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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 ECRIT H. Schulzrinne 3 Internet-Draft Columbia University 4 Intended status: Standards Track H. Tschofenig 5 Expires: April 17, 2014 Nokia Solutions and Networks 6 C. Holmberg 7 Ericsson 8 M. Patel 9 InterDigital Communications 10 October 14, 2013 12 Public Safety Answering Point (PSAP) Callback 13 draft-ietf-ecrit-psap-callback-13.txt 15 Abstract 17 After an emergency call is completed (either prematurely terminated 18 by the emergency caller or normally by the call taker) it is possible 19 that the call taker feels the need for further communication. For 20 example, the call may have been dropped by accident without the call 21 taker having sufficient information about the current situation of a 22 wounded person. A call taker may trigger a callback towards the 23 emergency caller using the contact information provided with the 24 initial emergency call. This callback could, under certain 25 circumstances, be treated like any other call and as a consequence it 26 may get blocked by authorization policies or may get forwarded to an 27 answering machine. 29 The IETF emergency services architecture specification already offers 30 a solution approach for allowing PSAP callbacks to bypass 31 authorization policies to reach the caller without unnecessary 32 delays. Unfortunately, the specified mechanism only supports limited 33 scenarios. This document discusses shortcomings of the current 34 mechanisms and illustrates additional scenarios where better-than- 35 normal call treatment behavior would be desirable. A solution based 36 on a new header field value, called "psap-callback", for the SIP 37 Priority header field is specified to accomplish the PSAP callback 38 marking. 40 Status of This Memo 42 This Internet-Draft is submitted in full conformance with the 43 provisions of BCP 78 and BCP 79. 45 Internet-Drafts are working documents of the Internet Engineering 46 Task Force (IETF). Note that other groups may also distribute 47 working documents as Internet-Drafts. The list of current Internet- 48 Drafts is at http://datatracker.ietf.org/drafts/current/. 50 Internet-Drafts are draft documents valid for a maximum of six months 51 and may be updated, replaced, or obsoleted by other documents at any 52 time. It is inappropriate to use Internet-Drafts as reference 53 material or to cite them other than as "work in progress." 55 This Internet-Draft will expire on April 17, 2014. 57 Copyright Notice 59 Copyright (c) 2013 IETF Trust and the persons identified as the 60 document authors. All rights reserved. 62 This document is subject to BCP 78 and the IETF Trust's Legal 63 Provisions Relating to IETF Documents 64 (http://trustee.ietf.org/license-info) in effect on the date of 65 publication of this document. Please review these documents 66 carefully, as they describe your rights and restrictions with respect 67 to this document. Code Components extracted from this document must 68 include Simplified BSD License text as described in Section 4.e of 69 the Trust Legal Provisions and are provided without warranty as 70 described in the Simplified BSD License. 72 Table of Contents 74 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 75 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 76 3. Callback Scenarios . . . . . . . . . . . . . . . . . . . . . 4 77 3.1. Routing Asymmetry . . . . . . . . . . . . . . . . . . . . 5 78 3.2. Multi-Stage Routing . . . . . . . . . . . . . . . . . . . 5 79 3.3. Call Forwarding . . . . . . . . . . . . . . . . . . . . . 6 80 3.4. Network-based Service URN Resolution . . . . . . . . . . 8 81 3.5. PSTN Interworking . . . . . . . . . . . . . . . . . . . . 9 82 4. SIP PSAP Callback Indicator . . . . . . . . . . . . . . . . . 10 83 4.1. General . . . . . . . . . . . . . . . . . . . . . . . . . 10 84 4.2. Usage . . . . . . . . . . . . . . . . . . . . . . . . . . 10 85 4.3. Syntax . . . . . . . . . . . . . . . . . . . . . . . . . 10 86 4.3.1. General . . . . . . . . . . . . . . . . . . . . . . . 10 87 4.3.2. ABNF . . . . . . . . . . . . . . . . . . . . . . . . 10 88 5. Security Considerations . . . . . . . . . . . . . . . . . . . 10 89 5.1. Security Threat . . . . . . . . . . . . . . . . . . . . . 10 90 5.2. Security Requirements . . . . . . . . . . . . . . . . . . 11 91 5.3. Security Solution . . . . . . . . . . . . . . . . . . . . 11 92 6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13 93 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 13 94 8. References . . . . . . . . . . . . . . . . . . . . . . . . . 14 95 8.1. Normative References . . . . . . . . . . . . . . . . . . 14 96 8.2. Informative References . . . . . . . . . . . . . . . . . 14 98 1. Introduction 100 Summoning police, the fire department or an ambulance in emergencies 101 is one of the fundamental and most-valued functions of the telephone. 102 As telephone functionality moves from circuit-switched telephony to 103 Internet telephony, its users rightfully expect that this core 104 functionality will continue to work at least as well as it has for 105 the legacy technology. New devices and services are being made 106 available that could be used to make a request for help, which are 107 not traditional telephones, and users are increasingly expecting them 108 to be used to place emergency calls. 110 An overview of the protocol interactions for emergency calling using 111 the IETF emergency services architecture are described in [RFC6443] 112 and [RFC6881] specifies the technical details. As part of the 113 emergency call setup procedure two important identifiers are conveyed 114 to the PSAP call taker's user agent, namely the Address-Of-Record 115 (AOR), and, if available, the Globally Routable User Agent (UA) URIs 116 (GRUU). RFC 3261 [RFC3261] defines the AOR as: 118 "An address-of-record (AOR) is a SIP or SIPS URI that points to a 119 domain with a location service that can map the URI to another URI 120 where the user might be available. Typically, the location 121 service is populated through registrations. An AOR is frequently 122 thought of as the "public address" of the user." 124 In SIP systems a single user can have a number of user agents 125 (handsets, softphones, voicemail accounts, etc.) which are all 126 referenced by the same AOR. There are a number of cases in which it 127 is desirable to have an identifier which addresses a single user 128 agent rather than the group of user agents indicated by an AOR. The 129 GRUU is such a unique user-agent identifier, which is still globally 130 routable. RFC 5627 [RFC5627] specifies how to obtain and use GRUUs. 131 [RFC6881] also makes use of the GRUU for emergency calls. 133 Regulatory requirements demand that the emergency call setup 134 procedure itself provides enough information to allow the call taker 135 to initiate a callback to the emergency caller. This is desirable in 136 those cases where the call got dropped prematurely or when further 137 communication need arises. The AOR and the GRUU serve this purpose. 139 The communication attempt by the PSAP call taker back to the 140 emergency caller is called 'PSAP callback'. 142 A PSAP callback may, however, be blocked by user configured 143 authorization policies or may be forwarded to an answering machine 144 since SIP entities (SIP proxies as well as the SIP user equipment 145 itself) cannot differentiate the PSAP callback from any other SIP 146 call. "Call barring", "do not disturb", or "call diversion"(aka call 147 forwarding) are features that prevent delivery of a call. It is 148 important to note that these features may be implemented by SIP 149 intermediaries as well as by the user agent. 151 Among the emergency services community there is the desire to offer 152 PSAP callbacks a treatment such that chances are increased that it 153 reaches the emergency caller. At the same time a design must deal 154 with the negative side-effects of allowing certain calls to bypass 155 call forwarding or other authorization policies. Ideally, the PSAP 156 callback has to relate to an earlier emergency call that was made 157 "not too long ago". An exact time interval is difficult to define in 158 a global IETF standard due to the variety of national regulatory 159 requirements but [RFC6881] suggests 30 minutes. 161 To nevertheless meet the needs from the emergency services community 162 a basic mechanism for preferential treatment of PSAP callbacks was 163 defined in Section 13 of [RFC6443]. The specification says: 165 "A UA may be able to determine a PSAP callback by examining the 166 domain of incoming calls after placing an emergency call and 167 comparing that to the domain of the answering PSAP from the 168 emergency call. Any call from the same domain and directed to the 169 supplied Contact header or AOR after an emergency call should be 170 accepted as a callback from the PSAP if it occurs within a 171 reasonable time after an emergency call was placed." 173 This approach mimics a stateful packet filtering firewall and is 174 indeed helpful in a number of cases. It is also relatively simple to 175 implement even though it requires call state to be maintained by the 176 user agent as well as by SIP intermediaries. Unfortunately, the 177 solution does not work in all deployment scenarios. In Section 3 we 178 describe cases where the currently standardized approach is 179 insufficient. 181 2. Terminology 183 Emergency services related terminology is borrowed from [RFC5012]. 184 This includes terminology like emergency caller, user equipment, call 185 taker, Emergency Service Routing Proxy (ESRP), and Public Safety 186 Answering Point (PSAP). 188 3. Callback Scenarios 190 This section illustrates a number of scenarios where the currently 191 specified solution, as specified in [RFC6881], for preferential 192 treatment of callbacks fails. As explained in Section 1 a SIP entity 193 examines an incoming PSAP callback by comparing the domain of the 194 PSAP with the destination domain of the outbound emergency call 195 placed earlier. 197 3.1. Routing Asymmetry 199 In some deployment environments it is common to have incoming and 200 outgoing SIP messaging routed through different SIP entities. Figure 201 1 shows this graphically whereby a VoIP provider uses different SIP 202 proxies for inbound and for outbound call handling. Unless the two 203 devices are synchronized, the callback hitting the inbound proxy 204 would get treated like any other call since the emergency call 205 established state information at the outbound proxy only. 207 ,-------. 208 ,' `. 209 ,-------. / Emergency \ 210 ,' `. | Services | 211 / VoIP \ I | Network | 212 | Provider | n | | 213 | | t | | 214 | | e | | 215 | +-------+ | r | | 216 +--+---|Inbound|<--+-----m | | 217 | | |Proxy | | e | +------+ | 218 | | +-------+ | d | |PSAP | | 219 | | | i | +--+---+ | 220 +----+ | | | a-+ | | | 221 | UA |<---+ | | t | | | | 222 | |----+ | | e | | | | 223 +----+ | | | | | | | 224 | | | P | | | | 225 | | | r | | | | 226 | | +--------+ | o | | | | 227 +--+-->|Outbound|--+---->v | | +--+---+ | 228 | |Proxy | | i | | +-+ESRP | | 229 | +--------+ | d | | | +------+ | 230 | | e || | | 231 | | r |+-+ | 232 \ / | | 233 `. ,' \ / 234 '-------' `. ,' 235 '-------' 237 Figure 1: Example for Routing Asymmetry. 239 3.2. Multi-Stage Routing 240 Consider the following emergency call routing scenario shown in 241 Figure 2 where routing towards the PSAP occurs in several stages. In 242 this scenario we consider a SIP UA that uses the Location-to-Service 243 Translation Protocol (LoST) [RFC5222] to learn the next hop 244 destination, namely esrp@example.net, to get the call closer to the 245 PSAP. This call is then sent to the proxy of the user's VoIP 246 provider (example.org). The user's VoIP provider receives the 247 emergency call and creates state based on the destination domain, 248 namely example.net. It then routes it to the indicated ESRP. When 249 the ESRP receives it it needs to decide what the next hop is to get 250 to the final PSAP. In our example the next hop is the PSAP with the 251 URI psap@example.com. 253 When a callback is sent from psap@example.com towards the emergency 254 caller the call will get normal treatment by the proxy of the VoIP 255 provider since the domain of the PSAP does not match the stored state 256 information. 258 ,-----------. 259 +----+ ,' `. 260 | UA |--- esrp@example.net / Emergency \ 261 +----+ \ | Services | 262 \ ,-------. | Network | 263 ,' `. | | 264 / VoIP \ | +------+ | 265 ( Provider ) | | PSAP | | 266 \ example.org / | +--+---+ | 267 `. ,' | | | 268 '---+---' | | | 269 | | psap@example.com | 270 esrp@example.net | | | 271 | | | | 272 | | | | 273 | | +--+---+ | 274 +------------+-----+ ESRP | | 275 | +------+ | 276 | | 277 \ / 278 `. ,' 279 '----------' 281 Figure 2: Example for Multi-Stage Routing. 283 3.3. Call Forwarding 285 Imagine the following case where an emergency call enters an 286 emergency network (state.example) via an ESRP but then gets forwarded 287 to a different emergency services network (in our example to 288 example.net, example.org or example.com). The same considerations 289 apply when the police, fire and ambulance networks are part of the 290 state.example sub-domains (e.g., police.state.example). 292 Similar to the previous scenario the problem here is with the wrong 293 state information being established during the emergency call setup 294 procedure. A callback would originate in the example.net, 295 example.org or example.com domains whereas the emergency caller's SIP 296 UA or the VoIP outbound proxy has stored state.example. 298 ,-------. 299 ,' `. 300 / Emergency \ 301 | Services | 302 | Network | 303 |(state.example)| 304 | | 305 | | 306 | +------+ | 307 | |PSAP +--+ | 308 | +--+---+ | | 309 | | | | 310 | | | | 311 | | | | 312 | | | | 313 | | | | 314 | +--+---+ | | 315 ------------------+---+ESRP | | | 316 esrp-a@state.org | +------+ | | 317 | | | 318 | Call Fwd | | 319 | +-+-+---+ | 320 \ | | | / 321 `. | | | ,' 322 '-|-|-|-' ,-------. 323 Police | | | Fire ,' `. 324 +------------+ | +----+ / Emergency \ 325 ,-------. | | | | Services | 326 ,' `. | | | | Network | 327 / Emergency \ | Ambulance | | (Fire) | 328 | Services | | | | | | 329 | Network | | +----+ | | +------+ | 330 | (Police) | | ,-------. | +----+---+PSAP | | 331 | | | ,' `. | | +------+ | 332 | +------+ | | / Emergency \ | | | 333 | |PSAP +----+--+ | Services | | | example.com , 334 | +------+ | | Network | | `~~~~~~~~~~~~~~~ 335 | | | (Ambulance) | | 336 | example.net , | | | 337 `~~~~~~~~~~~~~~~ | +------+ | | 338 | |PSAP +----+ + 339 | +------+ | 340 | | 341 | example.org , 342 `~~~~~~~~~~~~~~~ 344 Figure 3: Example for Call Forwarding. 346 3.4. Network-based Service URN Resolution 348 The IETF emergency services architecture also considers cases where 349 the resolution from the Service URN to the PSAP URI does not only 350 happen at the SIP UA itself but at intermediate SIP entities, such as 351 the user's VoIP provider. 353 Figure 4 shows this message exchange of the outgoing emergency call 354 and the incoming PSAP graphically. While the state information 355 stored at the VoIP provider is correct the state allocated at the SIP 356 UA is not. 358 ,-------. 359 ,' `. 360 / Emergency \ 361 | Services | 362 | Network | 363 | example.com | 364 | | 365 | +------+ | Invite to police@example.com 366 | |PSAP +<---+------------------------+ 367 | | +----+--------------------+ ^ 368 | +------+ |Invite from | | 369 | ,police@example.com | | 370 `~~~~~~~~~~~~~~~ | | 371 v | 372 +--------+ Query with location +--+---+-+ 373 | | + urn:service:sos | VoIP | 374 | LoST |<-----------------------|Service | 375 | Server | police@example.com |Provider| 376 | |----------------------->| | 377 +--------+ +--------+ 378 | ^ 379 Invite| | Invite 380 from| | to 381 police@example.com| | urn:service:sos 382 V | 384 +-------+ 385 | SIP | 386 | UA | 387 | Alice | 388 +-------+ 390 Figure 4: Example for Network-based Service URN Resolution. 392 3.5. PSTN Interworking 394 In case an emergency call enters the PSTN, as shown in Figure 5, 395 there is no guarantee that the callback some time later leaves the 396 same PSTN/VoIP gateway or that the same end point identifier is used 397 in the forward as well as in the backward direction making it 398 difficult to reliably detect PSAP callbacks. 400 +-----------+ 401 | PSTN |-------------+ 402 | Calltaker | | 403 | Bob |<--------+ | 404 +-----------+ | v 405 ------------------- 406 //// \\\\ +------------+ 407 | | |PSTN / VoIP | 408 | PSTN |---->|Gateway | 409 \\\\ //// | | 410 ------------------- +----+-------+ 411 ^ | 412 | | 413 +-------------+ | +--------+ 414 | | | |VoIP | 415 | PSTN / VoIP | +->|Service | 416 | Gateway | |Provider| 417 | |<------Invite----| Y | 418 +-------------+ +--------+ 419 | ^ 420 | | 421 Invite Invite 422 | | 423 V | 424 +-------+ 425 | SIP | 426 | UA | 427 | Alice | 428 +-------+ 430 Figure 5: Example for PSTN Interworking. 432 Note: This scenario is considered outside the scope of this document. 433 The specified solution does not support this use case. 435 4. SIP PSAP Callback Indicator 437 4.1. General 439 This section defines a new header field value, called "psap- 440 callback", for the SIP Priority header field defined in [RFC3261]. 441 The value is used to inform SIP entities that the request is 442 associated with a PSAP callback SIP session. 444 4.2. Usage 446 SIP entities that receive the header field value within an initial 447 request for a SIP session can, depending on local policies, apply 448 PSAP callback specific procedures for the session or request. 450 The PSAP callback specific procedures may be applied by SIP-based 451 network entities and by the callee. The specific procedures taken 452 when receiving such a PSAP callback marked call, such as bypassing 453 services and barring procedures, are outside the scope of this 454 document. 456 4.3. Syntax 458 4.3.1. General 460 This section defines the ABNF for the new SIP Priority header field 461 value "psap-callback". 463 4.3.2. ABNF 465 priority-value /= "psap-callback" 467 Figure 6: ABNF 469 5. Security Considerations 471 5.1. Security Threat 473 The PSAP callback functionality described in this document allows 474 marked calls to bypass blacklists, ignore call forwarding procedures 475 and other similar features used to raise the attention of emergency 476 callers when attempting to contact them. In the case where the SIP 477 Priority header value, 'psap-callback', is supported by the SIP UA, 478 it would override user interface configurations, such as vibrate-only 479 mode, to alert the caller of the incoming call. 481 5.2. Security Requirements 483 The security threat discussed in Section 5.1 leads to the requirement 484 to ensure that the mechanisms described in this document can not be 485 used for malicious purposes, including telemarketing. 487 Furthermore, if the newly defined extension is not recognized, not 488 verified adequately, or not obeyed by SIP intermediaries or SIP 489 endpoints then it must not lead to a failure of the call handling 490 procedure. Such call must be treated like a call that does not have 491 any marking attached. 493 The indicator described in Section 4 can be inserted by any SIP 494 entity, including attackers. So it is critical that the indicator 495 only lead to preferential call treatment in cases where the recipient 496 has some trust in the caller, as described in the next section. 498 5.3. Security Solution 500 The approach for dealing with implementing the security requirements 501 described in Section 5.2 can be differentiated between the behavior 502 applied by the UA and by SIP proxies. A UA that has made an 503 emergency call MUST keep state information so that it can recognize 504 and accepted a callback from the PSAP if it occurs within a 505 reasonable time after an emergency call was placed, as described in 506 Section 13 of [RFC6443]. Only a timer started at the time when the 507 original emergency call has ended is required; information about the 508 calling party identity is not needed since the callback may use a 509 different calling party identity, as described in Section 3. Since 510 these SIP UA considerations are described already in [RFC6443] as 511 well as in [RFC6881] the rest of this section focuses on the behavior 512 of SIP proxies. 514 Figure 7 shows the architecture that utilizes the identity of the 515 PSAP to decide whether a preferential treatment of callbacks should 516 be provided. To make this policy decision, the identity of the PSAP 517 (i.e., calling party identity) is compared with a PSAPs white list. 519 +----------+ 520 | List of |+ 521 | valid || 522 | PSAPs || 523 +----------+| 524 +----------+ 525 * 526 * white list 527 * 528 V 530 Incoming +----------+ Normal 531 SIP Msg | SIP |+ Treatment 532 -------------->| Entity ||======================> 533 + Identity | ||(if not in white list) 534 Info +----------+| 535 +----------+ 536 || 537 || 538 || Preferential 539 || Treatment 540 ++========================> 541 (if successfully verified) 543 Figure 7: Identity-based Authorization 545 The identity assurance in SIP can come in different forms, namely via 546 the SIP Identity [RFC4474] or the P-Asserted-Identity [RFC3325] 547 mechanisms. The former technique relies on a cryptographic assurance 548 and the latter on a chain of trust. Also the usage of TLS between 549 neighboring SIP entities may provide useful identity information. At 550 the time of writing these identity technologies are being revised in 551 the Secure Telephone Identity Revisited (stir) working group [STIR] 552 to offer better support for legacy technologies interworking and SIP 553 intermediaries that modify the content of various SIP headers and the 554 body. Once the work on these specifications has been completed they 555 will offer a stronger calling party identity mechanism that limits or 556 prevents identity spoofing. 558 An important aspect from a security point of view is the relationship 559 between the emergency services network (containing the PSAPs) and the 560 VoIP provider (assuming that the emergency call travels via the VoIP 561 provider and not directly between the SIP UA and the PSAP). 563 The establishment of a white list with PSAP identities may be 564 operationally complex and dependent on the relationship between the 565 emergency services operator and the VoIP provider. When there is a 566 relationship between the VoIP provider and the PSAP operator, for 567 example when they are both operating in the same geographical region, 568 then populating the white list is fairly simple and consequently the 569 identification of a PSAP callback is less problematic compared to the 570 case where the two entities have never interacted with each other 571 before. In the end, the VoIP provider has to verify whether the 572 marked callback message indeed came from a legitimate source. 574 VoIP providers MUST only give PSAP callbacks preferential treatment 575 when the calling party identity of the PSAP was successfully matched 576 against entries in the white list. If it cannot be verified (because 577 there was no match),then the VoIP provider MUST remove the PSAP 578 callback marking. Thereby, the callback is degenerated to a normal 579 call. As a second step, SIP UAs MUST maintain a timer that is 580 started with the original emergency call and this timer expires 581 within a reasonable amount of time, such as 30 minutes per [RFC6881]. 582 Such a timer also ensures that VoIP providers cannot misuse the PSAP 583 callback mechanism, for example to ensure that their support calls 584 reaches their customers. 586 Finally, a PSAP callback MUST use the same media as the original 587 emergency call. For example, when an initial emergency call 588 established a real-time text communication session then the PSAP 589 callback must also attempt to establish a real-time communication 590 interaction. The reason for this is two-fold. First, the person 591 seeking for help may have disabilities that prevent them from using 592 certain media and hence using the same media for the callback avoids 593 unpleasant surprises and delays. Second, the emergency caller may 594 have intentionally chosen a certain media and does not prefer to 595 communicate in a different way. For example, it would be unfortunate 596 if a hostage tries to seek for help using instant messaging to avoid 597 any noise when subsequently the ring-tone triggered by a PSAP 598 callback using a voice call gets the attention of the hostage-taker. 599 User interface designs need to cater to such situations. 601 6. IANA Considerations 603 This document adds the "psap-callback" value to the SIP Priority 604 header IANA registry allocated by [RFC6878]. The semantic of the 605 newly defined "psap-callback" value is defined in Section 4. 607 7. Acknowledgements 609 We would like to thank the following persons for their feedback: Paul 610 Kyzivat, Martin Thomson, Robert Sparks, Keith Drage, Cullen Jennings 611 Brian Rosen, Martin Dolly, Bernard Aboba, Andrew Allen, Atle Monrad, 612 John-Luc Bakker, John Elwell, Geoff Thompson, Dan Romascanu, James 613 Polk, John Medland, Hadriel Kaplan, Kenneth Carlberg, Timothy Dwight, 614 Janet Gunn 616 We would like to thank the ECRIT working group chairs, Marc Linsner 617 and Roger Marshall, for their support. Roger Marshall was the 618 document shepherd for this document. Vijay Gurbani provided the 619 general area review. 621 During IESG review the document received good feedback from Barry 622 Leiba, Spencer Dawkins, Richard Barnes, Joel Jaeggli, Stephen 623 Farrell, and Benoit Claise. 625 8. References 627 8.1. Normative References 629 [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, 630 A., Peterson, J., Sparks, R., Handley, M., and E. 631 Schooler, "SIP: Session Initiation Protocol", RFC 3261, 632 June 2002. 634 [RFC5627] Rosenberg, J., "Obtaining and Using Globally Routable User 635 Agent URIs (GRUUs) in the Session Initiation Protocol 636 (SIP)", RFC 5627, October 2009. 638 [RFC6878] Roach, A., "IANA Registry for the Session Initiation 639 Protocol (SIP) "Priority" Header Field", RFC 6878, March 640 2013. 642 8.2. Informative References 644 [RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private 645 Extensions to the Session Initiation Protocol (SIP) for 646 Asserted Identity within Trusted Networks", RFC 3325, 647 November 2002. 649 [RFC4474] Peterson, J. and C. Jennings, "Enhancements for 650 Authenticated Identity Management in the Session 651 Initiation Protocol (SIP)", RFC 4474, August 2006. 653 [RFC5012] Schulzrinne, H. and R. Marshall, "Requirements for 654 Emergency Context Resolution with Internet Technologies", 655 RFC 5012, January 2008. 657 [RFC5222] Hardie, T., Newton, A., Schulzrinne, H., and H. 658 Tschofenig, "LoST: A Location-to-Service Translation 659 Protocol", RFC 5222, August 2008. 661 [RFC6443] Rosen, B., Schulzrinne, H., Polk, J., and A. Newton, 662 "Framework for Emergency Calling Using Internet 663 Multimedia", RFC 6443, December 2011. 665 [RFC6881] Rosen, B. and J. Polk, "Best Current Practice for 666 Communications Services in Support of Emergency Calling", 667 BCP 181, RFC 6881, March 2013. 669 [STIR] IETF, "Secure Telephone Identity Revisited (stir) Working 670 Group", URL: http://datatracker.ietf.org/wg/stir/charter/, 671 Oct 2013. 673 Authors' Addresses 675 Henning Schulzrinne 676 Columbia University 677 Department of Computer Science 678 450 Computer Science Building 679 New York, NY 10027 680 US 682 Phone: +1 212 939 7004 683 EMail: hgs+ecrit@cs.columbia.edu 684 URI: http://www.cs.columbia.edu 686 Hannes Tschofenig 687 Nokia Solutions and Networks 688 Linnoitustie 6 689 Espoo 02600 690 Finland 692 Phone: +358 (50) 4871445 693 EMail: Hannes.Tschofenig@gmx.net 694 URI: http://www.tschofenig.priv.at 696 Christer Holmberg 697 Ericsson 698 Hirsalantie 11 699 Jorvas 02420 700 Finland 702 EMail: christer.holmberg@ericsson.com 704 Milan Patel 705 InterDigital Communications 707 EMail: Milan.Patel@interdigital.com