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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: June 20, 2016 J. Lennox 7 Vidyo 8 December 18, 2015 10 Sending Multiple Types of Media in a Single RTP Session 11 draft-ietf-avtcore-multi-media-rtp-session-13 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 June 20, 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 . . . . . . . . . . . . . . . . . . 9 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 . . . . . . . . . . . . . . . . . . . . . 13 72 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 13 73 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 13 74 11.1. Normative References . . . . . . . . . . . . . . . . . . 13 75 11.2. Informative References . . . . . . . . . . . . . . . . . 14 76 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 15 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 RTP 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 [RFC7656]. We also define the 114 following terms: 116 Media Type: The general type of media data used by a real-time 117 application. The media type corresponds to the value used in the 118 field of an SDP m= line. The media types defined at the 119 time of this writing are "audio", "video", "text", "image", 120 "application", and "message". [RFC4566] [RFC6466] 122 Quality of Service (QoS): Network mechanisms that are intended to 123 ensure that the packets within a flow or with a specific marking 124 are transported with certain properties. 126 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 127 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and 128 "OPTIONAL" in this document are to be interpreted as described in 129 [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 RTP 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 needs to ensure that their application and use case meets the 187 following criteria: 189 Equal treatment of media: The use of a single RTP session normally 190 results in similar network treatment for all types of media used 191 within the session. Applications that require significantly 192 different network quality of service (QoS) or RTCP configuration 193 for different RTP streams are better suited by sending those RTP 194 streams in separate RTP session, using separate transport layer 195 flows for each, since that gives greater flexibility. Further 196 guidance on how to provide differential treatment for some media 197 is given in [I-D.ietf-avtcore-multiplex-guidelines] and [RFC7657]. 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 on 204 how the RTCP session bandwidth, and hence reporting interval, is 205 configured. For example, it is likely infeasible to find a single 206 RTCP configuration that simultaneously suits both a low-rate audio 207 flow with no feedback, and a high-quality video flow with 208 sophisticated RTCP-based feedback. Thus, combining these into a 209 single RTP session is difficult and/or inadvisable. 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 is 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 using different media types and RTP payload formats within a 259 single RTP session. 261 Avoids incompatible Extensions: Some RTP and RTCP extensions rely on 262 the existence of multiple RTP sessions and relate RTP 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 specification's purpose is to override 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 sentence containing "SHALL NOT" in the above paragraph is changed 319 to: 321 Payload types of different media types SHALL NOT be interleaved or 322 multiplexed within a single RTP session unless [RFCXXXX] is used, 323 and the application conforms to the applicability constraints. 324 Multiple RTP sessions MAY be used in parallel to send multiple 325 media types. 327 RFC-Editor Note: Please replace RFCXXXX with the RFC number of this 328 specification when assigned. 330 5.2. Demultiplexing media types within an RTP session 332 When receiving packets from a transport layer flow, an endpoint will 333 first separate the RTP and RTCP packets from the non-RTP packets, and 334 pass them to the RTP/RTCP protocol handler. The RTP and RTCP packets 335 are then demultiplexed based on their SSRC into the different RTP 336 streams. For each RTP stream, incoming RTCP packets are processed, 337 and the RTP payload type is used to select the appropriate media 338 decoder. This process remains the same irrespective of whether 339 multiple media types are sent in a single RTP session or not. 341 As explained below, it is important to note that the RTP payload type 342 is never used to distinguish RTP streams. The RTP packets are 343 demultiplexed into RTP streams based on their SSRC, then the RTP 344 payload type is used to select the correct media decoding pathway for 345 each RTP stream. 347 5.3. Per-SSRC Media Type Restrictions 349 An SSRC in an RTP session can change between media formats of the 350 same type, subject to certain restrictions [RFC7160], but MUST NOT 351 change media type during its lifetime. For example, an SSRC can 352 change between different audio formats, but cannot start sending 353 audio then change to sending video. The lifetime of an SSRC ends 354 when an RTCP BYE packet for that SSRC is sent, or when it ceases 355 transmission for long enough that it times out for the other 356 participants in the session. 358 The main motivation is that a given SSRC has its own RTP timestamp 359 and sequence number spaces. The same way that you can't send two 360 encoded streams of audio with the same SSRC, you can't send one 361 encoded audio and one encoded video stream with the same SSRC. Each 362 encoded stream when made into an RTP stream needs to have the sole 363 control over the sequence number and timestamp space. If not, one 364 would not be able to detect packet loss for that particular encoded 365 stream. Nor can one easily determine which clock rate a particular 366 SSRCs timestamp will increase with. For additional arguments why RTP 367 payload type based multiplexing of multiple media sources doesn't 368 work, see [I-D.ietf-avtcore-multiplex-guidelines]. 370 Within an RTP session where multiple media types have been configured 371 for use, an SSRC can only send one type of media during its lifetime 372 (i.e., it can switch between different audio codecs, since those are 373 both the same type of media, but cannot switch between audio and 374 video). Different SSRCs MUST be used for the different media 375 sources, the same way multiple media sources of the same media type 376 already have to do. The payload type will inform a receiver which 377 media type the SSRC is being used for. Thus the payload type MUST be 378 unique across all of the payload configurations independent of media 379 type that is used in the RTP session. 381 5.4. RTCP Considerations 383 When sending multiple types of media that have different rates in a 384 single RTP session, endpoints MUST follow the guidelines for handling 385 RTCP described in Section 7 of [I-D.ietf-avtcore-rtp-multi-stream]. 387 6. Extension Considerations 389 This section outlines known issues and incompatibilities with RTP and 390 RTCP extensions when multiple media types are used in a single RTP 391 sessions. Future extensions to RTP and RTCP need to consider, and 392 document, any potential incompatibility. 394 6.1. RTP Retransmission Payload Format 396 The RTP Retransmission Payload Format [RFC4588] can operate in either 397 SSRC-multiplexed mode or session-multiplex mode. 399 In SSRC-multiplexed mode, retransmitted RTP packets are sent in the 400 same RTP session as the original packets, but use a different SSRC 401 with the same RTCP SDES CNAME. If each endpoint sends only a single 402 original RTP stream and a single retransmission RTP stream in the 403 session, this is sufficient. If an endpoint sends multiple original 404 and retransmission RTP streams, as would occur when sending multiple 405 media types in a single RTP session, then each original RTP stream 406 and the retransmission RTP stream have to be associated using 407 heuristics. By having retransmission requests outstanding for only 408 one SSRC not yet mapped, a receiver can determine the binding between 409 original and retransmission RTP stream. Another alternative is the 410 use of different RTP payload types, allowing the signalled "apt" 411 (associated payload type) parameter of the RTP retransmission payload 412 format to be used to associate retransmitted and original packets. 414 Session-multiplexed mode sends the retransmission RTP stream in a 415 separate RTP session to the original RTP stream, but using the same 416 SSRC for each, with association being done by matching SSRCs between 417 the two sessions. This is unaffected by the use of multiple media 418 types in a single RTP session, since each media type will be sent 419 using a different SSRC in the original RTP session, and the same 420 SSRCs can be used in the retransmission session, allowing the streams 421 to be associated. This can be signalled using SDP with the BUNDLE 422 [I-D.ietf-mmusic-sdp-bundle-negotiation] and FID grouping [RFC5888] 423 extensions. These SDP extensions require each "m=" line to only be 424 included in a single FID group, but the RTP retransmission payload 425 format uses FID groups to indicate the m= lines that form an original 426 and retransmission pair. Accordingly, when using the BUNDLE 427 extension to allow multiple media types to be sent in a single RTP 428 session, each original media source (m= line) that is retransmitted 429 needs a corresponding m= line in the retransmission RTP session. In 430 case there are multiple media lines for retransmission, these media 431 lines will form an independent BUNDLE group from the BUNDLE group 432 with the source streams. 434 An example SDP fragment showing the grouping structures is provided 435 in Figure 1. This example is not legal SDP and only the most 436 important attributes have been left in place. Note that this SDP is 437 not an initial BUNDLE offer. As can be seen there are two bundle 438 groups, one for the source RTP session and one for the 439 retransmissions. Then each of the media sources are grouped with its 440 retransmission flow using FID, resulting in three more groupings. 442 a=group:BUNDLE foo bar fiz 443 a=group:BUNDLE zoo kelp glo 444 a=group:FID foo zoo 445 a=group:FID bar kelp 446 a=group:FID fiz glo 447 m=audio 10000 RTP/AVP 0 448 a=mid:foo 449 a=rtpmap:0 PCMU/8000 450 m=video 10000 RTP/AVP 31 451 a=mid:bar 452 a=rtpmap:31 H261/90000 453 m=video 10000 RTP/AVP 31 454 a=mid:fiz 455 a=rtpmap:31 H261/90000 456 m=audio 40000 RTP/AVPF 99 457 a=rtpmap:99 rtx/90000 458 a=fmtp:99 apt=0;rtx-time=3000 459 a=mid:zoo 460 m=video 40000 RTP/AVPF 100 461 a=rtpmap:100 rtx/90000 462 a=fmtp:199 apt=31;rtx-time=3000 463 a=mid:kelp 464 m=video 40000 RTP/AVPF 100 465 a=rtpmap:100 rtx/90000 466 a=fmtp:199 apt=31;rtx-time=3000 467 a=mid:glo 469 Figure 1: SDP example of Session Multiplexed RTP Retransmission 471 6.2. RTP Payload Format for Generic FEC 473 The RTP Payload Format for Generic Forward Error Correction (FEC) 474 [RFC5109] (and its predecessor [RFC2733]) can either send the FEC 475 stream as a separate RTP stream, or it can send the FEC combined with 476 the original RTP stream as a redundant encoding [RFC2198]. 478 When sending FEC as a separate stream, the RTP Payload Format for 479 generic FEC requires that FEC stream to be sent in a separate RTP 480 session to the original stream, using the same SSRC, with the FEC 481 stream being associated by matching the SSRC between sessions. The 482 RTP session used for the original streams can include multiple RTP 483 streams, and those RTP streams can use multiple media types. The 484 repair session only needs one RTP Payload type to indicate FEC data, 485 irrespective of the number of FEC streams sent, since the SSRC is 486 used to associate the FEC streams with the original streams. Hence, 487 it is RECOMMENDED that the FEC stream use the "application/ulpfec" 488 media type for [RFC5109], and the "application/parityfec" media type 489 for [RFC2733]. It is legal, but NOT RECOMMENDED, to send FEC streams 490 using media specific payload format names (e.g., using both the 491 "audio/ulpfec" and "video/ulpfec" payload formats for a single RTP 492 session containing both audio and video flows), since this 493 unnecessarily uses up RTP payload type values, and adds no value for 494 demultiplexing since there might be multiple streams of the same 495 media type). 497 The combination of an original RTP session using multiple media types 498 with an associated generic FEC session can be signalled using SDP 499 with the BUNDLE extension [I-D.ietf-mmusic-sdp-bundle-negotiation]. 500 In this case, the RTP session carrying the FEC streams will be its 501 own BUNDLE group. The m= line for each original stream and the m= 502 line for the corresponding FEC stream are grouped using the SDP 503 grouping framework using either the FEC-FR [RFC5956] grouping or, for 504 backwards compatibility, the FEC [RFC4756] grouping. This is similar 505 to the situation that arises for RTP retransmission with session 506 multiplexing discussed in Section 6.1. 508 The Source-Specific Media Attributes [RFC5576] specification defines 509 an SDP extension (the "FEC" semantic of the "ssrc-group" attribute) 510 to signal FEC relationships between multiple RTP streams within a 511 single RTP session. This cannot be used with generic FEC, since the 512 FEC repair packets need to have the same SSRC value as the source 513 packets being protected. There was work on an Unequal Layer 514 Protection (ULP) extension to allow it be use FEC RTP streams within 515 the same RTP Session as the source stream 516 [I-D.lennox-payload-ulp-ssrc-mux]. 518 When the FEC is sent as a redundant encoding, the considerations in 519 Section 6.3 apply. 521 6.3. RTP Payload Format for Redundant Audio 523 The RTP Payload Format for Redundant Audio [RFC2198] can be used to 524 protect audio streams. It can also be used along with the generic 525 FEC payload format to send original and repair data in the same RTP 526 packets. Both are compatible with RTP sessions containing multiple 527 media types. 529 This payload format requires each different redundant encoding use a 530 different RTP payload type number. When used with generic FEC in 531 sessions that contain multiple media types, this requires each media 532 type to use a different payload type for the FEC stream. For 533 example, if audio and text are sent in a single RTP session with 534 generic ULP FEC sent as a redundant encoding for each, then payload 535 types need to be assigned for FEC using the audio/ulpfec and text/ 536 ulpfec payload formats. If multiple original payload types are used 537 in the session, different redundant payload types need to be 538 allocated for each one. This has potential to rapidly exhaust the 539 available RTP payload type numbers. 541 7. Signalling 543 Establishing a single RTP session using multiple media types requires 544 signalling. This signalling has to: 546 1. ensure that any participant in the RTP session is aware that this 547 is an RTP session with multiple media types; 549 2. ensure that the payload types in use in the RTP session are using 550 unique values, with no overlap between the media types; 552 3. ensure RTP session level parameters, for example the RTCP RR and 553 RS bandwidth modifiers, the RTP/AVPF trr-int parameter, transport 554 protocol, RTCP extensions in use, and any security parameters, 555 are consistent across the session; and 557 4. ensure that RTP and RTCP functions that can be bound to a 558 particular media type are reused where possible, rather than 559 configuring multiple code-points for the same thing. 561 When using SDP signalling, the BUNDLE extension 562 [I-D.ietf-mmusic-sdp-bundle-negotiation] is used to signal RTP 563 sessions containing multiple media types. 565 8. Security Considerations 567 RTP provides a range of strong security mechanisms that can be used 568 to secure sessions [RFC7201], [RFC7202]. The majority of these are 569 independent of the type of media sent in the RTP session; however it 570 is important to check that the security mechanism chosen is 571 compatible with all types of media sent within the session. 573 Sending multiple media types in a single RTP session will generally 574 require that all use the same security mechanism, whereas media sent 575 using different RTP sessions can be secured in different ways. When 576 different media types have different security requirements, it might 577 be necessary to send them using separate RTP sessions to meet those 578 different requirements. This can have significant costs in terms of 579 resource usage, session set-up time, etc. 581 9. IANA Considerations 583 This memo makes no request of IANA. 585 10. Acknowledgements 587 The authors would like to thank Christer Holmberg, Gunnar Hellstroem, 588 Charles Eckel, Tolga Asveren, Warren Kumari, and Meral Shirazipour 589 for their feedback on the document. 591 11. References 593 11.1. Normative References 595 [I-D.ietf-avtcore-rtp-multi-stream] 596 Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, 597 "Sending Multiple RTP Streams in a Single RTP Session", 598 draft-ietf-avtcore-rtp-multi-stream-11 (work in progress), 599 December 2015. 601 [I-D.ietf-mmusic-sdp-bundle-negotiation] 602 Holmberg, C., Alvestrand, H., and C. Jennings, 603 "Negotiating Media Multiplexing Using the Session 604 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- 605 negotiation-23 (work in progress), July 2015. 607 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 608 Requirement Levels", BCP 14, RFC 2119, 609 DOI 10.17487/RFC2119, March 1997, 610 . 612 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 613 Jacobson, "RTP: A Transport Protocol for Real-Time 614 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 615 July 2003, . 617 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 618 Video Conferences with Minimal Control", STD 65, RFC 3551, 619 DOI 10.17487/RFC3551, July 2003, 620 . 622 11.2. Informative References 624 [I-D.ietf-avtcore-multiplex-guidelines] 625 Westerlund, M., Perkins, C., and H. Alvestrand, 626 "Guidelines for using the Multiplexing Features of RTP to 627 Support Multiple Media Streams", draft-ietf-avtcore- 628 multiplex-guidelines-03 (work in progress), October 2014. 630 [I-D.lennox-payload-ulp-ssrc-mux] 631 Lennox, J., "Supporting Source-Multiplexing of the Real- 632 Time Transport Protocol (RTP) Payload for Generic Forward 633 Error Correction", draft-lennox-payload-ulp-ssrc-mux-00 634 (work in progress), February 2013. 636 [RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., 637 Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse- 638 Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, 639 DOI 10.17487/RFC2198, September 1997, 640 . 642 [RFC2733] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format 643 for Generic Forward Error Correction", RFC 2733, 644 DOI 10.17487/RFC2733, December 1999, 645 . 647 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 648 Description Protocol", RFC 4566, DOI 10.17487/RFC4566, 649 July 2006, . 651 [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. 652 Hakenberg, "RTP Retransmission Payload Format", RFC 4588, 653 DOI 10.17487/RFC4588, July 2006, 654 . 656 [RFC4756] Li, A., "Forward Error Correction Grouping Semantics in 657 Session Description Protocol", RFC 4756, 658 DOI 10.17487/RFC4756, November 2006, 659 . 661 [RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error 662 Correction", RFC 5109, DOI 10.17487/RFC5109, December 663 2007, . 665 [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific 666 Media Attributes in the Session Description Protocol 667 (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009, 668 . 670 [RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description 671 Protocol (SDP) Grouping Framework", RFC 5888, 672 DOI 10.17487/RFC5888, June 2010, 673 . 675 [RFC5956] Begen, A., "Forward Error Correction Grouping Semantics in 676 the Session Description Protocol", RFC 5956, 677 DOI 10.17487/RFC5956, September 2010, 678 . 680 [RFC6466] Salgueiro, G., "IANA Registration of the 'image' Media 681 Type for the Session Description Protocol (SDP)", 682 RFC 6466, DOI 10.17487/RFC6466, December 2011, 683 . 685 [RFC7160] Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple 686 Clock Rates in an RTP Session", RFC 7160, 687 DOI 10.17487/RFC7160, April 2014, 688 . 690 [RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP 691 Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014, 692 . 694 [RFC7202] Perkins, C. and M. Westerlund, "Securing the RTP 695 Framework: Why RTP Does Not Mandate a Single Media 696 Security Solution", RFC 7202, DOI 10.17487/RFC7202, April 697 2014, . 699 [RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and 700 B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms 701 for Real-Time Transport Protocol (RTP) Sources", RFC 7656, 702 DOI 10.17487/RFC7656, November 2015, 703 . 705 [RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services 706 (Diffserv) and Real-Time Communication", RFC 7657, 707 DOI 10.17487/RFC7657, November 2015, 708 . 710 Authors' Addresses 711 Magnus Westerlund 712 Ericsson 713 Farogatan 6 714 SE-164 80 Kista 715 Sweden 717 Phone: +46 10 714 82 87 718 Email: magnus.westerlund@ericsson.com 720 Colin Perkins 721 University of Glasgow 722 School of Computing Science 723 Glasgow G12 8QQ 724 United Kingdom 726 Email: csp@csperkins.org 728 Jonathan Lennox 729 Vidyo, Inc. 730 433 Hackensack Avenue 731 Seventh Floor 732 Hackensack, NJ 07601 733 US 735 Email: jonathan@vidyo.com