Internet DRAFT - draft-ietf-avt-rtp-retransmission

draft-ietf-avt-rtp-retransmission





                                                                        
   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 
   Task Force (IETF), its areas, and its working groups. Note that 
   other groups may also distribute working documents as Internet-
   Drafts. 
    
   Internet-Drafts are draft documents valid for a maximum of six 
   months and may be updated, replaced, or obsoleted by other 
   documents at any time. It is inappropriate to use Internet-Drafts 
   as reference material or to cite them other than as "work in 
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   The list of current Internet-Drafts can be accessed at 
   http://www.ietf.org/1id-abstracts.txt 
    
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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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   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 
     
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   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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   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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