<pre>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


                   <span class="h1">RTP Retransmission Payload Format</span>

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.













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Table of Contents

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










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<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a>.  Introduction</span>

   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.  <a href="./rfc2354">RFC 2354</a> [<a href="#ref-8" title="&quot;Options for Repair of Streaming Media&quot;">8</a>]
   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 [<a href="#ref-1" title="&quot;Extended RTP profile for Real-time Transport Control Protocol (RTCP)-Based feedback&quot;">1</a>].  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 [<a href="#ref-1" title="&quot;Extended RTP profile for Real-time Transport Control Protocol (RTCP)-Based feedback&quot;">1</a>]
   for details.

<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>.  Terminology</span>

   The following terms are used in this document:

   CSRC: contributing source.  See [<a href="#ref-3" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">3</a>].

   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.



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   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
   [<a href="#ref-1" title="&quot;Extended RTP profile for Real-time Transport Control Protocol (RTCP)-Based feedback&quot;">1</a>] 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 [<a href="#ref-3" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">3</a>].

   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 <a href="./rfc2119">RFC 2119</a> [<a href="#ref-2" title="&quot;Key words for use in RFCs to Indicate Requirement Levels&quot;">2</a>].

<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a>.  Requirements and Design Rationale for a Retransmission Scheme</span>

   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.



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   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
      [<a href="#ref-9" title="&quot;RTP Payload for Text Conversation&quot;">9</a>] 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 <a href="#section-5.2">Section 5.2</a> of RTP [<a href="#ref-3" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">3</a>] to RTP
   multiplexing do not apply in this case.





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   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.

<span class="h3"><a class="selflink" id="section-3.1" href="#section-3.1">3.1</a>.  Multiplexing Scheme Choice</span>

   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 [<a href="#ref-3" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">3</a>]) 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
   <a href="#section-5.3">Section 5.3</a> for motivation).





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<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>.  Retransmission Payload Format</span>

   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 <a href="#section-5.3">Section 5.3</a> 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.



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   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 <a href="#section-8.1">Section 8.1</a> 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



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   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 [<a href="#ref-3" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">3</a>].  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.

<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  Association of Retransmission and Original Streams</span>

<span class="h3"><a class="selflink" id="section-5.1" href="#section-5.1">5.1</a>.  Retransmission Session Sharing</span>

   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.

<span class="h3"><a class="selflink" id="section-5.2" href="#section-5.2">5.2</a>.  CNAME Use</span>

   In both the session-multiplexing and the SSRC-multiplexing cases, a
   sender MUST use the same RTCP CNAME [<a href="#ref-3" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">3</a>] for an original stream and
   its associated retransmission stream.

<span class="h3"><a class="selflink" id="section-5.3" href="#section-5.3">5.3</a>.  Association at the Receiver</span>

   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



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   several media streams may have the same payload type value.  The two
   sessions are themselves associated out-of-band.  See <a href="#section-8">Section 8</a> 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.









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<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a>.  Use with the Extended RTP Profile for RTCP-based Feedback</span>

   This section gives general hints for the usage of this payload format
   with the extended RTP profile for RTCP-based feedback, denoted AVPF
   [<a href="#ref-1" title="&quot;Extended RTP profile for Real-time Transport Control Protocol (RTCP)-Based feedback&quot;">1</a>].  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 [<a href="#ref-1" title="&quot;Extended RTP profile for Real-time Transport Control Protocol (RTCP)-Based feedback&quot;">1</a>] for details.

<span class="h3"><a class="selflink" id="section-6.1" href="#section-6.1">6.1</a>.  RTCP at the Sender</span>

   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.

<span class="h3"><a class="selflink" id="section-6.2" href="#section-6.2">6.2</a>.  RTCP Receiver Reports</span>

   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 [<a href="#ref-4" title="&quot;Session Description Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP) Bandwidth&quot;">4</a>]).

   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.








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<span class="h3"><a class="selflink" id="section-6.3" href="#section-6.3">6.3</a>.  Retransmission Requests</span>

   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)" [<a href="#ref-11" title="&quot;RTP Control Protocol Extended Reports (RTCP XR)&quot;">11</a>].

   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



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   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.

   <a href="#appendix-A">Appendix A</a> gives some guidelines as to how to control the number of
   retransmissions.

<span class="h3"><a class="selflink" id="section-6.4" href="#section-6.4">6.4</a>.  Timing Rules</span>

   The NACK feedback message may be sent in a regular full compound RTCP
   packet or in an early RTCP packet, as per AVPF [<a href="#ref-1" title="&quot;Extended RTP profile for Real-time Transport Control Protocol (RTCP)-Based feedback&quot;">1</a>].  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 [<a href="#ref-1" title="&quot;Extended RTP profile for Real-time Transport Control Protocol (RTCP)-Based feedback&quot;">1</a>] for details on the use of the minimum
   interval.

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a>.  Congestion Control</span>

   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.






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   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.









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<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a>.  Retransmission Payload Format MIME Type Registration</span>

<span class="h3"><a class="selflink" id="section-8.1" href="#section-8.1">8.1</a>.  Introduction</span>

   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:&lt;number&gt; apt=&lt;apt-value&gt;;rtx-time=&lt;rtx-time-val&gt;

   where

      &lt;number&gt;: indicates the dynamic payload type number assigned to
      the retransmission payload format in an rtpmap attribute.

      &lt;apt-value&gt;: is the value of the original stream payload type to
      which this retransmission stream payload type is associated.

      &lt;rtx-time-val&gt;: 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.











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<span class="h3"><a class="selflink" id="section-8.2" href="#section-8.2">8.2</a>.  Registration of audio/rtx</span>

   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 <a href="./rfc4588#section-12">Section&nbsp;12 of RFC 4588</a>

   Interoperability considerations: none

   Published specification: <a href="./rfc4588">RFC 4588</a>

   Applications which use this media type: multimedia streaming
   applications

   Additional information: none

   Person &amp; 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




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<span class="h3"><a class="selflink" id="section-8.3" href="#section-8.3">8.3</a>.  Registration of video/rtx</span>

   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 <a href="./rfc4588#section-12">Section&nbsp;12 of RFC 4588</a>

   Interoperability considerations: none

   Published specification: <a href="./rfc4588">RFC 4588</a>

   Applications which use this media type: multimedia streaming
   applications

   Additional information: none

   Person &amp; 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




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<span class="h3"><a class="selflink" id="section-8.4" href="#section-8.4">8.4</a>.  Registration of text/rtx</span>

   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 <a href="./rfc4588#section-12">Section&nbsp;12 of RFC 4588</a>

   Interoperability considerations: none

   Published specification: <a href="./rfc4588">RFC 4588</a>

   Applications which use this media type: multimedia streaming
   applications

   Additional information: none

   Person &amp; 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




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<span class="h3"><a class="selflink" id="section-8.5" href="#section-8.5">8.5</a>.  Registration of application/rtx</span>

   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 <a href="./rfc4588#section-12">Section&nbsp;12 of RFC 4588</a>

   Interoperability considerations: none

   Published specification: <a href="./rfc4588">RFC 4588</a>

   Applications which use this media type: multimedia streaming
   applications

   Additional information: none

   Person &amp; 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




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<span class="h3"><a class="selflink" id="section-8.6" href="#section-8.6">8.6</a>.  Mapping to SDP</span>

   The information carried in the MIME media type specification has a
   specific mapping to fields in SDP [<a href="#ref-5" title="&quot;SDP: Session Description Protocol&quot;">5</a>], which is commonly used to
   describe RTP sessions.  When SDP is used to specify retransmissions
   for an RTP stream, the mapping is done as follows:

   -  The MIME types ("video"), ("audio"), ("text"), and ("application")
      go in the SDP "m=" as the media name.

   -  The MIME subtype ("rtx") goes in SDP "a=rtpmap" as the encoding
      name.  The RTP clockrate in "a=rtpmap" MUST be that of the
      retransmission payload type.  See <a href="#section-4">Section 4</a> for details on this.

   -  The AVPF profile-specific parameters "ack" and "nack" go in SDP
      "a=rtcp-fb".  Several SDP "a=rtcp-fb" are used for several types
      of feedback.  See the AVPF profile [<a href="#ref-1" title="&quot;Extended RTP profile for Real-time Transport Control Protocol (RTCP)-Based feedback&quot;">1</a>] for details.

   -  The retransmission payload-format-specific parameters "apt" and
      "rtx-time" go in the SDP "a=fmtp" as a semicolon-separated list of
      parameter=value pairs.

   -  Any remaining parameters go in the SDP "a=fmtp" attribute by
      copying them directly from the MIME media type string as a
      semicolon-separated list of parameter=value pairs.

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

<span class="h3"><a class="selflink" id="section-8.7" href="#section-8.7">8.7</a>.  SDP Description with Session-Multiplexing</span>

   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
   <a href="./rfc3388">RFC 3388</a> [<a href="#ref-6" title="&quot;Grouping of Media Lines in the Session Description Protocol (SDP)&quot;">6</a>].

   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



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   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

<span class="h3"><a class="selflink" id="section-8.8" href="#section-8.8">8.8</a>.  SDP Description with SSRC-Multiplexing</span>

   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:




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   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

<span class="h2"><a class="selflink" id="section-9" href="#section-9">9</a>.  RTSP Considerations</span>

   The Real Time Streaming Protocol (RTSP), <a href="./rfc2326">RFC 2326</a> [<a href="#ref-7" title="&quot;Real Time Streaming Protocol (RTSP)&quot;">7</a>], 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.

<span class="h3"><a class="selflink" id="section-9.1" href="#section-9.1">9.1</a>.  RTSP Control with SSRC-Multiplexing</span>

   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.

<span class="h3"><a class="selflink" id="section-9.2" href="#section-9.2">9.2</a>.  RTSP Control with Session-Multiplexing</span>

   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
   <a href="#section-8">Section 8</a>.




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   The original and the retransmission streams are set up and torn down
   separately through their respective media "control" attribute.  The
   RTP profile contained in the Transport header MUST be the AVPF
   profile or another suitable profile allowing extended feedback for
   both the original and the retransmission 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.

<span class="h3"><a class="selflink" id="section-9.3" href="#section-9.3">9.3</a>.  RTSP Control of the Retransmission Stream</span>

   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.

<span class="h3"><a class="selflink" id="section-9.4" href="#section-9.4">9.4</a>.  Cache Control</span>

   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.

<span class="h2"><a class="selflink" id="section-10" href="#section-10">10</a>.  Implementation Examples</span>

   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.





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   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.

<span class="h3"><a class="selflink" id="section-10.1" href="#section-10.1">10.1</a>.  A Minimal Receiver Implementation Example</span>

   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 [<a href="#ref-1" title="&quot;Extended RTP profile for Real-time Transport Control Protocol (RTCP)-Based feedback&quot;">1</a>].  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 [<a href="#ref-1" title="&quot;Extended RTP profile for Real-time Transport Control Protocol (RTCP)-Based feedback&quot;">1</a>].  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.






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   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.

<span class="h3"><a class="selflink" id="section-10.2" href="#section-10.2">10.2</a>.  Retransmission of Layered Encoded Media in Multicast</span>

   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
   <a href="./rfc2887">RFC 2887</a> [<a href="#ref-10" title="&quot;The Reliable Multicast Design Space for Bulk Data Transfer&quot;">10</a>] 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 <a href="#section-5.3">Section 5.3</a> 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.







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<span class="h2"><a class="selflink" id="section-11" href="#section-11">11</a>.  IANA Considerations</span>

   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 <a href="#section-8">Section 8</a> for
   details.

<span class="h2"><a class="selflink" id="section-12" href="#section-12">12</a>.  Security Considerations</span>

   RTP packets using the payload format defined in this specification
   are subject to the general security considerations discussed in RTP
   [<a href="#ref-3" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">3</a>], 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 <a href="#section-5">Section 5</a> 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



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   the same key for the retransmitted stream and the original stream may
   lead to security problems, e.g., two-time pads.  Refer to <a href="#section-9.1">Section 9.1</a>
   of the Secure Real-Time Transport Protocol (SRTP) [<a href="#ref-12" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">12</a>] 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 <a href="#section-7">Section 7</a> of this document.

<span class="h2"><a class="selflink" id="section-13" href="#section-13">13</a>.  Acknowledgements</span>

   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.

<span class="h2"><a class="selflink" id="section-14" href="#section-14">14</a>.  References</span>

<span class="h3"><a class="selflink" id="section-14.1" href="#section-14.1">14.1</a>.  Normative References</span>

   [<a id="ref-1">1</a>]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
        "Extended RTP profile for Real-time Transport Control Protocol
        (RTCP)-Based feedback", <a href="./rfc4585">RFC 4585</a>, July 2006.

   [<a id="ref-2">2</a>]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
        Levels", <a href="https://www.rfc-editor.org/bcp/bcp14">BCP 14</a>, <a href="./rfc2119">RFC 2119</a>, March 1997.

   [<a id="ref-3">3</a>]  Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
        "RTP: A Transport Protocol for Real-Time Applications", STD 64,
        <a href="./rfc3550">RFC 3550</a>, July 2003.

   [<a id="ref-4">4</a>]  Casner, S., "Session Description Protocol (SDP) Bandwidth
        Modifiers for RTP Control Protocol (RTCP) Bandwidth", <a href="./rfc3556">RFC 3556</a>,
        July 2003.

   [<a id="ref-5">5</a>]  Handley, M. and V. Jacobson, "SDP: Session Description
        Protocol", <a href="./rfc2327">RFC 2327</a>, April 1998.

   [<a id="ref-6">6</a>]  Camarillo, G., Eriksson, G., Holler, J., and H. Schulzrinne,
        "Grouping of Media Lines in the Session Description Protocol
        (SDP)", <a href="./rfc3388">RFC 3388</a>, December 2002.




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   [<a id="ref-7">7</a>]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming
        Protocol (RTSP)", <a href="./rfc2326">RFC 2326</a>, April 1998.

<span class="h3"><a class="selflink" id="section-14.2" href="#section-14.2">14.2</a>.  Informative References</span>

   [<a id="ref-8">8</a>]  Perkins, C. and O. Hodson, "Options for Repair of Streaming
        Media", <a href="./rfc2354">RFC 2354</a>, June 1998.

   [<a id="ref-9">9</a>]  Hellstrom, G. and P. Jones, "RTP Payload for Text Conversation",
        <a href="./rfc4103">RFC 4103</a>, June 2005.

   [<a id="ref-10">10</a>] Handley, M., Floyd, S., Whetten, B., Kermode, R., Vicisano, L.,
        and M. Luby, "The Reliable Multicast Design Space for Bulk Data
        Transfer", <a href="./rfc2887">RFC 2887</a>, August 2000.

   [<a id="ref-11">11</a>] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol
        Extended Reports (RTCP XR)", <a href="./rfc3611">RFC 3611</a>, November 2003.

   [<a id="ref-12">12</a>] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
        Norrman, "The Secure Real-time Transport Protocol (SRTP)", <a href="./rfc3711">RFC</a>
        <a href="./rfc3711">3711</a>, March 2004.






























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<span class="h2"><a class="selflink" id="appendix-A" href="#appendix-A">Appendix A</a>.  How to Control the Number of Rtxs. per Packet</span>

   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.

<span class="h3"><a class="selflink" id="appendix-A.1" href="#appendix-A.1">A.1</a>.  Scenario and Assumptions</span>

   * 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 [<a href="#ref-1" title="&quot;Extended RTP profile for Real-time Transport Control Protocol (RTCP)-Based feedback&quot;">1</a>] 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



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     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-&gt;server)
   UL: uplink (server-&gt;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



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

<span class="h3"><a class="selflink" id="appendix-A.2" href="#appendix-A.2">A.2</a>.  Goal</span>

   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.

<span class="h3"><a class="selflink" id="appendix-A.3" href="#appendix-A.3">A.3</a>.  Solution</span>

   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 <a href="#appendix-A.7">Appendix A.7</a> of RTP [<a href="#ref-3" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">3</a>]),
   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)






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   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.

<span class="h3"><a class="selflink" id="appendix-A.4" href="#appendix-A.4">A.4</a>.  Numbers</span>

   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







<span class="grey">Rey, et al.                 Standards Track                    [Page 32]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-33" ></span>
<span class="grey"><a href="./rfc4588">RFC 4588</a>           RTP Retransmission Payload Format           July 2006</span>


   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












<span class="grey">Rey, et al.                 Standards Track                    [Page 33]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-34" ></span>
<span class="grey"><a href="./rfc4588">RFC 4588</a>           RTP Retransmission Payload Format           July 2006</span>


Authors' Addresses

   Jose Rey
   Panasonic R&amp;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&amp;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







<span class="grey">Rey, et al.                 Standards Track                    [Page 34]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-35" ></span>
<span class="grey"><a href="./rfc4588">RFC 4588</a>           RTP Retransmission Payload Format           July 2006</span>


Full Copyright Statement

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Rey, et al.                 Standards Track                    [Page 35]
</pre>