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Hutton 11 Unify 12 February 14, 2014 14 Session Recording Protocol 15 draft-ietf-siprec-protocol-12 17 Abstract 19 This document specifies the use of the Session Initiation Protocol 20 (SIP), the Session Description Protocol (SDP), and the Real Time 21 Protocol (RTP) for delivering real-time media and metadata from a 22 Communication Session (CS) to a recording device. The Session 23 Recording Protocol specifies the use of SIP, SDP, and RTP to 24 establish a Recording Session (RS) between the Session Recording 25 Client (SRC), which is on the path of the CS, and a Session Recording 26 Server (SRS) at the recording device. 28 Status of This Memo 30 This Internet-Draft is submitted in full conformance with the 31 provisions of BCP 78 and BCP 79. 33 Internet-Drafts are working documents of the Internet Engineering 34 Task Force (IETF). Note that other groups may also distribute 35 working documents as Internet-Drafts. The list of current Internet- 36 Drafts is at http://datatracker.ietf.org/drafts/current/. 38 Internet-Drafts are draft documents valid for a maximum of six months 39 and may be updated, replaced, or obsoleted by other documents at any 40 time. It is inappropriate to use Internet-Drafts as reference 41 material or to cite them other than as "work in progress." 43 This Internet-Draft will expire on August 18, 2014. 45 Copyright Notice 47 Copyright (c) 2014 IETF Trust and the persons identified as the 48 document authors. All rights reserved. 50 This document is subject to BCP 78 and the IETF Trust's Legal 51 Provisions Relating to IETF Documents 52 (http://trustee.ietf.org/license-info) in effect on the date of 53 publication of this document. Please review these documents 54 carefully, as they describe your rights and restrictions with respect 55 to this document. Code Components extracted from this document must 56 include Simplified BSD License text as described in Section 4.e of 57 the Trust Legal Provisions and are provided without warranty as 58 described in the Simplified BSD License. 60 Table of Contents 62 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 63 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 64 3. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 4 65 4. Scope . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 66 5. Overview of operations . . . . . . . . . . . . . . . . . . . 5 67 5.1. Delivering recorded media . . . . . . . . . . . . . . . . 5 68 5.2. Delivering recording metadata . . . . . . . . . . . . . . 7 69 5.3. Receiving recording indications and providing recording 70 preferences . . . . . . . . . . . . . . . . . . . . . . . 8 71 6. SIP Handling . . . . . . . . . . . . . . . . . . . . . . . . 9 72 6.1. Procedures at the SRC . . . . . . . . . . . . . . . . . . 10 73 6.1.1. Initiating a Recording Session . . . . . . . . . . . 10 74 6.1.2. SIP extensions for recording indication and 75 preference . . . . . . . . . . . . . . . . . . . . . 10 76 6.2. Procedures at the SRS . . . . . . . . . . . . . . . . . . 11 77 6.3. Procedures for Recording-aware User Agents . . . . . . . 11 78 7. SDP Handling . . . . . . . . . . . . . . . . . . . . . . . . 12 79 7.1. Procedures at the SRC . . . . . . . . . . . . . . . . . . 12 80 7.1.1. SDP handling in RS . . . . . . . . . . . . . . . . . 12 81 7.1.1.1. Handling media stream updates . . . . . . . . . . 13 82 7.1.2. Recording indication in CS . . . . . . . . . . . . . 14 83 7.1.3. Recording preference in CS . . . . . . . . . . . . . 15 84 7.2. Procedures at the SRS . . . . . . . . . . . . . . . . . . 15 85 7.3. Procedures for Recording-aware User Agents . . . . . . . 17 86 7.3.1. Recording indication . . . . . . . . . . . . . . . . 17 87 7.3.2. Recording preference . . . . . . . . . . . . . . . . 18 88 8. RTP Handling . . . . . . . . . . . . . . . . . . . . . . . . 19 89 8.1. RTP Mechanisms . . . . . . . . . . . . . . . . . . . . . 19 90 8.1.1. RTCP . . . . . . . . . . . . . . . . . . . . . . . . 19 91 8.1.2. RTP Profile . . . . . . . . . . . . . . . . . . . . . 20 92 8.1.3. SSRC . . . . . . . . . . . . . . . . . . . . . . . . 20 93 8.1.4. CSRC . . . . . . . . . . . . . . . . . . . . . . . . 21 94 8.1.5. SDES . . . . . . . . . . . . . . . . . . . . . . . . 21 95 8.1.5.1. CNAME . . . . . . . . . . . . . . . . . . . . . . 21 96 8.1.6. Keepalive . . . . . . . . . . . . . . . . . . . . . . 21 97 8.1.7. RTCP Feedback Messages . . . . . . . . . . . . . . . 22 98 8.1.7.1. Full Intra Request . . . . . . . . . . . . . . . 22 99 8.1.7.2. Picture Loss Indicator . . . . . . . . . . . . . 22 100 8.1.7.3. Temporary Maximum Media Stream Bit Rate Request . 23 101 8.1.8. Symmetric RTP/RTCP for Sending and Receiving . . . . 23 102 8.2. Roles . . . . . . . . . . . . . . . . . . . . . . . . . . 23 103 8.2.1. SRC acting as an RTP Translator . . . . . . . . . . . 24 104 8.2.1.1. Forwarding Translator . . . . . . . . . . . . . . 25 105 8.2.1.2. Transcoding Translator . . . . . . . . . . . . . 25 106 8.2.2. SRC acting as an RTP Mixer . . . . . . . . . . . . . 26 107 8.2.3. SRC acting as an RTP Endpoint . . . . . . . . . . . . 26 108 8.3. RTP Session Usage by SRC . . . . . . . . . . . . . . . . 27 109 8.3.1. SRC Using Multiple m-lines . . . . . . . . . . . . . 27 110 8.3.2. SRC Using Mixing . . . . . . . . . . . . . . . . . . 28 111 8.4. RTP Session Usage by SRS . . . . . . . . . . . . . . . . 29 112 9. Metadata . . . . . . . . . . . . . . . . . . . . . . . . . . 30 113 9.1. Procedures at the SRC . . . . . . . . . . . . . . . . . . 30 114 9.2. Procedures at the SRS . . . . . . . . . . . . . . . . . . 32 115 9.2.1. Formal Syntax . . . . . . . . . . . . . . . . . . . . 33 116 10. Persistent Recording . . . . . . . . . . . . . . . . . . . . 33 117 11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 34 118 11.1. Registration of Option Tags . . . . . . . . . . . . . . 34 119 11.1.1. siprec Option Tag . . . . . . . . . . . . . . . . . 34 120 11.1.2. record-aware Option Tag . . . . . . . . . . . . . . 34 121 11.2. Registration of media feature tags . . . . . . . . . . . 34 122 11.2.1. src feature tag . . . . . . . . . . . . . . . . . . 34 123 11.2.2. srs feature tag . . . . . . . . . . . . . . . . . . 35 124 11.3. New Content-Disposition Parameter Registrations . . . . 35 125 11.4. Media Type Registration . . . . . . . . . . . . . . . . 36 126 11.4.1. Registration of MIME Type application/rs-metadata . 36 127 11.4.2. Registration of MIME Type application/rs-metadata- 128 request . . . . . . . . . . . . . . . . . . . . . . 36 129 11.5. SDP Attributes . . . . . . . . . . . . . . . . . . . . . 36 130 11.5.1. 'record' SDP Attribute . . . . . . . . . . . . . . . 36 131 11.5.2. 'recordpref' SDP Attribute . . . . . . . . . . . . . 37 132 12. Security Considerations . . . . . . . . . . . . . . . . . . . 37 133 12.1. Authentication and Authorization . . . . . . . . . . . . 38 134 12.2. RTP handling . . . . . . . . . . . . . . . . . . . . . . 38 135 12.3. Metadata . . . . . . . . . . . . . . . . . . . . . . . . 39 136 12.4. Storage and playback . . . . . . . . . . . . . . . . . . 39 137 13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 39 138 14. References . . . . . . . . . . . . . . . . . . . . . . . . . 40 139 14.1. Normative References . . . . . . . . . . . . . . . . . . 40 140 14.2. Informative References . . . . . . . . . . . . . . . . . 40 142 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 42 144 1. Introduction 146 This document specifies the mechanism to record a Communication 147 Session (CS) by delivering real-time media and metadata from the CS 148 to a recording device. In accordance to the architecture 149 [I-D.ietf-siprec-architecture], the Session Recording Protocol 150 specifies the use of SIP, SDP, and RTP to establish a Recording 151 Session (RS) between the Session Recording Client (SRC), which is on 152 the path of the CS, and a Session Recording Server (SRS) at the 153 recording device. 155 SIP is also used to deliver metadata to the recording device, as 156 specified in [I-D.ietf-siprec-metadata]. Metadata is information 157 that describes recorded media and the CS to which they relate. 159 The Session Recording Protocol intends to satisfy the SIP-based Media 160 Recording requirements listed in [RFC6341]. 162 In addition to the Session Recording Protocol, this document 163 specifies extensions for user agents that are participants in a CS to 164 receive recording indications and to provide preferences for 165 recording. 167 2. Terminology 169 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 170 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 171 document are to be interpreted as described in [RFC2119]. 173 3. Definitions 175 This document refers to the core definitions provided in the 176 architecture document [I-D.ietf-siprec-architecture]. 178 The RTP Handling section uses the definitions provided in "RTP: A 179 Transport Protocol for Real-Time Application" [RFC3550]. 181 4. Scope 183 The scope of the Session Recording Protocol includes the 184 establishment of the recording sessions and the reporting of the 185 metadata. The scope also includes extensions supported by User 186 Agents participating in the CS such as indication of recording. The 187 user agents need not be recording-aware in order to participate in a 188 CS being recorded. 190 The following items, which are not an exhaustive list, do not 191 represent the protocol itself and are considered out of the scope of 192 the Session Recording Protocol: 194 o Delivering recorded media in real-time as the CS media 196 o Specifications of criteria to select a specific CS to be recorded 197 or triggers to record a certain CS in the future 199 o Recording policies that determine whether the CS should be 200 recorded and whether parts of the CS are to be recorded 202 o Retention policies that determine how long a recording is stored 204 o Searching and accessing the recorded media and metadata 206 o Policies governing how CS users are made aware of recording 208 o Delivering additional recording session metadata through non-SIP 209 mechanism 211 5. Overview of operations 213 This section is informative and provides a description of recording 214 operations. 216 Section 6 describes the SIP communication in a recording session 217 between a SRC and a SRS, and the procedures for recording-aware user 218 agents participating in a CS. Section 7 describes the SDP in a 219 recording session, and the procedures for recording indications and 220 recording preferences. Section 8 describes the RTP handling in a 221 recording session. Section 9 describes the mechanism to deliver 222 recording metadata from the SRC to the SRS. 224 As mentioned in the architecture document 225 [I-D.ietf-siprec-architecture], there are a number of types of call 226 flows based on the location of the Session Recording Client. The 227 following sample call flows provide a quick overview of the 228 operations between the SRC and the SRS. 230 5.1. Delivering recorded media 232 When a SIP Back-to-Back User Agent (B2BUA) with SRC functionality 233 routes a call from UA(A) to UA(B), the SRC has access to the media 234 path between the user agents. When the SRC is aware that it should 235 be recording the conversation, the SRC can cause the B2BUA to bridge 236 the media between UA(A) and UA(B). The SRC then establishes the 237 Recording Session with the SRS and sends replicated media towards the 238 SRS. 240 An endpoint may also have SRC functionality, where the endpoint 241 itself establishes the Recording Session to the SRS. Since the 242 endpoint has access to the media in the Communication Session, the 243 endpoint can send replicated media towards the SRS. 245 The following is a sample call flow that shows the SRC establishing a 246 recording session towards the SRS. The call flow is essentially 247 identical when the SRC is a B2BUA or as the endpoint itself. Note 248 that the SRC can choose when to establish the Recording Session 249 independent of the Communication Session, even though the following 250 call flow suggests that the SRC is establishing the Recording Session 251 (message #5) after the Communication Session is established. 253 UA A SRC UA B SRS 254 |(1)CS INVITE | | | 255 |------------->| | | 256 | |(2)CS INVITE | | 257 | |---------------------->| | 258 | | (3) 200 OK | | 259 | |<----------------------| | 260 | (4) 200 OK | | | 261 |<-------------| | | 262 | |(5)RS INVITE with SDP | | 263 | |--------------------------------------------->| 264 | | | (6) 200 OK with SDP | 265 | |<---------------------------------------------| 266 |(7)CS RTP | | | 267 |=============>|======================>| | 268 |<=============|<======================| | 269 | |(8)RS RTP | | 270 | |=============================================>| 271 | |=============================================>| 272 |(9)CS BYE | | | 273 |------------->| | | 274 | |(10)CS BYE | | 275 | |---------------------->| | 276 | |(11)RS BYE | | 277 | |--------------------------------------------->| 278 | | | | 280 Figure 1: Basic recording call flow 282 The above call flow can also apply to the case of a centralized 283 conference with a mixer. For clarity, ACKs to INVITEs and 200 OKs to 284 BYEs are not shown. The conference focus can provide the SRC 285 functionality since the conference focus has access to all the media 286 from each conference participant. When a recording is requested, the 287 SRC delivers the metadata and the media streams to the SRS. Since 288 the conference focus has access to a mixer, the SRC may choose to mix 289 the media streams from all participants as a single mixed media 290 stream towards the SRS. 292 An SRC can use a single recording session to record multiple 293 communication sessions. Every time the SRC wants to record a new 294 call, the SRC updates the recording session with a new SDP offer to 295 add new recorded streams to the recording session, and 296 correspondingly also update the metadata for the new call. 298 An SRS can also establish a recording session to an SRC, although it 299 is beyond the scope of this document to define how an SRS would 300 specify which calls to record. 302 5.2. Delivering recording metadata 304 The SRC is responsible for the delivery of metadata to the SRS. The 305 SRC may provide an initial metadata snapshot about recorded media 306 streams in the initial INVITE content in the recording session. 307 Subsequent metadata updates can be represented as a stream of events 308 in UPDATE or reINVITE requests sent by the SRC. These metadata 309 updates are normally incremental updates to the initial metadata 310 snapshot to optimize on the size of updates, however, the SRC may 311 also decide to send a new metadata snapshot anytime. 313 Metadata is transported in the body of INVITE or UPDATE messages. 314 Certain metadata, such as the attributes of the recorded media stream 315 are located in the SDP of the recording session. 317 The SRS has the ability to send a request to the SRC to request for a 318 new metadata snapshot update from the SRC. This can happen when the 319 SRS fails to understand the current stream of incremental updates for 320 whatever reason, for example, when SRS loses the current state due to 321 internal failure. The SRS may optionally attach a reason along with 322 the snapshot request. This request allows both SRC and SRS to 323 synchronize the states with a new metadata snapshot so that further 324 metadata incremental updates will be based on the latest metadata 325 snapshot. Similar to the metadata content, the metadata snapshot 326 request is transported as content in UPDATE or INVITE sent by the SRS 327 in the recording session. 329 SRC SRS 330 | | 331 |(1) INVITE (metadata snapshot) | 332 |---------------------------------------------------->| 333 | (2)200 OK | 334 |<----------------------------------------------------| 335 |(3) ACK | 336 |---------------------------------------------------->| 337 |(4) RTP | 338 |====================================================>| 339 |====================================================>| 340 |(5) UPDATE (metadata update 1) | 341 |---------------------------------------------------->| 342 | (6) 200 OK | 343 |<----------------------------------------------------| 344 |(7) UPDATE (metadata update 2) | 345 |---------------------------------------------------->| 346 | (8) 200 OK | 347 |<----------------------------------------------------| 348 | (9) UPDATE (metadata snapshot request) | 349 |<----------------------------------------------------| 350 | (10) 200 OK | 351 |---------------------------------------------------->| 352 | (11) INVITE (metadata snapshot 2 + SDP offer) | 353 |---------------------------------------------------->| 354 | (12) 200 OK (SDP answer) | 355 |<----------------------------------------------------| 356 | (13) UPDATE (metadata update 1 based on snapshot 2) | 357 |---------------------------------------------------->| 358 | (14) 200 OK | 359 |<----------------------------------------------------| 361 Figure 2: Delivering metadata via SIP UPDATE 363 5.3. Receiving recording indications and providing recording 364 preferences 366 The SRC is responsible to provide recording indications to the 367 participants in the CS. A recording-aware UA supports receiving 368 recording indications via the SDP attribute a=record, and it can 369 specify a recording preference in the CS by including the SDP 370 attribute a=recordpref. The recording attribute is a declaration by 371 the SRC in the CS to indicate whether recording is taking place. The 372 recording preference attribute is a declaration by the recording- 373 aware UA in the CS to indicate the recording preference. 375 To illustrate how the attributes are used, if a UA (A) is initiating 376 a call to UA (B) and UA (A) is also an SRC that is performing the 377 recording, then UA (A) provides the recording indication in the SDP 378 offer with a=record:on. Since UA (A) is the SRC, UA (A) receives the 379 recording indication from the SRC directly. When UA (B) receives the 380 SDP offer, UA (B) will see that recording is happening on the other 381 endpoint of this session. Since UA (B) is not an SRC and does not 382 provide any recording preference, the SDP answer does not contain 383 a=record nor a=recordpref. 385 UA A UA B 386 (SRC) | 387 | | 388 | [SRC recording starts] | 389 |(1) INVITE (SDP offer + a=record:on) | 390 |---------------------------------------------------->| 391 | (2) 200 OK (SDP answer) | 392 |<----------------------------------------------------| 393 |(3) ACK | 394 |---------------------------------------------------->| 395 |(4) RTP | 396 |<===================================================>| 397 | | 398 | [UA B wants to set preference to no recording] | 399 | (5) INVITE (SDP offer + a=recordpref:off) | 400 |<----------------------------------------------------| 401 | [SRC honors the preference and stops recording] | 402 |(6) 200 OK (SDP answer + a=record:off) | 403 |---------------------------------------------------->| 404 | (7) ACK | 405 |<----------------------------------------------------| 407 Figure 3: Recording indication and recording preference 409 After the call is established and recording is in progress, UA (B) 410 later decides to change the recording preference to no recording and 411 sends a reINVITE with the a=recordpref attribute. It is up to the 412 SRC to honor the preference, and in this case SRC decides to stop the 413 recording and updates the recording indication in the SDP answer. 415 6. SIP Handling 416 6.1. Procedures at the SRC 418 6.1.1. Initiating a Recording Session 420 A recording session is a SIP session with specific extensions 421 applied, and these extensions are listed in the procedures for SRC 422 and SRS below. When an SRC or an SRS receives a SIP session that is 423 not a recording session, it is up to the SRC or the SRS to determine 424 what to do with the SIP session. 426 The SRC can initiate a recording session by sending a SIP INVITE 427 request to the SRS. The SRC and the SRS are identified in the From 428 and To headers, respectively. 430 The SRC MUST include the '+sip.src' feature tag in the Contact URI, 431 defined in this specification as an extension to [RFC3840], for all 432 recording sessions. An SRS uses the presence of the '+sip.src' 433 feature tag in dialog creating and modifying requests and responses 434 to confirm that the dialog being created is for the purpose of a 435 Recording Session. In addition, when an SRC sends a REGISTER request 436 to a registrar, the SRC MAY include the '+sip.src' feature tag to 437 indicate the that it is a SRC. 439 Since SIP Caller Preferences extensions are optional to implement for 440 routing proxies, there is no guarantee that a recording session will 441 be routed to an SRC or SRS. A new options tag is introduced: 442 "siprec". As per [RFC3261], only an SRC or an SRS can accept this 443 option tag in a recording session. An SRC MUST include the "siprec" 444 option tag in the Require header when initiating a Recording Session 445 so that UA's which do not support the session recording protocol 446 extensions will simply reject the INVITE request with a 420 Bad 447 Extension. 449 When an SRC receives a new INVITE, the SRC MUST only consider the SIP 450 session as a recording session when both the '+sip.srs' feature tag 451 and 'siprec' option tag are included in the INVITE request. 453 6.1.2. SIP extensions for recording indication and preference 455 For the communication session, the SRC MUST provide recording 456 indication to all participants in the CS. A participant UA in a CS 457 can indicate that it is recording-aware by providing the "record- 458 aware" option tag, and the SRC MUST provide recording indications in 459 the new SDP a=record attribute described in the SDP Handling section. 460 In the absence of the "record-aware" option tag, meaning that the 461 participant UA is not recording-aware, an SRC MUST provide recording 462 indications through other means such as playing a tone inband, if the 463 SRC is required to do so (e.g. based on policies). 465 An SRC in the CS may also indicate itself as a session recording 466 client by including the '+sip.src' feature tag. A recording-aware 467 participant can learn that a SRC is in the CS, and can set the 468 recording preference for the CS with the new SDP a=recordpref 469 attribute described in the SDP Handling section below. 471 6.2. Procedures at the SRS 473 When an SRS receives a new INVITE, the SRS MUST only consider the SIP 474 session as a recording session when both the '+sip.src' feature tag 475 and 'siprec' option tag are included in the INVITE request. 477 The SRS can initiate a recording session by sending a SIP INVITE 478 request to the SRC. The SRS and the SRC are identified in the From 479 and To headers, respectively. 481 The SRS MUST include the '+sip.srs' feature tag in the Contact URI, 482 as per [RFC3840], for all recording sessions. An SRC uses the 483 presence of this feature tag in dialog creating and modifying 484 requests and responses to confirm that the dialog being created is 485 for the purpose of a Recording Session (REQ-30). In addition, when 486 an SRS sends a REGISTER request to a registrar, the SRS SHOULD 487 include the '+sip.srs' feature tag to indicate that it is a SRS. 489 An SRS MUST include the "siprec" option tag in the Require header as 490 per [RFC3261] when initiating a Recording Session so that UA's which 491 do not support the session recording protocol extensions will simply 492 reject the INVITE request with a 420 Bad Extension. 494 6.3. Procedures for Recording-aware User Agents 496 A recording-aware user agent is a participant in the CS that supports 497 the SIP and SDP extensions for receiving recording indication and for 498 requesting recording preferences for the call. A recording-aware UA 499 MUST indicate that it can accept reporting of recording indication 500 provided by the SRC with a new option tag "record-aware" when 501 initiating or establishing a CS, meaning including the "record-aware" 502 tag in the Supported header in the initial INVITE request or 503 response. 505 A recording-aware UA MUST be prepared to provide a recording 506 indication to the end user through an appropriate user interface, 507 indicating whether recording is on, off, or paused for each medium. 508 Some user agents that are automatons (e.g. IVR, media server, PSTN 509 gateway) may not have a user interface to render recording 510 indication. When such user agent indicates recording awareness, the 511 UA SHOULD render recording indication through other means, such as 512 passing an inband tone on the PSTN gateway, putting the recording 513 indication in a log file, or raising an application event in a 514 VoiceXML dialog. These user agents MAY also choose not to indicate 515 recording awareness, thereby relying on whatever mechanism an SRC 516 chooses to indicate recording, such as playing a tone inband. 518 7. SDP Handling 520 7.1. Procedures at the SRC 522 The SRC and SRS follows the SDP offer/answer model in [RFC3264]. The 523 procedures for SRC and SRS describe the conventions used in a 524 recording session. 526 7.1.1. SDP handling in RS 528 Since the SRC does not expect to receive media from the SRS, the SRC 529 typically sets each media stream of the SDP offer to only send media, 530 by qualifying them with the a=sendonly attribute, according to the 531 procedures in [RFC3264]. 533 The SRC sends recorded streams of participants to the SRS, and the 534 SRC MUST provide a label attribute (a=label), as per [RFC4574], on 535 each media stream in order to identify the recorded stream with the 536 rest of the metadata. The a=label attribute identifies each recorded 537 media stream, and the label name is mapped to the Media Stream 538 Reference in the metadata as per [I-D.ietf-siprec-metadata]. The 539 scope of the a=label attribute only applies to the SDP and Metadata 540 conveyed in the bodies of the SIP request or response that the label 541 appeared in. Note that a recorded stream is distinct from a CS 542 stream; the metadata provides a list of participants that contributes 543 to each recorded stream. 545 The following is an example SDP offer from SRC with both audio and 546 video recorded streams. Note that the following example contains 547 unfolded lines longer than 72 characters. These are captured between 548 tags. 550 v=0 551 o=SRC 2890844526 2890844526 IN IP4 198.51.100.1 552 s=- 553 c=IN IP4 198.51.100.1 554 t=0 0 555 m=audio 12240 RTP/AVP 0 4 8 556 a=sendonly 557 a=label:1 558 m=video 22456 RTP/AVP 98 559 a=rtpmap:98 H264/90000 560 561 a=fmtp:98 profile-level-id=42A01E; 562 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA== 563 564 a=sendonly 565 a=label:2 566 m=audio 12242 RTP/AVP 0 4 8 567 a=sendonly 568 a=label:3 569 m=video 22458 RTP/AVP 98 570 a=rtpmap:98 H264/90000 571 572 a=fmtp:98 profile-level-id=42A01E; 573 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA== 574 575 a=sendonly 576 a=label:4 578 Figure 4: Sample SDP offer from SRC with audio and video streams 580 7.1.1.1. Handling media stream updates 582 Over the lifetime of a recording session, the SRC can add and remove 583 recorded streams from the recording session for various reasons. For 584 example, when a CS stream is added or removed from the CS, or when a 585 CS is created or terminated if a recording session handles multiple 586 CSes. To remove a recorded stream from the recording session, the 587 SRC sends a new SDP offer where the port of the media stream to be 588 removed is set to zero, according to the procedures in [RFC3264]. To 589 add a recorded stream to the recording session, the SRC sends a new 590 SDP offer by adding a new media stream description or by reusing an 591 old media stream which had been previously disabled, according to the 592 procedures in [RFC3264]. 594 The SRC can temporarily discontinue streaming and collection of 595 recorded media from the SRC to the SRS for reasons such as masking 596 the recording. In this case, the SRC sends a new SDP offer and sets 597 the media stream to inactive (a=inactive) for each recorded stream to 598 be paused, as per the procedures in [RFC3264]. To resume streaming 599 and collection of recorded media, the SRC sends a new SDP offer and 600 sets the media stream to sendonly (a=sendonly). Note that a CS 601 itself may change the media stream direction by updating the SDP, for 602 example, by setting a=inactive for SDP hold. Media stream direction 603 changes in CS are conveyed in the metadata by the SRC. The SRC MUST 604 NOT modify the RS media stream with a=inactive for SDP hold on the CS 605 since this operation is reserved for pausing the RS media, however, 606 an SRC can have a local policy to pause the RS media when the CS is 607 placed on hold. 609 7.1.2. Recording indication in CS 611 While there are existing mechanisms for providing an indication that 612 a CS is being recorded, these mechanisms are usually delivered on the 613 CS media streams such as playing an in-band tone or an announcement 614 to the participants. A new 'record' SDP attribute is introduced to 615 allow the SRC to indicate recording state to a recording-aware UA in 616 CS. 618 The 'record' SDP attribute appears at the media level or session 619 level in either SDP offer or answer. When the attribute is applied 620 at the session level, the indication applies to all media streams in 621 the SDP. When the attribute is applied at the media level, the 622 indication applies to the media stream only, and that overrides the 623 indication if also set at the session level. Whenever the recording 624 indication needs to change, such as termination of recording, then 625 the SRC MUST initiate a reINVITE or UPDATE to update the SDP a=record 626 attribute. 628 The following is the ABNF of the 'record' attribute: 630 attribute /= record-attr 631 ; attribute defined in RFC 4566 633 record-attr = "record:" indication 634 indication = "on" / "off" / "paused" 636 on Recording is in progress. 638 off No recording is in progress. 640 paused Recording is in progress but media is paused. 642 7.1.3. Recording preference in CS 644 When the SRC receives the a=recordpref SDP in an SDP offer or answer, 645 the SRC chooses to honor the preference to record based on local 646 policy at the SRC. If the SRC makes a change in recording state, the 647 SRC MUST report the new recording state in the a=record attribute in 648 the SDP answer or in a subsequent SDP offer. 650 7.2. Procedures at the SRS 652 Typically the SRS only receives RTP streams from the SRC; therefore, 653 the SDP offer/answer from the SRS normally sets each media stream to 654 receive media, by setting them with the a=recvonly attribute, 655 according to the procedures of [RFC3264]. When the SRS is not ready 656 to receive a recorded stream, the SRS sets the media stream as 657 inactive in the SDP offer or answer by setting it with a=inactive 658 attribute, according to the procedures of [RFC3264]. When the SRS is 659 ready to receive recorded streams, the SRS sends a new SDP offer and 660 sets the media streams with a=recvonly attribute. 662 The following is an example of SDP answer from SRS for the SDP offer 663 from the above sample. Note that the following example contain 664 unfolded lines longer than 72 characters. These are captured between 665 tags. 667 v=0 668 o=SRS 0 0 IN IP4 198.51.100.20 669 s=- 670 c=IN IP4 198.51.100.20 671 t=0 0 672 m=audio 10000 RTP/AVP 0 673 a=recvonly 674 a=label:1 675 m=video 10002 RTP/AVP 98 676 a=rtpmap:98 H264/90000 677 678 a=fmtp:98 profile-level-id=42A01E; 679 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA== 680 681 a=recvonly 682 a=label:2 683 m=audio 10004 RTP/AVP 0 684 a=recvonly 685 a=label:3 686 m=video 10006 RTP/AVP 98 687 a=rtpmap:98 H264/90000 688 689 a=fmtp:98 profile-level-id=42A01E; 690 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA== 691 692 a=recvonly 693 a=label:4 695 Figure 5: Sample SDP answer from SRS with audio and video streams 697 Over the lifetime of a recording session, the SRS can remove recorded 698 streams from the recording session for various reasons. To remove a 699 recorded stream from the recording session, the SRS sends a new SDP 700 offer where the port of the media stream to be removed is set to 701 zero, according to the procedures in [RFC3264]. 703 The SRS SHOULD NOT add recorded streams in the recording session when 704 SRS sends a new SDP offer. Similarly, when the SRS starts a 705 recording session, the SRS SHOULD initiate the INVITE without an SDP 706 offer to let the SRC generate the SDP offer with recorded streams. 708 The following sequence diagram shows an example where the SRS is 709 initially not ready to receive recorded streams, and later updates 710 the recording session when the SRS is ready to record. 712 SRC SRS 713 | | 714 |(1) INVITE (SDP offer) | 715 |---------------------------------------------------->| 716 | [not ready to record] 717 | (2)200 OK with SDP inactive | 718 |<----------------------------------------------------| 719 |(3) ACK | 720 |---------------------------------------------------->| 721 | ... | 722 | [ready to record] 723 | (4) re-INVITE with SDP recvonly | 724 |<----------------------------------------------------| 725 |(5)200 OK with SDP sendonly | 726 |---------------------------------------------------->| 727 | (6) ACK | 728 |<----------------------------------------------------| 729 |(7) RTP | 730 |====================================================>| 731 | ... | 732 |(8) BYE | 733 |---------------------------------------------------->| 734 | (9) OK | 735 |<----------------------------------------------------| 737 Figure 6: SRS responding to offer with a=inactive 739 7.3. Procedures for Recording-aware User Agents 741 7.3.1. Recording indication 743 When a recording-aware UA receives an SDP offer or answer that 744 includes the a=record attribute, the UA MUST provide the recording 745 indication to the end user whether the recording is on, off, or 746 paused for each medium based on the most recently received a=record 747 SDP attribute for that medium. 749 When a CS is traversed through multiple UAs such as a B2BUA or a 750 conference focus, each UA involved in the CS that is aware that the 751 CS is being recorded MUST provide the recording indication through 752 the a=record attribute to all other parties in the CS. 754 It is possible that more than one SRC is in the call path of the same 755 CS, but the recording indication attribute does not provide any hint 756 as to which SRC or how many SRCs are recording. An endpoint knows 757 only that the call is being recorded. Furthermore, this attribute is 758 not used as a request for a specific SRC to start/stop recording. 760 7.3.2. Recording preference 762 A participant in a CS MAY set the recording preference in the CS to 763 be recorded or not recorded at session establishment or during the 764 session. A new 'recordpref' SDP attribute is introduced, and the 765 participant in CS may set this recording preference attribute in any 766 SDP offer/answer at session establishment time or during the session. 767 The SRC is not required to honor the recording preference from a 768 participant based on local policies at the SRC, and the participant 769 can learn the recording indication through the a=record SDP attribute 770 as described in the above section. 772 The SDP a=recordpref attribute can appear at the media level or 773 session level and can appear in an SDP offer or answer. When the 774 attribute is applied at the session level, the recording preference 775 applies to all media stream in the SDP. When the attribute is 776 applied at the media level, the recording preference applies to the 777 media stream only, and that overrides the recording preference if 778 also set at the session level. The user agent can change the 779 recording preference by changing the a=recordpref attribute in 780 subsequent SDP offer or answer. The absence of the a=recordpref 781 attribute in the SDP indicates that the UA has no recording 782 preference. 784 The following is the ABNF of the recordpref attribute: 786 attribute /= recordpref-attr 787 ; attribute defined in RFC 4566 789 recordpref-attr = "a=recordpref:" pref 790 pref = "on" / "off" / "pause" / "nopreference" 792 on Sets the preference to record if it has not already been started. 793 If the recording is currently paused, the preference is to resume 794 recording. 796 off Sets the preference for no recording. If recording has already 797 been started, then the preference is to stop the recording. 799 pause If the recording is currently in progress, sets the preference 800 to pause the recording. 802 nopreference To indicate that the UA has no preference on recording. 804 8. RTP Handling 806 This section provides recommendations and guidelines for RTP and RTCP 807 in the context of SIPREC. In order to communicate most effectively, 808 the Session Recording Client (SRC), the Session Recording Server 809 (SRS), and any Recording aware User Agents (UAs) SHOULD utilize the 810 mechanisms provided by RTP in a well-defined and predicable manner. 811 It is the goal of this document to make the reader aware of these 812 mechanisms and provide recommendations and guidelines. 814 8.1. RTP Mechanisms 816 This section briefly describes important RTP/RTCP constructs and 817 mechanisms that are particularly useful within the content of SIPREC. 819 8.1.1. RTCP 821 The RTP data transport is augmented by a control protocol (RTCP) to 822 allow monitoring of the data delivery. RTCP, as defined in 823 [RFC3550], is based on the periodic transmission of control packets 824 to all participants in the RTP session, using the same distribution 825 mechanism as the data packets. Support for RTCP is REQUIRED, per 826 [RFC3550], and it provides, among other things, the following 827 important functionality in relation to SIPREC: 829 1) Feedback on the quality of the data distribution 831 This feedback from the receivers may be used to diagnose faults in 832 the distribution. As such, RTCP is a well-defined and efficient 833 mechanism for the SRS to inform the SRC, and for the SRC to inform 834 Recording aware UAs, of issues that arise with respect to the 835 reception of media that is to be recorded. 837 2) Carries a persistent transport-level identifier for an RTP source 838 called the canonical name or CNAME 840 The SSRC identifier may change if a conflict is discovered or a 841 program is restarted; in which case receivers can use the CNAME to 842 keep track of each participant. Receivers may also use the CNAME to 843 associate multiple data streams from a given participant in a set of 844 related RTP sessions, for example to synchronize audio and video. 845 Synchronization of media streams is also facilitated by the NTP and 846 RTP timestamps included in RTCP packets by data senders. 848 8.1.2. RTP Profile 850 The RECOMMENDED RTP profiles for the SRC, SRS, and Recording aware 851 UAs are "Extended Secure RTP Profile for Real-time Transport Control 852 Protocol (RTCP)-Based Feedback (RTP/SAVPF)", [RFC5124] when using 853 encrypted RTP streams, and "Extended RTP Profile for Real-time 854 Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", 855 [RFC4585] when using non encrypted media streams. However, as this 856 is not a requirement, some implementations may use "The Secure Real- 857 time Transport Protocol (SRTP)", [RFC3711] and "RTP Profile for Audio 858 and Video Conferences with Minimal Control", AVP [RFC3551]. 859 Therefore, it is RECOMMENDED that the SRC, SRS, and Recording aware 860 UAs not rely entirely on SAVPF or AVPF for core functionality that 861 may be at least partially achievable using SAVP and AVP. 863 AVPF and SAVPF provide an improved RTCP timer model that allows more 864 flexible transmission of RTCP packets in response to events, rather 865 than strictly according to bandwidth. AVPF based codec control 866 messages provide efficient mechanisms for an SRC, SRS, and Recording 867 aware UAs to handle events such as scene changes, error recovery, and 868 dynamic bandwidth adjustments. These messages are discussed in more 869 detail later in this document. 871 SAVP and SAVPF provide media encryption, integrity protection, replay 872 protection, and a limited form of source authentication. They do not 873 contain or require a specific keying mechanism. 875 8.1.3. SSRC 877 The synchronization source (SSRC), as defined in [RFC3550] is carried 878 in the RTP header and in various fields of RTCP packets. It is a 879 random 32-bit number that is required to be globally unique within an 880 RTP session. It is crucial that the number be chosen with care in 881 order that participants on the same network or starting at the same 882 time are not likely to choose the same number. Guidelines regarding 883 SSRC value selection and conflict resolution are provided in 884 [RFC3550]. 886 The SSRC may also be used to separate different sources of media 887 within a single RTP session. For this reason as well as for conflict 888 resolution, it is important that the SRC, SRS, and Recording aware 889 UAs handle changes in SSRC values and properly identify the reason of 890 the change. The CNAME values carried in RTCP facilitate this 891 identification. 893 8.1.4. CSRC 895 The contributing source (CSRC), as defined in [RFC3550], identifies 896 the source of a stream of RTP packets that has contributed to the 897 combined stream produced by an RTP mixer. The mixer inserts a list 898 of the SSRC identifiers of the sources that contributed to the 899 generation of a particular packet into the RTP header of that packet. 900 This list is called the CSRC list. It is RECOMMENDED that a SRC or 901 Recording aware UA, when acting a mixer, sets the CSRC list 902 accordingly, and that the SRC and SRS interpret the CSRC list 903 appropriately when received. 905 8.1.5. SDES 907 The Source Description (SDES), as defined in [RFC3550], contains an 908 SSRC/CSRC identifier followed by a list of zero or more items, which 909 carry information about the SSRC/CSRC. End systems send one SDES 910 packet containing their own source identifier (the same as the SSRC 911 in the fixed RTP header). A mixer sends one SDES packet containing a 912 chunk for each contributing source from which it is receiving SDES 913 information, or multiple complete SDES packets if there are more than 914 31 such sources. 916 8.1.5.1. CNAME 918 The Canonical End-Point Identifier (CNAME), as defined in [RFC3550], 919 provides the binding from the SSRC identifier to an identifier for 920 the source (sender or receiver) that remains constant. It is 921 important the SRC and Recording aware UAs generate CNAMEs 922 appropriately and that the SRC and SRS interpret and use them for 923 this purpose. Guidelines for generating CNAME values are provided in 924 "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names 925 (CNAMEs)" [RFC7022]. 927 8.1.6. Keepalive 929 It is anticipated that media streams in SIPREC may exist in an 930 inactive state for extended periods of times for any of a number of 931 valid reasons. In order for the bindings and any pinholes in NATs/ 932 firewalls to remain active during such intervals, it is RECOMMENDED 933 that the SRC, SRS, and Recording aware UAs follow the keep-alive 934 procedure recommended in "Application Mechanism for Keeping Alive the 935 NAT Mappings Associated to RTP/RTP Control Protocol (RTCP) Flows" 936 [RFC6263] for all RTP media streams. 938 8.1.7. RTCP Feedback Messages 940 "Codec Control Messages in the RTP Audio-Visual Profile with Feedback 941 (AVPF)" [RFC5104] specifies extensions to the messages defined in 942 AVPF [RFC4585]. Support for and proper usage of these messages is 943 important to SRC, SRS, and Recording aware UA implementations. Note 944 that these messages are applicable only when using the AVFP or SAVPF 945 RTP profiles 947 8.1.7.1. Full Intra Request 949 A Full Intra Request (FIR) Command, when received by the designated 950 media sender, requires that the media sender sends a Decoder Refresh 951 Point at the earliest opportunity. Using a decoder refresh point 952 implies refraining from using any picture sent prior to that point as 953 a reference for the encoding process of any subsequent picture sent 954 in the stream. 956 Decoder refresh points, especially Intra or IDR pictures for H.264 957 video codecs, are in general several times larger in size than 958 predicted pictures. Thus, in scenarios in which the available bit 959 rate is small, the use of a decoder refresh point implies a delay 960 that is significantly longer than the typical picture duration. 962 8.1.7.1.1. SIP INFO for FIR 964 "XML Schema for Media Control" [RFC5168] defines an Extensible Markup 965 Language (XML) Schema for video fast update. Implementations are 966 discouraged from using the method described except for backward 967 compatibility purposes. Implementations SHOULD use FIR messages 968 instead. 970 8.1.7.2. Picture Loss Indicator 972 Picture Loss Indication (PLI), as defined in [RFC4585], informs the 973 encoder of the loss of an undefined amount of coded video data 974 belonging to one or more pictures. Using the FIR command to recover 975 from errors is explicitly disallowed, and instead the PLI message 976 SHOULD be used. FIR SHOULD be used only in situations where not 977 sending a decoder refresh point would render the video unusable for 978 the users. Examples where sending FIR is appropriate include a 979 multipoint conference when a new user joins the conference and no 980 regular decoder refresh point interval is established, and a video 981 switching MCU that changes streams. 983 8.1.7.3. Temporary Maximum Media Stream Bit Rate Request 985 A receiver, translator, or mixer uses the Temporary Maximum Media 986 Stream Bit Rate Request (TMMBR) to request a sender to limit the 987 maximum bit rate for a media stream to the provided value. 988 Appropriate use of TMMBR facilitates rapid adaptation to changes in 989 available bandwidth. 991 8.1.7.3.1. Renegotiation of SDP bandwidth attribute 993 If it is likely that the new value indicated by TMMBR will be valid 994 for the remainder of the session, the TMMBR sender is expected to 995 perform a renegotiation of the session upper limit using the session 996 signaling protocol. Therefore for SIPREC, implementations are 997 RECOMMENDED to use TMMBR for temporary changes, and renegotiation of 998 bandwidth via SDP offer/answer for more permanent changes. 1000 8.1.8. Symmetric RTP/RTCP for Sending and Receiving 1002 Within an SDP offer/answer exchange, RTP entities choose the RTP and 1003 RTCP transport addresses (i.e., IP addresses and port numbers) on 1004 which to receive packets. When sending packets, the RTP entities may 1005 use the same source port or a different source port as those signaled 1006 for receiving packets. When the transport address used to send and 1007 receive RTP is the same, it is termed "symmetric RTP" [RFC4961]. 1008 Likewise, when the transport address used to send and receive RTCP is 1009 the same, it is termed "symmetric RTCP" [RFC4961]. 1011 When sending RTP, it is REQUIRED to use symmetric RTP. When sending 1012 RTCP, it is REQUIRED to use symmetric RTCP. Although an SRS will not 1013 normally send RTP, it will send RTCP as well as receive RTP and RTCP. 1014 Likewise, although an SRC will not normally receive RTP from the SRS, 1015 it will receive RTCP as well as send RTP and RTCP. 1017 Note: Symmetric RTP and symmetric RTCP are different from RTP/RTCP 1018 multiplexing [RFC5761]. 1020 8.2. Roles 1022 An SRC has the task of gathering media from the various UAs in one or 1023 more Communication Sessions (CSs) and forwarding the information to 1024 the SRS within the context of a corresponding Recording Session (RS). 1025 There are numerous ways in which an SRC may do this, including but 1026 not limited to, appearing as a UA within a CS, or as a B2BUA between 1027 UAs within a CS. 1029 (Recording Session) +---------+ 1030 +------------SIP------->| | 1031 | +------RTP/RTCP----->| SRS | 1032 | | +-- Metadata -->| | 1033 | | | +---------+ 1034 v v | 1035 +---------+ 1036 | SRC | 1037 |---------| (Communication Session) +---------+ 1038 | |<----------SIP---------->| | 1039 | UA-A | | UA-B | 1040 | |<-------RTP/RTCP-------->| | 1041 +---------+ +---------+ 1043 Figure 7: UA as SRC 1045 (Recording Session) +---------+ 1046 +------------SIP------->| | 1047 | +------RTP/RTCP----->| SRS | 1048 | | +-- Metadata -->| | 1049 | | | +---------+ 1050 v v | 1051 +---------+ 1052 | SRC | 1053 +---------+ |---------| +---------+ 1054 | |<----SIP----->| |<----SIP----->| | 1055 | UA-A | | B2BUA | | UA-B | 1056 | |<--RTP/RTCP-->| |<--RTP/RTCP-->| | 1057 +---------+ +---------+ +---------+ 1058 |_______________________________________________| 1059 (Communication Session) 1061 Figure 8: B2BUA as SRC 1063 The following subsections define a set of roles an SRC may choose to 1064 play based on its position with respect to a UA within a CS, and an 1065 SRS within an RS. A CS and a corresponding RS are independent 1066 sessions; therefore, an SRC may play a different role within a CS 1067 than it does within the corresponding RS. 1069 8.2.1. SRC acting as an RTP Translator 1071 The SRC may act as a translator, as defined in [RFC3550]. A defining 1072 characteristic of a translator is that it forwards RTP packets with 1073 their SSRC identifier intact. There are two types of translators, 1074 one that simply forwards, and another that performs transcoding 1075 (e.g., from one codec to another) in addition to forwarding. 1077 8.2.1.1. Forwarding Translator 1079 When acting as a forwarding translator, RTP received as separate 1080 streams from different sources (e.g., from different UAs with 1081 different SSRCs) cannot be mixed by the SRC and MUST be sent 1082 separately to the SRS. All RTCP reports MUST be passed by the SRC 1083 between the UAs and the SRS, such that the UAs and SRS are able to 1084 detect any SSRC collisions. 1086 RTCP Sender Reports generated by a UA sending a stream MUST be 1087 forwarded to the SRS. RTCP Receiver Reports generated by the SRS 1088 MUST be forwarded to the relevant UA. 1090 UAs may receive multiple sets of RTCP Receiver Reports, one or more 1091 from other UAs participating in the CS, and one from the SRS 1092 participating in the RS. A Recording aware UA SHOULD be prepared to 1093 process the RTCP Receiver Reports from the SRS, whereas a recording 1094 unaware UA may discard such RTCP packets as not of relevance. 1096 If SRTP is used on both the CS and the RS, decryption and/or re- 1097 encryption may occur. For example, if different keys are used, it 1098 will occur. If the same keys are used, it need not occur. 1099 Section 12 provides additional information on SRTP and keying 1100 mechanisms. 1102 If packet loss occurs, either from the UA to the SRC or from the SRC 1103 to the SRS, the SRS SHOULD detect and attempt to recover from the 1104 loss. The SRC does not play a role in this other than forwarding the 1105 associated RTP and RTCP packets. 1107 8.2.1.2. Transcoding Translator 1109 When acting as a transcoding translator, an SRC MAY perform 1110 transcoding (e.g., from one codec to another), and this may result in 1111 a different rate of packets between what the SRC receives and what 1112 the SRC sends. As when acting as a forwarding translator, RTP 1113 received as separate streams from different sources (e.g., from 1114 different UAs with different SSRCs) cannot be mixed by the SRC and 1115 MUST be sent separately to the SRS. All RTCP reports MUST be passed 1116 by the SRC between the UAs and the SRS, such that the UAs and SRS are 1117 able to detect any SSRC collisions. 1119 RTCP Sender Reports generated by a UA sending a stream MUST be 1120 forwarded to the SRS. RTCP Receiver Reports generated by the SRS 1121 MUST be forwarded to the relevant UA. The SRC may need to manipulate 1122 the RTCP Receiver Reports to take account of any transcoding that has 1123 taken place. 1125 UAs may receive multiple sets of RTCP Receiver Reports, one or more 1126 from other UAs participating in the CS, and one from the SRS 1127 participating in the RS. A Recording aware UA SHOULD be prepared to 1128 process the RTCP Receiver Reports from the SRS, whereas a recording 1129 unaware UA may discard such RTCP packets as not of relevance. 1131 If SRTP is used on both the CS and the RS, decryption and/or re- 1132 encryption may occur. For example, if different keys are used, it 1133 will occur. If the same keys are used, it need not occur. 1134 Section 12 provides additional information on SRTP and keying 1135 mechanisms. 1137 If packet loss occurs, either from the UA to the SRC or from the SRC 1138 to the SRS, the SRS SHOULD detect and attempt to recover from the 1139 loss. The SRC does not play a role in this other than forwarding the 1140 associated RTP and RTCP packets. 1142 8.2.2. SRC acting as an RTP Mixer 1144 In the case of the SRC acting as a RTP mixer, as defined in 1145 [RFC3550], the SRC combines RTP streams from different UA and sends 1146 them towards the SRS using its own SSRC. The SSRCs from the 1147 contributing UA SHOULD be conveyed as CSRCs identifiers within this 1148 stream. The SRC may make timing adjustments among the received 1149 streams and generate its own timing on the stream sent to the SRS. 1150 Optionally an SRC acting as a mixer can perform transcoding, and can 1151 even cope with different codings received from different UAs. RTCP 1152 Sender Reports and Receiver Reports are not forwarded by an SRC 1153 acting as mixer, but there are requirements for forwarding RTCP 1154 Source Description (SDES) packets. The SRC generates its own RTCP 1155 Sender and Receiver reports toward the associated UAs and SRS. 1157 The use of SRTP between the SRC and the SRS for the RS is independent 1158 of the use of SRTP between the UAs and SRC for the CS. Section 12 1159 provides additional information on SRTP and keying mechanisms. 1161 If packet loss occurs from the UA to the SRC, the SRC SHOULD detect 1162 and attempt to recover from the loss. If packet loss occurs from the 1163 SRC to the SRS, the SRS SHOULD detect and attempt to recover from the 1164 loss. 1166 8.2.3. SRC acting as an RTP Endpoint 1168 The case of the SRC acting as an RTP endpoint, as defined in 1169 [RFC3550], is similar to the mixer case, except that the RTP session 1170 between the SRC and the SRS is considered completely independent from 1171 the RTP session that is part of the CS. The SRC can, but need not, 1172 mix RTP streams from different participants prior to sending to the 1173 SRS. RTCP between the SRC and the SRS is completely independent of 1174 RTCP on the CS. 1176 The use of SRTP between the SRC and the SRS for the RS is independent 1177 of the use of SRTP between the UAs and SRC for the CS. Section 12 1178 provides additional information on SRTP and keying mechanisms. 1180 If packet loss occurs from the UA to the SRC, the SRC SHOULD detect 1181 and attempt to recover from the loss. If packet loss occurs from the 1182 SRC to the SRS, the SRS SHOULD detect and attempt to recover from the 1183 loss. 1185 8.3. RTP Session Usage by SRC 1187 There are multiple ways that an SRC may choose to deliver recorded 1188 media to an SRS. In some cases, it may use a single RTP session for 1189 all media within the RS, whereas in others it may use multiple RTP 1190 sessions. The following subsections provide examples of basic RTP 1191 session usage by the SRC, including a discussion of how the RTP 1192 constructs and mechanisms covered previously are used. An SRC may 1193 choose to use one or more of the RTP session usages within a single 1194 RS. For the purpose of base interoperability between SRC and SRS, an 1195 SRC MUST support separate m-lines in SDP, one per CS media direction. 1196 The set of RTP session usages described is not meant to be 1197 exhaustive. 1199 8.3.1. SRC Using Multiple m-lines 1201 When using multiple m-lines, an SRC includes each m-line in an SDP 1202 offer to the SRS. The SDP answer from the SRS MUST include all 1203 m-lines, with any rejected m-lines indicated with a zero port, per 1204 [RFC3264]. Having received the answer, the SRC starts sending media 1205 to the SRS as indicated in the answer. Alternatively, if the SRC 1206 deems the level of support indicated in the answer to be 1207 unacceptable, it may initiate another SDP offer/answer exchange in 1208 which an alternative RTP session usage is negotiated. 1210 In order to preserve the mapping of media to participant within the 1211 CSs in the RS, the SRC SHOULD map each unique CNAME within the CSs to 1212 a unique CNAME within the RS. Additionally, the SRC SHOULD map each 1213 unique combination of CNAME/SSRC within the CSs to a unique CNAME/ 1214 SSRC within the RS. In doing to, the SRC may act as an RTP 1215 translator or as an RTP endpoint. 1217 The following figure illustrates a case in which each UA represents a 1218 participant contributing two RTP sessions (e.g. one for audio and one 1219 for video), each with a single SSRC. The SRC acts as an RTP 1220 translator and delivers the media to the SRS using four RTP sessions, 1221 each with a single SSRC. The CNAME and SSRC values used by the UAs 1222 within their media streams are preserved in the media streams from 1223 the SRC to the SRS. 1225 +---------+ 1226 +------------SSRC Aa--->| | 1227 | + --------SSRC Av--->| | 1228 | | +------SSRC Ba--->| SRS | 1229 | | | +---SSRC Bv--->| | 1230 | | | | +---------+ 1231 | | | | 1232 | | | | 1233 +---------+ +----------+ +---------+ 1234 | |---SSRC Aa-->| SRC |<--SSRC Ba---| | 1235 | UA-A | |(CNAME-A, | | UA-B | 1236 |(CNAME-A)|---SSRC Av-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)| 1237 +---------+ +----------+ +---------+ 1239 Figure 9: SRC Using Multiple m-lines 1241 8.3.2. SRC Using Mixing 1243 When using mixing, the SRC combines RTP streams from different 1244 participants and sends them towards the SRS using its own SSRC. The 1245 SSRCs from the contributing participants SHOULD be conveyed as CSRCs 1246 identifiers. The SRC includes one m-line for each RTP session in an 1247 SDP offer to the SRS. The SDP answer from the SRS MUST include all 1248 m-lines, with any rejected m-lines indicated with the zero port, per 1249 [RFC3264]. Having received the answer, the SRC starts sending media 1250 to the SRS as indicated in the answer. 1252 In order to preserve the mapping of media to participant within the 1253 CSs in the RS, the SRC SHOULD map each unique CNAME within the CSs to 1254 a unique CNAME within the RS. Additionally, the SRC SHOULD map each 1255 unique combination of CNAME/SSRC within the CSs to a unique CNAME/ 1256 SSRC within the RS. The SRC MUST avoid SSRC collisions, rewriting 1257 SSRCs if necessary when used as CSRCs in the RS. In doing to, the 1258 SRC acts as an RTP mixer. 1260 In the event the SRS does not support this usage of CSRC values, it 1261 relies entirely on the SIPREC metadata to determine the participants 1262 included within each mixed stream. 1264 The following figure illustrates a case in which each UA represents a 1265 participant contributing two RTP sessions (e.g. one for audio and one 1266 for video), each with a single SSRC. The SRC acts as an RTP mixer 1267 and delivers the media to the SRS using two RTP sessions, mixing 1268 media from each participant into a single RTP session containing a 1269 single SSRC and two CSRCs. 1271 SSRC Sa +---------+ 1272 +-------CSRC Aa,Ba--->| | 1273 | | | 1274 | SSRC Sa | SRS | 1275 | +---CSRC Av,Bv--->| | 1276 | | +---------+ 1277 | | 1278 +----------+ 1279 +---------+ | SRC | +---------+ 1280 | |---SSRC Aa-->|(CNAME-S, |<--SSRC Ba---| | 1281 | UA-A | | CNAME-A, | | UA-B | 1282 |(CNAME-A)|---SSRC Aa-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)| 1283 +---------+ +----------+ +---------+ 1285 Figure 10: SRC Using Mixing 1287 8.4. RTP Session Usage by SRS 1289 An SRS that supports recording an audio CS MUST support SRC usage of 1290 separate audio m-lines in SDP, one per CS media direction. An SRS 1291 that supports recording an video CS MUST support SRC usage of 1292 separate video m-lines in SDP, one per CS media direction. 1293 Therefore, for an SRS supporting a typical audio call, the SRS has to 1294 support receiving at least two audio m-lines. For an SRS supporting 1295 a typical audio and video call, the SRS has to support receiving at 1296 least four total m-lines in the SDP, two audio m-lines and two video 1297 m-lines. 1299 These requirements allow an SRS to be implemented that supports video 1300 only, without requiring support for audio recording. They also allow 1301 an SRS to be implemented that supports recording only one direction 1302 of one stream in a CS; for example, an SRS designed to record 1303 security monitoring cameras that only send (not receive) video 1304 without any audio. These requirements were not written to prevent 1305 other modes being implemented and used, such as using a single m-line 1306 and mixing the separate audio streams together. Rather, the 1307 requirements were written to provide a common base mode to implement 1308 for the sake of interoperability. It is important to note that an 1309 SRS implementation supporting the common base may not record all 1310 media streams in a CS if a participant supports more than one m-line 1311 in a video call, such as one for camera and one for presentation. 1312 SRS implementations may support other modes as well, but have to at 1313 least support the ones above such that they interoperate in the 1314 common base mode for basic interoperability. 1316 9. Metadata 1318 Some metadata attributes are contained in SDP, and others are 1319 contained in a new content type "application/rs-metadata". The 1320 format of the metadata is described as part of the mechanism in 1321 [I-D.ietf-siprec-metadata]. A new "disposition-type" of Content- 1322 Disposition is defined for the purpose of carrying metadata. The 1323 value is "recording-session", which indicates the "application/rs- 1324 metadata" content contains metadata to be handled by the SRS. 1326 9.1. Procedures at the SRC 1328 The SRC MUST send metadata to the SRS in an RS. The SRC SHOULD send 1329 metadata as soon as it becomes available and whenever it changes. 1330 Cases in which an SRC may be justified in waiting temporarily before 1331 sending metadata include: 1333 o waiting for previous metadata exchange to complete (i.e. cannot 1334 send another SDP offer until previous offer/answer completes, and 1335 may prefer not to send an UPDATE during this time either). 1337 o constraining the signaling rate on the RS. 1339 o sending metadata when key events occur rather than for every event 1340 that has any impact on metadata. 1342 o desire to suppress certain metadata out of concern for privacy or 1343 perceived lack of need for it to be included in the recording. 1345 Metadata sent by the SRC is categorized as either a full metadata 1346 snapshot or a partial update. A full metadata snapshot describes all 1347 metadata associated with the RS. The SRC MAY send a full metadata 1348 snapshot at any time. The SRC MAY send a partial update only if a 1349 full metadata snapshot has been sent previously. 1351 The SRC MAY send metadata (either a full metadata snapshot or a 1352 partial update) in an INVITE request, an UPDATE request [RFC3311], or 1353 an 200 response to an offerless INVITE from the SRS. If the metadata 1354 contains a reference to any SDP labels, the request containing the 1355 metadata MUST also contain an SDP offer that defines those labels. 1357 When a SIP message contains both an SDP offer and metadata, the 1358 request body MUST have content type "multipart/mixed", with one 1359 subordinate body part containing the SDP offer and another containing 1360 the metadata. When a SIP message contains only an SDP offer or 1361 metadata, the "multipart/mixed" container is optional. 1363 The SRC SHOULD include a full metadata snapshot in the initial INVITE 1364 request establishing the RS. If metadata is not yet available (e.g 1365 an RS established in absence of a CS), the SRC SHOULD send a full 1366 metadata snapshot as soon as metadata becomes available. 1368 If the SRC receives a snapshot request from the SRS, it MUST 1369 immediately send a full metadata snapshot. 1371 The following is an example of a full metadata snapshot sent by the 1372 SRC in the initial INVITE request: 1374 INVITE sip:recorder@example.com SIP/2.0 1375 Via: SIP/2.0/TCP src.example.com;branch=z9hG4bKdf6b622b648d9 1376 From: ;tag=35e195d2-947d-4585-946f-09839247 1377 To: 1378 Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a 1379 CSeq: 101 INVITE 1380 Max-Forwards: 70 1381 Require: siprec 1382 Accept: application/sdp, application/rs-metadata-request 1383 Contact: ;+sip.src 1384 Content-Type: multipart/mixed;boundary=foobar 1385 Content-Length: [length] 1387 --foobar 1388 Content-Type: application/sdp 1390 v=0 1391 o=SRS 2890844526 2890844526 IN IP4 198.51.100.1 1392 s=- 1393 c=IN IP4 198.51.100.1 1394 t=0 0 1395 m=audio 12240 RTP/AVP 0 4 8 1396 a=sendonly 1397 a=label:1 1399 --foobar 1400 Content-Type: application/rs-metadata 1401 Content-Disposition: recording-session 1403 [metadata content] 1405 Figure 11: Sample INVITE request for the recording session 1407 9.2. Procedures at the SRS 1409 The SRS receives metadata updates from the SRC in INVITE and UPDATE 1410 requests. Since the SRC can send partial updates based on the 1411 previous update, the SRS needs to keep track of the sequence of 1412 updates from the SRC. 1414 In the case of an internal failure at the SRS, the SRS may fail to 1415 recognize a partial update from the SRC. The SRS may be able to 1416 recover from the internal failure by requesting for a full metadata 1417 snapshot from the SRC. Certain errors, such as syntax errors or 1418 semantic errors in the metadata information, are likely caused by an 1419 error on the SRC side, and it is likely the same error will occur 1420 again even when a full metadata snapshot is requested. In order to 1421 avoid repeating the same error, the SRS can simply terminate the 1422 recording session when a syntax error or semantic error is detected 1423 in the metadata. 1425 The SRS MAY explicitly request a full metadata snapshot by sending an 1426 UPDATE request. This request MUST contain a body with content 1427 disposition type "recording-session", and MUST NOT contain an SDP 1428 body. The SRS MUST NOT request a full metadata snapshot in an UPDATE 1429 response or in any other SIP transaction. The format of the content 1430 is "application/rs-metadata-request", and the body format is a simple 1431 text-based format. The following shows an example: 1433 UPDATE sip:2000@src.exmaple.com SIP/2.0 1434 Via: SIP/2.0/UDP srs.example.com;branch=z9hG4bKdf6b622b648d9 1435 To: ;tag=35e195d2-947d-4585-946f-098392474 1436 From: ;tag=1234567890 1437 Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a 1438 CSeq: 1 UPDATE 1439 Max-Forwards: 70 1440 Require: siprec 1441 Contact: ;+sip.srs 1442 Accept: application/sdp, application/rs-metadata 1443 Content-Disposition: recording-session 1444 Content-Type: application/rs-metadata-request 1445 Content-Length: [length] 1447 SRS internal error 1449 Figure 12: Metadata Request 1451 The SRS MAY include the reason why a metadata snapshot request is 1452 being made to the SRC in the reason line. This reason line is free 1453 form text, mainly designed for logging purposes on the SRC side. The 1454 processing of the content by the SRC is entirely optional since the 1455 content is for logging only, and the snapshot request itself is 1456 indicated by the use of the application/rs-metadata-request content 1457 type. 1459 When the SRC receives a request for a metadata snapshot, it MUST 1460 immediately provide a full metadata snapshot in a separate INVITE or 1461 UPDATE transaction. Any subsequent partial updates will not be 1462 dependent on any metadata sent prior to this full metadata snapshot. 1464 The metadata received by the SRS can contain ID elements used to 1465 cross reference one element to another. An element containing the 1466 definition of an ID, and an element containing a reference to that ID 1467 will often be received from the same SRC. It is also valid for those 1468 elements to be received from different SRCs, for example, when each 1469 endpoint in the same CS act as an SRC to record the call and a common 1470 ID refers to the same CS. The SRS MUST NOT consider this an error. 1472 9.2.1. Formal Syntax 1474 The formal syntax for the application/rs-metadata-request MIME is 1475 described below using the augmented Backus-Naur Form (BNF) as 1476 described in [RFC5234]. 1478 snapshot-request = srs-reason-line CRLF 1479 srs-reason-line = [TEXT-UTF8-TRIM] 1480 ; TEXT-UTF8-TRIM defined in RFC3261 1482 10. Persistent Recording 1484 Persistent recording is a specific use case outlined in REQ-005 or 1485 Use Case 4 in [RFC6341], where a recording session can be established 1486 in the absence of a communication session. The SRC continuously 1487 records media in a recording session to the SRS even in the absence 1488 of a CS for all user agents that are part of persistent recording. 1489 By allocating recorded streams and continuously sending recorded 1490 media to the SRS, the SRC does not have to prepare new recorded 1491 streams with new SDP offer when a new communication session is 1492 created and also does not impact the timing of the CS. The SRC only 1493 needs to update the metadata when new communication sessions are 1494 created. 1496 When there is no communication sessions running on the devices with 1497 persistent recording, there is no recorded media to stream from the 1498 SRC to the SRS. In certain environments where Network Address 1499 Translator (NAT) is used, typically a minimum of flow activity is 1500 required to maintain the NAT binding for each port opened. Agents 1501 that support Interactive Connectivity Establishment (ICE) solves this 1502 problem. For non-ICE agents, in order not to lose the NAT bindings 1503 for the RTP/RTCP ports opened for the recorded streams, the SRC and 1504 SRS SHOULD follow the recommendations provided in [RFC6263] to 1505 maintain the NAT bindings. 1507 11. IANA Considerations 1509 11.1. Registration of Option Tags 1511 This specification registers two option tags. The required 1512 information for this registration, as specified in [RFC3261], is as 1513 follows. 1515 11.1.1. siprec Option Tag 1517 Name: siprec 1519 Description: This option tag is for identifying the SIP session 1520 for the purpose of recording session only. This is typically not 1521 used in a Supported header. When present in a Require header in a 1522 request, it indicates that the UAS MUST be either a SRC or SRS 1523 capable of handling the contexts of a recording session. 1525 11.1.2. record-aware Option Tag 1527 Name: record-aware 1529 Description: This option tag is to indicate the ability for the 1530 user agent to receive recording indicators in media level or 1531 session level SDP. When present in a Supported header, it 1532 indicates that the UA can receive recording indicators in media 1533 level or session level SDP. 1535 11.2. Registration of media feature tags 1537 This document registers two new media feature tags in the SIP tree 1538 per the process defined in [RFC2506] and [RFC3840] 1540 11.2.1. src feature tag 1542 Media feature tag name: sip.src 1544 ASN.1 Identifier: 25 1545 Summary of the media feature indicated by this tag: This feature 1546 tag indicates that the user agent is a Session Recording Client 1547 for the purpose for Recording Session. 1549 Values appropriate for use with this feature tag: boolean 1551 The feature tag is intended primarily for use in the following 1552 applications, protocols, services, or negotiation mechanisms: This 1553 feature tag is only useful for a Recording Session. 1555 Examples of typical use: Routing the request to a Session 1556 Recording Server. 1558 Security Considerations: Security considerations for this media 1559 feature tag are discussed in Section 11.1 of RFC 3840. 1561 11.2.2. srs feature tag 1563 Media feature tag name: sip.srs 1565 ASN.1 Identifier: 26 1567 Summary of the media feature indicated by this tag: This feature 1568 tag indicates that the user agent is a Session Recording Server 1569 for the purpose for Recording Session. 1571 Values appropriate for use with this feature tag: boolean 1573 The feature tag is intended primarily for use in the following 1574 applications, protocols, services, or negotiation mechanisms: This 1575 feature tag is only useful for a Recording Session. 1577 Examples of typical use: Routing the request to a Session 1578 Recording Client. 1580 Security Considerations: Security considerations for this media 1581 feature tag are discussed in Section 11.1 of RFC 3840. 1583 11.3. New Content-Disposition Parameter Registrations 1585 This document registers a new "disposition-type" value in Content- 1586 Disposition header: recording-session. 1588 recording-session the body describes the metadata information about 1589 the recording session 1591 11.4. Media Type Registration 1593 11.4.1. Registration of MIME Type application/rs-metadata 1595 This document registers the application/rs-metadata MIME media type 1596 in order to describe the recording session metadata. This media type 1597 is defined by the following information: 1599 Media type name: application 1601 Media subtype name: rs-metadata 1603 Required parameters: none 1605 Options parameters: none 1607 11.4.2. Registration of MIME Type application/rs-metadata-request 1609 This document registers the application/rs-metadata-request MIME 1610 media type in order to describe a recording session metadata snapshot 1611 request. This media type is defined by the following information: 1613 Media type name: application 1615 Media subtype name: rs-metadata-request 1617 Required parameters: none 1619 Options parameters: none 1621 11.5. SDP Attributes 1623 This document registers the following new SDP attributes. 1625 11.5.1. 'record' SDP Attribute 1627 Contact names: Leon Portman leon.portman@nice.com, Henry Lum 1628 henry.lum@genesyslab.com 1630 Attribute name: record 1632 Long form attribute name: Recording Indication 1634 Type of attribute: session or media level 1636 Subject to charset: no 1637 This attribute provides the recording indication for the session or 1638 media stream. 1640 Allowed attribute values: on, off, paused 1642 11.5.2. 'recordpref' SDP Attribute 1644 Contact names: Leon Portman leon.portman@nice.com, Henry Lum 1645 henry.lum@genesyslab.com 1647 Attribute name: recordpref 1649 Long form attribute name: Recording Preference 1651 Type of attribute: session or media level 1653 Subject to charset: no 1655 This attribute provides the recording preference for the session or 1656 media stream. 1658 Allowed attribute values: on, off, pause, nopreference 1660 12. Security Considerations 1662 The recording session is fundamentally a standard SIP dialog 1663 [RFC3261], therefore, the recording session can reuse any of the 1664 existing SIP security mechanisms available for securing the session 1665 signaling, the recorded media, and the metadata. The use cases and 1666 requirements document [RFC6341] outlines the general security 1667 considerations, and this document describes specific security 1668 recommendations. 1670 The SRC and SRS MUST support SIP with TLS and MAY support SIPS with 1671 TLS as per [RFC5630]. The Recording Session SHOULD be at least as 1672 secure as the Communication Session, meaning using at least the same 1673 strength of cipher suite as the CS if the CS is secured. For 1674 example, if the CS uses SIPS for signaling and RTP/SAVP for media, 1675 then the RS should not downgrade the level of security in the RS to 1676 SIP or plain RTP since doing so will mean an automatic security 1677 downgrade for the CS. In deployments where the SRC and the SRS are 1678 in the same administrative domain and the same physical switch that 1679 prevents outside user access, some SRCs may choose to lower the level 1680 of security when establishing a recording session. While physically 1681 securing the SRC and SRS may prevent an outside attacker from 1682 accessing important call recordings, this still does not prevent an 1683 inside attacker from accessing the internal network to gain access to 1684 the call recordings. 1686 12.1. Authentication and Authorization 1688 At the transport level, the recording session uses TLS authentication 1689 to validate the authenticity of the SRC and SRS. The SRC and SRS 1690 MUST implement TLS mutual authentication for establishing the 1691 recording session. Whether the SRC/SRS chooses to use TLS mutual 1692 authentication is a deployment decision. In deployments where the 1693 SRC and the SRS are in the same administrative domain, the SRC and 1694 SRS may choose not to authenticate each other, or to have the SRC 1695 authenticate the SRS only, as there is an inherent trust relation 1696 between the SRC and the SRS when they are hosted in the same 1697 administrative domain. In deployments where the SRS can be hosted on 1698 a different administrative domain, it is important to perform mutual 1699 authentication to ensure the authenticity of both the SRC and the SRS 1700 before transmitting any recorded media. The risk of not 1701 authenticating the SRS is that the recording may be sent to a 1702 compromised SRS and that a sensitive call recording will be obtained 1703 by an attacker. On the other hand, the risk of not authenticating 1704 the SRC is that an SRS will accept calls from an unknown SRC and 1705 allow potential forgery of call recordings. 1707 There may be scenarios in which the signaling between the SRC and SRS 1708 is not direct, e.g. a SIP proxy exists between the SRC and the SRS. 1709 In such scenarios, each hop is subject to the TLS mutual 1710 authentication constraint and transitive trust at each hop is 1711 utilized. Additionally, an SRC or SRS may use other existing SIP 1712 mechanisms available, including but not limited to, Digest 1713 Authentication [RFC3261], Asserted Identity [RFC3325], and Connected 1714 Identity [RFC4916]. 1716 The SRS may have its own set of recording policies to authorize 1717 recording requests from the SRC. The use of recording policies is 1718 outside the scope of the Session Recording Protocol. 1720 12.2. RTP handling 1722 In many scenarios it will be critical for the media transported 1723 between the SRC and the SRS to be protected. Media encryption is an 1724 important element in the overall SIPREC solution; therefore the SRC 1725 and the SRS MUST support RTP/SAVP [RFC3711] and RTP/SAVPF [RFC5124]. 1726 RTP/SAVP and RTP/SAVPF provide media encryption, integrity 1727 protection, replay protection, and a limited form of source 1728 authentication. They do not contain or require a specific keying 1729 mechanism. At a minimum, the SRC and SRS MUST support the SDP 1730 Security Descriptions (SDES) key negotiation mechanism [RFC4568]. 1731 For cases in which DTLS-SRTP is used to encrypt a CS media stream, an 1732 SRC may use SRTP Encrypted Key Transport (EKT) 1734 [I-D.ietf-avt-srtp-ekt] in order to use SRTP-SDES in the RS without 1735 needing to re-encrypt the media. 1737 When RTP/SAVP or RTP/SAVPF is used, an SRC can choose to use the same 1738 or different keys in the RS than the ones used in the CS. Some SRCs 1739 are designed to simply replicate RTP packets from a CS media stream 1740 to the SRS, in which case the SRC will use the same key in the RS as 1741 used in the CS. In this case, the SRC MUST secure the SDP containing 1742 the keying material in the RS with at least the same level of 1743 security as in the CS. The risk of lowering the level of security in 1744 the RS is that it will effectively become a downgrade attack on the 1745 CS since the same key is used for both CS and RS. 1747 SRCs that decrypt an encrypted CS media stream and re-encrypt it when 1748 sending it to the SRS MUST use a different key for the RS media 1749 stream than what is used for the CS media stream, to ensure that it 1750 is not possible for someone who has the key for the CS media stream 1751 to access recorded data they are not authorized to access. 1753 12.3. Metadata 1755 Metadata contains sensitive information such as the address of record 1756 of the participants and other extension data placed by the SRC. It 1757 is essential to protect the content of the metadata in the RS. Since 1758 metadata is a content type transmitted in SIP signaling, metadata 1759 SHOULD be protected at the transport level by SIPS/TLS. 1761 12.4. Storage and playback 1763 While storage and playback of the call recording is beyond the scope 1764 of this document, it is worthwhile to mention here that it is also 1765 important for the recording storage and playback to provide a level 1766 of security that is comparable to the communication session. It 1767 would defeat the purpose of securing both the communication session 1768 and the recording session mentioned in the previous sections if the 1769 recording can be easily played back with a simple unsecured HTTP 1770 interface without any form of authentication or authorization. 1772 13. Acknowledgements 1774 We want to thank John Elwell, Paul Kyzivat, Partharsarathi R, Ram 1775 Mohan R, Hadriel Kaplan, Adam Roach, Miguel Garcia, Thomas Stach, 1776 Muthu Perumal, Dan Wing, and Magnus Westerlund for their valuable 1777 comments and inputs to this document. 1779 14. References 1781 14.1. Normative References 1783 [I-D.ietf-siprec-metadata] 1784 R, R., Ravindran, P., and P. Kyzivat, "Session Initiation 1785 Protocol (SIP) Recording Metadata", draft-ietf-siprec- 1786 metadata-15 (work in progress), February 2014. 1788 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1789 Requirement Levels", BCP 14, RFC 2119, March 1997. 1791 [RFC2506] Holtman, K., Mutz, A., and T. Hardie, "Media Feature Tag 1792 Registration Procedure", BCP 31, RFC 2506, March 1999. 1794 [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, 1795 A., Peterson, J., Sparks, R., Handley, M., and E. 1796 Schooler, "SIP: Session Initiation Protocol", RFC 3261, 1797 June 2002. 1799 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 1800 with Session Description Protocol (SDP)", RFC 3264, June 1801 2002. 1803 [RFC3840] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, 1804 "Indicating User Agent Capabilities in the Session 1805 Initiation Protocol (SIP)", RFC 3840, August 2004. 1807 [RFC4574] Levin, O. and G. Camarillo, "The Session Description 1808 Protocol (SDP) Label Attribute", RFC 4574, August 2006. 1810 [RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax 1811 Specifications: ABNF", STD 68, RFC 5234, January 2008. 1813 14.2. Informative References 1815 [I-D.ietf-avt-srtp-ekt] 1816 Wing, D., McGrew, D., and K. Fischer, "Encrypted Key 1817 Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03 1818 (work in progress), October 2011. 1820 [I-D.ietf-siprec-architecture] 1821 Hutton, A., Portman, L., Jain, R., and K. Rehor, "An 1822 Architecture for Media Recording using the Session 1823 Initiation Protocol", draft-ietf-siprec-architecture-11 1824 (work in progress), December 2013. 1826 [RFC3311] Rosenberg, J., "The Session Initiation Protocol (SIP) 1827 UPDATE Method", RFC 3311, October 2002. 1829 [RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private 1830 Extensions to the Session Initiation Protocol (SIP) for 1831 Asserted Identity within Trusted Networks", RFC 3325, 1832 November 2002. 1834 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1835 Jacobson, "RTP: A Transport Protocol for Real-Time 1836 Applications", STD 64, RFC 3550, July 2003. 1838 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 1839 Video Conferences with Minimal Control", STD 65, RFC 3551, 1840 July 2003. 1842 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 1843 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 1844 RFC 3711, March 2004. 1846 [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session 1847 Description Protocol (SDP) Security Descriptions for Media 1848 Streams", RFC 4568, July 2006. 1850 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, 1851 "Extended RTP Profile for Real-time Transport Control 1852 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 1853 2006. 1855 [RFC4916] Elwell, J., "Connected Identity in the Session Initiation 1856 Protocol (SIP)", RFC 4916, June 2007. 1858 [RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)", 1859 BCP 131, RFC 4961, July 2007. 1861 [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, 1862 "Codec Control Messages in the RTP Audio-Visual Profile 1863 with Feedback (AVPF)", RFC 5104, February 2008. 1865 [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for 1866 Real-time Transport Control Protocol (RTCP)-Based Feedback 1867 (RTP/SAVPF)", RFC 5124, February 2008. 1869 [RFC5168] Levin, O., Even, R., and P. Hagendorf, "XML Schema for 1870 Media Control", RFC 5168, March 2008. 1872 [RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session 1873 Initiation Protocol (SIP)", RFC 5630, October 2009. 1875 [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and 1876 Control Packets on a Single Port", RFC 5761, April 2010. 1878 [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for 1879 Keeping Alive the NAT Mappings Associated with RTP / RTP 1880 Control Protocol (RTCP) Flows", RFC 6263, June 2011. 1882 [RFC6341] Rehor, K., Portman, L., Hutton, A., and R. Jain, "Use 1883 Cases and Requirements for SIP-Based Media Recording 1884 (SIPREC)", RFC 6341, August 2011. 1886 [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, 1887 "Guidelines for Choosing RTP Control Protocol (RTCP) 1888 Canonical Names (CNAMEs)", RFC 7022, September 2013. 1890 Authors' Addresses 1892 Leon Portman 1893 NICE Systems 1894 22 Zarhin Street 1895 P.O. Box 690 1896 Ra'anana 4310602 1897 Israel 1899 Email: leon.portman@gmail.com 1901 Henry Lum (editor) 1902 Genesys 1903 1380 Rodick Road, Suite 201 1904 Markham, Ontario L3R4G5 1905 Canada 1907 Email: henry.lum@genesyslab.com 1909 Charles Eckel 1910 Cisco 1911 170 West Tasman Drive 1912 San Jose, CA 95134 1913 United States 1915 Email: eckelcu@cisco.com 1916 Alan Johnston 1917 Avaya 1918 St. Louis, MO 63124 1920 Email: alan.b.johnston@gmail.com 1922 Andrew Hutton 1923 Unify 1924 Brickhill Street 1925 Milton Keynes MK15 0DJ 1926 United Kingdom 1928 Email: andrew.hutton@unify.com