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'10' on line 1135 looks like a reference -- Missing reference section? '12' on line 1209 looks like a reference Summary: 2 errors (**), 1 flaw (~~), 7 warnings (==), 14 comments (--). Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Internet Draft 3 draft-ietf-avt-rtp-retransmission- J. Rey/Matsushita 4 07.txt D. Leon/Nokia 5 A. Miyazaki/Matsushita 6 V. Varsa/Nokia 7 R. Hakenberg/Matsushita 9 Expires: November 2003 April 2003 11 RTP Retransmission Payload Format 13 Status of this Memo 15 This document is an Internet-Draft and is in full conformance 16 with all provisions of Section 10 of RFC 2026. 18 Internet-Drafts are working documents of the Internet Engineering 19 Task Force (IETF), its areas, and its working groups. Note that 20 other groups may also distribute working documents as Internet- 21 Drafts. 23 Internet-Drafts are draft documents valid for a maximum of six 24 months and may be updated, replaced, or obsoleted by other 25 documents at any time. It is inappropriate to use Internet-Drafts 26 as reference material or to cite them other than as "work in 27 progress." 29 The list of current Internet-Drafts can be accessed at 30 http://www.ietf.org/ietf/1id-abstracts.txt 31 The list of Internet-Draft Shadow Directories can be accessed at 32 http://www.ietf.org/shadow.html. 34 Copyright Notice 36 Copyright (C) The Internet Society (2003). All Rights Reserved. 38 [Note to RFC Editor: This paragraph is to be deleted when this 39 draft is published as an RFC. References in this draft to RFC 40 XXXX should be replaced with the RFC number assigned to this 41 document.] 43 Abstract 45 RTP retransmission is an effective packet loss recovery technique 46 for real-time applications with relaxed delay bounds. This 47 document describes an RTP payload format for performing 48 retransmissions. Retransmitted RTP packets are sent in a separate 49 stream from the original RTP stream. It is assumed that feedback 50 from receivers to senders is available. In particular, it is 51 assumed that RTCP feedback as defined in the extended RTP profile 52 for RTCP-based feedback (denoted RTP/AVPF), is available in this 53 memo. 55 Table of Contents 57 1. Introduction..................................................3 58 2. Terminology...................................................3 59 3. Requirements and design rationale for a retransmission scheme.4 60 4. Retransmission payload format.................................6 61 5. Asocciation of a retransmission stream to its original stream.8 62 6. Use with the extended RTP profile for RTCP-based feedback....10 63 7. Congestion control...........................................12 64 8. Retransmission Payload Format MIME type registration.........13 65 9. RTSP considerations..........................................19 66 10. Implementation examples.....................................21 67 11. IANA considerations.........................................24 68 12. Security considerations.....................................24 69 13. Acknowledgements............................................24 70 14. References..................................................25 71 15. Author's Addresses..........................................26 72 IPR Notices.....................................................26 73 Full Copyright Statement........................................27 75 1. Introduction 77 Packet losses between an RTP sender and receiver may significantly 78 degrade the quality of the received media. Several techniques, 79 such as forward error correction (FEC), retransmissions or 80 interleaving may be considered to increase packet loss resiliency. 81 RFC 2354 [8] discusses the different options. 83 When choosing a repair technique for a particular application, the 84 tolerable latency of the application has to be taken into account. 85 In the case of multimedia conferencing, the end-to-end delay has 86 to be at most a few hundred milliseconds in order to guarantee 87 interactivity, which usually excludes the use of retransmission. 89 However, in the case of multimedia streaming, the user can 90 tolerate an initial latency as part of the session set-up and thus 91 an end-to-end delay of several seconds may be acceptable. 92 Retransmission may thus be considered for such applications. 94 This document specifies a retransmission method for RTP applicable 95 to unicast and (small) multicast groups: it defines a payload 96 format for retransmitted RTP packets and provides protocol rules 97 for the sender and the receiver involved in retransmissions. 99 Furthermore, this retransmission payload format was designed for 100 use with the extended RTP profile for RTCP-based feedback, AVPF 101 [1]. It may also be used with other RTP profiles defined in the 102 future. 104 The AVPF profile allows for more frequent feedback and for early 105 feedback. It defines a small number of general-purpose feedback 106 messages, e.g. ACKs and NACKs, as well as codec and application- 107 specific feedback messages. See [1] for details. 109 2. Terminology 111 The following terms are used in this document: 113 Original packet: refers to an RTP packet which carries user data 114 sent for the first time by an RTP sender. 116 Original stream: refers to the RTP stream of original packets. 118 Retransmission packet: refers to an RTP packet which is to be used 119 by the receiver instead of a lost original packet. Such a 120 retransmission packet is said to be associated with the original 121 RTP packet. 123 Retransmission request: a means by which an RTP receiver is able 124 to request that the RTP sender should send a retransmission packet 125 for a given original packet. Usually, an RTCP NACK packet as 126 specified in [1] is used as retransmission request for lost 127 packets. 129 Retransmission stream: the stream of retransmission packets 130 associated with an original stream. 132 Session-multiplexing: scheme by which the original stream and the 133 associated retransmission stream are sent into two different RTP 134 sessions. 136 SSRC-multiplexing: scheme by which the original stream and the 137 retransmission stream are sent in the same RTP session with 138 different SSRC values. 140 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL 141 NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" 142 in this document are to be interpreted as described in RFC 2119 143 [2]. 145 3. Requirements and design rationale for a retransmission scheme 147 The use of retransmissions in RTP as a repair method for streaming 148 media is appropriate in those scenarios with relaxed delay bounds 149 and where full reliability is not a requirement. More 150 specifically, RTP retransmission allows to trade-off reliability 151 vs. delay, i.e. the endpoints may give up retransmitting a lost 152 packet after a given buffering time has elapsed. Unlike TCP 153 there is thus no head-of-line blocking caused by RTP 154 retransmissions. The implementer should be aware that in cases 155 where full reliability is required or higher delay and jitter can 156 be tolerated, TCP or other transport options should be considered. 158 The RTP retransmission scheme defined in this document is designed 159 to fulfil the following set of requirements: 161 1. It must not break general RTP and RTCP mechanisms. 162 2. It must be suitable for unicast and small multicast groups. 163 3. It must work with mixers and translators. 164 4. It must work with all known payload types. 165 5. It must not prevent the use of multiple payload types in a 166 session. 167 6. In order to support the largest variety of payload formats, the 168 RTP receiver must be able to derive how many and which RTP 169 packets were lost as a result of a gap in received RTP sequence 170 numbers. This requirement is referred to as sequence number 171 preservation. Without such a requirement, it would be 172 impossible to use retransmission with payload formats, such as 173 conversational text [9] or most audio/video streaming 174 applications, that use the RTP sequence number to detect lost 175 packets. 177 When designing a solution for RTP retransmission, several 178 approaches may be considered for the multiplexing of the original 179 RTP packets and the retransmitted RTP packets. 181 One approach may be to retransmit the RTP packet with its original 182 sequence number and send original and retransmission packets in 183 the same RTP stream. The retransmission packet would then be 184 identical to the original RTP packet, i.e. the same header (and 185 thus same sequence number) and the same payload. However, such an 186 approach is not acceptable because it would corrupt the RTCP 187 statistics. As a consequence, requirement 1 would not be met. 188 Correct RTCP statistics require that for every RTP packet within 189 the RTP stream, the sequence number be increased by one. 191 Another approach may be to multiplex original RTP packets and 192 retransmission packets in the same RTP stream using different 193 payload type values. With such an approach, the original packets 194 and the retransmission packets would share the same sequence 195 number space. As a result, the RTP receiver would not be able to 196 infer how many and which original packets (which sequence numbers) 197 were lost. 199 In other words, this approach does not satisfy the sequence number 200 preservation requirement (requirement 6). This in turn implies 201 that requirement 4 would not be met. Interoperability with mixers 202 and translators would also be more difficult if they did not 203 understand this new retransmission payload type in a sender RTP 204 stream. For these reasons, a solution based on payload type 205 multiplexing of original packets and retransmission packets in the 206 same RTP stream is excluded. 208 Finally, the original and retransmission packets may be sent in 209 two separate streams. These two streams may be multiplexed either 210 by sending them in two different sessions , i.e. session- 211 multiplexing, or in the same session using different SSRC values, 212 i.e. SSRC-multiplexing. Since original and retransmission packets 213 carry media of the same type, the objections in Section 5.2 of RTP 214 [3] to RTP multiplexing do not apply in this case. 216 Mixers and translators may process the original stream and simply 217 discard the retransmission stream if they are unable to utilise 218 it. Using two separate streams thus satisfies all the 219 requirements listed in this section. 221 3.1 Multiplexing scheme choice 223 Session-multiplexing and SSRC-multiplexing have different pros and 224 cons: 226 Session-multiplexing is based on sending the retransmission stream 227 in a different RTP session (as defined in RTP [3]) from that of 228 the original stream, i.e. the original and retransmission streams 229 are sent to different network addresses and/or port numbers. 231 Having a separate session allows more flexibility. In multicast, 232 using two separate sessions for the original and the 233 retransmission streams allows a receiver to choose whether or not 234 to subscribe to the RTP session carrying the retransmission 235 stream. The original session may also be single-source multicast 236 while separate unicast sessions are used to convey retransmissions 237 to each of the receivers, which as a result will receive only the 238 retransmission packets they request. 240 The use of separate sessions also facilitates differential 241 treatment by the network and may simplify processing in mixers, 242 translators and packet caches. 244 With SSRC-multiplexing, a single session is needed for the 245 original and the retransmission stream. This allows streaming 246 servers and middleware which are involved in a high number of 247 concurrent sessions to minimise their port usage. 249 This retransmission payload format allows both session- 250 multiplexing and SSRC-multiplexing for unicast sessions. From an 251 implementation point of view, there is little difference between 252 the two approaches. Hence, in order to maximise interoperability, 253 both multiplexing approaches SHOULD be supported by senders and 254 receivers. For multicast sessions, session-multiplexing MUST be 255 used because the association of the original stream and the 256 retransmission stream is problematic if SSRC-multiplexing is used 257 with multicast sessions(see Section 5.3 for motivation). 259 4. Retransmission payload format 261 The format of a retransmission packet is shown below: 263 0 1 2 3 264 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 1 265 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 266 | RTP Header | 267 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 268 | OSN | | 269 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 270 | Original RTP Packet Payload | 271 | | 272 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 274 The RTP header usage is as follows: 276 In the case of session-multiplexing, the same SSRC value MUST be 277 used for the original stream and the retransmission stream. In 278 the case of an SSRC collision in either the original session or 279 the retransmission session, the RTP specification requires that an 280 RTCP BYE packet MUST be sent in the session where the collision 281 happened. In addition, an RTCP BYE packet MUST also be sent for 282 the associated stream in its own session. After a new SSRC 283 identifier is obtained, the SSRC of both streams MUST be set to 284 this value. 286 In the case of SSRC-multiplexing, two different SSRC values MUST 287 be used for the original stream and the retransmission stream as 288 required by RTP. If an SSRC collision is detected for either the 289 original stream or the retransmission stream, the RTP 290 specification requires that an RTCP BYE packet MUST be sent for 291 this stream. No RTCP BYE packet MUST be sent for the associated 292 stream. Therefore, only the stream that experienced SSRC 293 collision will choose a new SSRC value. Refer to Section 5.3 for 294 the implications on the original and retransmission stream SSRC 295 association at the receiver. 297 For either multiplexing scheme, the sequence number has the 298 standard definition, i.e. it MUST be one higher than the sequence 299 number of the preceding packet sent in the retransmission stream. 301 The retransmission packet timestamp is set to the original 302 timestamp, i.e. to the timestamp of the original packet. As a 303 consequence, the initial RTP timestamp for the first packet of the 304 retransmission stream is not random but equal to the original 305 timestamp of the first packet that is retransmitted. See the 306 security considerations section in this document for security 307 implications. 309 Implementers have to be aware that the RTCP jitter value for the 310 retransmission stream does not reflect the actual network jitter 311 since there could be little correlation between the time a packet 312 is retransmitted and its original timestamp. 314 The payload type is dynamic. Each payload type of the original 315 stream MUST map to a different payload type value in the 316 retransmission stream. Therefore, when multiple payload types are 317 used in the original stream, multiple dynamic payload types will 318 be mapped to the retransmission payload format. See Section 8.1 319 for the specification of how the mapping between original and 320 retransmission payload types is done with SDP. 322 As the retransmission packet timestamp carries the original media 323 timestamp, the timestamp clockrate used by the retransmission 324 payload type is the same as the one used by the associated 325 original payload type. It is thus possible to send retransmission 326 packets whose original payload types have different timestamp 327 clockrates in the same retransmission stream. Note that an RTP 328 stream does not usually carry payload types of different 329 clockrates. 331 The payload of the RTP retransmission packet comprises the 332 retransmission payload header followed by the payload of the 333 original RTP packet. The length of the retransmission payload 334 header is 2 octets. This payload header contains only one field, 335 OSN (original sequence number), which MUST be set to the sequence 336 number of the associated original RTP packet. The original RTP 337 packet payload, including any possible payload headers specific to 338 the original payload type, is placed right after the 339 retransmission payload header. 341 For payload types that support encoding at multiple rates, instead 342 of retransmitting the same payload as the original RTP packet the 343 sender MAY retransmit the same data encoded at a lower rate. This 344 aims at limiting the bandwidth usage of the retransmission stream. 345 When doing so, the sender MUST ensure that the receiver will still 346 be able to decode the payload of the already sent original packets 347 that might have been encoded based on the payload of the lost 348 original packet. In addition, if the sender chooses to retransmit 349 at a lower rate, the values in the payload header of the original 350 RTP packet may not longer apply to the retransmission packet and 351 may need to be modified in the retransmission packet to reflect 352 the change in rate. The sender should trade-off the decrease in 353 bandwidth usage with the decrease in quality caused by resending 354 at a lower rate. 356 If the original RTP header carried any profile-specific 357 extensions, the retransmission packet SHOULD include the same 358 extensions immediately following the fixed RTP header as expected 359 by applications running under this profile. In this case, the 360 retransmission payload header is thus placed after the profile- 361 specific extensions. 363 If the original RTP header carried an RTP header extension, the 364 retransmission packet SHOULD carry the same header extension. 365 This header extension MUST be placed right after the fixed RTP 366 header, as specified in RTP [3]. In this case, the retransmission 367 payload header is thus placed after the header extension. 369 If the original RTP packet contained RTP padding, that padding 370 MUST be removed before constructing the retransmission packet. If 371 padding of the retransmission packet is needed, padding is 372 performed as with any RTP packets and the padding bit is set. 374 The marker bit (M), the CSRC count (CC) and the CSRC list of the 375 original RTP header MUST be copied "as is" into the RTP header of 376 the retransmission packet. 378 5. Association of a retransmission stream to its original stream 380 5.1 Retransmission session sharing 382 In the case of session-multiplexing, a retransmission session MUST 383 map to exactly one original session, i.e. the same retransmission 384 session cannot be used for different original sessions. 386 If retransmission session sharing were allowed, it would be a 387 problem for receivers, since they would receive retransmissions 388 for original sessions they might not have joined. For example, a 389 receiver wishing to receive only audio would receive also 390 retransmitted video packets if an audio and video session shared 391 the same retransmission session. 393 5.2 CNAME use 395 In both the session-multiplexing and the SSRC-multiplexing cases, 396 a sender MUST use the same CNAME for an original stream and its 397 associated retransmission stream. 399 5.3 Association at the receiver 401 A receiver receiving multiple original and retransmission streams 402 needs to associate each retransmission stream with its original 403 stream. The association is done differently depending on whether 404 session-multiplexing or SSRC-multiplexing is used. 406 If session-multiplexing is used, the receiver associates the two 407 streams having the same SSRC in the two sessions. Note that the 408 payload type field cannot be used to perform the association as 409 several media streams may have the same payload type value. The 410 two sessions are themselves associated out-of-band. See Section 8 411 for how the grouping of the two sessions is done with SDP. 413 If SSRC-multiplexing is used, the receiver should first of all 414 look for two streams that have the same CNAME in the session. In 415 some cases, the CNAME may not be enough to determine the 416 association as multiple original streams in the same session may 417 share the same CNAME. For example, there can be in the same video 418 session multiple video streams mapping to different SSRCs and 419 still using the same CNAME and possibly the same PT values. Each 420 (or some) of these streams may have an associated retransmission 421 stream. 423 In this case, in order to find out the association between 424 original and retransmission streams having the same CNAME, the 425 receiver SHOULD behave as follows. 427 The association can generally be resolved when the receiver 428 receives a retransmission packet matching a retransmission request 429 which had been sent earlier. Upon reception of a retransmission 430 packet whose original sequence number has been previously 431 requested, the receiver can derive that the SSRC of the 432 retransmission packet is associated to the sender SSRC from which 433 the packet was requested. 435 However, this mechanism might fail if there are two outstanding 436 requests for the same packet sequence number in two different 437 original streams of the session. Note that since the initial 438 packet sequence numbers are random, the probability of having two 439 outstanding requests for the same packet sequence number would be 440 very small. Nevertheless, in order to avoid ambiguity in the 441 unicast case, the receiver MUST NOT have two outstanding requests 442 for the same packet sequence number in two different original 443 streams before the association is resolved. In multicast, this 444 ambiguity cannot be completely avoided, because another receiver 445 may have requested the same sequence number from another stream. 446 Therefore, SSRC-multiplexing MUST NOT be used in multicast 447 sessions. 449 If the receiver discovers that two senders are using the same SSRC 450 or if it receives an RTCP BYE packet, it MUST stop requesting 451 retransmissions for that SSRC. Upon reception of original RTP 452 packets with a new SSRC, the receiver MUST perform the SSRC 453 association again as described in this section. 455 6. Use with the extended RTP profile for RTCP-based feedback 457 This section gives general hints for the usage of this payload 458 format with the extended RTP profile for RTCP-based feedback, 459 denoted AVPF [1]. Note that the general RTCP send and receive 460 rules and the RTCP packet format as specified in RTP apply, except 461 for the changes that the AVPF profile introduces. In short, the 462 AVPF profile relaxes the RTCP timing rules and specifies 463 additional general-purpose RTCP feedback messages. See [1] for 464 details. 466 6.1 RTCP at the sender 468 In the case of session-multiplexing, Sender Report (SR) packets 469 for the original stream are sent in the original session and SR 470 packets for the retransmission stream are sent in the 471 retransmission session according to the rules of RTP. 473 In the case of SSRC-multiplexing, SR packets for both original and 474 retransmission streams are sent in the same session according to 475 the rules of RTP. The original and retransmission streams are 476 seen, as far the RTCP bandwidth calculation is concerned, as 477 independent senders belonging to the same RTP session and are thus 478 equally sharing the RTCP bandwidth assigned to senders. 480 Note that in both cases, session- and SSRC-multiplexing, BYE 481 packets MUST still be sent for both streams as specified in RTP. 482 In other words, it is not enough to send BYE packets for the 483 original stream only. 485 6.2 RTCP Receiver Reports 487 In the case of session-multiplexing, the receiver will send report 488 blocks for the original stream and the retransmission stream in 489 separate Receiver Report (RR) packets belonging to separate RTP 490 sessions. RR packets reporting on the original stream are sent in 491 the original RTP session while RR packets reporting on the 492 retransmission stream are sent in the retransmission session. The 493 RTCP bandwidth for these two sessions may be chosen independently 494 (for example through RTCP bandwidth modifiers [4]). 496 In the case of SSRC-multiplexing, the receiver sends report blocks 497 for the original and the retransmission streams in the same RR 498 packet since there is a single session. 500 6.3 Retransmission requests 502 The NACK feedback message format defined in the AVPF profile 503 SHOULD be used by receivers to send retransmission requests. 504 Whether a receiver chooses to request a packet or not is an 505 implementation issue. An actual receiver implementation should 506 take into account such factors as the tolerable application delay, 507 the network environment and the media type. 509 The receiver should generally assess whether the retransmitted 510 packet would still be useful at the time it is received. The 511 timestamp of the missing packet can be estimated from the 512 timestamps of packets preceding and/or following the sequence 513 number gap caused by the missing packet in the original stream. 514 In most cases, some form of linear estimate of the timestamp is 515 good enough. 517 Furthermore, a receiver should compute an estimate of the round- 518 trip time (RTT) to the sender. This can be done, for example, by 519 measuring the retransmission delay to receive a retransmission 520 packet after a NACK has been sent for that packet. This estimate 521 may also be obtained from past observations, RTCP report round- 522 trip time if available or any other means. A standard mechanism 523 for the receiver to estimate the RTT is specified in RTP Extended 524 Reports [11]. 526 The receiver should not send a retransmission request as soon as 527 it detects a missing sequence number but should add some extra 528 delay to compensate for packet reordering. This extra delay may, 529 for example, be based on past observations of the experienced 530 packet reordering. 532 To increase the robustness to the loss of a NACK or of a 533 retransmission packet, a receiver may send a new NACK for the same 534 packet. This is referred to as multiple retransmissions. Before 535 sending a new NACK for a missing packet, the receiver should rely 536 on a timer to be reasonably sure that the previous retransmission 537 attempt has failed in order to avoid unnecessary retransmissions. 539 NACKs MUST be sent only for the original RTP stream. Otherwise, 540 if a receiver wanted to perform multiple retransmissions by 541 sending a NACK in the retransmission stream, it would not be able 542 to know the original sequence number and a timestamp estimation of 543 the packet it requests. 545 6.4 Timing rules 547 The NACK feedback message may be sent in a regular full compound 548 RTCP packet or in an early RTCP packet, as per AVPF [1]. Sending 549 a NACK in an early packet allows to react more quickly to a given 550 packet loss. However, in that case if a new packet loss occurs 551 right after the early RTCP packet was sent, the receiver will then 552 have to wait for the next regular RTCP compound packet after the 553 early packet. Sending NACKs only in regular RTCP compound 554 decreases the maximum delay between detecting an original packet 555 loss and being able to send a NACK for that packet. Implementers 556 should consider the possible implications of this fact for the 557 application being used. 559 Furthermore, receivers may make use of the minimum interval 560 between regular RTCP compound packets. This interval can be used 561 to keep regular receiver reporting down to a minimum, while still 562 allowing receivers to send early RTCP packets during periods 563 requiring more frequent feedback, e.g. times of higher packet loss 564 rate.. Note that although RTCP packets may be suppressed because 565 they do not contain NACKs, the same RTCP bandwidth as if they were 566 sent needs to be available. See AVPF [1] for details on the use 567 of the minimum interval. 569 7. Congestion control 571 RTP retransmission poses a risk of increasing network congestion. 572 In a best-effort environment, packet loss is caused by congestion. 573 Reacting to loss by retransmission of older data without 574 decreasing the rate of the original stream would thus further 575 increase congestion. Implementations SHOULD follow the 576 recommendations below in order to use retransmission. 578 The RTP profile under which the retransmission scheme is used 579 defines an appropriate congestion control mechanism in different 580 environments. Following the rules under the profile, an RTP 581 application can determine its acceptable bitrate and packet rate 582 in order to be fair to other TCP or RTP flows. 584 If an RTP application uses retransmission, the acceptable packet 585 rate and bitrate includes both the original and retransmitted 586 data. This guarantees that an application using retransmission 587 achieves the same fairness as one that does not. Such a rule 588 would translate in practice into the following actions: 590 If enhanced service is used, it should be made sure that the total 591 bitrate and packet rate do not exceed that of the requested 592 service. It should be further monitored that the requested 593 services are actually delivered. In a best-effort environment, 594 the sender SHOULD NOT send retransmission packets without reducing 595 the packet rate and bitrate of the original stream (for example by 596 encoding the data at a lower rate). 598 In addition, the sender MAY selectively retransmit only the 599 packets that it deems important and ignore NACK messages for other 600 packets in order to limit the bitrate. 602 These congestion control mechanisms should keep the packet loss 603 rate within acceptable parameters. Packet loss is considered 604 acceptable if a TCP flow across the same network path and 605 experiencing the same network conditions would achieve, on a 606 reasonable timescale, an average throughput, that is not less than 607 the one the RTP flow achieves. If the packet loss rate exceeds an 608 acceptable level, it should be concluded that congestion is not 609 kept under control and retransmission should then not be used. It 610 may further be necessary to adapt the transmission rate (or the 611 number of layers subscribed for a layered multicast session), or 612 to arrange for the receiver to leave the session. 614 8. Retransmission Payload Format MIME type registration 616 8.1 Introduction 618 The following MIME subtype name and parameters are introduced in 619 this document: "rtx", "rtx-time" and "apt". 621 The binding used for the retransmission stream to the payload type 622 number is indicated by an rtpmap attribute. The MIME subtype name 623 used in the binding is "rtx". 625 The "apt" (associated payload type) parameter MUST be used to map 626 the retransmission payload type to the associated original stream 627 payload type. If multiple original payload types are used, then 628 multiple "apt" parameters MUST be included to map each original 629 payload type to a different retransmission payload type. 631 An OPTIONAL payload-format-specific parameter, "rtx-time", 632 indicates the maximum time a sender will keep an original RTP 633 packet in its buffers available for retransmission. This time 634 starts with the first transmission of the packet. 636 The syntax is as follows: 638 a=fmtp: apt=;rtx-time= 639 where, 641 : indicates the dynamic payload type number assigned 642 to the retransmission payload format in an rtpmap attribute. 644 : the value of the original stream payload type to 645 which this retransmission stream payload type is associated. 647 : specifies the time in milliseconds (measured 648 from the time a packet was first sent) that a sender keeps an 649 RTP packet in its buffers available for retransmission. The 650 absence of the rtx-time parameter for a retransmission stream 651 means that the maximum retransmission time is not defined, 652 but MAY be negotiated by other means. 654 8.2 Registration of audio/rtx 656 MIME type: audio 658 MIME subtype: rtx 660 Required parameters: 662 rate: the RTP timestamp clockrate is equal to the RTP 663 timestamp clockrate of the media that is retransmitted. 665 apt: associated payload type. The value of this parameter is 666 the payload type of the associated original stream. 668 Optional parameters: 670 rtx-time: indicates the time in milliseconds (measured from 671 the time a packet was first sent) that the sender keeps an 672 RTP packet in its buffers available for retransmission. 674 Encoding considerations: this type is only defined for transfer 675 via RTP. 677 Security considerations: see Section 12 of RFC XXXX 679 Interoperability considerations: none 681 Published specification: RFC XXXX 683 Applications which use this media type: multimedia streaming 684 applications 686 Additional information: none 688 Person & email address to contact for further information: 689 rey@panasonic.de 690 david.leon@nokia.com 691 avt@ietf.org 693 Intended usage: COMMON 695 Author/Change controller: 696 Jose Rey 697 David Leon 698 IETF AVT WG 700 8.3 Registration of video/rtx 702 MIME type: video 704 MIME subtype: rtx 706 Required parameters: 708 rate: the RTP timestamp clockrate is equal to the RTP 709 timestamp clockrate of the media that is retransmitted. 711 apt: associated payload type. The value of this parameter is 712 the payload type of the associated original stream. 714 Optional parameters: 716 rtx-time: indicates the time in milliseconds (measured from 717 the time a packet was first sent) that the sender keeps an 718 RTP packet in its buffers available for retransmission. 720 Encoding considerations: this type is only defined for transfer 721 via RTP. 723 Security considerations: see Section 12 of RFC XXXX 725 Interoperability considerations: none 727 Published specification: RFC XXXX 729 Applications which use this media type: multimedia streaming 730 applications 732 Additional information: none 734 Person & email address to contact for further information: 735 rey@panasonic.de 736 david.leon@nokia.com 737 avt@ietf.org 739 Intended usage: COMMON 741 Author/Change controller: 742 Jose Rey 743 David Leon 744 IETF AVT WG 746 8.4 Registration of text/rtx 748 MIME type: text 750 MIME subtype: rtx 751 Required parameters: 753 rate: the RTP timestamp clockrate is equal to the RTP 754 timestamp clockrate of the media that is retransmitted. 756 apt: associated payload type. The value of this parameter is 757 the payload type of the associated original stream. 759 Optional parameters: 761 rtx-time: indicates the time in milliseconds (measured from 762 the time a packet was first sent) that the sender keeps an 763 RTP packet in its buffers available for retransmission. 765 Encoding considerations: this type is only defined for transfer 766 via RTP. 768 Security considerations: see Section 12 of RFC XXXX 770 Interoperability considerations: none 772 Published specification: RFC XXXX 774 Applications which use this media type: multimedia streaming 775 applications 777 Additional information: none 779 Person & email address to contact for further information: 780 rey@panasonic.de 781 david.leon@nokia.com 782 avt@ietf.org 784 Intended usage: COMMON 786 Author/Change controller: 787 Jose Rey 788 David Leon 789 IETF AVT WG 791 8.5 Registration of application/rtx 793 MIME type: application 795 MIME subtype: rtx 797 Required parameters: 799 rate: the RTP timestamp clockrate is equal to the RTP 800 timestamp clockrate of the media that is retransmitted. 802 apt: associated payload type. The value of this parameter is 803 the payload type of the associated original stream. 805 Optional parameters: 807 rtx-time: indicates the time in milliseconds (measured from 808 the time a packet was first sent) that the sender keeps an 809 RTP packet in its buffers available for retransmission. 811 Encoding considerations: this type is only defined for transfer 812 via RTP. 814 Security considerations: see Section 12 of RFC XXXX 816 Interoperability considerations: none 818 Published specification: RFC XXXX 820 Applications which use this media type: multimedia streaming 821 applications 823 Additional information: none 825 Person & email address to contact for further information: 826 rey@panasonic.de 827 david.leon@nokia.com 828 avt@ietf.org 830 Intended usage: COMMON 832 Author/Change controller: 833 Jose Rey 834 David Leon 835 IETF AVT WG 837 8.6 Mapping to SDP 839 The information carried in the MIME media type specification has a 840 specific mapping to fields in SDP [5], which is commonly used to 841 describe RTP sessions. When SDP is used to specify 842 retransmissions for an RTP stream, the mapping is done as 843 follows: 845 - The MIME types ("video"), ("audio"), ("text") and 846 ("application") go in the SDP "m=" as the media name. 848 - The MIME subtype ("rtx") goes in SDP "a=rtpmap" as the encoding 849 name. The RTP clock rate in "a=rtpmap" MUST be that of the 850 retransmission payload type. See Section 4 for details on this. 852 - The AVPF profile-specific parameters "ack" and "nack" go in SDP 853 "a=rtcp-fb". Several SDP "a=rtcp-fb" are used for several types 854 of feedback. See the AVPF profile [1] for details. 856 - The retransmission payload-format-specific parameters "apt" and 857 "rtx-time" go in the SDP "a=fmtp" as a semicolon separated list of 858 parameter=value pairs. 860 - Any remaining parameters go in the SDP "a=fmtp" attribute by 861 copying them directly from the MIME media type string as a 862 semicolon separated list of parameter=value pairs. 864 In the following sections some example SDP descriptions are 865 presented. In some of these examples, long lines are folded to 866 meet the column width constraints of this document; the backslash 867 ("\") at the end of a line and the carriage return that follows it 868 should be ignored. 870 8.7 SDP description with session-multiplexing 872 In the case of session-multiplexing, the SDP description contains 873 one media specification "m" line per RTP session. The SDP MUST 874 provide the grouping of the original and associated retransmission 875 sessions' "m" lines, using the Flow Identification (FID) semantics 876 defined in RFC 3388 [6]. 878 The following example specifies two original, AMR and MPEG-4, 879 streams on ports 49170 and 49174 and their corresponding 880 retransmission streams on ports 49172 and 49176, respectively: 882 v=0 883 o=mascha 2980675221 2980675778 IN IP4 host.example.net 884 c=IN IP4 192.0.2.0 885 a=group:FID 1 2 886 a=group:FID 3 4 887 m=audio 49170 RTP/AVPF 96 888 a=rtpmap:96 AMR/8000 889 a=fmtp:96 octet-align=1 890 a=rtcp-fb:96 nack 891 a=mid:1 892 m=audio 49172 RTP/AVPF 97 893 a=rtpmap:97 rtx/8000 894 a=fmtp:97 apt=96;rtx-time=3000 895 a=mid:2 896 m=video 49174 RTP/AVPF 98 897 a=rtpmap:98 MP4V-ES/90000 898 a=rtcp-fb:98 nack 899 a=fmtp:98 profile-level-id=8;config=01010000012000884006682C209\ 900 0A21F 901 a=mid:3 902 m=video 49176 RTP/AVPF 99 903 a=rtpmap:99 rtx/90000 904 a=fmtp:99 apt=98;rtx-time=3000 905 a=mid:4 907 A special case of the SDP description is a description that 908 contains only one original session "m" line and one retransmission 909 session "m" line, the grouping is then obvious and FID semantics 910 MAY be omitted in this special case only. 912 This is illustrated in the following example, which is an SDP 913 description for a single original MPEG-4 stream and its 914 corresponding retransmission session: 916 v=0 917 o=mascha 2980675221 2980675778 IN IP4 host.example.net 918 c=IN IP4 192.0.2.0 919 m=video 49170 RTP/AVPF 96 920 a=rtpmap:96 MP4V-ES/90000 921 a=rtcp-fb:96 nack 922 a=fmtp:96 profile-level-id=8;config=01010000012000884006682C209\ 923 0A21F 924 m=video 49172 RTP/AVPF 97 925 a=rtpmap:97 rtx/90000 926 a=fmtp:97 apt=96;rtx-time=3000 928 8.8 SDP description with SSRC-multiplexing 930 The following is an example of an SDP description for an RTP video 931 session using SSRC-multiplexing with similar parameters as in the 932 single-session example above: 934 v=0 935 o=mascha 2980675221 2980675778 IN IP4 host.example.net 936 c=IN IP4 192.0.2.0 937 m=video 49170 RTP/AVPF 96 97 938 a=rtpmap:96 MP4V-ES/90000 939 a=rtcp-fb:96 nack 940 a=fmtp:96 profile-level-id=8;config=01010000012000884006682C209\ 941 0A21F 942 a=rtpmap:97 rtx/90000 943 a=fmtp:97 apt=96;rtx-time=3000 945 9. RTSP considerations 947 The Real-time Streaming Protocol (RTSP), RFC 2326 [7] is an 948 application-level protocol for control over the delivery of data 949 with real-time properties. This section looks at the issues 950 involved in controlling RTP sessions that use retransmissions. 952 9.1 RTSP control with SSRC-multiplexing 954 In the case of SSRC-multiplexing, the "m" line includes both 955 original and retransmission payload types and has a single RTSP 956 "control" attribute. The receiver uses the "m" line to request 957 SETUP and TEARDOWN of the whole media session. The RTP profile 958 contained in the Transport header MUST be the AVPF profile or 959 another suitable profile allowing extended feedback. If the SSRC 960 value is included in the SETUP response's Transport header, it 961 MUST be that of the original stream. 963 In order to control the sending of the session original media 964 stream, the receiver sends as usual PLAY and PAUSE requests to the 965 sender for the session. The RTP-info header that is used to set 966 RTP-specific parameters in the PLAY response MUST be set according 967 to the RTP information of the original stream. 969 When the receiver starts receiving the original stream, it can 970 then request retransmission through RTCP NACKs without additional 971 RTSP signalling. 973 9.2 RTSP control with session-multiplexing 975 In the case of session-multiplexing, each SDP "m" line has an RTSP 976 "control" attribute. Hence, when retransmission is used, both the 977 original session and the retransmission have their own "control" 978 attributes. The receiver can associate the original session and 979 the retransmission session through the FID semantics as specified 980 in Section 8. 982 The original and the retransmission streams are set up and torn 983 down separately through their respective media "control" 984 attribute. The RTP profile contained in the Transport header MUST 985 be the AVPF profile or another suitable profile allowing extended 986 feedback for both the original and the retransmission session. 988 The RTSP presentation SHOULD support aggregate control and SHOULD 989 contain a session level RTSP URL. The receiver SHOULD use 990 aggregate control for an original session and its associated 991 retransmission session. Otherwise, there would need to be two 992 different 'session-id' values, i.e. different values for the 993 original and retransmission sessions, and the sender would not 994 know how to associate them. 996 The session-level "control" attribute is then used as usual to 997 control the playing of the original stream. When the receiver 998 starts receiving the original stream, it can then request 999 retransmissions through RTCP without additional RTSP signalling. 1001 9.3 RTSP control of the retransmission stream 1003 Because of the nature of retransmissions, the sending of 1004 retransmission packets SHOULD NOT be controlled through RTSP PLAY 1005 and PAUSE requests. The PLAY and PAUSE requests SHOULD NOT affect 1006 the retransmission stream. Retransmission packets are sent upon 1007 receiver requests in the original RTCP stream, regardless of the 1008 state. 1010 9.4 Cache control 1012 Retransmission streams SHOULD NOT be cached. 1014 In the case of session-multiplexing, the "Cache-Control" header 1015 SHOULD be set to "no-cache" for the retransmission stream. 1017 In the case of SSRC-multiplexing, RTSP cannot specify independent 1018 caching for the retransmission stream, because there is a single 1019 "m" line in SDP. Therefore, the implementer should take this fact 1020 into account when deciding whether to cache an SSRC-multiplexed 1021 session or not. 1023 10. Implementation examples 1025 This document mandates only the sender and receiver behaviours 1026 that are necessary for interoperability. In addition, certain 1027 algorithms, such as rate control or buffer management when 1028 targeted at specific environments, may enhance the retransmission 1029 efficiency. 1031 This section gives an overview of different implementation options 1032 allowed within this specification. 1034 The first example describes a minimal receiver implementation. 1035 With this implementation, it is possible to retransmit lost RTP 1036 packets, detect efficiently the loss of retransmissions and 1037 perform multiple retransmissions, if needed. Most of the 1038 necessary processing is done at the server. 1040 The second example shows how a receiver may implement additional 1041 enhancements that might help reduce sender buffer requirements and 1042 optimise the retransmission efficiency 1044 The third example shows how retransmissions may be used in (small) 1045 multicast groups in conjunction with layered encoding. It 1046 illustrates that retransmissions and layered encoding may be 1047 complementary techniques. 1049 10.1 A minimal receiver implementation example 1051 This section gives an example of an implementation supporting 1052 multiple retransmissions. The sender transmits the original data 1053 in RTP packets using the MPEG-4 video RTP payload format. 1054 It is assumed that NACK feedback messages are used, as per 1055 [1]. An SDP description example with SSRC-multiplexing is given 1056 below: 1058 v=0 1059 o=mascha 2980675221 2980675778 IN IP4 host.example.net 1060 c=IN IP4 192.0.2.0 1061 m=video 49170 RTP/AVPF 96 97 1062 a=rtpmap:96 MP4V-ES/90000 1063 a=rtcp-fb:96 nack 1064 a=rtpmap:97 rtx/90000 1065 a=fmtp:97 apt=96;rtx-time=3000 1067 The format-specific parameter "rtx-time" indicates that the server 1068 will buffer the sent packets in a retransmission buffer for 3.0 1069 seconds, after which the packets are deleted from the 1070 retransmission buffer and will never be sent again. 1072 In this implementation example, the required RTP receiver 1073 processing to handle retransmission is kept to a minimum. The 1074 receiver detects packet loss from the gaps observed in the 1075 received sequence numbers. It signals lost packets to the sender 1076 through NACKs as defined in the AVPF profile [1]. The receiver 1077 should take into account the signalled sender retransmission 1078 buffer length in order to dimension its own reception buffer. It 1079 should also derive from the buffer length the maximum number of 1080 times the retransmission of a packet can be requested. 1082 The sender should retransmit the packets selectively, i.e. it 1083 should choose whether to retransmit a requested packet depending 1084 on the packet importance, the observed QoS and congestion state of 1085 the network connection to the receiver. Obviously, the sender 1086 processing increases with the number of receivers as state 1087 information and processing load must be allocated to each 1088 receiver. 1090 10.2 An enhanced receiver implementation example 1092 The receiver may have more accurate information than the sender 1093 about the current network QoS such as available bandwidth, packet 1094 loss rate, delay and jitter. In addition, other receiver-specific 1095 parameters such as buffer level, estimated importance of the lost 1096 packet and application level QoS may be used by the receiver to 1097 make a more efficient use of RTP retransmission by selectively 1098 sending NACKs for important lost packets and not for others. For 1099 example, a receiver may decide to suppress a request for a packet 1100 loss that could be concealed locally, or for a retransmission that 1101 would arrive late. 1103 Furthermore, a receiver may acknowledge the received packets. 1104 This can be done by sending ACKs, as per [1]. Upon receiving an 1105 ACK, the sender may delete all the acknowledged packets from its 1106 retransmission buffer. Note that this would also require only 1107 limited increase in the required RTCP bandwidth as long as ACK 1108 packets are sent seldom enough. 1110 This implementation may help reduce buffer requirements at the 1111 sender and optimise the performance of the implementation by using 1112 selective requests. 1114 Note that these receiver enhancements do not need to be negotiated 1115 as they do not affect the sender implementation. However, in 1116 order to allow the receiver to acknowledge packets, it is needed 1117 to allow the use of ACKs in the SDP description, by means of an 1118 additional SDP "a=rtcp-fb" line, as follows: 1120 v=0 1121 o=mascha 2980675221 2980675778 IN IP4 host.example.net 1122 c=IN IP4 192.0.2.0 1123 m=video 49170 RTP/AVPF 96 97 1124 a=rtpmap:96 MP4V-ES/90000 1125 a=rtcp-fb:96 nack 1126 a=rtcp-fb:96 ack 1127 a=rtpmap:97 rtx/90000 1128 a=fmtp:97 apt=96;rtx-time=3000 1130 10.3 Retransmission of Layered Encoded Media in Multicast 1132 This section shows how to combine retransmissions with layered 1133 encoding in multicast sessions. Note that the retransmission 1134 framework is not intended as a complete solution to reliable 1135 multicast. Refer to RFC 2887 [10], for an overview of the 1136 problems related with reliable multicast transmission. 1138 Packets of different importance are sent in different RTP 1139 sessions. The retransmission streams corresponding to the 1140 different layers can themselves be seen as different 1141 retransmission layers. The relative importance of the different 1142 retransmission streams should reflect the relative importance of 1143 the different original streams. 1145 In multicast, SSRC-multiplexing of the original and retransmission 1146 streams is not allowed as per Section 5.3 of this document. For 1147 this reason, the retransmission stream(s) MUST be sent in 1148 different RTP session(s) using session-multiplexing. 1150 An SDP description example of multicast retransmissions for 1151 layered encoded media is given below: 1153 m=video 8000 RTP/AVPF 98 1154 c=IN IP4 192.0.2.0/127/3 1155 a=rtpmap:98 MP4V-ES/90000 1156 a=rtcp-fb:98 nack 1157 m=video 8000 RTP/AVPF 99 1158 c=IN IP4 192.0.2.4/127/3 1159 a=rtpmap:99 rtx/90000 1160 a=fmtp:99 apt=98;rtx-time=3000 1162 The server and the receiver may implement the retransmission 1163 methods illustrated in the previous examples. In addition, they 1164 may choose to request and retransmit a lost packet depending on 1165 the layer it belongs to. 1167 11. IANA considerations 1169 A new MIME subtype name, "rtx", has been registered for four 1170 different media types, as follows: "video", "audio", "text" and 1171 "application". An additional REQUIRED parameter, "apt", and an 1172 OPTIONAL parameter, "rtx-time", are defined. See Section 8 for 1173 details. 1175 12. Security considerations 1177 If cryptography is used to provide security services on the 1178 original stream, then the same services, with equivalent 1179 cryptographic strength, MUST be provided on the retransmission 1180 stream. Old keys will likely need to be cached so that when the 1181 keys change for the original stream, the old key is used until it 1182 is determined that the key has changed on the retransmission 1183 packets as well. 1185 The use of the same key for the retransmitted stream and the 1186 original stream may lead to security problems, e.g. two-time pads. 1187 This sharing has to be evaluated towards the chosen security 1188 protocol and security algorithms. 1190 Furthermore, it is RECOMMENDED that the cryptography mechanisms 1191 used for this payload format provide protection against known 1192 plaintext attacks. RTP recommends that the initial RTP timestamp 1193 SHOULD be random to secure the stream against known plaintext 1194 attacks. This payload format does not follow this recommendation 1195 as the initial timestamp will be the media timestamp of the first 1196 retransmitted packet. However, since the initial timestamp of the 1197 original stream is itself random, if the original stream is 1198 encrypted, the first retransmitted packet timestamp would also be 1199 random to an attacker. Therefore, confidentiality would not be 1200 compromised. 1202 Congestion control considerations with the use of retransmission 1203 are dealt with in Section 7 of this document. 1205 Any other security considerations of the profile under which the 1206 retransmission scheme is used should be applied. The 1207 retransmission payload format MUST NOT be used under the SAVP 1208 profile defined by the Secure Real-Time Transport Protocol 1209 (SRTP)[12] but instead an extension of SRTP should be defined to 1210 secure the AVPF profile. The definition of such a profile is out 1211 of the scope of this document. 1213 13. Acknowledgements 1215 We would like to express our gratitude to Carsten Burmeister for 1216 his participation in the development of this document. Our thanks 1217 also go to Koichi Hata, Colin Perkins, Stephen Casner, Magnus 1218 Westerlund, Go Hori and Rahul Agarwal for their helpful comments. 1220 14. References 1222 14.1 Normative References 1224 1 J. Ott, S. Wenger, N. Sato, C. Burmeister, J. Rey, "Extended RTP 1225 profile for RTCP-based feedback", draft-ietf-avt-rtcp-feedback- 1226 04.txt, September 2002. 1228 2 S. Bradner, "Key words for use in RFCs to Indicate Requirement 1229 Levels", BCP 14, RFC 2119, March 1997 1231 3 H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson, "RTP: A 1232 Transport Protocol for Real-Time Applications", draft-ietf-avt- 1233 rtp-new-12.txt, March 2003. 1235 4 S. Casner, "SDP bandwidth modifiers for RTCP bandwidth", draft- 1236 ietf-avt-rtcp-bw-05.txt, May 2002. 1238 5 M. Handley, V. Jacobson, "SDP: Session Description Protocol", 1239 RFC 2327, April 1998. 1241 6 G. Camarillo, J. Holler, G. AP. Eriksson, "Grouping of media 1242 lines in the Session Description Protocol (SDP)", RFC 3388, 1243 December 2002. 1245 7 H. Schulzrinne, A. Rao, R. Lanphier, "Real Time Streaming 1246 Protocol (RTSP)", RFC 2326, April 1998. 1248 14.2 Informative References 1250 8 C. Perkins, O. Hodson, "Options for Repair of Streaming Media", 1251 RFC 2354, June 1998. 1253 9 G. Hellstrom, "RTP for conversational text", RFC 2793, May 2000 1255 10 M. Handley, et al., "The Reliable Multicast Design Space for 1256 Bulk Data Transfer", RFC 2887, August 2000. 1258 11 Friedman, et. al., "RTP Extended Reports", Work in Progress. 1260 12 M. Baugher, D. A. McGrew, D. Oran, R. Blom, E. Carrara, M. 1261 Naslund, K. Norrman, "The Secure Real-Time Transport Protocol", 1262 draft-ietf-avt-srtp-05.txt, June 2002. 1264 13 R. Hovey and S. Bradner, "The Organizations Involved in the IETF 1265 Standards Process," BCP 11, RFC 2028, IETF, October 1996. 1267 15. Author's Addresses 1269 Jose Rey rey@panasonic.de 1270 Panasonic European Laboratories GmbH 1271 Monzastr. 4c 1272 D-63225 Langen, Germany 1273 Phone: +49-6103-766-134 1274 Fax: +49-6103-766-166 1276 David Leon david.leon@nokia.com 1277 Nokia Research Center 1278 6000 Connection Drive 1279 Irving, TX. USA 1280 Phone: 1-972-374-1860 1282 Akihiro Miyazaki akihiro@isl.mei.co.jp 1283 Core Software Development Center 1284 Corporate Software Development Division 1285 Matsushita Electric Industrial Co., Ltd. 1286 1006 Kadoma, Kadoma City, Osaka 571-8501, Japan 1287 Phone: +81-6-6900-9192 1288 Fax: +81-6-6900-9193 1290 Viktor Varsa viktor.varsa@nokia.com 1291 Nokia Research Center 1292 6000 Connection Drive 1293 Irving, TX. USA 1294 Phone: 1-972-374-1861 1296 Rolf Hakenberg hakenberg@panasonic.de 1297 Panasonic European Laboratories GmbH 1298 Monzastr. 4c 1299 D-63225 Langen, Germany 1300 Phone: +49-6103-766-162 1301 Fax: +49-6103-766-166 1303 IPR Notices 1305 The IETF takes no position regarding the validity or scope of any 1306 intellectual property or other rights that might be claimed to 1307 pertain to the implementation or use of the technology described 1308 in this document or the extent to which any license under such 1309 rights might or might not be available; neither does it represent 1310 that it has made any effort to identify any such rights. 1311 Information on the IETF's procedures with respect to rights in 1312 standards-track and standards-related documentation can be found 1313 in BCP 11 [13]. 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