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Versions: 00 01 02 03 04 05 06 07 08 09 10 11 12 RFC 4588

   Internet Draft
   draft-ietf-avt-rtp-retransmission-                  J. Rey/Panasonic
   12.txt                                                 D. Leon/Nokia
                                                  A. Miyazaki/Panasonic
                                                         V. Varsa/Nokia
                                                 R. Hakenberg/Panasonic



   Expires: March 15, 2006                          September 15, 2005


                   RTP Retransmission Payload Format

   Status of this Memo

   By submitting this Internet-Draft, each author represents that any
   applicable patent or other IPR claims of which he or she is aware
   have been or will be disclosed, and any of which he or she becomes
   aware will be disclosed, in accordance with Section 6 of BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
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   [Note to RFC Editor:  This paragraph shall be deleted upon
   publication as an RFC.  References in this draft to RFC XXXX
   should be replaced with the RFC number assigned to this document.]

   Abstract

   RTP retransmission is an effective packet loss recovery technique
   for real-time applications with relaxed delay bounds.  This
   document describes an RTP payload format for performing
   retransmissions.  Retransmitted RTP packets are sent in a separate
   stream from the original RTP stream.  It is assumed that feedback
   from receivers to senders is available.  In particular, it is
   assumed that RTCP 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

   1. Introduction..................................................3
   2. Terminology...................................................3
   3. Requirements and design rationale for a retransmission scheme.4
    3.1 Multiplexing scheme choice..................................6
   4. Retransmission payload format.................................7
   5. Association of retransmission and original streams............9
    5.1 Retransmission session sharing..............................9
    5.2 CNAME use...................................................9
    5.3 Association at the receiver.................................9
   6. Use with the extended RTP profile for RTCP-based feedback....10
    6.1 RTCP at the sender.........................................11
    6.2 RTCP Receiver Reports......................................11
    6.3 Retransmission requests....................................11
    6.4 Timing rules...............................................12
   7. Congestion control...........................................13
   8. Retransmission Payload Format MIME type registration.........14
    8.1 Introduction...............................................14
    8.2 Registration of audio/rtx..................................15
    8.3 Registration of video/rtx..................................16
    8.4 Registration of text/rtx...................................17
    8.5 Registration of application/rtx............................17
    8.6 Mapping to SDP.............................................18
    8.7 SDP description with session-multiplexing..................19
    8.8 SDP description with SSRC-multiplexing.....................20
   9. RTSP considerations..........................................20
    9.1 RTSP control with SSRC-multiplexing........................21
    9.2 RTSP control with session-multiplexing.....................21
    9.3 RTSP control of the retransmission stream..................22
    9.4 Cache control..............................................22
   10. Implementation examples.....................................22
    10.1 A minimal receiver implementation example.................22
    10.2 Retransmission of Layered Encoded Media in Multicast......23
   11. IANA considerations.........................................24
   12. Security considerations.....................................24
   13. Acknowledgements............................................25
   14. References..................................................25
    14.1 Normative References......................................25
    14.2 Informative References....................................26
   15. Author's Addresses..........................................26
   Appendix A. How to control the number of rtxs. per packet.......27
   IPR Notices.....................................................31
   Full Copyright Statement........................................32









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1. Introduction

   Packet losses between an RTP sender and receiver may significantly
   degrade the quality of the received media.  Several techniques,
   such as forward error correction (FEC), retransmissions or
   interleaving may be considered to increase packet loss resiliency.
   RFC 2354 [8] discusses the different options.

   When choosing a repair technique for a particular application, the
   tolerable latency of the application has to be taken into account.
   In the case of multimedia conferencing, the end-to-end delay has
   to be at most a few hundred milliseconds in order to guarantee
   interactivity, which usually excludes the use of retransmission.

   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:

   Original packet: refers to an RTP packet which carries user data
   sent for the first time by an RTP sender.

   Original stream: refers to the RTP stream of original packets.

   Retransmission packet: refers to an RTP packet which is to be used
   by the receiver instead of a lost original packet.  Such a


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   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-multiplexing: scheme by which the original stream and the
   retransmission stream are sent in the same RTP session with
   different SSRC values.


   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
   NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL"
   in this document are to be interpreted as described in RFC 2119
   [2].


3. Requirements and design rationale for a retransmission scheme

   The use of retransmissions in RTP as a repair method for streaming
   media is appropriate in those scenarios with relaxed delay bounds
   and where full reliability is not a requirement.  More
   specifically, RTP retransmission allows to trade-off reliability
   vs. delay, i.e. the endpoints may give up retransmitting a lost
   packet after a given buffering time has elapsed.  Unlike TCP there
   is thus no head-of-line blocking caused by RTP retransmissions.
   The implementer should be aware that in cases where full
   reliability is required or higher delay and jitter can be
   tolerated, TCP or other transport options should be considered.

   The RTP retransmission scheme defined in this document is designed
   to fulfil the following set of requirements:

   1. It must not break general RTP and RTCP mechanisms.
   2. It must be suitable for unicast and small multicast groups.
   3. It must work with mixers and translators.
   4. It must work with all known payload types.
   5. It must not prevent the use of multiple payload types in a
      session.
   6. In order to support the largest variety of payload formats, the
      RTP receiver must be able to derive how many and which RTP
      packets were lost as a result of a gap in received RTP sequence
      numbers.  This requirement is referred to as sequence number

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      preservation.  Without such a requirement, it would be
      impossible to use retransmission with payload formats, such as
      conversational text [9] or most audio/video streaming
      applications, that use the RTP sequence number to detect lost
      packets.

   When designing a solution for RTP retransmission, several
   approaches may be considered for the multiplexing of the original
   RTP packets and the retransmitted RTP packets.

   One approach may be to retransmit the RTP packet with its original
   sequence number and send original and retransmission packets in
   the same RTP stream.  The retransmission packet would then be
   identical to the original RTP packet, i.e. the same header (and
   thus same sequence number) and the same payload.  However, such an
   approach is not acceptable because it would corrupt the RTCP
   statistics.  As a consequence, requirement 1 would not be met.
   Correct RTCP statistics require that for every RTP packet within
   the RTP stream, the sequence number be increased by one.

   Another approach may be to multiplex original RTP packets and
   retransmission packets in the same RTP stream using different
   payload type values.  With such an approach, the original packets
   and the retransmission packets would share the same sequence
   number space.  As a result, the RTP receiver would not be able to
   infer how many and which original packets (which sequence numbers)
   were lost.

   In other words, this approach does not satisfy the sequence number
   preservation requirement (requirement 6).  This in turn implies
   that requirement 4 would not be met.  Interoperability with mixers
   and translators would also be more difficult if they did not
   understand this new retransmission payload type in a sender RTP
   stream.  For these reasons, a solution based on payload type
   multiplexing of original packets and retransmission packets in the
   same RTP stream is excluded.

   Finally, the original and retransmission packets may be sent in
   two separate streams.  These two streams may be multiplexed either
   by sending them in two different sessions , i.e., session-
   multiplexing, or in the same session using different SSRC values,
   i.e. SSRC-multiplexing.  Since original and retransmission packets
   carry media of the same type, the objections in Section 5.2 of RTP
   [3] to RTP multiplexing do not apply in this case.

   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.

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   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 stream.  This allows streaming
   servers and middleware which are involved in a high number of
   concurrent sessions to minimise their port usage.

   This retransmission payload format allows both session-
   multiplexing and SSRC-multiplexing for unicast sessions.  From an
   implementation point of view, there is little difference between
   the two approaches.  Hence, in order to maximise interoperability,
   both multiplexing approaches SHOULD be supported by senders and
   receivers.  For multicast sessions, session-multiplexing MUST be
   used because the association of the original stream and the
   retransmission stream is problematic if SSRC-multiplexing is used
   with multicast sessions(see Section 5.3 for motivation).














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4. Retransmission payload format

   The format of a retransmission packet is shown below:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         RTP Header                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |            OSN                |                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               |
   |                  Original RTP Packet Payload                  |
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   The RTP header usage is as follows:

   In the case of session-multiplexing, the same SSRC value MUST be
   used for the original stream and the retransmission stream.  In
   the case of an SSRC collision in either the original session or
   the retransmission session, the RTP specification requires that an
   RTCP BYE packet MUST be sent in the session where the collision
   happened.  In addition, an RTCP BYE packet MUST also be sent for
   the associated stream in its own session.  After a new SSRC
   identifier is obtained, the SSRC of both streams MUST be set to
   this value.

   In the case of SSRC-multiplexing, two different SSRC values MUST
   be used for the original stream and the retransmission stream as
   required by RTP.  If an SSRC collision is detected for either the
   original stream or the retransmission stream, the RTP
   specification requires that an RTCP BYE packet MUST be sent for
   this stream.  An RTCP BYE packet MUST NOT be sent for the
   associated stream.  Therefore, only the stream that experienced
   SSRC collision MUST choose a new SSRC value.  Refer to Section 5.3
   for the implications on the original 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.

   Implementers have to be aware that the RTCP jitter value for the
   retransmission stream does not reflect the actual network jitter

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   since there could be little correlation between the time a packet
   is retransmitted and its original timestamp.

   The payload type is dynamic.  If multiple payload types using
   retransmission are present in the original stream, then for each
   of these, a dynamic payload type MUST be mapped to the
   retransmission payload format.  See Section 8.1 for the
   specification of how the mapping between original and
   retransmission payload types is done with 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 not longer apply to
   the retransmission packet and may need to be modified in the
   retransmission packet to reflect the change in rate.  The sender
   SHOULD trade-off the decrease in bandwidth usage with the decrease
   in quality caused by resending at a lower rate.

   If the original RTP header carried any profile-specific
   extensions, the retransmission packet SHOULD include the same
   extensions immediately following the fixed RTP header as expected
   by applications running under this profile.  In this case, the
   retransmission payload header 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

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   header, as specified in RTP [3].  In this case, the retransmission
   payload header MUST be placed after the header extension.

   If the original RTP packet contained RTP padding, that padding
   MUST be removed before constructing the retransmission packet.  If
   padding of the retransmission packet is needed, padding MUST be
   performed as with any RTP packets and the padding bit MUST be set.

   The marker bit (M), the CSRC count (CC) and the CSRC list of the
   original RTP header MUST be copied "as is" into the RTP header of
   the retransmission packet.


5. Association of retransmission and original streams

5.1 Retransmission session sharing

   In the case of session-multiplexing, a retransmission session MUST
   map to exactly one original session, i.e. the same retransmission
   session cannot be used for different original sessions.

   If retransmission session sharing were allowed, it would be a
   problem for receivers, since they would receive retransmissions
   for original sessions they might not have joined.  For example, a
   receiver wishing to receive only audio would receive also
   retransmitted video packets if an audio and video session shared
   the same retransmission session.


5.2 CNAME use

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

5.3 Association at the receiver

   A receiver receiving multiple original and retransmission streams
   needs to associate each retransmission stream with its original
   stream.  The association is done differently depending on whether
   session-multiplexing or SSRC-multiplexing is used.

   If session-multiplexing is used, the receiver associates the two
   streams having the same SSRC in the two sessions.  Note that the
   payload type field cannot be used to perform the association as
   several media streams may have the same payload type value.  The
   two sessions are themselves associated out-of-band.  See Section 8
   for how the grouping of the two sessions is done with SDP.

   If SSRC-multiplexing is used, the receiver should first of all
   look for two streams that have the same CNAME in the session.  In
   some cases, the CNAME may not be enough to determine the
   association as multiple original streams in the same session may

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   share the same CNAME.  For example, there can be in the same video
   session multiple video streams mapping to different SSRCs and
   still using the same CNAME and possibly the same PT values.  Each
   (or some) of these streams may have an associated retransmission
   stream.

   In this case, in order to find out the association between
   original and retransmission streams having the same CNAME, the
   receiver SHOULD behave as follows.

   The association can generally be resolved when the receiver
   receives a retransmission packet matching a retransmission request
   which had been sent earlier.  Upon reception of a retransmission
   packet whose original sequence number has been previously
   requested, the receiver can derive that the SSRC of the
   retransmission packet is associated to the sender SSRC from which
   the packet was requested.

   However, this mechanism might fail if there are two outstanding
   requests for the same packet sequence number in two different
   original streams of the session.  Note that since the initial
   packet sequence numbers are random, the probability of having two
   outstanding requests for the same packet sequence number would be
   very small.  Nevertheless, in order to avoid ambiguity in the
   unicast case, the receiver MUST NOT have two outstanding requests
   for the same packet sequence number in two different original
   streams before the association is resolved.  In multicast, this
   ambiguity cannot be completely avoided, because another receiver
   may have requested the same sequence number from another stream.
   Therefore, SSRC-multiplexing MUST NOT be used in multicast
   sessions.

   If the receiver discovers that two senders are using the same SSRC
   or if it receives an RTCP BYE packet, it MUST stop requesting
   retransmissions for that SSRC.  Upon reception of original RTP
   packets with a new SSRC, the receiver MUST perform the SSRC
   association again as described in this section.


6. Use with the extended RTP profile for RTCP-based feedback

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





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6.1 RTCP at the sender

   In the case of session-multiplexing, Sender Report (SR) packets
   for the original stream are sent in the original session and SR
   packets for the retransmission stream are sent in the
   retransmission session according to the rules of RTP.

   In the case of SSRC-multiplexing, SR packets for both original and
   retransmission streams are sent in the same session according to
   the rules of RTP.  The original and retransmission streams are
   seen, as far the RTCP bandwidth calculation is concerned, as
   independent senders belonging to the same RTP session and are thus
   equally sharing the RTCP bandwidth assigned to senders.

   Note that in both cases, session- and SSRC-multiplexing, BYE
   packets MUST still be sent for both streams as specified in RTP.
   In other words, it is not enough to send BYE packets for the
   original stream only.

6.2 RTCP Receiver Reports

   In the case of session-multiplexing, the receiver will send report
   blocks for the original stream and the retransmission stream in
   separate Receiver Report (RR) packets belonging to separate RTP
   sessions.  RR packets reporting on the original stream are sent in
   the original RTP session while RR packets reporting on the
   retransmission stream are sent in the retransmission session.  The
   RTCP bandwidth for these two sessions may be chosen independently
   (for example through RTCP bandwidth modifiers [4]).

   In the case of SSRC-multiplexing, the receiver sends report blocks
   for the original and the retransmission streams in the same RR
   packet since there is a single session.

6.3 Retransmission requests

   The NACK feedback message format defined in the AVPF profile
   SHOULD be used by receivers to send retransmission requests.
   Whether a receiver chooses to request a packet or not is an
   implementation issue.  An actual receiver implementation should
   take into account 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

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   measuring the retransmission delay to receive a retransmission
   packet after a NACK has been sent for that packet.  This estimate
   may also be obtained from past observations, RTCP report round-
   trip time if available or any other means.  A standard mechanism
   for the receiver to estimate the RTT is specified in RTP Extended
   Reports [11].

   The receiver should not send a retransmission request as soon as
   it detects a missing sequence number but should add some extra
   delay to compensate for packet reordering. This extra delay may,
   for example, be based on past observations of the experienced
   packet reordering. 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.  E.g., 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. Both,
   a static or an adaptive value MAY be used. E.g.: an adaptive timer
   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 to react more quickly to a given
   packet loss.  However, in that case if a new packet loss occurs
   right after the early RTCP packet was sent, the receiver will then
   have to wait for the next regular RTCP compound packet after the
   early packet.  Sending NACKs only in regular RTCP compound

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   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.

   The RTP profile under which the retransmission scheme is used
   defines an appropriate congestion control mechanism in different
   environments.  Following the rules under the profile, an RTP
   application can determine its acceptable bitrate and packet rate
   in order to be fair to other TCP or RTP flows.

   If an RTP application uses retransmission, the acceptable packet
   rate and bitrate includes both the original and retransmitted
   data.  This guarantees that an application using retransmission
   achieves the same fairness as one that does not.  Such a rule
   would translate in practice into the following actions:

   If enhanced service is used, it should be made sure that the total
   bitrate and packet rate do not exceed that of the requested
   service.  It should be further monitored that the requested
   services are actually delivered.  In a best-effort environment,
   the sender SHOULD NOT send retransmission packets without reducing
   the packet rate and bitrate of the original stream (for example by
   encoding the data at a lower rate).

   In addition, the sender MAY selectively retransmit only the
   packets that it deems important and ignore NACK messages for other
   packets in order to limit the bitrate.

   These congestion control mechanisms should keep the packet loss
   rate within acceptable parameters. 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

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   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.


8. Retransmission Payload Format MIME type registration

8.1 Introduction

   The following MIME subtype name and parameters are introduced in
   this document: "rtx", "rtx-time" and "apt".

   The binding used for the retransmission stream to the payload type
   number is indicated by an rtpmap attribute.  The MIME subtype name
   used in the binding is "rtx".

   The "apt" (associated payload type) parameter MUST be used to map
   the retransmission payload type to the associated original stream
   payload type.  If multiple original payload types are used, then
   multiple "apt" parameters MUST be included to map each original
   payload type to a different retransmission payload type.

   An OPTIONAL payload-format-specific parameter, "rtx-time",
   indicates the maximum time a sender will keep an original RTP
   packet in its buffers available for retransmission.  This time
   starts with the first transmission of the packet.

   The syntax is as follows:

        a=fmtp:<number> apt=<apt-value>;rtx-time=<rtx-time-val>
   where,

        <number>: indicates the dynamic payload type number assigned
        to the retransmission payload format in an rtpmap attribute.

        <apt-value>: 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

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        absence of the rtx-time parameter for a retransmission stream
        means that the maximum retransmission time is not defined,
        but MAY be negotiated by other means.


8.2 Registration of audio/rtx

   MIME type: audio

   MIME subtype: rtx

   Required parameters:

        rate: the RTP timestamp clockrate is equal to the RTP
        timestamp clockrate of the media that is retransmitted.

        apt: associated payload type.  The value of this parameter is
        the payload type of the associated original stream.


   Optional parameters:

        rtx-time: indicates the time in milliseconds (measured from
        the time a packet was first sent) that the sender keeps an
        RTP packet in its buffers available for retransmission.


   Encoding considerations: this type is only defined for transfer
   via RTP.

   Security considerations: see Section 12 of RFC XXXX

   Interoperability considerations: none

   Published specification: RFC XXXX

   Applications which use this media type: multimedia streaming
   applications

   Additional information: none

   Person & email address to contact for further information:
   jose.rey@eu.panasonic.com
   david.leon@nokia.com
   avt@ietf.org

   Intended usage: COMMON

   Authors:
   Jose Rey
   David Leon

   Change controller:

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   IETF AVT WG delegated from the IESG

8.3 Registration of video/rtx

   MIME type: video

   MIME subtype: rtx

   Required parameters:

        rate: the RTP timestamp clockrate is equal to the RTP
        timestamp clockrate of the media that is retransmitted.

        apt: associated payload type.  The value of this parameter is
        the payload type of the associated original stream.


   Optional parameters:

        rtx-time: indicates the time in milliseconds (measured from
        the time a packet was first sent) that the sender keeps an
        RTP packet in its buffers available for retransmission.


   Encoding considerations: this type is only defined for transfer
   via RTP.

   Security considerations: see Section 12 of RFC XXXX

   Interoperability considerations: none

   Published specification: RFC XXXX

   Applications which use this media type: multimedia streaming
   applications

   Additional information: none

   Person & email address to contact for further information:
   jose.rey@eu.panasonic.com
   david.leon@nokia.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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8.4 Registration of text/rtx

   MIME type: text

   MIME subtype: rtx

   Required parameters:

        rate: the RTP timestamp clockrate is equal to the RTP
        timestamp clockrate of the media that is retransmitted.

        apt: associated payload type.  The value of this parameter is
        the payload type of the associated original stream.

   Optional parameters:

        rtx-time: indicates the time in milliseconds (measured from
        the time a packet was first sent) that the sender keeps an
        RTP packet in its buffers available for retransmission.


   Encoding considerations: this type is only defined for transfer
   via RTP.

   Security considerations: see Section 12 of RFC XXXX

   Interoperability considerations: none

   Published specification: RFC XXXX

   Applications which use this media type: multimedia streaming
   applications

   Additional information: none

   Person & email address to contact for further information:
   jose.rey@eu.panasonic.com
   david.leon@nokia.com
   avt@ietf.org

   Intended usage: COMMON

   Authors:
   Jose Rey
   David Leon

   Change controller:
   IETF AVT WG delegated from the IESG

8.5 Registration of application/rtx

   MIME type: application


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   MIME subtype: rtx

   Required parameters:

        rate: the RTP timestamp clockrate is equal to the RTP
        timestamp clockrate of the media that is retransmitted.

        apt: associated payload type.  The value of this parameter is
        the payload type of the associated original stream.

   Optional parameters:

        rtx-time: indicates the time in milliseconds (measured from
        the time a packet was first sent) that the sender keeps an
        RTP packet in its buffers available for retransmission.

   Encoding considerations: this type is only defined for transfer
   via RTP.

   Security considerations: see Section 12 of RFC XXXX

   Interoperability considerations: none

   Published specification: RFC XXXX

   Applications which use this media type: multimedia streaming
   applications

   Additional information: none

   Person & email address to contact for further information:
   jose.rey@eu.panasonic.com
   david.leon@nokia.com
   avt@ietf.org

   Intended usage: COMMON

   Authors:
   Jose Rey
   David Leon

   Change controller:
   IETF AVT WG delegated from the IESG

8.6 Mapping to SDP

   The information carried in the MIME media type specification has a
   specific mapping to fields in SDP [5], which is commonly used to
   describe RTP sessions.  When SDP is used to specify
   retransmissions for an RTP  stream, the mapping is done as
   follows:



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   -  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 clock rate in "a=rtpmap" MUST be that of the
   retransmission payload type.  See Section 4 for details on this.

   -  The AVPF profile-specific parameters "ack" and "nack" go in SDP
   "a=rtcp-fb".  Several SDP "a=rtcp-fb" are used for several types
   of feedback.  See the AVPF profile [1] for details.

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

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

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

8.7 SDP description with session-multiplexing

   In the case of session-multiplexing, the SDP description contains
   one media specification "m" line per RTP session.  The SDP MUST
   provide the grouping of the original and associated retransmission
   sessions' "m" lines, using the Flow Identification (FID) semantics
   defined in RFC 3388 [6].

   The following example specifies two original, AMR and MPEG-4,
   streams on ports 49170 and 49174 and their corresponding
   retransmission streams on ports 49172 and 49176, respectively:

   v=0
   o=mascha 2980675221 2980675778 IN IP4 host.example.net
   c=IN IP4 192.0.2.0
   a=group:FID 1 2
   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

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   a=rtcp-fb:98 nack
   a=fmtp:98 profile-level-id=8;config=01010000012000884006682C209\
   0A21F
   a=mid:3
   m=video 49176 RTP/AVPF 99
   a=rtpmap:99 rtx/90000
   a=fmtp:99 apt=98;rtx-time=3000
   a=mid:4

   A special case of the SDP description is a description that
   contains only one original session "m" line and one retransmission
   session "m" line, the grouping is then obvious and FID semantics
   MAY be omitted in this special case only.

   This is illustrated in the following example, which is an SDP
   description for a single original MPEG-4 stream and its
   corresponding retransmission session:

   v=0
   o=mascha 2980675221 2980675778 IN IP4 host.example.net
   c=IN IP4 192.0.2.0
   m=video 49170 RTP/AVPF 96
   a=rtpmap:96 MP4V-ES/90000
   a=rtcp-fb:96 nack
   a=fmtp:96 profile-level-id=8;config=01010000012000884006682C209\
   0A21F
   m=video 49172 RTP/AVPF 97
   a=rtpmap:97 rtx/90000
   a=fmtp:97 apt=96;rtx-time=3000

8.8 SDP description with SSRC-multiplexing

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

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


9. RTSP considerations

   The Real-time Streaming Protocol (RTSP), RFC 2326 [7] is an
   application-level protocol for control over the delivery of data


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   with real-time properties.  This section looks at the issues
   involved in controlling RTP sessions that use retransmissions.

9.1 RTSP control with SSRC-multiplexing

   In the case of SSRC-multiplexing, the "m" line includes both
   original and retransmission payload types and has a single RTSP
   "control" attribute.  The receiver uses the "m" line to request
   SETUP and TEARDOWN of the whole media session.  The RTP profile
   contained in the Transport header MUST be the AVPF profile or
   another suitable profile allowing extended feedback.  If the SSRC
   value is included in the SETUP response's Transport header, it
   MUST be that of the original stream.

   In order to control the sending of the session original media
   stream, the receiver sends as usual PLAY and PAUSE requests to the
   sender for the session.  The RTP-info header that is used to set
   RTP-specific parameters in the PLAY response MUST be set according
   to the RTP information of the original stream.

   When the receiver starts receiving the original stream, it can
   then request retransmission through RTCP NACKs without additional
   RTSP signalling.

9.2 RTSP control with session-multiplexing

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

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

   The RTSP presentation SHOULD support aggregate control and SHOULD
   contain a session level RTSP URL.  The receiver SHOULD use
   aggregate control for an original session and its associated
   retransmission session.  Otherwise, there would need to be two
   different 'session-id' values, i.e. different values for the
   original and retransmission sessions, and the sender would not
   know how to associate them.

   The session-level "control" attribute is then used as usual to
   control the playing of the original stream.  When the receiver
   starts receiving the original stream, it can then request
   retransmissions through RTCP without additional RTSP signalling.



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9.3 RTSP control of the retransmission stream

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

9.4 Cache control

   Retransmission streams SHOULD NOT be cached.

   In the case of session-multiplexing, the "Cache-Control" header
   SHOULD be set to "no-cache" for the retransmission stream.

   In the case of SSRC-multiplexing, RTSP cannot specify independent
   caching for the retransmission stream, because there is a single
   "m" line in SDP.  Therefore, the implementer should take this fact
   into account when deciding whether to cache an SSRC-multiplexed
   session or not.


10. Implementation examples

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

   This section gives an overview of different implementation options
   allowed within this specification.

   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:

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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=rtpmap:97 rtx/90000
   a=fmtp:97 apt=96;rtx-time=3000

   The format-specific parameter "rtx-time" indicates that the server
   will buffer the sent packets in a retransmission buffer for 3.0
   seconds, after which the packets are deleted from the
   retransmission buffer and will never be sent again.

   In this implementation example, the required RTP receiver
   processing to handle retransmission is kept to a minimum.  The
   receiver detects packet loss from the gaps observed in the
   received sequence numbers.  It signals lost packets to the sender
   through NACKs as defined in the AVPF profile [1].  The receiver
   should take into account the signalled sender retransmission
   buffer length in order to dimension its own reception buffer.  It
   should also derive from the buffer length the maximum number of
   times the retransmission of a packet can be requested.

   The sender should retransmit the packets selectively, i.e. it
   should choose whether to retransmit a requested packet depending
   on the packet importance, the observed QoS and congestion state of
   the network connection to the receiver.  Obviously, the sender
   processing 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 not intended as a complete solution to reliable
   multicast.  Refer to RFC 2887 [10], for an overview of the
   problems related with reliable multicast transmission.

   Packets of different importance are sent in different RTP
   sessions.  The retransmission streams corresponding to the
   different layers can themselves be seen as different
   retransmission layers.  The relative importance of the different
   retransmission streams should reflect the relative importance of
   the different original streams.

   In multicast, SSRC-multiplexing of the original and retransmission
   streams is not allowed as per Section 5.3 of this document.  For
   this reason, the retransmission stream(s) MUST be sent in
   different RTP session(s) using session-multiplexing.


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   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.


11. IANA considerations

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


12. Security considerations

   RTP packets using the payload format defined in this specification
   are subject to the general security considerations discussed in
   RTP, 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, tampered
   retransmission requests may invalidate the SSRC association
   mechanism described in Section 5 of this document.  On the other
   hand, replayed packets could lead to false re-ordering and RTT
   measurements (required for the retransmission request strategy)
   and may cause the receiver buffer to overflow.

   Further, 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.


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   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 the same key for the retransmitted stream and
   the original stream may lead to security problems, e. g., two-time
   pads.  Refer to Section 9.1 of the Secure Real-Time Transport
   Protocol (SRTP)[12] for a discussion the implications of two-time
   pads and how to avoid them.

   At the time of writing this document, SRTP does not provide all
   the security services mentioned. There are, at least, two reasons
   for this: 1) the occurrence of two-time pads and 2) the fact that
   this payload format typically works under the RTP/AVPF profile
   while SRTP only supports RTP/AVP.  An adapted variant of SRTP
   shall solve these shortcomings in the future.

   Congestion control considerations with the use of retransmission
   are dealt with in Section 7 of this document.


13. Acknowledgements

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

14. References

14.1 Normative References

   1 J. Ott, S. Wenger, N. Sato, C. Burmeister, J. Rey, "Extended RTP
     profile for RTCP-based feedback", draft-ietf-avt-rtcp-feedback-
     11.txt, August 2004.

   2 S. Bradner, "Key words for use in RFCs to Indicate Requirement
     Levels", BCP 14, RFC 2119, March 1997

   3 H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson, "RTP: A
     Transport Protocol for Real-Time Applications", RFC 3550, July
     2003.

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   4 S. Casner, "SDP bandwidth modifiers for RTCP bandwidth", RFC
     3556, July 2003.

   5 M. Handley, V. Jacobson, "SDP: Session Description Protocol",
     RFC 2327, April 1998.

   6 G. Camarillo, J. Holler, G. AP. Eriksson, "Grouping of media
     lines in the Session Description Protocol (SDP)", RFC 3388,
     December 2002.

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

14.2 Informative References

   8 C. Perkins, O. Hodson, "Options for Repair of Streaming Media",
     RFC 2354, June 1998.

   9 G. Hellstrom, "RTP for conversational text", RFC 2793, May 2000

   10 M. Handley, et al., "The Reliable Multicast Design Space for
     Bulk Data Transfer", RFC 2887, August 2000.

   11 Friedman, et. al., "RTP Extended Reports", Work in Progress.

   12 M. Baugher, D. A. McGrew, D. Oran, R. Blom, E. Carrara, M.
     Naslund, K. Norrman, "The Secure Real-Time Transport Protocol",
     RFC 3711, March 2004.

   13 R. Hovey and S. Bradner, "The Organizations Involved in the IETF
     Standards Process," BCP 11, RFC 2028, IETF, October 1996.


15. Author's Addresses

   Jose Rey                                jose.rey@eu.panasonic.com
   Panasonic R&D Center Germany GmbH
   Monzastr. 4c
   D-63225 Langen, Germany
   Phone: +49-6103-766-134
   Fax:   +49-6103-766-166

   David Leon                                   david.leon@nokia.com
   Nokia Research Center
   6000 Connection Drive
   Irving, TX. USA
   Phone:  1-972-374-1860

   Akihiro Miyazaki                miyazaki.akihiro@jp.panasonic.com
   CE Architecture Development Center
   Matsushita Electric Industrial Co., Ltd.
   1006 Kadoma, Kadoma City, Osaka 571-8501, Japan

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   Internet Draft   RTP Retransmission Payload Format  September 2005


   Phone: +81-6-6900-9172
   Fax:   +81-6-6900-9173

   Viktor Varsa                               viktor.varsa@nokia.com
   Nokia Research Center
   6000 Connection Drive
   Irving, TX. USA
   Phone:  1-972-374-1861

   Rolf Hakenberg                    rolf.hakenberg@eu.panasonic.com
   Panasonic R&D Center Germany GmbH
   Monzastr. 4c
   D-63225 Langen, Germany
   Phone: +49-6103-766-162
   Fax:   +49-6103-766-166

Appendix A. How to control the number of rtxs. per packet

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

   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.

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   * 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 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 may be requested for
   retransmission a maximum of 2 times, the corresponding generic
   NACK report block requesting that particular packet is sent in two
   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 approx. 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 feedback processing and queuing
   time for retransmissions (T5).  Implementers shall assign
   appropriate values to these two parameters .

   Graphically, we have:

           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 bit 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

   Rey, et al.                                               [Page 28]

   Internet Draft   RTP Retransmission Payload Format  September 2005



   Time to report packet loss      : T3= time-to-next-rtcp-report

   UL transmission time            : T4= physical-delay-UL +
    transmission-delay-UL + interarrival-delay-jitter

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

   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.

   Solution
   ========
   First we find the value of the estimate for 1 retransmission,
   T(1)=T:

     T = T1 + T2 + T3 + T4 + T5

   Since T1 + T4 ~= RTT,

     T = RTT + T2 + T3 + T5

   The worst case for T3 would be that we assume that reporting has
   to wait a whole RTCP interval and that the maximum randomization
   factor of 1.5 is applied.  Therefore, after applying the
   subsequent compensation to avoid traffic bursts (see RTP Section
   A.7 [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.
   This includes  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=1/3)*(12 + 4*N) =
                   = 124 + 4*N/3.


   Numbers
   ========
   If we ignore the effect of T2 and T5, i.e., assume 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                   |
   |============|=====|======================================|

                        1,00    2,00    5,00    7,00    10,00
   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


   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,

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   Internet Draft   RTP Retransmission Payload Format  September 2005


   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                   |
   |============|=====|======================================|
                        1,00    2,00    5,00    7,00    10,00
   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


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   Rey, et al.                                               [Page 31]

   Internet Draft   RTP Retransmission Payload Format  September 2005


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   Rey, et al.                                               [Page 32]

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