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'86') (Obsoleted by RFC 5245) == Outdated reference: A later version (-16) exists of draft-ietf-mmusic-ice-tcp-00 == Outdated reference: A later version (-04) exists of draft-ietf-mmusic-securityprecondition-01 == Outdated reference: A later version (-07) exists of draft-ietf-mmusic-connectivity-precon-02 == Outdated reference: A later version (-06) exists of draft-ietf-mmusic-sdp-media-content-03 == Outdated reference: A later version (-08) exists of draft-ietf-sipping-policy-package-00 == Outdated reference: A later version (-03) exists of draft-ietf-sip-hop-limit-diagnostics-02 == Outdated reference: A later version (-15) exists of draft-ietf-sip-certs-00 Summary: 3 errors (**), 0 flaws (~~), 22 warnings (==), 16 comments (--). Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 SIP J. Rosenberg 3 Internet-Draft Cisco Systems 4 Expires: December 21, 2006 June 19, 2006 6 A Hitchhikers Guide to the Session Initiation Protocol (SIP) 7 draft-ietf-sip-hitchhikers-guide-00 9 Status of this Memo 11 By submitting this Internet-Draft, each author represents that any 12 applicable patent or other IPR claims of which he or she is aware 13 have been or will be disclosed, and any of which he or she becomes 14 aware will be disclosed, in accordance with Section 6 of BCP 79. 16 Internet-Drafts are working documents of the Internet Engineering 17 Task Force (IETF), its areas, and its working groups. Note that 18 other groups may also distribute working documents as Internet- 19 Drafts. 21 Internet-Drafts are draft documents valid for a maximum of six months 22 and may be updated, replaced, or obsoleted by other documents at any 23 time. It is inappropriate to use Internet-Drafts as reference 24 material or to cite them other than as "work in progress." 26 The list of current Internet-Drafts can be accessed at 27 http://www.ietf.org/ietf/1id-abstracts.txt. 29 The list of Internet-Draft Shadow Directories can be accessed at 30 http://www.ietf.org/shadow.html. 32 This Internet-Draft will expire on December 21, 2006. 34 Copyright Notice 36 Copyright (C) The Internet Society (2006). 38 Abstract 40 The Session Initiation Protocol (SIP) is the subject of numerous 41 specifications that have been produced by the IETF. It can be 42 difficult to locate the right document, or even to determine the set 43 of Request for Comments (RFC) about SIP. Don't Panic! This 44 specification serves as a guide to the SIP RFC series. It lists the 45 specifications under the SIP umbrella, briefly summarizes each, and 46 groups them into categories. 48 Table of Contents 50 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 51 2. Scope of this Document . . . . . . . . . . . . . . . . . . . . 3 52 3. Core SIP Specifications . . . . . . . . . . . . . . . . . . . 4 53 4. Public Switched Telephone Network (PSTN) Interworking . . . . 7 54 5. General Purpose Infrastructure Extensions . . . . . . . . . . 8 55 6. Minor Extensions . . . . . . . . . . . . . . . . . . . . . . . 10 56 7. Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . 12 57 8. Call Control Primitives . . . . . . . . . . . . . . . . . . . 12 58 9. Event Framework and Packages . . . . . . . . . . . . . . . . . 13 59 10. Quality of Service . . . . . . . . . . . . . . . . . . . . . . 15 60 11. Operations and Management . . . . . . . . . . . . . . . . . . 15 61 12. SIP Compression . . . . . . . . . . . . . . . . . . . . . . . 16 62 13. SIP Service URIs . . . . . . . . . . . . . . . . . . . . . . . 16 63 14. Security Mechanisms . . . . . . . . . . . . . . . . . . . . . 17 64 15. Instant Messaging and Presence . . . . . . . . . . . . . . . . 18 65 16. Emergency Services . . . . . . . . . . . . . . . . . . . . . . 18 66 17. Security Considerations . . . . . . . . . . . . . . . . . . . 19 67 18. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 19 68 19. Informative References . . . . . . . . . . . . . . . . . . . . 19 69 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 28 70 Intellectual Property and Copyright Statements . . . . . . . . . . 29 72 1. Introduction 74 The Session Initiation Protocol (SIP) [1] is the subject of numerous 75 specifications that have been produced by the IETF. It can be 76 difficult to locate the right document, or even to determine the set 77 of Request for Comments (RFC) about SIP. Don't Panic! [42] This 78 specification serves as a guide to the SIP RFC series. It lists the 79 specifications under the SIP umbrella. For each specification, a 80 paragraph or so description is included that summarizes the purpose 81 of the specification. Each specification also includes a letter that 82 designates its category in the standards track [2]. These values 83 are: 85 S: Standards Track (Proposed Standard, Draft Standard, or Standard) 87 E: Experimental 89 B: Best Current Practice 91 I: Informational 93 The specifications are grouped together by topic. Typically, SIP 94 extensions fit naturally into topic areas, and implementations 95 interested in a particular topic often implement many or all of the 96 specifications in that area. There are some specifications which 97 fall into multiple topic areas, in which case they are listed more 98 than once. 100 This document itself is not an update to RFC 3261 or an extension to 101 SIP. It is an informational document, meant to guide newcomers and 102 implementors to the SIP suite of specifications. 104 2. Scope of this Document 106 It is very difficult to enumerate the set of SIP specifications. 107 This is because there are many protocols that are intimately related 108 to SIP and used by nearly all SIP implementations, but are not 109 formally SIP extensions. As such, this document formally defines a 110 "SIP specification" as: 112 o Any specification that defines an extension to SIP itself, where 113 an extension is a mechanism that changes or updates in some way a 114 behavior specified in RFC 3261 116 o Any specification that defines an extension to SDP whose primary 117 purpose is to support SIP 119 o Any specification that defines a MIME object whose primary purpose 120 is to support SIP 122 Excluded from this list are requirements, architectures, registry 123 definitions, non-normative frameworks, and processes. Best Current 124 Practices are included when they are effectively standard mechanisms 125 for accomplishing a task. 127 The SIP change process [8] defines two types of extensions to SIP. 128 These are normal extensions and the so-called P-headers, which are 129 meant to be used in areas of limited applicability. P-headers cannot 130 be defined in the standards track. For the most part, P-headers are 131 not included in the listing here, with the exception of those which 132 have seen general usage despite their P-header status. 134 3. Core SIP Specifications 136 The core SIP specifications represent the set of specifications whose 137 functionality is broadly applicable. An extension is broadly 138 applicable if it fits into one of the following categories: 140 o For specifications that impact SIP session management, the 141 extension would be used for almost every session initiated by a 142 user agent 144 o For specifications that impact SIP registrations, the extension 145 would be used for almost every registration initiated by a user 146 agent 148 o For specifications that impact SIP subscriptions, the extension 149 would be used for almost every subscription initiated by a user 150 agent 152 In other words, these are not specifications that are used just for 153 some requests and not others; they are specifications that would 154 apply to each and every request that the extension is relevant for. 156 RFC 3261, The Session Initiation Protocol (S): RFC 3261 [1] is the 157 core SIP protocol itself. RFC 3261 is an update to RFC 2543 [9]. 158 It is the president of the galaxy as far as the suite of SIP 159 specifications is concerned. 161 RFC 3263, Locating SIP Servers (S): RFC 3263 [10] provides DNS 162 procedures for taking a SIP URI, and determining a SIP server that 163 is associated with that SIP URI. RFC 3263 is essential for any 164 implementation using SIP with DNS. RFC 3263 makes use of both DNS 165 SRV records [11] and NAPTR records [12]. 167 RFC 3264, An Offer/Answer Model with the Session Description Protocol 168 (S): RFC 3264 [4] defines how the Session Description Protocol (SDP) 169 [77] is used with SIP to negotiate the parameters of a media 170 session. It is in widespread usage and an integral part of the 171 behavior of RFC 3261. 173 RFC 3265, SIP-Specific Event Notification (S): RFC 3265 [13] defines 174 the SUBSCRIBE and NOTIFY methods. These two methods provide a 175 general event notification framework for SIP. To actually use the 176 framework, extensions need to be defined for specific event 177 packages. An event package defines a schema for the event data, 178 and describes other aspects of event processing specific to that 179 schema. An RFC 3265 implementation is required when any event 180 package is used. 182 RFC 3325, Private Extensions to SIP for Asserted Identity within 183 Trusted Networks (I): Though its P-header status implies that it has 184 limited applicability, RFC 3325 [15], which defines the 185 P-Asserted-ID header field has been widely deployed. It is used 186 as the basic mechanism for providing secure caller ID services. 188 RFC 3327, SIP Extension Header Field for Registering Non-Adjacent 189 Contacts (S): RFC 3327 [16] defines the Path header field. This 190 field is inserted by proxies between a client and their registrar. 191 It allows inbound requests towards that client to traverse these 192 proxies prior to being delivered to the user agent. It is 193 essential in any SIP deployment that has edge proxies, which are 194 proxies between the client and the home proxy or SIP registrar. 195 It is also instrumental in the SIP NAT traversal specifications. 197 RFC 3581, An Extension to SIP for Symmetric Response Routing (S): RFC 198 3581 [17] defines the rport parameter of the Via header. It is an 199 essential piece of getting SIP through NAT. NAT traversal for SIP 200 is considered a core part of the specifications. 202 RFC 3840, Indicating User Agent Capabilities in SIP (S): RFC 3840 203 [33] defines a mechanism for carrying capability information about 204 a user agent in REGISTER requests and in dialog-forming requests 205 like INVITE. It has found use with conferencing (the isfocus 206 parameter declares that a user agent is a conference server) and 207 with applications like push-to-talk. 209 RFC 4320, Actions Addressing Issues Identified with the Non-INVITE 210 Transaction in SIP (S): RFC 4320 [18] formally updates RFC 3261, and 211 modifies some of the behaviors associated with non-INVITE 212 transactions. These address some problems found in timeout and 213 failure cases. 215 RFC XXXX, Enhancements for Authenticated Identity Management in SIP 216 (S): RFC XXXX [19] defines a mechanism for providing a 217 cryptographically verifiable identity of the calling party in a 218 SIP request. Also known as "SIP Identity", this mechanism 219 provides an alternative to RFC 3325. It has seen little 220 deployment so far, but its importance as a key construct for 221 almost also anti-spam techniques makes it a core part of the SIP 222 specifications. 224 RFC XXXX, Obtaining and Using Globally Routable User Agent 225 Identifiers (GRUU) in SIP (S): RFC XXXX [20] defines a mechanism for 226 directing requests towards a specific UA instance. GRUU is 227 essential for features like transfer and provides another piece of 228 the SIP NAT traversal story. 230 RFC XXXX, Managing Client Initiated Connections through SIP (S): RFC 231 XXXX [21], also known as SIP outbound, defines important changes 232 to the SIP registration mechanism which enable delivery of SIP 233 messages towards a UA when it is behind a NAT. This specification 234 is the cornerstone of the SIP NAT traversal strategy. 236 RFC XXXX, Session Description Protocol (S): RFC XXXX [77] defines a 237 format for representing multimedia sessions. SDP objects are 238 carried in the body of SIP messages, and based on the offer/answer 239 model, are used to negotiate the media characteristics of a 240 session between users. 242 RFC 3388, Grouping of Media Lines in the Session Description Protocol 243 (S): RFC 3388 [78] defines a framework for grouping together media 244 streams in an SDP message. Such a grouping allows relationships 245 between these streams, such as which stream is the audio for a 246 particular video feed, to be expressed. 248 RFC XXXX, Interactive Connectivity Establishment (ICE) (S): RFC XXXX 249 [5] defines a technique for NAT traversal of media sessions for 250 protocols that make use of the offer/answer model. This 251 specification is the IETF recommended mechanism for NAT traversal 252 for SIP media streams, and is meant to be used even by endpoints 253 which are themselves never behind a NAT. 255 RFC 3605, Real Time Control Protocol (RTCP) Attribute in the Session 256 Description Protocol (SDP) (S): RFC 3605 [79] defines a way to 257 explicitly signal, within an SDP message, the IP address and port 258 for RTCP, rather than using the port+1 rule in the Real Time 259 Transport Protocol (RTP) [3]. It is needed for devices behind NAT 260 and used by ICE. 262 RFC XXXX, Connected Identity in the Session Initiation Protocol (SIP) 263 (S): RFC XXXX [80] defines an extension to SIP that allows a UAC to 264 determine the identity of the UAS. Due to forwarding and 265 retargeting services, this may not be the same as the user that 266 the UAC was originally trying to reach. The mechanism works in 267 tandem with the SIP identity specification [19] to provide 268 signatures over the connected party identity. 270 RFC XXXX, Addressing an Amplification Vulnerability in Forking 271 Proxies (S): RFC XXXX [81] makes a small normative change to RFC 272 3261, requiring loop detection in any proxy that forks a request. 273 It addresses a vulnerability uncovered in RFC 3261. 275 4. Public Switched Telephone Network (PSTN) Interworking 277 Numerous extensions and usages of SIP related to interoperability and 278 communications with or through the PSTN. 280 RFC 2848, The PINT Service Protocol (S): RFC 2848 [22] is one of the 281 earliest extensions to SIP. It defines procedures for using SIP 282 to invoke services that actually execute on the PSTN. Its main 283 application is for third party call control, allowing an IP host 284 to set up a call between two PSTN endpoints. PINT has a 285 relatively narrow focus and has not seen widespread deployment. 287 RFC 3910, The SPIRITS Protocol (S): Continuing the trend of naming 288 PSTN related extensions with alcohol references, SPIRITS [23] 289 defines the inverse of PINT. It allows a switch in the PSTN to 290 ask an IP element about how to proceed with call waiting. It was 291 developed primarily to support Internet Call Waiting (ICW). 292 Perhaps the next specification will be called the PGGB. 294 RFC 3372, SIP for Telephones (SIP-T): Context and Architectures 295 (I): SIP-T [24] defines a mechanism for using SIP between pairs of 296 PSTN gateways. Its essential idea is to tunnel ISUP signaling 297 between the gateways in the body of SIP messages. SIP-T motivated 298 the development of INFO [30]. SIP-T has seen widespread 299 implementation. 301 RFC 3398, ISUP to SIP Mapping (S): RFC 3398 [25] defines how to do 302 protocol mapping from the SS7 ISDN User Part (ISUP) signaling to 303 SIP. It is widely used in SS7 to SIP gateways and is part of the 304 SIP-T framework. 306 RFC 3578, Mapping of ISUP Overlap Signaling to SIP (S): RFC 3578 [26] 307 defines a mechanism to map overlap dialing into SIP. This 308 specification is widely regarded as the ugliest SIP specification, 309 as the introduction to the specification itself advises that it 310 has many problems. Overlap signaling (the practice of sending 311 digits into the network as dialed instead of waiting for complete 312 collection of the called party number) is largely incompatible 313 with SIP at some fairly fundamental levels. That said, RFC 3578 314 is mostly harmless and has seen some usage. 316 RFC 3960, Early Media and Ringtone Generation in SIP (I): RFC 3960 317 [27] defines some guidelines for handling early media - the 318 practice of sending media from the called party towards the caller 319 - prior to acceptance of the call. Early media is generated only 320 from the PSTN. 322 RFC 3959, Early Session Disposition Type for the Session Initiation 323 Protocol (SIP) (S): RFC 3959 [82] defines a new session disposition 324 type for use with early media. It indicates that the SDP in the 325 body is for a special early media session. 327 RFC 3204, MIME Media Types for ISUP and QSIG Objects (S): RFC 3204 328 [83] defines MIME objects for representing SS7 signaling messages. 329 These are carried in the body of SIP messages when SIP-T is used. 331 5. General Purpose Infrastructure Extensions 333 These extensions are general purpose enhancements to SIP, SDP and 334 MIME that can serve a wide variety of uses. However, they are not as 335 widely used or as essential as the core specifications. 337 RFC 3262, Reliability of Provisional Responses in SIP (S): SIP 338 defines two types of responses to a request - final and 339 provisional. Provisional responses are numbered from 100 to 199. 340 In SIP, these responses are not sent reliably. This choice was 341 made in RFC 2543 since the messages were meant to just be truly 342 informational, and rendered to the user. However, subsequent work 343 on PSTN interworking demonstrated a need to map provisional 344 responses to PSTN messages that needed to be sent reliably. RFC 345 3262 [28] was developed to allow reliability of provisional 346 responses. The specification defines the PRACK method, used for 347 indicating that a provisional response was received. Though it 348 provides a generic capability for SIP, RFC 3262 implementations 349 have been most common in PSTN interworking devices. However, 350 PRACK brings a great deal of complication for relatively small 351 benefit. As such, it has seen only mild levels of deployment. 353 RFC 3323, A Privacy Mechanism for the Session Initiation Protocol 354 (SIP) (S): RFC 3323 [14] defines the Privacy header field, used by 355 clients to request anonymity for their requests. Though it 356 defines numerous privacy services, the only one broadly used is 357 the one that supports privacy of the P-Asserted-ID header field 358 [15]. 360 RFC 3311, The SIP UPDATE Method (S): RFC 3311 [29] defines the UPDATE 361 method for SIP. This method is meant as a means for updating 362 session information prior to the completion of the initial INVITE 363 transaction. It was developed primarily to support RFC 3312 [59]. 365 RFC 2976, The INFO Method (S): RFC 2976 [30] was defined as an 366 extension to RFC 2543. It defines a method, INFO, used to 367 transport mid-dialog information that has no impact on SIP itself. 368 Its driving application was the transport of PSTN related 369 information when using SIP between a pair of gateways. Though 370 originally conceived for broader use, it only found standardized 371 usage with SIP-T [24]. It has been used to support numerous 372 proprietary and non-interoperable extensions due to its poorly 373 defined scope. 375 RFC 3326, The Reason header field for SIP (S): RFC 3326 [31] defines 376 the Reason header field. It is used in requests, such as BYE, to 377 indicate the reason that the request is being sent. 379 RFC 3420, Internet Media Type message/sipfrag (S): RFC 3420 [84] 380 defines a MIME object that contains a SIP message fragment. Only 381 certain header fields and parts of the SIP message are present. 382 For example, it is used to report back on the responses received 383 to a request sent as a consequence of a REFER. 385 RFC 3608, SIP Extension Header Field for Service Route Discovery 386 During Registration (S): RFC 3608 [32] allows a client to determine, 387 from a REGISTER response, a path of proxies to use in requests it 388 sends outside of a dialog. In many respects, it is the inverse of 389 the Path header field, but has seen less usage since default 390 outbound proxies have been sufficient in many deployments. 392 RFC 3841, Caller Preferences for SIP (S): RFC 3841 [34] defines a set 393 of headers that a client can include in a request to control the 394 way in which the request is routed downstream. It allows a client 395 to direct a request towards a UA with specific capabilities. 397 RFC 4028, Session Timers in SIP (S): RFC 4028 [35] defines a 398 keepalive mechanism for SIP signaling. It is primarily meant to 399 provide a way to cleanup old state in proxies that are holding 400 call state for calls from failed endpoints which were never 401 terminated normally. Despite its name, the session timer is not a 402 mechanism for detecting a network failure mid-call. Session 403 timers introduces a fair bit of complexity for relatively little 404 gain, and has thus seen little deployment. 406 RFC 4168, SCTP as a Transport for SIP (S): RFC 4168 [36] defines how 407 to carry SIP messages over the Stream Control Transmission 408 Protocol (SCTP). SCTP has seen very limited usage for SIP 409 transport. 411 RFC 4244, An Extension to SIP for Request History Information 412 (S): RFC 4244 [37] defines the History-Info header field, which 413 indicates information on how a call came to be routed to a 414 particular destination. Its primary application was in support of 415 voicemail services. 417 RFC 4145, TCP-Based Media Transport in the Session Description 418 Protocol (SDP) (S): RFC 4145 [85] defines an extension to SDP for 419 setting up TCP-based sessions between user agents. It defines who 420 sets up the connection and how its lifecycle is managed. It has 421 seen relatively little usage due to the small number of media 422 types to date which use TCP. 424 RFC 4091, The Alternative Network Address Types (ANAT) Semantics for 425 the Session Description Protocol (SDP) Grouping Framework (S): RFC 426 4091 [86] defines a mechanism for including both IPv4 and IPv6 427 addresses for a media session as alternates. 429 RFC XXXX, TCP Candidates with Interactive Connectivity Establishment 430 (ICE) (S): RFC XXXX [87] specifies the usage of ICE for TCP streams. 431 This allows for selection of RTP-based voice ontop of TCP only 432 when NAT or firewalls would prevent UDP-based voice from working. 434 6. Minor Extensions 436 These SIP extensions don't fit easily into a single specific use 437 case. They have somewhat general applicability, but they solve a 438 relatively small problem or provide an optimization. 440 RFC 4488, Suppression of the SIP REFER Implicit Subscription (S): RFC 441 4488 [38] defines an enhancement to REFER. REFER normally creates 442 an implicit subscription to the target of the REFER. This 443 subscription is used to pass back updates on the progress of the 444 referral. This extension allows that implicit subscription to be 445 bypassed as an optimization. 447 RFC 4538, Request Authorization through Dialog Identification in SIP 448 (S): RFC 4538 [39] provides a mechanism that allows a UAS to 449 authorize a request because the requestor proves it knows a dialog 450 that is in progress with the UAS. The specification is useful in 451 conjunction with the SIP application interaction framework [76]. 453 RFC 4508, Conveying Feature Tags with the REFER Method in SIP 454 (S): RFC 4508 [40] defines a mechanism for carrying RFC 3840 455 feature tags in REFER. It is useful for informing the target of 456 the REFER about the characteristics of the REFER target. 458 RFC XXXX, Requesting Answer Modes for SIP (S): RFC XXXX [41] defines 459 an extension for indicating to the called party whether or not the 460 phone should ring and/or be answered immediately. This is useful 461 for push-to-talk and for diagnostic applications. 463 RFC XXXX, Rejecting Anonymous Requests in SIP (S): RFC XXXX [43] 464 defines a mechanism for a called party to indicate to the calling 465 party that a call was rejected since the caller was anonymous. 466 This is needed for implementation of the Anonymous Call Rejection 467 (ACR) feature in SIP. 469 RFC XXXX, Referring to Multiple Resources in SIP (S): RFC XXXX [44] 470 allows a UA sending a REFER to ask the recipient of the REFER to 471 generate multiple SIP requests, not just one. This is useful for 472 conferencing, where a client would like to ask a conference server 473 to eject multiple users. 475 RFC 4483, A Mechanism for Content Indirection in Session Initiation 476 Protocol (SIP) Messages (S): RFC 4483 [88] defines a mechanism for 477 content indirection. Instead of carrying an object within a SIP 478 body, a URL reference is carried instead, and the recipient 479 dereferences the URL to obtain the object. The specification has 480 potential applicability for sending large instant messages, but 481 has yet to find much actual use. 483 RFC 3890, A Transport Independent Bandwidth Modifier for the Session 484 Description Protocol (SDP) (S): RFC 3890 [89] specifies an SDP 485 extension that allows for the description of the bandwidth for a 486 media session that is independent of the underlying transport 487 mechanism. It has seen relatively little usage. 489 RFC XXXX, Session Description Protocol (SDP) Format for Binary Floor 490 Control Protocol (BFCP) Streams (S): RFC XXXX [90] defines a 491 mechanism in SDP to signal floor control streams that use BFCP. 492 It is used for Push-To-Talk and conference floor control. 494 RFC XXXX, Connectivity Preconditions for Session Description Protocol 495 Media Streams (S): RFC XXXX [92] defines a usage of the precondition 496 framework [59]. The connectivity precondition makes sure that the 497 session doesn't get established until actual packet connectivity 498 is checked. 500 RFC XXXX, The SDP (Session Description Protocol) Content Attribute 501 (S): RFC XXXX [93] defines an SDP attribute for describing the 502 purpose of a media stream. Examples include a slide view, the 503 speaker, a sign language feed, and so on. 505 7. Conferencing 507 Numerous SIP and SDP extensions are aimed at conferencing as their 508 primary application. 510 RFC XXXX, The SDP (Session Description Protocol) Label Attribute 511 (S): RFC XXXX [94] defines an SDP attribute for providing an opaque 512 label for media streams. These labels can be referred to by 513 external documents, and in particular, by conference policy 514 documents. This allows a UA to tie together documents it may 515 obtain through conferencing mechanisms to media streams to which 516 they refer. 518 RFC 3911, The SIP Join Header Field (S): RFC 3911 [49] defines the 519 Join header field. When sent in an INVITE, it causes the 520 recipient to join the resulting dialog into a conference with 521 another dialog in progress. 523 RFC XXXX, A SIP Event Package for Conference State (S): RFC XXXX [56] 524 defines a mechanism for learning about changes in conference 525 state, including group membership. 527 RFC XXXX, Conference Establishment Using Request-Contained Lists in 528 SIP (S): RFC XXXX [69] is similar to [67]. However, instead of 529 subscribing to the resource, an INVITE request is sent to the 530 resource, and it will act as a conference focus and generate an 531 invitation to each recipient in the list. 533 8. Call Control Primitives 535 Numerous SIP extensions provide a toolkit of dialog and call 536 management techniques. These techniques have been combined together 537 to build many SIP-based services. 539 RFC 3515, The REFER Method (S): REFER [45] defines a mechanism for 540 asking a user agent to send a SIP request. Its a form of SIP 541 remote control, and is the primary tool used for call transfer in 542 SIP. 544 RFC 3725, Best Current Practices for Third Party Call Control (3pcc) 545 (B): RFC 3725 [46] defines a number of different call flows that 546 allow one SIP entity, called the controller, to create SIP 547 sessions amongst other SIP user agents. 549 RFC 3911, The SIP Join Header Field (S): RFC 3911 [49] defines the 550 Join header field. When sent in an INVITE, it causes the 551 recipient to join the resulting dialog into a conference with 552 another dialog in progress. 554 RFC 3891, The SIP Replaces Header (S): RFC 3891 [47] defines a 555 mechanism that allows a new dialog to replace an existing dialog. 556 It is useful for certain advanced transfer services. 558 RFC 3892, The SIP Referred-By Mechanism (S): RFC 3892 [48] defines 559 the Referred-By header field. It is used in requests triggered by 560 REFER, and provides the identity of the referring party to the 561 referred-to party. 563 RFC 4117, Transcoding Services Invocation in SIP Using Third Party 564 Call Control (I): RFC 4117 [50] defines how to use 3pcc for the 565 purposes of invoking transcoding services for a call. 567 9. Event Framework and Packages 569 RFC 3265 defines a basic framework for event notification in SIP. It 570 introduces the notion of an event package, which is a collection of 571 related state and event information. Much of the state and events in 572 SIP systems have event packages, allowing other entities to learn 573 about changes in that state. 575 RFC 3903, SIP Extension for Event State Publication (S): RFC 3903 576 [51] defines the PUBLISH method. It is not an event package, but 577 is used by all event packages as a mechanism for pushing an event 578 into the system. 580 RFC XXXX, A Session Initiation Protocol (SIP) Event Notification 581 Extension for Resource Lists (S): RFC XXXX [66] defines an extension 582 to RFC 3265 that allows a client to subscribe to a list of 583 resources using a single subscription. The server, called a 584 Resource List Server (RLS) will "expand" the subscription and 585 subscribe to each individual member of the list. It has found 586 applicability primarily in the area of presence, but can be used 587 with any event package. 589 RFC 3680, A SIP Event Package for Registrations (S): RFC 3680 [52] 590 defines an event package for finding out about changes in 591 registration state. 593 RFC 3842, A Message Summary and Message Waiting Indication Event 594 Package for SIP (S): RFC 3842 [64] defines a way for a user agent to 595 find out about voicemails and other messages that are waiting for 596 it. Its primary purpose is to enable the voicemail waiting lamp 597 on most business telephones. 599 RFC 3856, A Presence Event Package for SIP (S): RFC 3856 [53] defines 600 an event package for indicating user presence through SIP. 602 RFC 3857, A Watcher Information Event Template Package for SIP 603 (S): RFC 3857 [54], also known as winfo, provides a mechanism for 604 a user agent to find out what subscriptions are in place for a 605 particular event package. Its primary usage is with presence, but 606 it can be used with any event package. 608 RFC 4235, An INVITE Initiated Dialog Event Package for SIP (S): RFC 609 4235 [55] defines an event package for learning the state of the 610 dialogs in progress at a user agent. 612 RFC XXXX, A SIP Event Package for Conference State (S): RFC XXXX [56] 613 defines a mechanism for learning about changes in conference 614 state, including group membership. 616 RFC XXXX, A SIP Event Package for Keypress Stimulus (KPML) (S): RFC 617 XXXX [57] defines a way for an application in the network to 618 subscribe to the set of keypresses made on the keypad of a 619 traditional telephone. 621 RFC XXXX, SIP Event Package for Voice Quality Reporting (S): RFC XXXX 622 [58] defines a SIP event package that enables the collection and 623 reporting of metrics that measure the quality for Voice over 624 Internet Protocol (VoIP) sessions. 626 RFC XXXX, A Session Initiation Protocol (SIP) Event Package for 627 Session-Specific Session Policies (S): RFC XXXX [95] defines a SIP 628 event package that allows a proxy to notify a user agent about its 629 desire for the UA to use certain codecs or generally obey certain 630 media session policies. 632 10. Quality of Service 634 Several specifications concern themselves with the interactions of 635 SIP with network Quality of Service (QoS) mechanisms. 637 RFC 3312, Integration of Resource Management and SIP (S): RFC 3312 638 [59], updated by RFC 4032 [60] defines a way to make sure that the 639 phone of the called party doesn't ring until a QoS reservation has 640 been installed in the network. It does so by defining a general 641 preconditions framework, which defines conditions that must be 642 true in order for a SIP session to proceed 644 RFC 3313, Private SIP Extensions for Media Authorization (I): RFC 645 3313 [61] defines a P-header that provides a mechanism for passing 646 an authorization token between SIP and a network QoS reservation 647 protocol like RSVP. Its purpose is to make sure network QoS is 648 only granted if a client has made a SIP call through the same 649 providers network. This specification is sometimes referred to as 650 the SIP walled garden specification by the truly paranoid androids 651 in the SIP community. This is because it requires coupling of 652 signaling and the underlying IP network. 654 RFC 3524, Mapping of Media Streams to Resource Reservation Flows 655 (S): RFC 3524 [96] defines a usage of the SDP grouping framework for 656 indicating that a set of media streams should be handled by a 657 single resource reservation. 659 11. Operations and Management 661 Several specifications have been defined to support operations and 662 management of SIP systems. These include mechanisms for 663 configuration and network diagnostics. 665 RFC XXXX, Diagnostic Responses for SIP Hop Limit Errors (S): RFC XXXX 666 [97] defines a mechanism for including diagnostic information in a 667 483 response. This response is sent when the hop-count of a SIP 668 request was exceeded. 670 RFC XXXX, A Framework for SIP User Agent Profile Delivery (S): RFC 671 XXXX [62] defines a mechanism that allows a SIP user agent to 672 bootstrap its configuration from the network, and receive updates 673 to its configuration should it change. This is considered an 674 essential piece of deploying a usable SIP network. 676 RFC XXXX, SIP Event Package for Voice Quality Reporting (S): RFC XXXX 677 [58] defines a SIP event package that enables the collection and 678 reporting of metrics that measure the quality for Voice over 679 Internet Protocol (VoIP) sessions. 681 12. SIP Compression 683 Sigcomp [6] was defined to allow compression of SIP messages over low 684 bandwidth links. Sigcomp is not formally part of SIP. However, 685 usage of Sigcomp with SIP has required extensions to SIP. 687 RFC 3486, Compressing SIP (S): RFC 3486 [63] defines a SIP URI 688 parameter that can be used to indicate that a SIP server supports 689 Sigcomp. 691 13. SIP Service URIs 693 Several extensions define well-known services that can be invoked by 694 constructing requests with the specific structures for the Request 695 URI, resulting in specific behaviors at the UAS. 697 RFC 3087, Control of Service Context using Request URI (I): RFC 3087 698 [65] introduced the context of using Request URIs, encoded 699 appropriately, to invoke services. 701 RFC XXXX, A SIP Event Notification Extension for Resource Lists 702 (S): RFC XXXX [66] defines a resource called a Resource List 703 Server. A client can send a subscribe to this server. The server 704 will generate a series of subscriptions, and compile the resulting 705 information and send it back to the subscriber. The set of 706 resources that the RLS will subscribe to is a property of the 707 request URI in the SUBSCRIBE request. 709 RFC XXXX, Subscriptions To Request-Contained Resource Lists in SIP 710 (S): RFC XXXX [67] allows a client to subscribe to a resource called 711 a Resource List Server. This server will generate a series of 712 subscriptions, and compile the resulting information and send it 713 back to the subscriber. For this specification, the list of 714 things to subscribe to is in the body of the SUBSCRIBE request. 716 RFC XXXX, Multiple-Recipient MESSAGE Requests in SIP (S): RFC XXXX 717 [68] is similar to [67]. However, instead of subscribing to the 718 resource, a MESSAGE request is sent to the resource, and it will 719 send a copy to each recipient. 721 RFC XXXX, Conference Establishment Using Request-Contained Lists in 722 SIP (S): RFC XXXX [69] is similar to [67]. However, instead of 723 subscribing to the resource, an INVITE request is sent to the 724 resource, and it will act as a conference focus and generate an 725 invitation to each recipient in the list. 727 RFC 4240, Basic Network Media Services with SIP (I): RFC 4240 [98] 728 defines a way for SIP application servers to invoke announcement 729 and conferencing services from a media server. This is 730 accomplished through a set of defined URI parameters which tell 731 the media server what to do, such as what file to play and what 732 language to render it in. 734 14. Security Mechanisms 736 Several extensions provide additional security features to SIP. 738 RFC 3853, S/MIME AES Requirement for SIP (S): RFC 3853 [70] is a 739 brief specification that updates the cryptography mechanisms used 740 in SIP S/MIME. However, SIP S/MIME has seen very little 741 deployment. 743 RFC XXXX, Certificate Management Service for The Session Initiation 744 Protocol (SIP) (S): RFC XXXX [99] defines a certificate service for 745 SIP whose purpose is to facilitate the deployment of S/MIME. The 746 certificate service allows clients to store and retrieve their own 747 certificates, in addition to obtaining the certificates for other 748 users. 750 RFC 3893, Session Initiation Protocol (SIP) Authenticated Identity 751 Body (AIB) Format (S): RFC 3893 [7] defines a SIP message fragment 752 which can be signed in order to provide an authenticated identity 753 over a request. It was an early predecessor to [19], and 754 consequently AIB has seen no deployment. 756 RFC 3329, Security Mechanism Agreement for SIP (S): RFC 3329 [71] 757 defines a mechanism to prevent bid-down attacks in conjunction 758 with SIP authentication. The mechanism has seen very limited 759 deployment. It was defined as part of the 3gpp IMS specification 760 suite, and is needed only when there are a multiplicity of 761 security mechanisms deployed at a particular server. In practice, 762 this has not been the case. 764 RFC XXXX, End-to-Middle Security in SIP (S): RFC XXXX [72] defines 765 mechanisms for encrypting content from user agents to specific 766 network intermediaries. 768 RFC XXXX, Connection-Oriented Media Transport over the Transport 769 Layer Security (TLS) Protocol in the Session Description Protocol 770 (SDP) (S): RFC XXXX [100] specifies a mechanism for signaling TLS- 771 based media streams between endpoints. It expands the TCP-based 772 media signaling parameters defined in [85] to include fingerprint 773 information for TLS streams, so that TLS can operate between end 774 hosts using self-signed certificates. 776 RFC XXXX, Security Preconditions for Session Description Protocol 777 Media Streams (S): RFC XXXX [91] defines a precondition for use with 778 the preconditions framework [59]. The security precondition 779 prevents a session from being established until a security media 780 stream is set up. 782 15. Instant Messaging and Presence 784 SIP provides extensions for instant messaging and presence. 786 RFC 3428, SIP Extension for Instant Messaging (S): RFC 3428 [73] 787 defines the MESSAGE method, used for sending a page mode instant 788 message. 790 RFC 3856, A Presence Event Package for SIP (S): RFC 3856 [53] defines 791 an event package for indicating user presence through SIP. 793 RFC 3857, A Watcher Information Event Template Package for SIP 794 (S): RFC 3857 [54], also known as winfo, provides a mechanism for 795 a user agent to find out what subscriptions are in place for a 796 particular event package. Its primary usage is with presence, but 797 it can be used with any event package. 799 16. Emergency Services 801 Emergency services here covers both emergency calling (for example, 802 911 in the United States), and pre-emption services, which allow 803 authorized individuals to gain access to network resources in time of 804 emergency. 806 RFC 4411, Extending the SIP Reason Header for Preemption Events 807 (S): RFC 4411 [74] defines an extension to the Reason header, 808 allowing a UA to know that its dialog was torn down because a 809 higher priority session came through. 811 RFC 4412, Communications Resource Priority for SIP (S): RFC 4412 [75] 812 defines a new header field, Resource-Priority, that allows a 813 session to get priority treatment from the network. 815 17. Security Considerations 817 This specification is an overview of existing specifications, and 818 does not introduce any security considerations on its own. 820 18. IANA Considerations 822 None. 824 19. Informative References 826 [1] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., 827 Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: 828 Session Initiation Protocol", RFC 3261, June 2002. 830 [2] Bradner, S., "The Internet Standards Process -- Revision 3", 831 BCP 9, RFC 2026, October 1996. 833 [3] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, 834 "RTP: A Transport Protocol for Real-Time Applications", 835 RFC 3550, July 2003. 837 [4] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with 838 Session Description Protocol (SDP)", RFC 3264, June 2002. 840 [5] Rosenberg, J., "Interactive Connectivity Establishment (ICE): 841 A Methodology for Network Address Translator (NAT) Traversal 842 for Offer/Answer Protocols", draft-ietf-mmusic-ice-08 (work in 843 progress), March 2006. 845 [6] Price, R., Bormann, C., Christoffersson, J., Hannu, H., Liu, 846 Z., and J. Rosenberg, "Signaling Compression (SigComp)", 847 RFC 3320, January 2003. 849 [7] Peterson, J., "Session Initiation Protocol (SIP) Authenticated 850 Identity Body (AIB) Format", RFC 3893, September 2004. 852 [8] Mankin, A., Bradner, S., Mahy, R., Willis, D., Ott, J., and B. 853 Rosen, "Change Process for the Session Initiation Protocol 854 (SIP)", BCP 67, RFC 3427, December 2002. 856 [9] Handley, M., Schulzrinne, H., Schooler, E., and J. Rosenberg, 857 "SIP: Session Initiation Protocol", RFC 2543, March 1999. 859 [10] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol 860 (SIP): Locating SIP Servers", RFC 3263, June 2002. 862 [11] Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS RR for 863 specifying the location of services (DNS SRV)", RFC 2782, 864 February 2000. 866 [12] Mealling, M. and R. Daniel, "The Naming Authority Pointer 867 (NAPTR) DNS Resource Record", RFC 2915, September 2000. 869 [13] Roach, A., "Session Initiation Protocol (SIP)-Specific Event 870 Notification", RFC 3265, June 2002. 872 [14] Peterson, J., "A Privacy Mechanism for the Session Initiation 873 Protocol (SIP)", RFC 3323, November 2002. 875 [15] Jennings, C., Peterson, J., and M. Watson, "Private Extensions 876 to the Session Initiation Protocol (SIP) for Asserted Identity 877 within Trusted Networks", RFC 3325, November 2002. 879 [16] Willis, D. and B. Hoeneisen, "Session Initiation Protocol 880 (SIP) Extension Header Field for Registering Non-Adjacent 881 Contacts", RFC 3327, December 2002. 883 [17] Rosenberg, J. and H. Schulzrinne, "An Extension to the Session 884 Initiation Protocol (SIP) for Symmetric Response Routing", 885 RFC 3581, August 2003. 887 [18] Sparks, R., "Actions Addressing Identified Issues with the 888 Session Initiation Protocol's (SIP) Non-INVITE Transaction", 889 RFC 4320, January 2006. 891 [19] Peterson, J. and C. Jennings, "Enhancements for Authenticated 892 Identity Management in the Session Initiation Protocol 893 (SIP)", draft-ietf-sip-identity-06 (work in progress), 894 October 2005. 896 [20] Rosenberg, J., "Obtaining and Using Globally Routable User 897 Agent (UA) URIs (GRUU) in the Session Initiation Protocol 898 (SIP)", draft-ietf-sip-gruu-07 (work in progress), May 2006. 900 [21] Jennings, C. and R. Mahy, "Managing Client Initiated 901 Connections in the Session Initiation Protocol (SIP)", 902 draft-ietf-sip-outbound-03 (work in progress), March 2006. 904 [22] Petrack, S. and L. Conroy, "The PINT Service Protocol: 906 Extensions to SIP and SDP for IP Access to Telephone Call 907 Services", RFC 2848, June 2000. 909 [23] Gurbani, V., Brusilovsky, A., Faynberg, I., Gato, J., Lu, H., 910 and M. Unmehopa, "The SPIRITS (Services in PSTN requesting 911 Internet Services) Protocol", RFC 3910, October 2004. 913 [24] Vemuri, A. and J. Peterson, "Session Initiation Protocol for 914 Telephones (SIP-T): Context and Architectures", BCP 63, 915 RFC 3372, September 2002. 917 [25] Camarillo, G., Roach, A., Peterson, J., and L. Ong, 918 "Integrated Services Digital Network (ISDN) User Part (ISUP) 919 to Session Initiation Protocol (SIP) Mapping", RFC 3398, 920 December 2002. 922 [26] Camarillo, G., Roach, A., Peterson, J., and L. Ong, "Mapping 923 of Integrated Services Digital Network (ISDN) User Part (ISUP) 924 Overlap Signalling to the Session Initiation Protocol (SIP)", 925 RFC 3578, August 2003. 927 [27] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing 928 Tone Generation in the Session Initiation Protocol (SIP)", 929 RFC 3960, December 2004. 931 [28] Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional 932 Responses in Session Initiation Protocol (SIP)", RFC 3262, 933 June 2002. 935 [29] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE 936 Method", RFC 3311, October 2002. 938 [30] Donovan, S., "The SIP INFO Method", RFC 2976, October 2000. 940 [31] Schulzrinne, H., Oran, D., and G. Camarillo, "The Reason 941 Header Field for the Session Initiation Protocol (SIP)", 942 RFC 3326, December 2002. 944 [32] Willis, D. and B. Hoeneisen, "Session Initiation Protocol 945 (SIP) Extension Header Field for Service Route Discovery 946 During Registration", RFC 3608, October 2003. 948 [33] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating 949 User Agent Capabilities in the Session Initiation Protocol 950 (SIP)", RFC 3840, August 2004. 952 [34] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller 953 Preferences for the Session Initiation Protocol (SIP)", 954 RFC 3841, August 2004. 956 [35] Donovan, S. and J. Rosenberg, "Session Timers in the Session 957 Initiation Protocol (SIP)", RFC 4028, April 2005. 959 [36] Rosenberg, J., Schulzrinne, H., and G. Camarillo, "The Stream 960 Control Transmission Protocol (SCTP) as a Transport for the 961 Session Initiation Protocol (SIP)", RFC 4168, October 2005. 963 [37] Barnes, M., "An Extension to the Session Initiation Protocol 964 (SIP) for Request History Information", RFC 4244, 965 November 2005. 967 [38] Levin, O., "Suppression of Session Initiation Protocol (SIP) 968 REFER Method Implicit Subscription", RFC 4488, May 2006. 970 [39] Rosenberg, J., "Request Authorization through Dialog 971 Identification in the Session Initiation Protocol (SIP)", 972 RFC 4538, June 2006. 974 [40] Levin, O. and A. Johnston, "Conveying Feature Tags with the 975 Session Initiation Protocol (SIP) REFER Method", RFC 4508, 976 May 2006. 978 [41] Willis, D. and A. Allen, "Requesting Answering Modes for the 979 Session Initiation Protocol (SIP)", 980 draft-ietf-sip-answermode-01 (work in progress), May 2006. 982 [42] Adams, D., "The Hitchhikers Guide to the Galaxy", 983 September 1979. 985 [43] Rosenberg, J., "Rejecting Anonymous Requests in the Session 986 Initiation Protocol (SIP)", draft-ietf-sip-acr-code-00 (work 987 in progress), January 2006. 989 [44] Camarillo, G., "Refering to Multiple Resources in the Session 990 Initiation Protocol (SIP)", 991 draft-ietf-sipping-multiple-refer-05 (work in progress), 992 May 2006. 994 [45] Sparks, R., "The Session Initiation Protocol (SIP) Refer 995 Method", RFC 3515, April 2003. 997 [46] Rosenberg, J., Peterson, J., Schulzrinne, H., and G. 998 Camarillo, "Best Current Practices for Third Party Call 999 Control (3pcc) in the Session Initiation Protocol (SIP)", 1000 BCP 85, RFC 3725, April 2004. 1002 [47] Mahy, R., Biggs, B., and R. Dean, "The Session Initiation 1003 Protocol (SIP) "Replaces" Header", RFC 3891, September 2004. 1005 [48] Sparks, R., "The Session Initiation Protocol (SIP) Referred-By 1006 Mechanism", RFC 3892, September 2004. 1008 [49] Mahy, R. and D. Petrie, "The Session Initiation Protocol (SIP) 1009 "Join" Header", RFC 3911, October 2004. 1011 [50] Camarillo, G., Burger, E., Schulzrinne, H., and A. van Wijk, 1012 "Transcoding Services Invocation in the Session Initiation 1013 Protocol (SIP) Using Third Party Call Control (3pcc)", 1014 RFC 4117, June 2005. 1016 [51] Niemi, A., "Session Initiation Protocol (SIP) Extension for 1017 Event State Publication", RFC 3903, October 2004. 1019 [52] Rosenberg, J., "A Session Initiation Protocol (SIP) Event 1020 Package for Registrations", RFC 3680, March 2004. 1022 [53] Rosenberg, J., "A Presence Event Package for the Session 1023 Initiation Protocol (SIP)", RFC 3856, August 2004. 1025 [54] Rosenberg, J., "A Watcher Information Event Template-Package 1026 for the Session Initiation Protocol (SIP)", RFC 3857, 1027 August 2004. 1029 [55] Santesson, S. and R. Housley, "Internet X.509 Public Key 1030 Infrastructure Authority Information Access Certificate 1031 Revocation List (CRL) Extension", RFC 4325, December 2005. 1033 [56] Rosenberg, J., "A Session Initiation Protocol (SIP) Event 1034 Package for Conference State", 1035 draft-ietf-sipping-conference-package-12 (work in progress), 1036 July 2005. 1038 [57] Burger, E., "A Session Initiation Protocol (SIP) Event Package 1039 for Key Press Stimulus (KPML)", draft-ietf-sipping-kpml-07 1040 (work in progress), December 2004. 1042 [58] Pendleton, A., "Session Initiation Protocol Package for Voice 1043 Quality Reporting Event", draft-ietf-sipping-rtcp-summary-01 1044 (work in progress), February 2006. 1046 [59] Camarillo, G., Marshall, W., and J. Rosenberg, "Integration of 1047 Resource Management and Session Initiation Protocol (SIP)", 1048 RFC 3312, October 2002. 1050 [60] Camarillo, G. and P. Kyzivat, "Update to the Session 1051 Initiation Protocol (SIP) Preconditions Framework", RFC 4032, 1052 March 2005. 1054 [61] Marshall, W., "Private Session Initiation Protocol (SIP) 1055 Extensions for Media Authorization", RFC 3313, January 2003. 1057 [62] Petrie, D., "A Framework for Session Initiation Protocol User 1058 Agent Profile Delivery", 1059 draft-ietf-sipping-config-framework-08 (work in progress), 1060 March 2006. 1062 [63] Camarillo, G., "Compressing the Session Initiation Protocol 1063 (SIP)", RFC 3486, February 2003. 1065 [64] Foster, M., McGarry, T., and J. Yu, "Number Portability in the 1066 Global Switched Telephone Network (GSTN): An Overview", 1067 RFC 3482, February 2003. 1069 [65] Campbell, B. and R. Sparks, "Control of Service Context using 1070 SIP Request-URI", RFC 3087, April 2001. 1072 [66] Roach, A., Rosenberg, J., and B. Campbell, "A Session 1073 Initiation Protocol (SIP) Event Notification Extension for 1074 Resource Lists", draft-ietf-simple-event-list-07 (work in 1075 progress), January 2005. 1077 [67] Camarillo, G., "Subscriptions to Request-Contained Resource 1078 Lists in the Session Initiation Protocol (SIP)", 1079 draft-ietf-sipping-uri-list-subscribe-05 (work in progress), 1080 May 2006. 1082 [68] Garcia-Martin, M. and G. Camarillo, "Multiple-Recipient 1083 MESSAGE Requests in the Session Initiation Protocol (SIP)", 1084 draft-ietf-sipping-uri-list-message-07 (work in progress), 1085 February 2006. 1087 [69] Camarillo, G. and A. Johnston, "Conference Establishment Using 1088 Request-Contained Lists in the Session Initiation Protocol 1089 (SIP)", draft-ietf-sipping-uri-list-conferencing-05 (work in 1090 progress), February 2006. 1092 [70] Peterson, J., "S/MIME Advanced Encryption Standard (AES) 1093 Requirement for the Session Initiation Protocol (SIP)", 1094 RFC 3853, July 2004. 1096 [71] Arkko, J., Torvinen, V., Camarillo, G., Niemi, A., and T. 1097 Haukka, "Security Mechanism Agreement for the Session 1098 Initiation Protocol (SIP)", RFC 3329, January 2003. 1100 [72] Ono, K. and S. Tachimoto, "End-to-middle Security in the 1101 Session Initiation Protocol (SIP)", draft-ietf-sip-e2m-sec-01 1102 (work in progress), October 2005. 1104 [73] Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C., and 1105 D. Gurle, "Session Initiation Protocol (SIP) Extension for 1106 Instant Messaging", RFC 3428, December 2002. 1108 [74] Polk, J., "Extending the Session Initiation Protocol (SIP) 1109 Reason Header for Preemption Events", RFC 4411, February 2006. 1111 [75] Schulzrinne, H. and J. Polk, "Communications Resource Priority 1112 for the Session Initiation Protocol (SIP)", RFC 4412, 1113 February 2006. 1115 [76] Rosenberg, J., "A Framework for Application Interaction in the 1116 Session Initiation Protocol (SIP)", 1117 draft-ietf-sipping-app-interaction-framework-05 (work in 1118 progress), July 2005. 1120 [77] Handley, M., "SDP: Session Description Protocol", 1121 draft-ietf-mmusic-sdp-new-26 (work in progress), January 2006. 1123 [78] Camarillo, G., Eriksson, G., Holler, J., and H. Schulzrinne, 1124 "Grouping of Media Lines in the Session Description Protocol 1125 (SDP)", RFC 3388, December 2002. 1127 [79] Huitema, C., "Real Time Control Protocol (RTCP) attribute in 1128 Session Description Protocol (SDP)", RFC 3605, October 2003. 1130 [80] Elwell, J., "Connected Identity in the Session Initiation 1131 Protocol (SIP)", draft-ietf-sip-connected-identity-00 (work in 1132 progress), April 2006. 1134 [81] Sparks, R., "Addressing an Amplification Vulnerability in 1135 Forking Proxies", draft-ietf-sip-fork-loop-fix-01 (work in 1136 progress), April 2006. 1138 [82] Camarillo, G., "The Early Session Disposition Type for the 1139 Session Initiation Protocol (SIP)", RFC 3959, December 2004. 1141 [83] Zimmerer, E., Peterson, J., Vemuri, A., Ong, L., Audet, F., 1142 Watson, M., and M. Zonoun, "MIME media types for ISUP and QSIG 1143 Objects", RFC 3204, December 2001. 1145 [84] Sparks, R., "Internet Media Type message/sipfrag", RFC 3420, 1146 November 2002. 1148 [85] Yon, D. and G. Camarillo, "TCP-Based Media Transport in the 1149 Session Description Protocol (SDP)", RFC 4145, September 2005. 1151 [86] Camarillo, G. and J. Rosenberg, "The Alternative Network 1152 Address Types (ANAT) Semantics for the Session Description 1153 Protocol (SDP) Grouping Framework", RFC 4091, June 2005. 1155 [87] Rosenberg, J., "TCP Candidates with Interactive Connectivity 1156 Establishment (ICE)", draft-ietf-mmusic-ice-tcp-00 (work in 1157 progress), March 2006. 1159 [88] Burger, E., "A Mechanism for Content Indirection in Session 1160 Initiation Protocol (SIP) Messages", RFC 4483, May 2006. 1162 [89] Westerlund, M., "A Transport Independent Bandwidth Modifier 1163 for the Session Description Protocol (SDP)", RFC 3890, 1164 September 2004. 1166 [90] Camarillo, G., "Session Description Protocol (SDP) Format for 1167 Binary Floor Control Protocol (BFCP) Streams", 1168 draft-ietf-mmusic-sdp-bfcp-03 (work in progress), 1169 December 2005. 1171 [91] Andreasen, F. and D. Wing, "Security Preconditions for Session 1172 Description Protocol Media Streams", 1173 draft-ietf-mmusic-securityprecondition-01 (work in progress), 1174 October 2005. 1176 [92] Andreasen, F., "Connectivity Preconditions for Session 1177 Description Protocol Media Streams", 1178 draft-ietf-mmusic-connectivity-precon-02 (work in progress), 1179 June 2006. 1181 [93] Hautakorpi, J. and G. Camarillo, "The SDP (Session Description 1182 Protocol) Content Attribute", 1183 draft-ietf-mmusic-sdp-media-content-03 (work in progress), 1184 April 2006. 1186 [94] Levin, O. and G. Camarillo, "The SDP (Session Description 1187 Protocol) Label Attribute", 1188 draft-ietf-mmusic-sdp-media-label-01 (work in progress), 1189 January 2005. 1191 [95] Hilt, V. and G. Camarillo, "A Session Initiation Protocol 1192 (SIP) Event Package for Session-Specific Session Policies.", 1193 draft-ietf-sipping-policy-package-00 (work in progress), 1194 April 2006. 1196 [96] Camarillo, G. and A. Monrad, "Mapping of Media Streams to 1197 Resource Reservation Flows", RFC 3524, April 2003. 1199 [97] Lawrence, S., "Diagnostic Responses for SIP Hop Limit Errors", 1200 draft-ietf-sip-hop-limit-diagnostics-02 (work in progress), 1201 June 2006. 1203 [98] Burger, E., Van Dyke, J., and A. Spitzer, "Basic Network Media 1204 Services with SIP", RFC 4240, December 2005. 1206 [99] Jennings, C., "Certificate Management Service for The Session 1207 Initiation Protocol (SIP)", draft-ietf-sip-certs-00 (work in 1208 progress), May 2006. 1210 [100] Lennox, J., "Connection-Oriented Media Transport over the 1211 Transport Layer Security (TLS) Protocol in the Session 1212 Description Protocol (SDP)", draft-ietf-mmusic-comedia-tls-06 1213 (work in progress), March 2006. 1215 Author's Address 1217 Jonathan Rosenberg 1218 Cisco Systems 1219 600 Lanidex Plaza 1220 Parsippany, NJ 07054 1221 US 1223 Phone: +1 973 952-5000 1224 Email: jdrosen@cisco.com 1225 URI: http://www.jdrosen.net 1227 Intellectual Property Statement 1229 The IETF takes no position regarding the validity or scope of any 1230 Intellectual Property Rights or other rights that might be claimed to 1231 pertain to the implementation or use of the technology described in 1232 this document or the extent to which any license under such rights 1233 might or might not be available; nor does it represent that it has 1234 made any independent effort to identify any such rights. 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