27] RFC 4588 RTP Retransmission Payload Format July 2006

 [7]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming
      Protocol (RTSP)", RFC 2326, April 1998.

14.2. Informative References

 [8]  Perkins, C. and O. Hodson, "Options for Repair of Streaming
      Media", RFC 2354, June 1998.
 [9]  Hellstrom, G. and P. Jones, "RTP Payload for Text Conversation",
      RFC 4103, June 2005.
 [10] Handley, M., Floyd, S., Whetten, B., Kermode, R., Vicisano, L.,
      and M. Luby, "The Reliable Multicast Design Space for Bulk Data
      Transfer", RFC 2887, August 2000.
 [11] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol
      Extended Reports (RTCP XR)", RFC 3611, November 2003.
 [12] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
      Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
      3711, March 2004.

Rey, et al. Standards Track [Page 28] RFC 4588 RTP Retransmission Payload Format July 2006

Appendix A. How to Control the Number of Rtxs. per Packet

 Finding out the number of retransmissions (rtxs.) per packet for
 achieving the best possible transmission is a difficult task.  Of
 course, the absolute minimum should be one (1); otherwise, do not use
 this payload format.  Moreover, as of date of publication, the
 authors were not aware of any studies on the number of
 retransmissions per packet that should be used for best performance.
 To help implementers and researchers on this item, this section
 describes an estimate of the buffering time required for achieving a
 given number of retransmissions.  Once this time has been calculated,
 it can be communicated to the client via SDP parameter "rtx-time", as
 defined in this document.

A.1. Scenario and Assumptions

  • Streaming scenario with relaxed delay bounds. Client and server

are provided with buffering space as indicated by the parameter

   "rtx-time" in SDP.
  • RTP AVPF profile used with SSRC-multiplexing retransmission scheme:

1 SSRC for original packets, 1 for retransmission packets.

  • Default RTCP bandwidth share for SRs and RRs, i.e., SR+RR = 0.05.

We have senders (2) and receivers (1). Receivers and senders get

   equally 1/3 of the RTCP bandwidth share because the proportion of
   senders is greater than 1/4 of session members.
  • avg-rtcp-size is approximated by 120 bytes. This is a rounded-up

average of 2 SRs, one for each SSRC, containing 40/8/28/32 bytes

   for IPv6/UDP/SR/SDES with CNAME, thus making 105 bytes each; and a
   RR with 40/8/64/32 bytes for IPv6/UDP/2*RR/SDES, making 157 bytes.
   Since senders and receivers share the RTCP bandwidth equally, then
   avg-rtcp-size = (157+105+105)/3 = 117.3 ~= 120 bytes.  The
   important characteristic of this value is that it is something over
   100 bytes, which seems to be a representative figure for typical
   configurations.
  • The profile used is AVPF [1] and Generic NACKs are used for

requesting retransmissions. This adds 16 bytes of overhead for 1

   NACK and 4 bytes more for every additional NACK Feedback Control
   Information (FCI) field.
  • We assume a worst-case scenario in which each packet exhausts its

corresponding number of available retransmissions, N, before it is

   received.  This means that if a packet is requested for
   retransmission a maximum of 2 times, the corresponding generic NACK
   report block requesting that particular packet is sent in two

Rey, et al. Standards Track [Page 29] RFC 4588 RTP Retransmission Payload Format July 2006

   consecutive RTCP compounds; likewise, if it is requested for
   retransmission 10 times, then the generic NACK is sent 10 times.
   This assumption makes the RTCP packet size approximately constant
   after N*RTCP intervals (seconds), namely, to avg-rtcp-size = 120 +
   (receiver-RTCP-bw-share)*(12 + 4*N).  In our case, the receiver
   RTCP bandwidth share is 1/3; thus, avg-rtcp-size = 124 + 4*N/3.
  • Two delay parameters are difficult to approximate and may be

implementation dependent. Therefore, we list them here explicitly

   without assigning them a particular value: one is the packet loss
   detection time (T2), and the other is feedback processing and
   queuing time for retransmissions (T5).  Implementers shall assign
   appropriate values to these two parameters.
 Graphically, we have the following:
       Sender
     +-+---------------------------------^-----\-----------------
      \ \                               /       \
       \ \                             |         |
 SN=0   \ \ SN=1                       /         \  RTX(SN=0)
         \ \                          /           \
          X \                        /             \
             `.                     /               \
               \                   /                 \
                \                 |                   |
                 \                /                   \    ......
                  \              /                     \
     -------------V----D--------/-----------------------V--------
            T1      T2    T3         T4    T5     T1   ........
      Receiver
 Legend:
 =======
 DL: downlink (client->server)
 UL: uplink (server->client)
 Time unit is seconds, s.
 Bitrate unit is bits per second, bps.
 DL transmission time:            T1 = physical-delay-DL +
    tx-delay-DL(=avg-pkt-size/DL-bitrate) + interarrival-delay-jitter
 Time to detect packet loss:      T2 = pkt-loss-detect-time
 Time to report packet loss:      T3 = time-to-next-rtcp-report
 UL transmission time:            T4 = physical-delay-UL +
    transmission-delay-UL + interarrival-delay-jitter

Rey, et al. Standards Track [Page 30] RFC 4588 RTP Retransmission Payload Format July 2006

 Retransmissions processing time: T5 = feedback-processing-time +
    rtx-queuing-time

A.2. Goal

 To find an estimate of the buffering time, T(), that a streaming
 server shall use in order to enable a given number of retransmissions
 for each packet, N.  This time is approximately equal at the server
 and at the client, if one considers that the client starts buffering
 T1 seconds later.

A.3. Solution

 First, we find the value of the estimate for 1 retransmission,
 T(1)=T:
    T = T1 + T2 + T3 + T4 + T5
 Since T1 + T4 ~= RTT,
    T = RTT + T2 + T3 + T5
 The worst case for T3 would be that we assume that reporting has to
 wait a whole RTCP interval and that the maximum randomization factor
 of 1.5 is applied.  Therefore, after applying the subsequent
 compensation to avoid traffic bursts (see Appendix A.7 of RTP [3]),
 we have that T3 = 1.5/1.21828*RTCP-Interval.  Thus,
    T = RTT + 1.2312*RTCP-Interval + T2 + T5
 On the other hand, RTCP-Interval = avg-rtcp-size*8*(senders +
 receivers)/(RR+RS).  In this scenario: sender + receivers = 3; RR+RS
 is the receiver report plus sender report bandwidth share, in this
 case, equal to the default 5% of session bandwidth, bw.  We assume an
 average RTCP packet size, avg-rtcp-size = 120 bytes.  Thus:
    T = RTT + 1.2312*avg-rtcp-size*8*3/(0.05*bw) + T2 + T5
 for 1 retransmission.
 For enabling N retransmissions, the available buffering time in a
 streaming server or client is approximately:
    T(N) = N*(RTT+1.2312*avg-rtcp-size*8*3/(0.05*bw) + T2 + T5)

Rey, et al. Standards Track [Page 31] RFC 4588 RTP Retransmission Payload Format July 2006

 where, as per above,
    avg-rtcp-size = 120 + (receiver-RTCP-bw-share)*(12 + 4*N)
                  = 120 + (1/3)*(12 + 4*N)
                  = 124 + 4*N/3.

A.4. Numbers

 If we ignore the effect of T2 and T5, i.e., assume that all losses
 are detected immediately and that there is no additional delay due to
 feedback processing or retransmission queuing, we have the following
 buffering times for different values of N:
 RTCP w/ several Generic NACKs; variable packet size = 124 + 4*N/3
 bytes
 |============|=====|======================================|
 |  RTP BW    | RTT |            N value                   |
 |============|=====|   1      2       5       7       10  |
                    |======================================|
 64000         0,05   1,21    2,44    6,28    8,97    13,18
 128000        0,05   0,63    1,27    3,27    4,66    6,84
 256000        0,05   0,34    0,68    1,76    2,50    3,67
 512000        0,05   0,19    0,39    1,00    1,43    2,09
 1024000       0,05   0,12    0,25    0,63    0,89    1,29
 5000000       0,05   0,06    0,13    0,33    0,46    0,66
 10000000      0,05   0,06    0,11    0,29    0,41    0,58
 64000         0,2    1,36    2,74    7,03    10,02   14,68
 128000        0,2    0,78    1,57    4,02    5,71    8,34
 256000        0,2    0,49    0,98    2,51    3,55    5,17
 512000        0,2    0,34    0,69    1,75    2,48    3,59
 1024000       0,2    0,27    0,55    1,38    1,94    2,79
 5000000       0,2    0,21    0,43    1,08    1,51    2,16
 10000000      0,2    0,21    0,41    1,04    1,46    2,08
 64000         1      2,16    4,34    11,03   15,62   22,68
 128000        1      1,58    3,17    8,02    11,31   16,34
 256000        1      1,29    2,58    6,51    9,15    13,17
 512000        1      1,14    2,29    5,75    8,08    11,59
 1024000       1      1,07    2,15    5,38    7,54    10,79
 5000000       1      1,01    2,03    5,08    7,11    10,16
 10000000      1      1,01    2,01    5,04    7,06    10,08

Rey, et al. Standards Track [Page 32] RFC 4588 RTP Retransmission Payload Format July 2006

 To quantify the error of not taking the Generic NACKS into account,
 we can do the same numbers, but ignoring the Generic NACK
 contribution, avg-rtcp-size ~= 120 bytes.  As we see from below, this
 may result in a buffer estimation error of 1-1.5 seconds (5-10%) for
 lower bandwidth values and higher number of retransmissions.  This
 effect is low in this case.  Nevertheless, it should be carefully
 evaluated for the particular scenario; that is why the formula
 includes it.
 RTCP w/o Generic NACK, fixed packet size ~= 120 bytes
 |============|=====|======================================|
 |  RTP BW    | RTT |            N value                   |
 |============|=====|   1      2       5       7       10  |
                    |======================================|
 64000         0,05   1,16    2,32    5,79    8,11    11,58
 128000        0,05   0,60    1,21    3,02    4,23    6,04
 256000        0,05   0,33    0,65    1,64    2,29    3,27
 512000        0,05   0,19    0,38    0,94    1,32    1,89
 1024000       0,05   0,12    0,24    0,60    0,83    1,19
 5000000       0,05   0,06    0,13    0,32    0,45    0,64
 10000000      0,05   0,06    0,11    0,29    0,40    0,57
 64000         0,2    1,31    2,62    6,54    9,16    13,08
 128000        0,2    0,75    1,51    3,77    5,28    7,54
 256000        0,2    0,48    0,95    2,39    3,34    4,77
 512000        0,2    0,34    0,68    1,69    2,37    3,39
 1024000       0,2    0,27    0,54    1,35    1,88    2,69
 5000000       0,2    0,21    0,43    1,07    1,50    2,14
 10000000      0,2    0,21    0,41    1,04    1,45    2,07
 64000         1      2,11    4,22    10,54   14,76   21,08
 128000        1      1,55    3,11    7,77    10,88   15,54
 256000        1      1,28    2,55    6,39    8,94    12,77
 512000        1      1,14    2,28    5,69    7,97    11,39
 1024000       1      1,07    2,14    5,35    7,48    10,69
 5000000       1      1,01    2,03    5,07    7,10    10,14
 10000000      1      1,01    2,01    5,04    7,05    10,07

Rey, et al. Standards Track [Page 33] RFC 4588 RTP Retransmission Payload Format July 2006

Authors' Addresses

 Jose Rey
 Panasonic R&D Center Germany GmbH
 Monzastr. 4c
 D-63225 Langen, Germany
 Phone: +49-6103-766-134
 Fax:   +49-6103-766-166
 EMail: jose.rey@eu.panasonic.com
 David Leon
 Consultant
 EMail: davidleon123@yahoo.com
 Akihiro Miyazaki
 Matsushita Electric Industrial Co., Ltd
 1006, Kadoma, Kadoma City, Osaka, Japan
 Phone: +81-6-6900-9172
 Fax:   +81-6-6900-9173
 EMail: miyazaki.akihiro@jp.panasonic.com
 Viktor Varsa
 Nokia Research Center
 6000 Connection Drive
 Irving, TX. USA
 Phone:  1-972-374-1861
 EMail: viktor.varsa@nokia.com
 Rolf Hakenberg
 Panasonic R&D Center Germany GmbH
 Monzastr. 4c
 D-63225 Langen, Germany
 Phone: +49-6103-766-162
 Fax:   +49-6103-766-166
 EMail: rolf.hakenberg@eu.panasonic.com

Rey, et al. Standards Track [Page 34] RFC 4588 RTP Retransmission Payload Format July 2006

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Rey, et al. Standards Track [Page 35]

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