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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: June 21, 2013 Nokia Siemens Networks 6 C. Holmberg 7 Ericsson 8 M. Patel 9 InterDigital Communications 10 December 18, 2012 12 Public Safety Answering Point (PSAP) Callback 13 draft-ietf-ecrit-psap-callback-07.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. 37 Note that this version of the document does not yet specify a 38 solution due to the lack of the working group participants agreeing 39 on the requirements. 41 Status of This Memo 43 This Internet-Draft is submitted in full conformance with the 44 provisions of BCP 78 and BCP 79. 46 Internet-Drafts are working documents of the Internet Engineering 47 Task Force (IETF). Note that other groups may also distribute 48 working documents as Internet-Drafts. The list of current Internet- 49 Drafts is at http://datatracker.ietf.org/drafts/current/. 51 Internet-Drafts are draft documents valid for a maximum of six months 52 and may be updated, replaced, or obsoleted by other documents at any 53 time. It is inappropriate to use Internet-Drafts as reference 54 material or to cite them other than as "work in progress." 56 This Internet-Draft will expire on June 21, 2013. 58 Copyright Notice 60 Copyright (c) 2012 IETF Trust and the persons identified as the 61 document authors. All rights reserved. 63 This document is subject to BCP 78 and the IETF Trust's Legal 64 Provisions Relating to IETF Documents 65 (http://trustee.ietf.org/license-info) in effect on the date of 66 publication of this document. Please review these documents 67 carefully, as they describe your rights and restrictions with respect 68 to this document. Code Components extracted from this document must 69 include Simplified BSD License text as described in Section 4.e of 70 the Trust Legal Provisions and are provided without warranty as 71 described in the Simplified BSD License. 73 Table of Contents 75 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 76 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6 77 3. Callback Scenarios . . . . . . . . . . . . . . . . . . . . . . 7 78 3.1. Routing Asymmetry . . . . . . . . . . . . . . . . . . . . 7 79 3.2. Multi-Stage Routing . . . . . . . . . . . . . . . . . . . 8 80 3.3. Call Forwarding . . . . . . . . . . . . . . . . . . . . . 9 81 3.4. Network-based Service URN Resolution . . . . . . . . . . . 11 82 3.5. PSTN Interworking . . . . . . . . . . . . . . . . . . . . 12 83 4. SIP PSAP Callback Indicator . . . . . . . . . . . . . . . . . 13 84 4.1. General . . . . . . . . . . . . . . . . . . . . . . . . . 13 85 4.2. Usage . . . . . . . . . . . . . . . . . . . . . . . . . . 13 86 4.3. Syntax . . . . . . . . . . . . . . . . . . . . . . . . . . 13 87 4.3.1. General . . . . . . . . . . . . . . . . . . . . . . . 13 88 4.3.2. ABNF . . . . . . . . . . . . . . . . . . . . . . . . . 13 89 5. Security Considerations . . . . . . . . . . . . . . . . . . . 14 90 5.1. Security Threat . . . . . . . . . . . . . . . . . . . . . 14 91 5.2. Security Requirements . . . . . . . . . . . . . . . . . . 14 92 5.3. Security Solution . . . . . . . . . . . . . . . . . . . . 14 93 6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 16 94 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 17 95 8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 18 96 8.1. Normative References . . . . . . . . . . . . . . . . . . . 18 97 8.2. Informative References . . . . . . . . . . . . . . . . . . 19 99 1. Introduction 101 Summoning police, the fire department or an ambulance in emergencies 102 is one of the fundamental and most-valued functions of the telephone. 103 As telephone functionality moves from circuit-switched telephony to 104 Internet telephony, its users rightfully expect that this core 105 functionality will continue to work at least as well as it has for 106 the legacy technology. New devices and services are being made 107 available that could be used to make a request for help, which are 108 not traditional telephones, and users are increasingly expecting them 109 to be used to place emergency calls. 111 An overview of the protocol interactions for emergency calling using 112 the IETF emergency services architecture are described in [RFC6444] 113 and [I-D.ietf-ecrit-phonebcp] specifies the technical details. As 114 part of the emergency call setup procedure two important identifiers 115 are conveyed to the PSAP call taker's user agent, namely the Address- 116 Of-Record (AoR), and, if available, the Globally Routable User Agent 117 (UA) URIs (GRUU). RFC 3261 [RFC3261] defines the AoR as: 119 'An address-of-record (AOR) is a SIP or SIPS URI that points to a 120 domain with a location service that can map the URI to another URI 121 where the user might be available. Typically, the location 122 service is populated through registrations. An AOR is frequently 123 thought of as the "public address" of the user.' 125 In SIP systems a single user can have a number of user agents 126 (handsets, softphones, voicemail accounts, etc.) which are all 127 referenced by the same AOR. There are a number of cases in which it 128 is desirable to have an identifier which addresses a single user 129 agent rather than the group of user agents indicated by an AOR. The 130 GRUU is such a unique user- agent identifier, which is still globally 131 routable. RFC 5627 [RFC5627] specifies how to obtain and use GRUUs. 132 [I-D.ietf-ecrit-phonebcp] also makes use of the GRUU for emergency 133 calls. 135 Regulatory requirements demand that the emergency call setup 136 procedure itself provides enough information to allow the call taker 137 to initiate a call back to the emergency caller. This is desirable 138 in those cases where the call got dropped prematurely or when further 139 communication need arises. The AoR and the GRUU serve this purpose. 141 The communication attempt by the PSAP call taker back to the 142 emergency caller is called 'PSAP callback'. 144 A PSAP callback may, however, be blocked by user configured 145 authorization policies or may be forwarded to an answering machine 146 since SIP entities (SIP proxies as well as the SIP user equipment 147 itself) cannot differentiate the PSAP callback from any other SIP 148 call. "Call barring", "do not disturb", or "call diversion"(aka call 149 forwarding) are features that prevent delivery of a call. It is 150 important to note that these features may be implemented by SIP 151 intermediaries as well as by the user agent. 153 Among the emergency services community there is the desire to offer 154 PSAP callbacks a treatment such that chances are increased that it 155 reaches the emergency caller. At the same time a design must deal 156 with the negative side-effects of allowing certain calls to bypass 157 call forwarding or other authorization policies. Ideally, the PSAP 158 callback has to relate to an earlier emergency call that was made 159 "not too long ago". An exact time interval is difficult to define in 160 a global IETF standard due to the variety of national regulatory 161 requirements. 163 To nevertheless meet the needs from the emergency services community 164 a basic mechanism for preferential treatment of PSAP callbacks was 165 defined in Section 13 of [RFC6444]. The specification says: 167 'A UA may be able to determine a PSAP call back by examining the 168 domain of incoming calls after placing an emergency call and 169 comparing that to the domain of the answering PSAP from the 170 emergency call. Any call from the same domain and directed to the 171 supplied Contact header or AoR after an emergency call should be 172 accepted as a callback from the PSAP if it occurs within a 173 reasonable time after an emergency call was placed.' 175 This approach mimics a stateful packet filtering firewall and is 176 indeed helpful in a number of cases. It is also relatively simple to 177 implement even though it requires state to be maintained by the user 178 agent as well as by SIP intermediaries. Unfortunately, the solution 179 does not work in all deployment scenarios. In Section 3 we describe 180 cases where the currently standardized approach is insufficient. 182 2. Terminology 184 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 185 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 186 document are to be interpreted as described in [RFC2119]. 188 Emergency services related terminology is borrowed from [RFC5012]. 189 This includes terminology like emergency caller, user equipment, and 190 call taker. 192 3. Callback Scenarios 194 This section illustrates a number of scenarios where the currently 195 specified solution, as specified in [I-D.ietf-ecrit-phonebcp], for 196 preferential treatment of callbacks fails. As explained in Section 1 197 a SIP entity examines an incoming PSAP call back by comparing the 198 domain of the PSAP with the destination domain of the emergency call. 200 3.1. Routing Asymmetry 202 In some deployment environments it is common to have incoming and 203 outgoing SIP messaging routed through different SIP entities. 204 Figure 1 shows this graphically whereby a VoIP provider uses 205 different SIP proxies for inbound and for outbound call handling. 206 Unless they two devices are state synchronized the callback hitting 207 the inbound proxy would get treated like any other call since the 208 emergency call established state information at the outbound proxy 209 only. 211 ,-------. 212 ,' `. 213 ,-------. / Emergency \ 214 ,' `. | Services | 215 / VoIP \ I | Network | 216 | Provider | n | | 217 | | t | | 218 | | e | | 219 | +-------+ | r | | 220 +--+---|Inbound|<--+-----m | | 221 | | |Proxy | | e | +------+ | 222 | | +-------+ | d | |PSAP | | 223 | | | i | +--+---+ | 224 +----+ | | | a-+ | | | 225 | UA |<---+ | | t | | | | 226 | |----+ | | e | | | | 227 +----+ | | | | | | | 228 | | | P | | | | 229 | | | r | | | | 230 | | +--------+ | o | | | | 231 +--+-->|Outbound|--+---->v | | +--+---+ | 232 | |Proxy | | i | | +-+ESRP | | 233 | +--------+ | d | | | +------+ | 234 | | e || | | 235 | | r |+-+ | 236 \ / | | 237 `. ,' \ / 238 '-------' `. ,' 239 '-------' 241 Figure 1: Example for Routing Asymmetry 243 3.2. Multi-Stage Routing 245 Consider the following emergency call routing scenario shown in 246 Figure 2 where routing towards the PSAP occurs in several stages. In 247 this scenario we consider a SIP UA that uses LoST to learn the next 248 hop destination closer to the PSAP. This call is then sent to the 249 user's VoIP provider. The user's VoIP provider receives the 250 emergency call and creates state based on the destination domain, 251 namely state.com. It then routes it to the indicated ESRP. When the 252 ESRP receives it it needs to decide what the next hop is to get it 253 closer to the PSAP. In our example the next hop is the PSAP with the 254 URI psap@town.com. 256 When a callback is sent from psap@town.com towards the emergency 257 caller the call will get normal treatment by the VoIP providers 258 inbound proxy since the domain of the PSAP does not match the stored 259 state information. 261 ,-------. 262 +----+ ,' `. 263 | UA |--- esrp1@foobar.com / Emergency \ 264 +----+ \ | Services | 265 \ ,-------. | Network | 266 ,' `. | | 267 / VoIP \ | +------+ | 268 ( Provider ) | |PSAP | | 269 \ / | +--+---+ | 270 `. ,' | | 271 '---+---' | | | 272 | |psap@town.com | 273 esrp@state.com | | | 274 | | | | 275 | | | | 276 | | +--+---+ | 277 +------------+---+ESRP | | 278 | +------+ | 279 | | 280 \ / 281 `. ,' 282 '-------' 284 Figure 2: Example for Multi-Stage Routing 286 3.3. Call Forwarding 288 Imagine the following case where an emergency call enters an 289 emergency network (state.org) via an ERSP but then gets forwarded to 290 a different emergency services network (in our example to police- 291 town.org, fire-town.org or medic-town.org). The same considerations 292 apply when the police, fire and ambulance networks are part of the 293 state.org sub-domains (e.g., police.state.org). 295 Similarly to the previous scenario the problem here is with the wrong 296 state information being established during the emergency call setup 297 procedure. A callback would originate in the police-town.org, fire- 298 town.org or medic-town.org domain whereas the emergency caller's SIP 299 UA or the VoIP outbound proxy has stored state.org. 301 ,-------. 302 ,' `. 303 / Emergency \ 304 | Services | 305 | Network | 306 | (state.org) | 307 | | 308 | | 309 | +------+ | 310 | |PSAP +--+ | 311 | +--+---+ | | 312 | | | | 313 | | | | 314 | | | | 315 | | | | 316 | | | | 317 | +--+---+ | | 318 ------------------+---+ESRP | | | 319 esrp-a@state.org | +------+ | | 320 | | | 321 | Call Fwd | | 322 | +-+-+---+ | 323 \ | | | / 324 `. | | | ,' 325 '-|-|-|-' ,-------. 326 Police | | | Fire ,' `. 327 +------------+ | +----+ / Emergency \ 328 ,-------. | | | | Services | 329 ,' `. | | | | Network | 330 / Emergency \ | Ambulance | | fire-town.org | 331 | Services | | | | | | 332 | Network | | +----+ | | +------+ | 333 |police-town.org| | ,-------. | +----+---+PSAP | | 334 | | | ,' `. | | +------+ | 335 | +------+ | | / Emergency \ | | | 336 | |PSAP +----+--+ | Services | | | , 337 | +------+ | | Network | | `~~~~~~~~~~~~~~~ 338 | | |medic-town.org | | 339 | , | | | 340 `~~~~~~~~~~~~~~~ | +------+ | | 341 | |PSAP +----+ + 342 | +------+ | 343 | | 344 | , 345 `~~~~~~~~~~~~~~~ 347 Figure 3: Example for Call Forwarding 349 3.4. Network-based Service URN Resolution 351 The IETF emergency services architecture also considers cases where 352 the resolution from the Service URN to the PSAP URI does not only 353 happen at the SIP UA itself but at intermediate SIP entities, such as 354 the user's VoIP provider. 356 Figure 4 shows this message exchange of the outgoing emergency call 357 and the incoming PSAP graphically. While the state information 358 stored at the VoIP provider is correct the state allocated at the SIP 359 UA is not. 361 ,-------. 362 ,' `. 363 / Emergency \ 364 | Services | 365 | Network | 366 |police-town.org| 367 | | 368 | +------+ | Invite to police.example.com 369 | |PSAP +<---+------------------------+ 370 | | +----+------------------+ ^ 371 | +------+ |Invite from | | 372 | ,police.example.com| | 373 `~~~~~~~~~~~~~~~ v | 374 +--------+ ++-----+-+ 375 | | query |VoIP | 376 | LoST |<-----------------------|Service | 377 | Server | police.example.com |Provider| 378 | |----------------------->| | 379 +--------+ +--------+ 380 | ^ 381 Invite| | Invite 382 from| | to 383 police.example.com| | urn:service:sos 384 V | 385 +-------+ 386 | SIP | 387 | UA | 388 | Alice | 389 +-------+ 391 Figure 4: Example for Network-based Service URN Resolution 393 3.5. PSTN Interworking 395 In case an emergency call enters the PSTN, as shown in Figure 5, 396 there is no guarantee that the callback some time later does leave 397 the same PSTN/VoIP gateway or that the same end point identifier is 398 used in the forward as well as in the backward direction making it 399 difficult to reliably detect PSAP callbacks. 401 +-----------+ 402 | PSTN |-------------+ 403 | Calltaker | | 404 | Bob |<--------+ | 405 +-----------+ | v 406 ------------------- 407 //// \\\\ +------------+ 408 | | |PSTN / VoIP | 409 | PSTN |---->|Gateway | 410 \\\\ //// | | 411 ------------------- +----+-------+ 412 ^ | 413 | | 414 +-------------+ | +--------+ 415 | | | |VoIP | 416 | PSTN / VoIP | +->|Service | 417 | Gateway | |Provider| 418 | |<------Invite----| Y | 419 +-------------+ +--------+ 420 | ^ 421 | | 422 Invite Invite 423 | | 424 V | 425 +-------+ 426 | SIP | 427 | UA | 428 | Alice | 429 +-------+ 431 Figure 5: Example for PSTN Interworking 433 Note: This scenario is considered outside the scope of this document. 434 The specified solution does not support this use case. 436 4. SIP PSAP Callback Indicator 438 4.1. General 440 This section defines a new header field value, called "psap- 441 callback", for the SIP Priority header field defined in [RFC3261]. 442 The value is used to inform SIP entities that the request is 443 associated with a PSAP callback SIP session. 445 4.2. Usage 447 SIP entities that receive the header field value within an initial 448 request for a SIP session can, depending on local policies, apply 449 PSAP callback specific procedures for the session or request. 451 The PSAP callback specific procedures may be applied by SIP-based 452 network entities and by the callee. The specific procedures taken 453 when receiving such a PSAP callback marked call, such as bypassing 454 services and barring procedures, are outside the scope of this 455 document. 457 4.3. Syntax 459 4.3.1. General 461 This section defines the ABNF for the new SIP Priority header field 462 value "psap-callback". 464 4.3.2. ABNF 466 priority-value /= "psap-callback" 468 Figure 6: ABNF 470 5. Security Considerations 472 5.1. Security Threat 474 The PSAP callback functionality described in this document allows 475 marked calls to bypass blacklists, ignore call forwarding procedures 476 and similar features to contact emergency callers and to raise their 477 attention. Regarding the latter aspect a callback, if understood by 478 the SIP UA would allow to override user interface configurations, 479 such as vibrate-only mode, to alert the caller of the incoming call. 481 5.2. Security Requirements 483 The requirement is to ensure that the mechanisms described in this 484 document can not be used for malicious purposes, including 485 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 5.3. Security Solution 495 Figure 7 shows the architecture that utilizes the identity of the 496 PSAP to decide whether a preferential treatment of callbacks should 497 be provided. To make this policy decision the identity of the PSAP 498 is compared with a whitelist of valid PSAPs available to the SIP 499 entity. The identity assurance in SIP can come in different forms, 500 such as SIP Identity [RFC4474] or with P-Asserted-Identity [RFC3325]. 501 The former technique relies on a cryptographic assurance and the 502 latter on a chain of trust. Also the usage of TLS between 503 neighboring SIP entities may provide useful identity information. 505 +----------+ 506 | List of |+ 507 | valid || 508 | PSAPs || 509 +----------+| 510 +----------+ 511 * 512 * whitelist 513 * 514 V 515 Incoming +----------+ Normal 516 SIP Msg | SIP |+ Treatment 517 -------------->| Entity ||======================> 518 + Identity | ||(if not in whitelist) 519 Info +----------+| 520 +----------+ 521 || 522 || 523 || Preferential 524 || Treatment 525 ++========================> 526 (if successfully verified) 528 Figure 7: Identity-based Authorization 530 An important aspect from a security point of view is the relationship 531 between the emergency services network (containing PSAPs) and the VSP 532 (assuming that the emergency call travels via the VSP and not 533 directly between the SIP UA and the PSAP). 535 If there is some form of relationship between the emergency services 536 operator and the VSP then the identification of a PSAP call back is 537 less problematic than in the case where the two entities have not 538 entered in some form of relationship that would allow the VSP to 539 verify whether the marked callback message indeed came from a 540 legitimate source. 542 The establishment of a whitelist with PSAP identities maybe be 543 operationally complex. When there is a local relationship between 544 the VSP/ASP and the PSAP then populating the whitelist is fairly 545 simple. For SIP UAs there is no need to maintain a list of PSAPs. 546 Instead SIP UAs are assumed to trust the correct processing of their 547 VSP/ASP, i.e., the VSP/ASP processes the PSAP callback marking and, 548 if it cannot be verified, the PSAP callback marking is removed. If 549 it is left untouched then the SIP UA should assume that it has been 550 verified successfully by the VSP/ASP and it should therefore be 551 obeyed. 553 6. IANA Considerations 555 This document adds the "psap-callback" value to the SIP Priority 556 header IANA registry allocated by [I-D.ietf-sipcore-priority]. The 557 semantic of the newly defined "psap-callback" value is defined in 558 Section 4. 560 7. Acknowledgements 562 We would like to thank members from the ECRIT working group, in 563 particular Brian Rosen, for their discussions around PSAP callbacks. 564 The working group discussed the topic of callbacks at their virtual 565 interim meeting in February 2010 and the following persons provided 566 valuable input: John Elwell, Bernard Aboba, Cullen Jennings, Keith 567 Drage, Marc Linsner, Roger Marshall, Dan Romascanu, Geoff Thompson, 568 Janet Gunn. 570 At IETF#81 a small group of people got to together to continue the 571 discussions started at the working group meeting to explore a GRUU- 572 based solution approach. Martin Thomson, Marc Linsner, Andrew Allen, 573 Brian Rosen, Martin Dolly, and Atle Monrad participated at this side- 574 meeting. 576 We would like to thank the following persons for their feedback on 577 the solution discussion in 2012: Paul Kyzivat, Martin Thomson, Robert 578 Sparks, Keith Drage, Brian Rosen, Roger Marshall, Martin Dolly, 579 Bernard Aboba, Andrew Allen, John-Luc Bakker, James Polk, John 580 Medland, Hadriel Kaplan, Kenneth Carlberg, Timothy Dwight, Janet Gunn 582 8. References 584 8.1. Normative References 586 [I-D.ietf-sipcore-priority] Roach, A., "IANA Registry for the 587 Session Initiation Protocol (SIP) 588 "Priority" Header Field", 589 draft-ietf-sipcore-priority-00 (work in 590 progress), December 2012. 592 [RFC2119] Bradner, S., "Key words for use in RFCs 593 to Indicate Requirement Levels", BCP 14, 594 RFC 2119, March 1997. 596 [RFC3261] Rosenberg, J., Schulzrinne, H., 597 Camarillo, G., Johnston, A., Peterson, 598 J., Sparks, R., Handley, M., and E. 599 Schooler, "SIP: Session Initiation 600 Protocol", RFC 3261, June 2002. 602 [RFC3325] Jennings, C., Peterson, J., and M. 603 Watson, "Private Extensions to the 604 Session Initiation Protocol (SIP) for 605 Asserted Identity within Trusted 606 Networks", RFC 3325, November 2002. 608 [RFC3966] Schulzrinne, H., "The tel URI for 609 Telephone Numbers", RFC 3966, 610 December 2004. 612 [RFC3969] Camarillo, G., "The Internet Assigned 613 Number Authority (IANA) Uniform Resource 614 Identifier (URI) Parameter Registry for 615 the Session Initiation Protocol (SIP)", 616 BCP 99, RFC 3969, December 2004. 618 [RFC4474] Peterson, J. and C. Jennings, 619 "Enhancements for Authenticated Identity 620 Management in the Session Initiation 621 Protocol (SIP)", RFC 4474, August 2006. 623 [RFC5627] Rosenberg, J., "Obtaining and Using 624 Globally Routable User Agent URIs 625 (GRUUs) in the Session Initiation 626 Protocol (SIP)", RFC 5627, October 2009. 628 8.2. Informative References 630 [I-D.ietf-ecrit-phonebcp] Rosen, B. and J. Polk, "Best Current 631 Practice for Communications Services in 632 support of Emergency Calling", 633 draft-ietf-ecrit-phonebcp-20 (work in 634 progress), September 2011. 636 [RFC4484] Peterson, J., Polk, J., Sicker, D., and 637 H. Tschofenig, "Trait-Based 638 Authorization Requirements for the 639 Session Initiation Protocol (SIP)", 640 RFC 4484, August 2006. 642 [RFC5012] Schulzrinne, H. and R. Marshall, 643 "Requirements for Emergency Context 644 Resolution with Internet Technologies", 645 RFC 5012, January 2008. 647 [RFC5031] Schulzrinne, H., "A Uniform Resource 648 Name (URN) for Emergency and Other Well- 649 Known Services", RFC 5031, January 2008. 651 [RFC5234] Crocker, D. and P. Overell, "Augmented 652 BNF for Syntax Specifications: ABNF", 653 STD 68, RFC 5234, January 2008. 655 [RFC6444] Schulzrinne, H., Liess, L., Tschofenig, 656 H., Stark, B., and A. Kuett, "Location 657 Hiding: Problem Statement and 658 Requirements", RFC 6444, January 2012. 660 Authors' Addresses 662 Henning Schulzrinne 663 Columbia University 664 Department of Computer Science 665 450 Computer Science Building 666 New York, NY 10027 667 US 669 Phone: +1 212 939 7004 670 EMail: hgs+ecrit@cs.columbia.edu 671 URI: http://www.cs.columbia.edu 673 Hannes Tschofenig 674 Nokia Siemens Networks 675 Linnoitustie 6 676 Espoo 02600 677 Finland 679 Phone: +358 (50) 4871445 680 EMail: Hannes.Tschofenig@gmx.net 681 URI: http://www.tschofenig.priv.at 683 Christer Holmberg 684 Ericsson 685 Hirsalantie 11 686 Jorvas 02420 687 Finland 689 EMail: christer.holmberg@ericsson.com 691 Milan Patel 692 InterDigital Communications 694 EMail: Milan.Patel@interdigital.com