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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 AVTCORE WG M. Westerlund 3 Internet-Draft Ericsson 4 Updates: 3550, 3551 (if approved) C. Perkins 5 Intended status: Standards Track University of Glasgow 6 Expires: January 21, 2016 J. Lennox 7 Vidyo 8 July 20, 2015 10 Sending Multiple Types of Media in a Single RTP Session 11 draft-ietf-avtcore-multi-media-rtp-session-09 13 Abstract 15 This document specifies how an RTP session can contain RTP Streams 16 with media from multiple media types such as audio, video, and text. 17 This has been restricted by the RTP Specification, and thus this 18 document updates RFC 3550 and RFC 3551 to enable this behaviour for 19 applications that satisfy the applicability for using multiple media 20 types in a single RTP session. 22 Status of This Memo 24 This Internet-Draft is submitted in full conformance with the 25 provisions of BCP 78 and BCP 79. 27 Internet-Drafts are working documents of the Internet Engineering 28 Task Force (IETF). Note that other groups may also distribute 29 working documents as Internet-Drafts. The list of current Internet- 30 Drafts is at http://datatracker.ietf.org/drafts/current/. 32 Internet-Drafts are draft documents valid for a maximum of six months 33 and may be updated, replaced, or obsoleted by other documents at any 34 time. It is inappropriate to use Internet-Drafts as reference 35 material or to cite them other than as "work in progress." 37 This Internet-Draft will expire on January 21, 2016. 39 Copyright Notice 41 Copyright (c) 2015 IETF Trust and the persons identified as the 42 document authors. All rights reserved. 44 This document is subject to BCP 78 and the IETF Trust's Legal 45 Provisions Relating to IETF Documents 46 (http://trustee.ietf.org/license-info) in effect on the date of 47 publication of this document. Please review these documents 48 carefully, as they describe your rights and restrictions with respect 49 to this document. Code Components extracted from this document must 50 include Simplified BSD License text as described in Section 4.e of 51 the Trust Legal Provisions and are provided without warranty as 52 described in the Simplified BSD License. 54 Table of Contents 56 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 57 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 58 3. Background and Motivation . . . . . . . . . . . . . . . . . . 3 59 4. Applicability . . . . . . . . . . . . . . . . . . . . . . . . 4 60 5. Using Multiple Media Types in a Single RTP Session . . . . . 6 61 5.1. Allowing Multiple Media Types in an RTP Session . . . . . 6 62 5.2. Demultiplexing media types within an RTP session . . . . 7 63 5.3. Per-SSRC Media Type Restrictions . . . . . . . . . . . . 8 64 5.4. RTCP Considerations . . . . . . . . . . . . . . . . . . . 8 65 6. Extension Considerations . . . . . . . . . . . . . . . . . . 8 66 6.1. RTP Retransmission Payload Format . . . . . . . . . . . . 9 67 6.2. RTP Payload Format for Generic FEC . . . . . . . . . . . 10 68 6.3. RTP Payload Format for Redundant Audio . . . . . . . . . 11 69 7. Signalling . . . . . . . . . . . . . . . . . . . . . . . . . 12 70 8. Security Considerations . . . . . . . . . . . . . . . . . . . 12 71 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12 72 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 13 73 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 13 74 11.1. Normative References . . . . . . . . . . . . . . . . . . 13 75 11.2. Informative References . . . . . . . . . . . . . . . . . 13 76 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 16 78 1. Introduction 80 The Real-time Transport Protocol [RFC3550] was designed to use 81 separate RTP sessions to transport different types of media. This 82 implies that different transport layer flows are used for different 83 media streams. For example, a video conferencing application might 84 send audio and video traffic RTP flows on separate UDP ports. With 85 increased use of network address/port translation, firewalls, and 86 other middleboxes it is, however, becoming difficult to establish 87 multiple transport layer flows between endpoints. Hence, there is 88 pressure to reduce the number of concurrent transport flows used by 89 RTP applications. 91 This memo updates [RFC3550] and [RFC3551] to allow multiple media 92 types to be sent in a single RTP session in certain cases, thereby 93 reducing the number of transport layer flows that are needed. It 94 makes no changes to RTP behaviour when using multiple RTP streams 95 containing media of the same type (e.g., multiple audio streams or 96 multiple video streams) in a single RTP session, however 98 [I-D.ietf-avtcore-rtp-multi-stream] provides important clarifications 99 to RTP behaviour in that case. 101 This memo is structured as follows. Section 2 defines terminology. 102 Section 3 further describes the background to, and motivation for, 103 this memo and Section 4 describes the scenarios where this memo is 104 applicable. Section 5 discusses issues arising from the base RTP and 105 RTCP specification when using multiple types of media in a single RTP 106 session, while Section 6 considers the impact of RTP extensions. We 107 discuss signalling in Section 7. Finally, security considerations 108 are discussed in Section 8. 110 2. Terminology 112 The terms Encoded Stream, Endpoint, Media Source, RTP Session, and 113 RTP Stream are used as defined in 114 [I-D.ietf-avtext-rtp-grouping-taxonomy]. We also define the 115 following terms: 117 Media Type: The general type of media data used by a real-time 118 application. The media type corresponds to the value used in the 119 field of an SDP m= line. The media types defined at the 120 time of this writing are "audio", "video", "text", "application", 121 and "message". 123 Quality of Service (QoS): Network mechanisms that are intended to 124 ensure that the packets within a flow or with a specific marking 125 are transported with certain properties. 127 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 128 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 129 document are to be interpreted as described in [RFC2119]. 131 3. Background and Motivation 133 RTP was designed to support multimedia sessions, containing multiple 134 types of media sent simultaneously, by using multiple transport layer 135 flows. The existence of network address translators, firewalls, and 136 other middleboxes complicates this, however, since a mechanism is 137 needed to ensure that all the transport layer flows needed by the 138 application can be established. This has three consequences: 140 1. increased delay to establish a complete session, since each of 141 the transport layer flows needs to be negotiated and established; 143 2. increased state and resource consumption in the middleboxes that 144 can lead to unexpected behaviour when middlebox resource limits 145 are reached; and 147 3. increased risk that a subset of the transport layer flows will 148 fail to be established, thus preventing the application from 149 communicating. 151 Using fewer transport layer flows can hence be seen to reduce the 152 risk of communication failure, and can lead to improved reliability 153 and performance. 155 One of the benefits of using multiple transport layer flows is that 156 it makes it easy to use network layer quality of service (QoS) 157 mechanisms to give differentiated performance for different flows. 158 However, we note that many RTP-using application don't use network 159 QoS features, and don't expect or desire any separation in network 160 treatment of their media packets, independent of whether they are 161 audio, video or text. When an application has no such desire, it 162 doesn't need to provide a transport flow structure that simplifies 163 flow based QoS. 165 Given the above issues, it might seem appropriate for RTP-based 166 applications to send all their media streams bundled into one RTP 167 session, running over a single transport layer flow. However, this 168 is prohibited by the RTP specification, because the design of RTP 169 makes certain assumptions that can be incompatible with sending 170 multiple media types in a single RTP session. Specifically, the RTP 171 control protocol (RTCP) timing rules assume that all RTP media flows 172 in a single RTP session have broadly similar RTCP reporting and 173 feedback requirements, which can be problematic when different types 174 of media are multiplexed together. Various RTP extensions also make 175 assumptions about SSRC use and RTCP reporting that are incompatible 176 with sending different media types in a single RTP session. 178 This memo updates [RFC3550] and [RFC3551] to allow RTP sessions to 179 contain more than one media type in certain circumstances, and gives 180 guidance on when it is safe to send multiple media types in a single 181 RTP session. 183 4. Applicability 185 This specification has limited applicability, and anyone intending to 186 use it MUST ensure that their application and use meets the following 187 criteria: 189 Equal treatment of media: The use of a single RTP session enforces 190 similar treatment on all types of media used within the session. 191 Applications that require significantly different network QoS or 192 RTCP configuration for different media streams are better suited 193 by sending those media streams on separate RTP session, using 194 separate transport layer flows for each, since that gives greater 195 flexibility. Further guidance is given in 196 [I-D.ietf-avtcore-multiplex-guidelines] and 197 [I-D.ietf-dart-dscp-rtp]. 199 Compatible RTCP Behaviour: The RTCP timing rules enforce a single 200 RTCP reporting interval for all participants in an RTP session. 201 Flows with very different media sending rate or RTCP feedback 202 requirements cannot be multiplexed together, since this leads to 203 either excessive or insufficient RTCP for some flows, depending 204 how the RTCP session bandwidth, and hence reporting interval, is 205 configured. For example, it is likely not feasible to find a 206 single RTCP configuration that simultaneously suits both a low- 207 rate audio flow with no feedback and a high-quality video flow 208 with sophisticated RTCP-based feedback needs, making it difficult 209 to combine these into a single RTP session. 211 Signalled Support: The extensions defined in this memo are not 212 compatible with unmodified [RFC3550]-compatible endpoints. Their 213 use requires signalling and mutual agreement by all participants 214 within an RTP session. This requirement can be a problem for 215 signalling solutions that can't negotiate with all participants. 216 For declarative signalling solutions, mandating that the session 217 is using multiple media types in one RTP session can be a way of 218 attempting to ensure that all participants in the RTP session 219 follow the requirement. However, for signalling solutions that 220 lack methods for enforcing that a receiver supports a specific 221 feature, this can still cause issues. 223 Consistent support for multiparty RTP sessions: If it is desired to 224 send multiple types of media in a multiparty RTP session, then all 225 participants in that session need to support sending multiple type 226 of media in a single RTP session. It is not possible, in the 227 general case, to implement a gateway that can interconnect an 228 endpoint using multiple types of media sent using separate RTP 229 sessions, with one or more endpoints that send multiple types of 230 media in a single RTP session. 232 One reason for this is that the same SSRC value can safely be used 233 for different streams in multiple RTP sessions, but when collapsed 234 to a single RTP session there is an SSRC collision. This would 235 not be an issue, since SSRC collision detection will resolve the 236 conflict, except that some RTP payload formats and extensions use 237 matching SSRCs to identify related flows, and break when a single 238 RTP session is used. 240 A middlebox that remaps SSRC values when combining multiple RTP 241 sessions into one also needs to be aware of all possible RTCP 242 packet types that might be used, so that it can remap the SSRC 243 values in those packets. This is impossible to do without 244 restricting the set of RTCP packet types that can be used to those 245 that are known by the middlebox. Such a middlebox might also have 246 difficulty due to differences in configured RTCP bandwidth and 247 other parameters between the RTP sessions. 249 Finally, the use of a middlebox that translates SSRC values can 250 negatively impact the possibility for loop detection, as SSRC/CSRC 251 can't be used to detect the loops, instead some other RTP stream 252 or media source identity name space that is common across all 253 interconnect parts are needed. 255 Ability to operate with limited payload type space: An RTP session 256 has only a single 7-bit payload type space for all its payload 257 type numbers. Some applications might find this space limiting 258 when media different media types and RTP payload formats are using 259 within a single RTP session. 261 Avoids incompatible Extensions: Some RTP and RTCP extensions rely on 262 the existence of multiple RTP sessions and relate media streams 263 between sessions. Others report on particular media types, and 264 cannot be used with other media types. Applications that send 265 multiple types of media into a single RTP session need to avoid 266 such extensions. 268 5. Using Multiple Media Types in a Single RTP Session 270 This section defines what needs to be done or avoided to make an RTP 271 session with multiple media types function without issues. 273 5.1. Allowing Multiple Media Types in an RTP Session 275 Section 5.2 of "RTP: A Transport Protocol for Real-Time Applications" 276 [RFC3550] states: 278 For example, in a teleconference composed of audio and video media 279 encoded separately, each medium SHOULD be carried in a separate 280 RTP session with its own destination transport address. 282 Separate audio and video streams SHOULD NOT be carried in a single 283 RTP session and demultiplexed based on the payload type or SSRC 284 fields. 286 This specification changes both of these sentences. The first 287 sentence is changed to: 289 For example, in a teleconference composed of audio and video media 290 encoded separately, each medium SHOULD be carried in a separate 291 RTP session with its own destination transport address, unless 292 specification [RFCXXXX] is followed and the application meets the 293 applicability constraints. 295 The second sentence is changed to: 297 Separate audio and video media sources SHOULD NOT be carried in a 298 single RTP session, unless the guidelines specified in [RFCXXXX] 299 are followed. 301 Second paragraph of Section 6 in RTP Profile for Audio and Video 302 Conferences with Minimal Control [RFC3551] says: 304 The payload types currently defined in this profile are assigned 305 to exactly one of three categories or media types: audio only, 306 video only and those combining audio and video. The media types 307 are marked in Tables 4 and 5 as "A", "V" and "AV", respectively. 308 Payload types of different media types SHALL NOT be interleaved or 309 multiplexed within a single RTP session, but multiple RTP sessions 310 MAY be used in parallel to send multiple media types. An RTP 311 source MAY change payload types within the same media type during 312 a session. See the section "Multiplexing RTP Sessions" of RFC 313 3550 for additional explanation. 315 This specifications purpose is to violate that existing SHALL NOT 316 under certain conditions. Thus this sentence also has to be changed 317 to allow for multiple media type's payload types in the same session. 318 The above sentence is changed to: 320 Payload types of different media types SHALL NOT be interleaved or 321 multiplexed within a single RTP session unless [RFCXXXX] is used, 322 and the application conforms to the applicability constraints. 323 Multiple RTP sessions MAY be used in parallel to send multiple 324 media types. 326 RFC-Editor Note: Please replace RFCXXXX with the RFC number of this 327 specification when assigned. 329 5.2. Demultiplexing media types within an RTP session 331 When receiving packets from a transport layer flow, an endpoint will 332 first separate the RTP and RTCP packets from the non-RTP packets, and 333 pass them to the RTP/RTCP protocol handler. The RTP and RTCP packets 334 are then demultiplexed based on their SSRC into the different media 335 streams. For each media stream, incoming RTCP packets are processed, 336 and the RTP payload type is used to select the appropriate media 337 decoder. This process remains the same irrespective of whether 338 multiple media types are sent in a single RTP session or not. 340 It is important to note that the RTP payload type is never used to 341 distinguish media streams. The RTP packets are demultiplexed into 342 media streams based on their SSRC, then the RTP payload type is used 343 to select the correct media decoding pathway for each media stream. 345 5.3. Per-SSRC Media Type Restrictions 347 An SSRC in an RTP session can change between media formats of the 348 same type, subject to certain restrictions [RFC7160], but MUST NOT 349 change media type during its lifetime. For example, an SSRC can 350 change between different audio formats, but cannot start sending 351 audio then change to sending video. The lifetime of an SSRC ends 352 when an RTCP BYE packet for that SSRC is sent, or when it ceases 353 transmission for long enough that it times out for the other 354 participants in the session. 356 The main motivation is that a given SSRC has its own RTP timestamp 357 and sequence number spaces. The same way that you can't send two 358 encoded streams of audio on the same SSRC, you can't send one encoded 359 audio and one encoded video stream on the same SSRC. Each encoded 360 stream when made into an RTP stream needs to have the sole control 361 over the sequence number and timestamp space. If not, one would not 362 be able to detect packet loss for that particular encoded stream. 363 Nor can one easily determine which clock rate a particular SSRCs 364 timestamp will increase with. For additional arguments why RTP 365 payload type based multiplexing of multiple media sources doesn't 366 work see [I-D.ietf-avtcore-multiplex-guidelines]. 368 Within an RTP session where multiple media types have been configured 369 for use, an SSRC can only send one type of media during its lifetime 370 (i.e., it can switch between different audio codecs, since those are 371 both the same type of media, but cannot switch between audio and 372 video). Different SSRCs MUST be used for the different media 373 sources, the same way multiple media sources of the same media type 374 already have to do. The payload type will inform a receiver which 375 media type the SSRC is being used for. Thus the payload type MUST be 376 unique across all of the payload configurations independent of media 377 type that is used in the RTP session. 379 5.4. RTCP Considerations 381 When sending multiple types of media that have different rates in a 382 single RTP session, endpoints MUST follow the guidelines for handling 383 RTCP described in Section 7 of [I-D.ietf-avtcore-rtp-multi-stream]. 385 6. Extension Considerations 386 This section outlines known issues and incompatibilities with RTP and 387 RTCP extensions when multiple media types are used in a single RTP 388 sessions. Future extensions to RTP and RTCP need to consider, and 389 document, any potential incompatibility. 391 6.1. RTP Retransmission Payload Format 393 The RTP Retransmission Payload Format [RFC4588] can operate in either 394 SSRC-multiplexed mode or session-multiplex mode. 396 In SSRC-multiplexed mode, retransmitted RTP packets are sent in the 397 same RTP session as the original packets, but use a different SSRC 398 with the same RTCP SDES CNAME. If each endpoint sends only a single 399 original RTP stream and a single retransmission RTP stream in the 400 session, this is sufficient. If an endpoint sends multiple original 401 and retransmission RTP streams, as would occur when sending multiple 402 media types in a single RTP session, then each original RTP stream 403 and the retransmission RTP stream have to be associated using 404 heuristics. By having retransmission requests outstanding for only 405 one SSRC not yet mapped, a receiver can determine the binding between 406 original and retransmission RTP stream. Another alternative is the 407 use of different RTP payload types, allowing the signalled "apt" 408 (associated payload type) parameter of the RTP retransmission payload 409 format to be used to associate retransmitted and original packets. 411 Session-multiplexed mode sends the retransmission RTP stream in a 412 separate RTP session to the original RTP stream, but using the same 413 SSRC for each, with association being done by matching SSRCs between 414 the two sessions. This is unaffected by the use of multiple media 415 types in a single RTP session, since each media type will be sent 416 using a different SSRC in the original RTP session, and the same 417 SSRCs can be used in the retransmission session, allowing the streams 418 to be associated. This can be signalled using SDP with the BUNDLE 419 [I-D.ietf-mmusic-sdp-bundle-negotiation] and FID grouping [RFC5888] 420 extensions. These SDP extensions require each "m=" line to only be 421 included in a single FID group, but the RTP retransmission payload 422 format uses FID groups to indicate the m= lines that form an original 423 and retransmission pair. Accordingly, when using the BUNDLE 424 extension to allow multiple media types to be sent in a single RTP 425 session, each original media source (m= line) that is retransmitted 426 needs a corresponding m= line in the retransmission RTP session. In 427 case there are multiple media lines for retransmission, these media 428 lines will form a independent BUNDLE group from the BUNDLE group with 429 the source streams. 431 An example SDP fragment showing the grouping structures is provided 432 in Figure 1. This example is not legal SDP and only the most 433 important attributes have been left in place. Note that this SDP is 434 not an initial BUNDLE offer. As can be seen there are two bundle 435 groups, one for the source RTP session and one for the 436 retransmissions. Then each of the media sources are grouped with its 437 retransmission flow using FID, resulting in three more groupings. 439 a=group:BUNDLE foo bar fiz 440 a=group:BUNDLE zoo kelp glo 441 a=group:FID foo zoo 442 a=group:FID bar kelp 443 a=group:FID fiz glo 444 m=audio 10000 RTP/AVP 0 445 a=mid:foo 446 a=rtpmap:0 PCMU/8000 447 m=video 10000 RTP/AVP 31 448 a=mid:bar 449 a=rtpmap:31 H261/90000 450 m=video 10000 RTP/AVP 31 451 a=mid:fiz 452 a=rtpmap:31 H261/90000 453 m=audio 40000 RTP/AVPF 99 454 a=rtpmap:99 rtx/90000 455 a=fmtp:99 apt=0;rtx-time=3000 456 a=mid:zoo 457 m=video 40000 RTP/AVPF 100 458 a=rtpmap:100 rtx/90000 459 a=fmtp:199 apt=31;rtx-time=3000 460 a=mid:kelp 461 m=video 40000 RTP/AVPF 100 462 a=rtpmap:100 rtx/90000 463 a=fmtp:199 apt=31;rtx-time=3000 464 a=mid:glo 466 Figure 1: SDP example of Session Multiplexed RTP Retransmission 468 6.2. RTP Payload Format for Generic FEC 470 The RTP Payload Format for Generic Forward Error Correction (FEC) 471 [RFC5109] (and its predecessor [RFC2733]) can either send the FEC 472 stream as a separate RTP stream, or it can send the FEC combined with 473 the original RTP stream as a redundant encoding [RFC2198]. 475 When sending FEC as a separate stream, the RTP Payload Format for 476 generic FEC requires that FEC stream to be sent in a separate RTP 477 session to the original stream, using the same SSRC, with the FEC 478 stream being associated by matching the SSRC between sessions. The 479 RTP session used for the original streams can include multiple RTP 480 streams, and those RTP stream can use multiple media types. The 481 repair session only needs one RTP Payload type to indicate FEC data, 482 irrespective of the number of FEC streams sent, since the SSRC is 483 used to associate the FEC streams with the original streams. Hence, 484 it is RECOMMENDED that FEC stream use the "application/ulpfec" media 485 type for [RFC5109], and the "application/parityfec" media type for 486 [RFC2733]. It is legal, but NOT RECOMMENDED, to send FEC streams 487 using media specific payload format names (e.g., if an original RTP 488 session contains audio and video flows, for the associated FEC RTP 489 session where to use the "audio/ulpfec" and "video/ulpfec" payload 490 formats), since this unnecessarily uses up RTP payload type values, 491 and adds no value for demultiplexing since there might be multiple 492 streams of the same media type). 494 The combination of an original RTP session using multiple media types 495 with a associated generic FEC session can be signalled using SDP with 496 the BUNDLE extension [I-D.ietf-mmusic-sdp-bundle-negotiation]. In 497 this case, the RTP session carrying the FEC streams will be its own 498 BUNDLE group. The m= line for each original stream and the m= line 499 for the corresponding FEC stream are grouped using the SDP grouping 500 framework with either the FEC [RFC4756] or the FEC-FR [RFC5956] 501 grouping. This is similar to the situation that arises for RTP 502 retransmission with session multiplexing discussed in Section 6.1. 504 The Source-Specific Media Attributes [RFC5576] specification defines 505 an SDP extension (the "FEC" semantic of the "ssrc-group" attribute) 506 to signal FEC relationships between multiple RTP streams within a 507 single RTP session. This cannot be used with generic FEC, since the 508 FEC repair packets need to have the same SSRC value as the source 509 packets being protected. There is ongoing work on an ULP extension 510 to allow it be use FEC RTP streams within the same RTP Session as the 511 source stream [I-D.lennox-payload-ulp-ssrc-mux]. 513 When the FEC is sent as a redundant encoding, the considerations in 514 Section 6.3 apply. 516 6.3. RTP Payload Format for Redundant Audio 518 The RTP Payload Format for Redundant Audio [RFC2198] can be used to 519 protect audio streams. It can also be used along with the generic 520 FEC payload format to send original and repair data in the same RTP 521 packets. Both are compatible with RTP sessions containing multiple 522 media types. 524 This payload format requires each different redundant encoding use a 525 different RTP payload type number. When used with generic FEC in 526 sessions that contain multiple media types, this requires each media 527 type use a different payload type for the FEC stream. For example, 528 if audio and text are sent in a single RTP session with generic ULP 529 FEC sent as a redundant encoding for each, then payload types need to 530 be assigned for FEC using the audio/ulpfec and text/ulpfec payload 531 formats. If multiple original payload types of used in the session, 532 different redundant payload types need to be allocated for each one. 533 This has potential to rapidly exhaust the available RTP payload type 534 numbers. 536 7. Signalling 538 Establishing a single RTP session using multiple media types requires 539 signalling. This signalling has to: 541 1. ensure that any participant in the RTP session is aware that this 542 is an RTP session with multiple media types; 544 2. ensure that the payload types in use in the RTP session are using 545 unique values, with no overlap between the media types; 547 3. ensure RTP session level parameters, for example the RTCP RR and 548 RS bandwidth modifiers, the RTP/AVPF trr-int parameter, transport 549 protocol, RTCP extensions in use, and any security parameters, 550 are consistent across the session; and 552 4. ensure that RTP and RTCP functions that can be bound to a 553 particular media type are reused where possible, rather than 554 configuring multiple code-points for the same thing. 556 When using SDP signalling, the BUNDLE extension 557 [I-D.ietf-mmusic-sdp-bundle-negotiation] is used to signal RTP 558 sessions containing multiple media types. 560 8. Security Considerations 562 RTP provides a range of strong security mechanisms that can be used 563 to secure sessions [RFC7201], [RFC7202]. The majority of these are 564 independent of the type of media sent in the RTP session, however it 565 is important to check that the security mechanism chosen is 566 compatible with all types of media sent within the session. 568 Sending multiple media types in a single RTP session will generally 569 require that all use the same security mechanism, whereas media sent 570 using different RTP sessions can be secured in different ways. When 571 different media types have different security requirements, it might 572 be necessary to send them using separate RTP sessions to meet those 573 different requirements. This can have significant costs in terms of 574 resource usage, session set-up time, etc. 576 9. IANA Considerations 577 This memo makes no request of IANA. 579 10. Acknowledgements 581 The authors would like to thank Christer Holmberg, Gunnar Hellstroem, 582 and Charles Eckel for the feedback on the document. 584 11. References 586 11.1. Normative References 588 [I-D.ietf-avtcore-rtp-multi-stream] 589 Lennox, J., Westerlund, M., Wu, W., and C. Perkins, 590 "Sending Multiple Media Streams in a Single RTP Session", 591 draft-ietf-avtcore-rtp-multi-stream-08 (work in progress), 592 July 2015. 594 [I-D.ietf-mmusic-sdp-bundle-negotiation] 595 Holmberg, C., Alvestrand, H., and C. Jennings, 596 "Negotiating Media Multiplexing Using the Session 597 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- 598 negotiation-22 (work in progress), June 2015. 600 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 601 Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/ 602 RFC2119, March 1997, 603 . 605 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 606 Jacobson, "RTP: A Transport Protocol for Real-Time 607 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 608 July 2003, . 610 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 611 Video Conferences with Minimal Control", STD 65, RFC 3551, 612 DOI 10.17487/RFC3551, July 2003, 613 . 615 11.2. Informative References 617 [I-D.ietf-avtcore-multiplex-guidelines] 618 Westerlund, M., Perkins, C., and H. Alvestrand, 619 "Guidelines for using the Multiplexing Features of RTP to 620 Support Multiple Media Streams", draft-ietf-avtcore- 621 multiplex-guidelines-03 (work in progress), October 2014. 623 [I-D.ietf-avtcore-rtp-topologies-update] 624 Westerlund, M. and S. Wenger, "RTP Topologies", draft- 625 ietf-avtcore-rtp-topologies-update-10 (work in progress), 626 July 2015. 628 [I-D.ietf-avtext-rtp-grouping-taxonomy] 629 Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and 630 B. Burman, "A Taxonomy of Semantics and Mechanisms for 631 Real-Time Transport Protocol (RTP) Sources", draft-ietf- 632 avtext-rtp-grouping-taxonomy-07 (work in progress), June 633 2015. 635 [I-D.ietf-dart-dscp-rtp] 636 Black, D. and P. Jones, "Differentiated Services 637 (DiffServ) and Real-time Communication", draft-ietf-dart- 638 dscp-rtp-10 (work in progress), November 2014. 640 [I-D.lennox-payload-ulp-ssrc-mux] 641 Lennox, J., "Supporting Source-Multiplexing of the Real- 642 Time Transport Protocol (RTP) Payload for Generic Forward 643 Error Correction", draft-lennox-payload-ulp-ssrc-mux-00 644 (work in progress), February 2013. 646 [I-D.westerlund-avtcore-transport-multiplexing] 647 Westerlund, M. and C. Perkins, "Multiplexing Multiple RTP 648 Sessions onto a Single Lower-Layer Transport", draft- 649 westerlund-avtcore-transport-multiplexing-07 (work in 650 progress), October 2013. 652 [RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., 653 Handley, M., Bolot, J.C., Vega-Garcia, A., and S. Fosse- 654 Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, 655 DOI 10.17487/RFC2198, September 1997, 656 . 658 [RFC2733] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format 659 for Generic Forward Error Correction", RFC 2733, DOI 660 10.17487/RFC2733, December 1999, 661 . 663 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 664 Description Protocol", RFC 4566, DOI 10.17487/RFC4566, 665 July 2006, . 667 [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. 668 Hakenberg, "RTP Retransmission Payload Format", RFC 4588, 669 DOI 10.17487/RFC4588, July 2006, 670 . 672 [RFC4756] Li, A., "Forward Error Correction Grouping Semantics in 673 Session Description Protocol", RFC 4756, DOI 10.17487/ 674 RFC4756, November 2006, 675 . 677 [RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error 678 Correction", RFC 5109, DOI 10.17487/RFC5109, December 679 2007, . 681 [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific 682 Media Attributes in the Session Description Protocol 683 (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009, 684 . 686 [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and 687 Control Packets on a Single Port", RFC 5761, DOI 10.17487/ 688 RFC5761, April 2010, 689 . 691 [RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description 692 Protocol (SDP) Grouping Framework", RFC 5888, DOI 10.17487 693 /RFC5888, June 2010, 694 . 696 [RFC5956] Begen, A., "Forward Error Correction Grouping Semantics in 697 the Session Description Protocol", RFC 5956, DOI 10.17487/ 698 RFC5956, September 2010, 699 . 701 [RFC7160] Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple 702 Clock Rates in an RTP Session", RFC 7160, DOI 10.17487/ 703 RFC7160, April 2014, 704 . 706 [RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP 707 Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014, 708 . 710 [RFC7202] Perkins, C. and M. Westerlund, "Securing the RTP 711 Framework: Why RTP Does Not Mandate a Single Media 712 Security Solution", RFC 7202, DOI 10.17487/RFC7202, April 713 2014, . 715 Authors' Addresses 717 Magnus Westerlund 718 Ericsson 719 Farogatan 6 720 SE-164 80 Kista 721 Sweden 723 Phone: +46 10 714 82 87 724 Email: magnus.westerlund@ericsson.com 726 Colin Perkins 727 University of Glasgow 728 School of Computing Science 729 Glasgow G12 8QQ 730 United Kingdom 732 Email: csp@csperkins.org 734 Jonathan Lennox 735 Vidyo, Inc. 736 433 Hackensack Avenue 737 Seventh Floor 738 Hackensack, NJ 07601 739 US 741 Email: jonathan@vidyo.com