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Network Working Group                                             J. ReyRequest for Comments: 4588                                     PanasonicCategory: Standards Track                                        D. Leon                                                              Consultant                                                             A. Miyazaki                                                               Panasonic                                                                V. Varsa                                                                   Nokia                                                            R. Hakenberg                                                               Panasonic                                                               July 2006RTP Retransmission Payload FormatStatus 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 2006Table of Contents1. Introduction ....................................................32. Terminology .....................................................33. Requirements and Design Rationale for a Retransmission Scheme ...43.1. Multiplexing Scheme Choice .................................64. Retransmission Payload Format ...................................75. Association of Retransmission and Original Streams ..............95.1. Retransmission Session Sharing .............................95.2. CNAME Use ..................................................95.3. Association at the Receiver ................................96. Use with the Extended RTP Profile for RTCP-based Feedback ......116.1. RTCP at the Sender ........................................116.2. RTCP Receiver Reports .....................................116.3. Retransmission Requests ...................................126.4. Timing Rules ..............................................137. Congestion Control .............................................138. Retransmission Payload Format MIME Type Registration ...........158.1. Introduction ..............................................158.2. Registration of audio/rtx .................................168.3. Registration of video/rtx .................................178.4. Registration of text/rtx ..................................188.5. Registration of application/rtx ...........................198.6. Mapping to SDP ............................................208.7. SDP Description with Session-Multiplexing .................208.8. SDP Description with SSRC-Multiplexing ....................219. RTSP Considerations ............................................229.1. RTSP Control with SSRC-Multiplexing .......................229.2. RTSP Control with Session-Multiplexing ....................229.3. RTSP Control of the Retransmission Stream .................239.4. Cache Control .............................................2310. Implementation Examples .......................................2310.1. A Minimal Receiver Implementation Example ................2410.2. Retransmission of Layered Encoded Media in Multicast .....2511. IANA Considerations ...........................................2612. Security Considerations .......................................2613. Acknowledgements ..............................................2714. References ....................................................2714.1. Normative References .....................................2714.2. Informative References ...................................28Appendix A. How to Control the Number of Rtxs. per Packet .........29Rey, et al.                 Standards Track                     [Page 2]

RFC 4588           RTP Retransmission Payload Format           July 20061.  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 inRFC 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 inSection 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(seeSection 5.3 for motivation).Rey, et al.                 Standards Track                     [Page 6]

RFC 4588           RTP Retransmission Payload Format           July 20064.  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 toSection 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.  SeeSection 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, theRey, 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 Streams5.1.  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 asRey, 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.  SeeSection 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 20066.  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 20066.3.  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 timerRey, 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 20068.  Retransmission Payload Format MIME Type Registration8.1.  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 20068.2.  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: seeSection 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 IESGRey, et al.                 Standards Track                    [Page 16]

RFC 4588           RTP Retransmission Payload Format           July 20068.3.  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: seeSection 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 IESGRey, et al.                 Standards Track                    [Page 17]

RFC 4588           RTP Retransmission Payload Format           July 20068.4.  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: seeSection 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 IESGRey, et al.                 Standards Track                    [Page 18]

RFC 4588           RTP Retransmission Payload Format           July 20068.5.  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: seeSection 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 IESGRey, et al.                 Standards Track                    [Page 19]

RFC 4588           RTP Retransmission Payload Format           July 20068.6.  Mapping to SDP   The information carried in the MIME media type specification has a   specific mapping to fields in SDP [5], which is commonly used to   describe RTP sessions.  When SDP is used to specify retransmissions   for an RTP stream, the mapping is done as follows:   -  The MIME types ("video"), ("audio"), ("text"), and ("application")      go in the SDP "m=" as the media name.   -  The MIME subtype ("rtx") goes in SDP "a=rtpmap" as the encoding      name.  The RTP clockrate in "a=rtpmap" MUST be that of the      retransmission payload type.  SeeSection 4 for details on this.   -  The AVPF profile-specific parameters "ack" and "nack" go in SDP      "a=rtcp-fb".  Several SDP "a=rtcp-fb" are used for several types      of feedback.  See the AVPF profile [1] for details.   -  The retransmission payload-format-specific parameters "apt" and      "rtx-time" go in the SDP "a=fmtp" as a semicolon-separated list of      parameter=value pairs.   -  Any remaining parameters go in the SDP "a=fmtp" attribute by      copying them directly from the MIME media type string as a      semicolon-separated list of parameter=value pairs.   In the following sections, some example SDP descriptions are   presented.  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 inRFC 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 2Rey, 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=30008.8.  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=30009.  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 inSection 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 toRFC 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 perSection 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 200611.  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.  SeeSection 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 inSection 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 ofRey, 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 toSection 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 inSection 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.  References14.1.  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)",RFC3711, March 2004.Rey, et al.                 Standards Track                    [Page 28]

RFC 4588           RTP Retransmission Payload Format           July 2006Appendix 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 twoRey, 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-jitterRey, et al.                 Standards Track                    [Page 30]

RFC 4588           RTP Retransmission Payload Format           July 2006   Retransmissions processing time: T5 = feedback-processing-time +      rtx-queuing-timeA.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 (seeAppendix 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    10,08Rey, 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    10,07Rey, et al.                 Standards Track                    [Page 33]

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

RFC 4588           RTP Retransmission Payload Format           July 2006Full Copyright Statement   Copyright (C) The Internet Society (2006).   This document is subject to the rights, licenses and restrictions   contained inBCP 78, and except as set forth therein, the authors   retain all their rights.   This document and the information contained herein are provided on an   "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS   OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET   ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,   INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE   INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED   WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.Intellectual Property   The IETF takes no position regarding the validity or scope of any   Intellectual Property Rights or other rights that might be claimed to   pertain to the implementation or use of the technology described in   this document or the extent to which any license under such rights   might or might not be available; nor does it represent that it has   made any independent effort to identify any such rights.  Information   on the procedures with respect to rights in RFC documents can be   found inBCP 78 andBCP 79.   Copies of IPR disclosures made to the IETF Secretariat and any   assurances of licenses to be made available, or the result of an   attempt made to obtain a general license or permission for the use of   such proprietary rights by implementers or users of this   specification can be obtained from the IETF on-line IPR repository athttp://www.ietf.org/ipr.   The IETF invites any interested party to bring to its attention any   copyrights, patents or patent applications, or other proprietary   rights that may cover technology that may be required to implement   this standard.  Please address the information to the IETF at   ietf-ipr@ietf.org.Acknowledgement   Funding for the RFC Editor function is provided by the IETF   Administrative Support Activity (IASA).Rey, et al.                 Standards Track                    [Page 35]