idnits 2.17.1 draft-ietf-martini-gin-13.txt: Checking boilerplate required by RFC 5378 and the IETF Trust (see https://trustee.ietf.org/license-info): ---------------------------------------------------------------------------- No issues found here. Checking nits according to https://www.ietf.org/id-info/1id-guidelines.txt: ---------------------------------------------------------------------------- No issues found here. Checking nits according to https://www.ietf.org/id-info/checklist : ---------------------------------------------------------------------------- -- The draft header indicates that this document updates RFC3680, but the abstract doesn't seem to mention this, which it should. Miscellaneous warnings: ---------------------------------------------------------------------------- == The copyright year in the IETF Trust and authors Copyright Line does not match the current year (Using the creation date from RFC3680, updated by this document, for RFC5378 checks: 2002-10-31) -- The document seems to lack a disclaimer for pre-RFC5378 work, but may have content which was first submitted before 10 November 2008. If you have contacted all the original authors and they are all willing to grant the BCP78 rights to the IETF Trust, then this is fine, and you can ignore this comment. If not, you may need to add the pre-RFC5378 disclaimer. (See the Legal Provisions document at https://trustee.ietf.org/license-info for more information.) -- The document date (January 20, 2011) is 4844 days in the past. Is this intentional? Checking references for intended status: Proposed Standard ---------------------------------------------------------------------------- (See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) ** Downref: Normative reference to an Informational RFC: RFC 2104 (ref. '1') ** Obsolete normative reference: RFC 3265 (ref. '5') (Obsoleted by RFC 6665) -- Obsolete informational reference (is this intentional?): RFC 5246 (ref. '18') (Obsoleted by RFC 8446) Summary: 2 errors (**), 0 flaws (~~), 1 warning (==), 4 comments (--). Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 MARTINI WG A. B. Roach 3 Internet-Draft Tekelec 4 Updates: 3680 (if approved) January 20, 2011 5 Intended status: Standards Track 6 Expires: July 24, 2011 8 Registration for Multiple Phone Numbers in the Session Initiation 9 Protocol (SIP) 10 draft-ietf-martini-gin-13 12 Abstract 14 This document defines a mechanism by which a Session Initiation 15 Protocol (SIP) server acting as a traditional Private Branch Exchange 16 (SIP-PBX) can register with a SIP Service Provider (SSP) to receive 17 phone calls for SIP User Agents (UAs). In order to function 18 properly, this mechanism requires that each of the Addresses of 19 Record (AORs) registered in bulk map to a unique set of contacts. 20 This requirement is satisfied by AORs representing phone numbers 21 regardless of the domain, since phone numbers are fully qualified and 22 globally unique. This document therefore focuses on this use case. 24 Status of this Memo 26 This Internet-Draft is submitted in full conformance with the 27 provisions of BCP 78 and BCP 79. 29 Internet-Drafts are working documents of the Internet Engineering 30 Task Force (IETF). Note that other groups may also distribute 31 working documents as Internet-Drafts. The list of current Internet- 32 Drafts is at http://datatracker.ietf.org/drafts/current/. 34 Internet-Drafts are draft documents valid for a maximum of six months 35 and may be updated, replaced, or obsoleted by other documents at any 36 time. It is inappropriate to use Internet-Drafts as reference 37 material or to cite them other than as "work in progress." 39 This Internet-Draft will expire on July 24, 2011. 41 Copyright Notice 43 Copyright (c) 2011 IETF Trust and the persons identified as the 44 document authors. All rights reserved. 46 This document is subject to BCP 78 and the IETF Trust's Legal 47 Provisions Relating to IETF Documents 48 (http://trustee.ietf.org/license-info) in effect on the date of 49 publication of this document. Please review these documents 50 carefully, as they describe your rights and restrictions with respect 51 to this document. Code Components extracted from this document must 52 include Simplified BSD License text as described in Section 4.e of 53 the Trust Legal Provisions and are provided without warranty as 54 described in the Simplified BSD License. 56 Table of Contents 58 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 59 2. Constraints . . . . . . . . . . . . . . . . . . . . . . . . . 3 60 3. Terminology and Conventions . . . . . . . . . . . . . . . . . 4 61 4. Mechanism Overview . . . . . . . . . . . . . . . . . . . . . . 5 62 5. Registering for Multiple Phone Numbers . . . . . . . . . . . . 5 63 5.1. SIP-PBX Behavior . . . . . . . . . . . . . . . . . . . . . 5 64 5.2. Registrar Behavior . . . . . . . . . . . . . . . . . . . . 6 65 5.3. SIP URI "user" Parameter Handling . . . . . . . . . . . . 8 66 6. SSP Processing of Inbound Requests . . . . . . . . . . . . . . 8 67 7. Interaction with Other Mechanisms . . . . . . . . . . . . . . 9 68 7.1. Globally Routable User-Agent URIs (GRUU) . . . . . . . . . 9 69 7.1.1. Public GRUUs . . . . . . . . . . . . . . . . . . . . . 9 70 7.1.2. Temporary GRUUs . . . . . . . . . . . . . . . . . . . 11 71 7.2. Registration Event Package . . . . . . . . . . . . . . . . 15 72 7.2.1. SIP-PBX Aggregate Registration State . . . . . . . . . 16 73 7.2.2. Individual AOR Registration State . . . . . . . . . . 16 74 7.3. Client-Initiated (Outbound) Connections . . . . . . . . . 18 75 7.4. Non-Adjacent Contact Registration (Path) and Service 76 Route Discovery . . . . . . . . . . . . . . . . . . . . . 18 77 8. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 20 78 8.1. Usage Scenario: Basic Registration . . . . . . . . . . . . 20 79 8.2. Usage Scenario: Using Path to Control Request URI . . . . 22 80 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 24 81 9.1. New SIP Option Tag . . . . . . . . . . . . . . . . . . . . 24 82 9.2. New SIP URI Parameters . . . . . . . . . . . . . . . . . . 24 83 9.2.1. 'bnc' SIP URI parameter . . . . . . . . . . . . . . . 25 84 9.2.2. 'sg' SIP URI parameter . . . . . . . . . . . . . . . . 25 85 9.3. New SIP Header Field Parameter . . . . . . . . . . . . . . 25 86 10. Security Considerations . . . . . . . . . . . . . . . . . . . 25 87 11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 27 88 12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 28 89 12.1. Normative References . . . . . . . . . . . . . . . . . . . 28 90 12.2. Informative References . . . . . . . . . . . . . . . . . . 28 91 Appendix A. Requirements Analysis . . . . . . . . . . . . . . . . 30 92 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 34 94 1. Introduction 96 The Session Initiation Protocol (SIP) is an application-layer control 97 (signaling) protocol for creating, modifying, and terminating 98 sessions with one or more participants. One of SIP's primary 99 functions is providing rendezvous between users. By design, this 100 rendezvous has been provided through a combination of the server 101 look-up procedures defined in RFC 3263 [4], and the registrar 102 procedures described in RFC 3261 [3]. 104 The intention of the original protocol design was that any user's AOR 105 (Address of Record) would be handled by the authority indicated by 106 the hostport portion of the AOR. The users would register individual 107 reachability information with this authority, which would then route 108 incoming requests accordingly. 110 In actual deployments, some SIP servers have been deployed in 111 architectures that, for various reasons, have requirements to provide 112 dynamic routing information for large blocks of AORs, where all of 113 the AORs in the block were to be handled by the same server. For 114 purposes of efficiency, many of these deployments do not wish to 115 maintain separate registrations for each of the AORs in the block. 116 This leads to the desire for an alternate mechanism for providing 117 dynamic routing information for blocks of AORs. 119 Although the use of SIP REGISTER request messages to update 120 reachability information for multiple users simultaneously is 121 somewhat beyond the original semantics defined for REGISTER requests 122 by RFC 3261 [3], this approach has seen significant deployment in 123 certain environments. In particular, deployments in which small to 124 medium SIP-PBX servers are addressed using E.164 numbers have used 125 this mechanism to avoid the need to maintain DNS entries or static IP 126 addresses for the SIP-PBX servers. 128 In recognition of the momentum that REGISTER-based approaches have 129 seen in deployments, this document defines a REGISTER-based approach. 130 Since E.164-addressed UAs are very common today in SIP-PBX 131 environments, and since SIP URIs in which the user portion is an 132 E.164 number are always globally unique regardless of the domain, 133 this document focuses on registration of SIP URIs in which the user 134 portion is an E.164 number. 136 2. Constraints 138 Within the problem space that has been established for this work, 139 several constraints shape our solution. These are defined in the 140 MARTINI requirements document [22], and analyzed in Appendix A. In 141 terms of impact to the solution at hand, the following two 142 constraints have the most profound effect: (1) The SIP-PBX cannot be 143 assumed to be assigned a static IP address; and (2) No DNS entry can 144 be relied upon to consistently resolve to the IP address of the SIP- 145 PBX. 147 3. Terminology and Conventions 149 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 150 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 151 document are to be interpreted as described in RFC 2119 [2]. 153 Further, the term "SSP" is meant as an acronym for a "SIP Service 154 Provider," while the term "SIP-PBX" is used to indicate a SIP Private 155 Branch Exchange. 157 Indented portions of the document, such as this one, form non- 158 normative, explanatory sections of the document. 160 Although SIP is a text-based protocol, some of the examples in this 161 document cannot be unambiguously rendered without additional markup 162 due to the constraints placed on the formatting of RFCs. This 163 document uses the markup convention established in RFC 164 4475 [17] to avoid ambiguity and meet the RFC layout requirements. 165 For the sake of completeness, the text defining this markup from 166 Section 2.1 of RFC 4475 [17] is reproduced in its entirety below: 168 Several of these examples contain unfolded lines longer than 72 169 characters. These are captured between tags. The 170 single unfolded line is reconstructed by directly concatenating 171 all lines appearing between the tags (discarding any line feeds or 172 carriage returns). There will be no whitespace at the end of 173 lines. Any whitespace appearing at a fold-point will appear at 174 the beginning of a line. 176 The following represent the same string of bits: 178 Header-name: first value, reallylongsecondvalue, third value 180 181 Header-name: first value, 182 reallylongsecondvalue 183 , third value 184 185 186 Header-name: first value, 187 reallylong 188 second 189 value, 190 third value 191 193 Note that this is NOT SIP header-line folding, where different 194 strings of bits have equivalent meaning. 196 4. Mechanism Overview 198 The overall mechanism is achieved using a REGISTER request with a 199 specially-formatted Contact URI. This document also defines an 200 option tag that can be used to ensure a registrar and any 201 intermediaries understand the mechanism described herein. 203 The Contact URI itself is tagged with a URI parameter to indicate 204 that it actually represents multiple phone-number-associated 205 contacts. 207 We also define some lightweight extensions to the Globally Routable 208 UA URIs (GRUU) mechanism defined by RFC 5627 [20] to allow the use of 209 public and temporary GRUUs assigned by the SSP. 211 Aside from these extensions, the REGISTER request itself is processed 212 by a registrar in the same way as normal registrations: by updating 213 its location service with additional AOR-to-Contact bindings. 215 Note that the list of AORs associated with a SIP-PBX is a matter of 216 local provisioning at the SSP and at the SIP-PBX. The mechanism 217 defined in this document does not provide any means to detect or 218 recover from provisioning mismatches (although the registration event 219 package can be used as a standardized means for auditing such AORs; 220 see Section 7.2.1). 222 5. Registering for Multiple Phone Numbers 224 5.1. SIP-PBX Behavior 226 To register for multiple AORs, the SIP-PBX sends a REGISTER request 227 to the SSP. This REGISTER request varies from a typical REGISTER 228 request in two important ways. First, it MUST contain an option tag 229 of "gin" in both a "Require" header field and a "Proxy-Require" 230 header field. (The option tag "gin" is an acronym for "generate 231 implicit numbers".) Second, in at least one "Contact" header field, 232 it MUST include a Contact URI that contains the URI parameter "bnc" 233 (which stands for "bulk number contact"), and no user portion (hence 234 no "@" symbol). A URI with a "bnc" parameter MUST NOT contain a user 235 portion. Except for the SIP URI "user" parameter, this URI MAY 236 contain any other parameters that the SIP-PBX desires. These 237 parameters will be echoed back by the SSP in any requests bound for 238 the SIP-PBX. 240 Because of the constraints discussed in Section 2, the host portion 241 of the Contact URI will generally contain an IP address, although 242 nothing in this mechanism enforces or relies upon that fact. If the 243 SIP-PBX operator chooses to maintain DNS entries that resolve to the 244 IP address of his SIP-PBX via RFC 3263 resolution procedures, then 245 this mechanism works just fine with domain names in the Contact 246 header field. 248 The "bnc" URI parameter indicates that special interpretation of the 249 Contact URI is necessary: instead of indicating the insertion of a 250 single Contact URI into the location service, it indicates that 251 multiple URIs (one for each associated AOR) should be inserted. 253 Any SIP-PBX implementing the registration mechanism defined in this 254 document MUST also support the Path mechanism defined by RFC 3327 255 [10], and MUST include a 'path' option-tag in the Supported header 256 field of the REGISTER request (which is a stronger requirement than 257 imposed by the Path mechanism itself). This behavior is necessary 258 because proxies between the SIP-PBX and the Registrar may need to 259 insert Path header field values in the REGISTER request for this 260 document's mechanism to function properly, and per RFC 3327 [10], 261 they can only do so if the User Agent Client (UAC) inserted the 262 option-tag in the Supported header field. In accordance with the 263 procedures defined in RFC 3327 [10], the SIP-PBX is allowed to ignore 264 the Path header fields returned in the REGISTER response. 266 5.2. Registrar Behavior 268 The registrar, upon receipt of a REGISTER request containing at least 269 one Contact header field with a "bnc" parameter will use the value in 270 the "To" header field to identify the SIP-PBX for which registration 271 is being requested. It then authenticates the SIP-PBX (using, e.g., 272 SIP Digest authentication, mutual TLS [18], or some other 273 authentication mechanism). After the SIP-PBX is authenticated, the 274 registrar updates its location service with a unique AOR-to-Contact 275 mapping for each of the AORs associated with the SIP-PBX. 276 Semantically, each of these mappings will be treated as a unique row 277 in the location service. The actual implementation may, of course, 278 perform internal optimizations to reduce the amount of memory used to 279 store such information. 281 For each of these unique rows, the AOR will be in the format that the 282 SSP expects to receive from external parties (e.g. 283 "sip:+12145550102@ssp.example.com"), and the corresponding Contact 284 will be formed by adding to the REGISTER request's Contact URI a user 285 portion containing the fully-qualified, E.164-formatted number 286 (including the preceding "+" symbol) and removing the "bnc" 287 parameter. Aside from the initial "+" symbol, this E.164-formatted 288 number MUST consist exclusively of digits from 0 through 9, and 289 explicitly MUST NOT contain any visual separator symbols (e.g., "-", 290 ".", "(", or ")"). For example, if the "Contact" header field 291 contains the URI , then the Contact value 292 associated with the aforementioned AOR will be 293 . 295 Although the SSP treats this registration as a number of discrete 296 rows for the purpose of re-targeting incoming requests, the renewal, 297 expiration, and removal of these rows is bound to the registered 298 contact. In particular, this means that REGISTER requests that 299 attempt to de-register a single AOR that has been implicitly 300 registered MUST NOT remove that AOR from the bulk registration. In 301 this circumstance, the registrar simply acts as if the UA attempted 302 to unregister a contact that wasn't actually registered (e.g., return 303 the list of presently registered contacts in a success response). A 304 further implication of this property is that an individual extension 305 that is implicitly registered may also be explicitly registered using 306 a normal, non-bulk registration (subject to SSP policy). If such a 307 registration exists, it is refreshed independently of the bulk 308 registration, and is not removed when the bulk registration is 309 removed. 311 A registrar that receives a REGISTER request containing a Contact URI 312 with both a "bnc" parameter and a user portion MUST NOT send a 200- 313 class (success) response. If no other error is applicable, the 314 registrar can use a 400 (Bad Request) response to indicate this error 315 condition. 317 Note that the preceding paragraph is talking about the user 318 portion of a URI: 320 sip:+12145550100@example.com 321 ^^^^^^^^^^^^ 323 A Registrar compliant with this document MUST support the Path 324 mechanism defined in RFC 3327 [10]. The rationale for support of 325 this mechanism is given in section Section 5.1. 327 Aside from the "bnc" parameter, all URI parameters present on the 328 "Contact" URI in the REGISTER request MUST be copied to the Contact 329 value stored in the location service. 331 If the SSP servers perform processing based on User Agent 332 Capabilities (as defined in RFC 3840 [13]), they will treat any 333 feature tags present on a Contact header field with a "bnc" parameter 334 in its URI as applicable to all of the resulting AOR-to-Contact 335 mappings. Similarly, any option tags present on the REGISTER request 336 that indicate special handling for any subsequent requests are also 337 applicable to all of the AOR-to-Contact mappings. 339 5.3. SIP URI "user" Parameter Handling 341 This document does not modify the behavior specified in RFC 3261 [3] 342 for inclusion of the "user" parameter on request URIs. However, to 343 avoid any ambiguity in handling at the SIP-PBX, the following 344 normative behavior is imposed on its interactions with the SSP. 346 When a SIP-PBX registers with an SSP using a contact containing a 347 "bnc" parameter, that contact MUST NOT include a "user" parameter. A 348 registrar that receives a REGISTER request containing a Contact URI 349 with both a "bnc" parameter and a "user" parameter MUST NOT send a 350 200-class (success) response. If no other error is applicable, the 351 registrar can use a 400 (Bad Request) response to indicate this error 352 condition. 354 Note that the preceding paragraph is talking about the "user" 355 parameter of a URI: 357 sip:+12145550100@example.com;user=phone 358 ^^^^^^^^^^ 360 When a SIP-PBX receives a request from an SSP, and the Request-URI 361 contains a user portion corresponding to an AOR registered using a 362 contact containing a "bnc" parameter, then the SIP-PBX MUST NOT 363 reject the request (or otherwise cause the request to fail) due to 364 the absence, presence, or value of a "user" parameter on the Request- 365 URI. 367 6. SSP Processing of Inbound Requests 369 In general, after processing the AOR-to-Contact mapping described in 370 the preceding section, the SSP Proxy/Registrar (or equivalent entity) 371 performs traditional Proxy/Registrar behavior, based on the mapping. 372 For any inbound SIP requests whose AOR indicates an E.164 number 373 assigned to one of the SSP's customers, this will generally involve 374 setting the target set to the registered contacts associated with 375 that AOR, and performing request forwarding as described in section 376 16.6 of RFC 3261 [3]. An SSP using the mechanism defined in this 377 document MUST perform such processing for inbound INVITE requests and 378 SUBSCRIBE requests to the "reg" event package (see Section 7.2.2), 379 and SHOULD perform such processing for all other method types, 380 including unrecognized SIP methods. 382 7. Interaction with Other Mechanisms 384 The following sections describe the means by which this mechanism 385 interacts with relevant REGISTER-related extensions currently defined 386 by the IETF. 388 7.1. Globally Routable User-Agent URIs (GRUU) 390 To enable advanced services to work with UAs behind a SIP-PBX, it is 391 important that the GRUU mechanism defined by RFC 5627 [20] work 392 correctly with the mechanism defined by this document -- that is, 393 that User Agents served by the SIP-PBX can acquire and use GRUUs for 394 their own use. 396 7.1.1. Public GRUUs 398 Support of public GRUUs is OPTIONAL in SSPs and SIP-PBXes. 400 When a SIP-PBX registers a Bulk Number Contact (a Contact with a 401 "bnc" parameter), and also invokes GRUU procedures for that Contact 402 during registration, then the SSP will assign a public GRUU to the 403 SIP-PBX in the normal fashion. Because the URI being registered 404 contains a "bnc" parameter, the GRUU will also contain a "bnc" 405 parameter. In particular, this means that the GRUU will not contain 406 a user portion. 408 When a UA registers a contact with the SIP-PBX using GRUU procedures, 409 the SIP-PBX provides to the UA a public GRUU formed by adding an "sg" 410 parameter to the GRUU parameter it received from the SSP. This "sg" 411 parameter contains a disambiguation token that the SIP-PBX can use to 412 route inbound requests to the proper UA. 414 So, for example, when the SIP-PBX registers with the following 415 contact header field: 417 Contact: ; 418 +sip.instance="" 420 Then the SSP may choose to respond with a Contact header field that 421 looks like this: 423 424 Contact: ; 425 pub-gruu="sip:ssp.example.com;bnc;gr=urn: 426 uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6"; 427 +sip.instance="" 428 ;expires=7200 429 431 When its own UAs register using GRUU procedures, the SIP-PBX can then 432 add whatever device identifier it feels appropriate in an "sg" 433 parameter, and present this value to its own UAs. For example, 434 assume the UA associated with the AOR "+12145550102" sent the 435 following Contact header field in its REGISTER request: 437 Contact: ; 438 +sip.instance="" 440 The SIP-PBX will add an "sg" parameter to the pub-gruu it received 441 from the SSP with a token that uniquely identifies the device 442 (possibly the URN itself; possibly some other identifier); insert a 443 user portion containing the fully-qualified E.164 number associated 444 with the UA; and return the result to the UA as its public GRUU. The 445 resulting Contact header field sent from the SIP-PBX to the 446 registering UA would look something like this: 448 449 Contact: ; 450 pub-gruu="sip:+12145550102@ssp.example.com;gr=urn: 451 uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6;sg=00:05:03:5e:70:a6"; 452 +sip.instance="" 453 ;expires=3600 454 456 When an incoming request arrives at the SSP for a GRUU corresponding 457 to a bulk number contact ("bnc"), the SSP performs slightly different 458 processing for the GRUU than it would for a URI without a "bnc" 459 parameter. When the GRUU is re-targeted to the registered bulk 460 number contact, the SSP MUST copy the "sg" parameter from the GRUU to 461 the new target. The SIP-PBX can then use this "sg" parameter to 462 determine which user agent the request should be routed to. For 463 example, the first line of an INVITE request that has been re- 464 targeted to the SIP-PBX for the UA shown above would look like this: 466 INVITE sip:+12145550102@198.51.100.3;sg=00:05:03:5e:70:a6 SIP/2.0 468 7.1.2. Temporary GRUUs 470 In order to provide support for privacy, the SSP SHOULD implement the 471 temporary GRUU mechanism described in this section. Reasons for not 472 doing so would include systems with an alternative privacy mechanism 473 which maintains the integrity of public GRUUs (i.e., if public GRUUs 474 are anonymized then the anonymizer function would need to be capable 475 of providing as the anonymized URI a globally routable URI that 476 routes back only to the target identified by the original public 477 GRUU). 479 Temporary GRUUs are used to provide anonymity for the party creating 480 and sharing the GRUU. Being able to correlate two temporary GRUUs as 481 having originated from behind the same SIP-PBX violates this 482 principle of anonymity. Consequently, rather than relying upon a 483 single, invariant identifier for the SIP-PBX in its UA's temporary 484 GRUUs, we define a mechanism whereby the SSP provides the SIP-PBX 485 with sufficient information for the SIP-PBX to mint unique temporary 486 GRUUs. These GRUUs have the property that the SSP can correlate them 487 to the proper SIP-PBX, but no other party can do so. To achieve this 488 goal, we use a slight modification of the procedure described in 489 appendix A.2 of RFC 5627 [20]. 491 The SIP-PBX needs to be able to construct a temp-gruu in a way that 492 the SSP can decode. In order to ensure that the SSP can decode 493 GRUUs, we need to standardize the algorithm for creation of temp- 494 gruus at the SIP-PBX. This allows the SSP to reverse the algorithm 495 to identify the registration entry that corresponds to the GRUU. 497 It is equally important that no party other than the SSP is capable 498 of decoding a temporary GRUU, including other SIP-PBXes serviced by 499 the SSP. To achieve this property, an SSP that supports temporary 500 GRUUs MUST create and store an asymmetric key pair, {K_e1,K_e2}. 501 K_e1 is kept secret by the SSP, while K_e2 is shared with the SIP- 502 PBXes via provisioning. 504 All base64 encoding discussed in the following sections MUST use the 505 character set and encoding defined in Section 4 of RFC 4648 [8], 506 except that any trailing "=" characters are discarded on encoding, 507 and added as necessary to decode. 509 The following sections make use of the term "HMAC-SHA256-80" to 510 describe a particular HMAC algorithm. In this document, 511 HMAC-SHA256-80 is defined to mean the application of the SHA-256 [24] 512 secure hashing algorithm, and truncating the results to 80 bits by 513 discarding the trailing (least significant) bits. 515 7.1.2.1. Generation of temp-gruu-cookie by the SSP 517 An SSP that supports temporary GRUUs MUST include a "temp-gruu- 518 cookie" parameter on all Contact header fields containing a "bnc" 519 parameter in a 200-class REGISTER response. This "temp-gruu-cookie" 520 MUST have the following properties: 522 1. It can be used by the SSP to uniquely identify the registration 523 to which it corresponds. 524 2. It is encoded using base64. This allows the SIP-PBX to decode it 525 into as compact a form as possible for use in its calculations. 526 3. It is of a fixed length. This allows for extraction of it once 527 the SIP-PBX has concatenated a distinguisher onto it. 528 4. The temp-gruu-cookie MUST NOT be forgeable by any party. In 529 other words, the SSP needs to be able to examine the cookie and 530 validate that it was generated by the SSP. 531 5. The temp-gruu-cookie MUST be invariant during the course of a 532 registration, including any refreshes to that registration. This 533 property is important, as it allows the SIP-PBX to examine the 534 temp-gruu-cookie to determine whether the temp-gruus it has 535 issued to its UAs are still valid. 537 The above properties can be met using the following algorithm, which 538 is non-normative. Implementors may chose to implement any algorithm 539 of their choosing for generation of the temp-gruu-cookie, as long as 540 it fulfills the five properties listed above. 542 The registrar maintains a counter, I. This counter is 48 bits 543 long, and initialized to zero. This counter is persistently 544 stored, using a back-end database or similar technique. When the 545 registrar creates the first temporary GRUU for a particular SIP- 546 PBX and instance ID (as defined by [20]), the registrar notes the 547 current value of the counter, I_i, and increments the counter in 548 the database. The registrar then maps I_i to the Contact and 549 instance ID using the database, a persistent hash-map or similar 550 technology. If the registration expires such that there are no 551 longer any contacts with that particular instance ID bound to the 552 GRUU, the registrar removes the mapping. Similarly, if the 553 temporary GRUUs are invalidated due to a change in Call-ID, the 554 registrar removes the current mapping from I_i to the AOR and 555 instance ID, notes the current value of the counter I_j, and 556 stores a mapping from I_j to the contact containing a "bnc" 557 parameter and instance ID. Based on these rules, the hash-map 558 will contain a single mapping for each contact containing a "bnc" 559 parameter and instance ID for which there is a currently valid 560 registration. 562 The registrar maintains a symmetric key SK_a, which is regenerated 563 every time the counter rolls over or is is reset. When the 564 counter rolls over or is reset, the registrar remembers the old 565 value of SK_a for a while. To generate a temp-gruu-cookie, the 566 registrar computes: 568 SA = HMAC(SK_a, I_i) 569 temp-gruu-cookie = base64enc(I_i || SA) 571 where || denotes concatenation. "HMAC" represents any suitably 572 strong HMAC algorithm; see RFC 2104 [1] for a discussion of HMAC 573 algorithms. One suitable HMAC algorithm for this purpose is HMAC- 574 SHA256-80. 576 7.1.2.2. Generation of temp-gruu by the SIP-PBX 578 According to RFC5627 [20] section 3.2, every registration refresh 579 generates a new temp-gruu that is valid for as long as the contact 580 remains registered. This property is both critical for the privacy 581 properties of temp-gruu and is expected by UAs that implement the 582 temp-gruu procedures. Nothing in this document should be construed 583 as changing this fundamental temp-gruu property in any way. SIP- 584 PBXes that implement temporary GRUUs MUST generate a new temp-gruu 585 according to the procedures in this section for every registration or 586 registration refresh from GRUU-supporting UAs attached to the SIP- 587 PBX. 589 Similarly, if the registration that a SIP-PBX has with its SSP 590 expires or is terminated, then the temp-gruu cookie it maintains with 591 the SSP will change. This change will invalidate all the temp-gruus 592 the SIP-PBX has issued to its UAs. If the SIP-PBX tracks this 593 information (e.g., to include elements in registration 594 event bodies, as described in RFC 5628 [9]), it can determine that 595 previously issued temp-gruus are invalid by observing a change in the 596 temp-gruu-cookie provided to it by the SSP. 598 A SIP-PBX that issues temporary GRUUs to its UAs MUST maintain an 599 HMAC key, PK_a. This value is used to validate that incoming GRUUs 600 were generated by the SIP-PBX. 602 To generate a new temporary GRUU for use by its own UAs, the SIP-PBX 603 MUST generate a random distinguisher value D. The length of this 604 value is up to implementors, but MUST be long enough to prevent 605 collisions among all the temporary GRUUs issued by the SIP-PBX. A 606 size of 80 bits or longer is RECOMMENDED. See RFC 4086 [16] for 607 further considerations on the generation of random numbers in a 608 security context. After generating the distinguisher D, the SIP-PBX 609 then MUST calculate: 611 M = base64dec(SSP-cookie) || D 612 E = RSA-Encrypt(K_e2, M) 613 PA = HMAC(PK_a, E) 615 Temp-Gruu-userpart = "tgruu." || base64(E) || "." || base64(PA) 617 where || denotes concatenation. "HMAC" represents any suitably 618 strong HMAC algorithm; see RFC 2104 [1] for a discussion of HMAC 619 algorithms. One suitable HMAC algorithm for this purpose is HMAC- 620 SHA256-80. 622 Finally, the SIP-PBX adds a "gr" parameter to the temporary GRUU that 623 can be used to uniquely identify the UA registration record to which 624 the GRUU corresponds. The means of generation of the "gr" parameter 625 are left to the implementor, as long as they satisfy the properties 626 of a GRUU as described in RFC 5627 [20]. 628 One valid approach for generation of the "gr" parameter is 629 calculation of "E" and "A" as described in Appendix A.2 of RFC 630 5627 [20], and forming the "gr" parameter as: 632 gr = base64enc(E) || base64enc(A) 634 Using this procedure may result in a temporary GRUU returned to the 635 registering UA by the SIP-PBX that looks similar to this: 637 638 Contact: 639 ;temp-gruu="sip:tgruu.MQyaRiLEd78RtaWkcP7N8Q.5qVbsasdo2pkKw@ 640 ssp.example.com;gr=YZGSCjKD42ccxO08pA7HwAM4XNDIlMSL0HlA" 641 ;+sip.instance="" 642 ;expires=3600 643 645 7.1.2.3. Decoding of temp-gruu by the SSP 647 When the SSP proxy receives a request in which the user part begins 648 with "tgruu.", it extracts the remaining portion, and splits it at 649 the "." character into E' and PA'. It discards PA'. It then 650 computes E by performing a base64 decode of E'. Next, it computes: 652 M = RSA-Decrypt(K_e1, E) 654 The SSP proxy extracts the fixed-length temp-gruu-cookie information 655 from the beginning of this M, and discards the remainder (which will 656 be the distinguisher added by the SIP-PBX). It then validates this 657 temp-gruu-cookie. If valid, it uses it to locate the corresponding 658 SIP-PBX registration record, and routes the message appropriately. 660 If the non-normative, exemplary algorithm described in 661 Section 7.1.2.1 is used to generate the temp-gruu-cookie, then 662 this identification is performed by splitting the temp-gruu-cookie 663 information into its 48-bit counter I and 80-bit HMAC. It 664 validates that the HMAC matches the counter I, and then uses 665 counter I to locate the SIP-PBX registration record in its map. 666 If the counter has rolled over or reset, this computation is 667 performed with the current and previous SK_a. 669 7.1.2.4. Decoding of temp-gruu by the SIP-PBX 671 When the SIP-PBX receives a request in which the user part begins 672 with "tgruu.", it extracts the remaining portion, and splits it at 673 the "." character into E' and PA'. It then computes E and PA by 674 performing a base64 decode of E' and PA' respectively. Next, it 675 computes: 677 PAc = HMAC(PK_a, E) 679 where HMAC is the HMAC algorithm used for the steps in 680 Section 7.1.2.2. If this computed value for PAc does not match the 681 value of PA extracted from the GRUU, then the GRUU is rejected as 682 invalid. 684 The SIP-PBX then uses the value of the "gr" parameter to locate the 685 UA registration to which the GRUU corresponds, and routes the message 686 accordingly. 688 7.2. Registration Event Package 690 Neither the SSP nor the SIP-PBX is required to support the 691 Registration event package defined by RFC 3680 [12]. However, if 692 they do support the Registration event package, they MUST conform to 693 the behavior described in this section and its subsections. 695 As this mechanism inherently deals with REGISTER transaction 696 behavior, it is imperative to consider its impact on the Registration 697 Event Package defined by RFC 3680 [12]. In practice, there will be 698 two main use cases for subscribing to registration data: learning 699 about the overall registration state for the SIP-PBX, and learning 700 about the registration state for a single SIP-PBX AOR. 702 7.2.1. SIP-PBX Aggregate Registration State 704 If the SIP-PBX (or another interested and authorized party) wishes to 705 monitor or audit the registration state for all of the AORs currently 706 registered to that SIP-PBX, it can subscribe to the SIP registration 707 event package at the SIP-PBX's main URI -- that is, the URI used in 708 the "To" header field of the REGISTER request. 710 The NOTIFY messages for such a subscription will contain a body that 711 contains one record for each AOR associated with the SIP-PBX. The 712 AORs will be in the format expected to be received by the SSP (e.g., 713 "sip:+12145550105@ssp.example.com"), and the Contacts will correspond 714 to the mapped Contact created by the registration (e.g., 715 "sip:+12145550105@98.51.100.3"). 717 In particular, the "bnc" parameter is forbidden from appearing in the 718 body of a reg-event NOTIFY request unless the subscriber has 719 indicated knowledge of the semantics of the "bnc" parameter. The 720 means for indicating this support are out of scope of this document. 722 Because the SSP does not necessarily know which GRUUs have been 723 issued by the SIP-PBX to its associated UAs, these records will not 724 generally the contain or elements defined in 725 RFC 5628 [9]. This information can be learned, if necessary, by 726 subscribing to the individual AOR registration state, as described in 727 Section 7.2.2. 729 7.2.2. Individual AOR Registration State 731 As described in Section 6, the SSP will generally retarget all 732 requests addressed to an AOR owned by a SIP-PBX to that SIP-PBX 733 according to the mapping established at registration time. Although 734 policy at the SSP may override this generally expected behavior, 735 proper behavior of the registration event package requires that all 736 "reg" event SUBSCRIBE requests are processed by the SIP-PBX. As a 737 consequence, the requirements on an SSP for processing registration 738 event package SUBSCRIBE requests are not left to policy. 740 If the SSP receives a SUBSCRIBE request for the registration event 741 package with a Request-URI that indicates an AOR registered via the 742 "Bulk Number Contact" mechanism defined in this document, then the 743 SSP MUST proxy that SUBSCRIBE to the SIP-PBX in the same way that it 744 would proxy an INVITE bound for that AOR, unless the SSP has and can 745 maintain a copy of complete, accurate, and up-to-date information 746 from the SIP-PBX (e.g., through an active back-end subscription). 748 If the Request-URI in a SUBSCRIBE request for the registration event 749 package indicates a contact that is registered by more than one SIP- 750 PBX, then the SSP proxy will fork the SUBSCRIBE request to all the 751 applicable SIP-PBXes. Similarly, if the Request-URI corresponds to a 752 contact that is both implicitly registered by a SIP-PBX and 753 explicitly registered directly with the SSP proxy, then the SSP proxy 754 will semantically fork the SUBSCRIBE request to the applicable SIP- 755 PBX or SIP-PBXes and to the registrar function (which will respond 756 with registration data corresponding to the explicit registrations at 757 the SSP). The forking in both of these cases can be avoided if the 758 SSP has and can maintain a copy of up-to-date information from the 759 PBXes. 761 Section 4.9 of RFC 3680 [12] indicates that "a subscriber MUST NOT 762 create multiple dialogs as a result of a single [registration event] 763 subscription request." Consequently, subscribers who are not aware 764 of the extension described by this document will accept only one 765 dialog in response to such requests. In the case described in the 766 preceding paragraph, this behavior will result in such client 767 receiving accurate but incomplete information about the registration 768 state of an AOR. As an explicit change to the normative behavior of 769 RFC 3680, this document stipulates that subscribers to the 770 registration event package MAY create multiple dialogs as the result 771 of a single subscription request. This will allow subscribers to 772 create a complete view of an AOR's registration state. 774 Defining the behavior as described above is important, since the reg- 775 event subscriber is interested in finding out about the comprehensive 776 list of devices associated with the AOR. Only the SIP-PBX will have 777 authoritative access to this information. For example, if the user 778 has registered multiple UAs with differing capabilities, the SSP will 779 not know about the devices or their capabilities. By contrast, the 780 SIP-PBX will. 782 If the SIP-PBX is not registered with the SSP when a registration 783 event subscription for a contact that would be implicitly registered 784 if the SIP-PBX were registered, then the SSP SHOULD accept the 785 subscription and indicate that the user is not currently registered. 786 Once the associated SIP-PBX is registered, the SSP SHOULD use the 787 subscription migration mechanism defined in RFC 3265 [5] to migrate 788 the subscription to the SIP-PBX. 790 When a SIP-PBX receives a registration event subscription addressed 791 to an AOR that has been registered using the bulk registration 792 mechanism described in this document, then each resulting 793 registration information document SHOULD contain an 'aor' attribute 794 in its element that corresponds to the AOR at the 795 SSP. 797 For example, consider a SIP-PBX that has registered with an SSP 798 that has a domain of "ssp.example.com" The SIP-PBX used a contact 799 of "sip:198.51.100.3:5060;bnc". After such registration is 800 complete, a registration event subscription arriving at the SSP 801 with a Request-URI of "sip:+12145550102@ssp.example.com" will be 802 re-targeted to the SIP-PBX, with a Request-URI of 803 "sip:+12145550102@198.51.100.3:5060". The resulting registration 804 document created by the SIP-PBX would contain a 805 element with an "aor" attribute of 806 "sip:+12145550102@ssp.example.com". 808 This behavior ensures that subscribers external to the system (and 809 unaware of GIN procedures) will be able to find the relevant 810 information in the registration document (since they will be 811 looking for the publicly-visible AOR, not the address used for 812 sending information from the SSP to the SIP-PBX). 814 A SIP-PBX that supports both GRUU procedures and the registration 815 event packages SHOULD implement the extension defined in RFC 5628 816 [9]. 818 7.3. Client-Initiated (Outbound) Connections 820 RFC 5626 [19] defines a mechanism that allows UAs to establish long- 821 lived TCP connections or UDP associations with a proxy in a way that 822 allows bidirectional traffic between the proxy and the UA. This 823 behavior is particularly important in the presence of NATs, and 824 whenever TLS [18] security is required. Neither the SSP nor the SIP- 825 PBX is required to support client-initiated connections. 827 The outbound mechanism generally works with the solution defined in 828 this document without any modifications. Implementors should note 829 that the instance ID used between the SIP-PBX and the SSP's registrar 830 identifies the SIP-PBX itself, and not any of the UAs registered with 831 the SIP-PBX. As a consequence, any attempts to use caller 832 preferences (defined in RFC 3841[14]) to target a specific instance 833 are likely to fail. This shouldn't be an issue, as the preferred 834 mechanism for targeting specific instances of a user agent is GRUU 835 (see Section 7.1). 837 7.4. Non-Adjacent Contact Registration (Path) and Service Route 838 Discovery 840 RFC 3327 [10] defines a means by which a registrar and its associated 841 proxy can be informed of a route that is to be used between the proxy 842 and the registered user agent. The scope of the route created by a 843 "Path" header field is contact-specific; if an AOR has multiple 844 contacts associated with it, the routes associated with each contact 845 may be different from each other. Support for non-adjacent contact 846 registration is required in all SSPs and SIP-PBXes implementing the 847 multiple-AOR-registration protocol described in this document. 849 At registration time, any proxies between the user agent and the 850 registrar may add themselves to the Path. By doing so, they request 851 that any requests destined to the user agent as a result of the 852 associated registration include them as part of the Route towards the 853 User Agent. Although the Path mechanism does deliver the final Path 854 value to the registering UA, UAs typically ignore the value of the 855 Path. 857 To provide similar functionality in the opposite direction -- that 858 is, to establish a route for requests sent by a registering UA -- RFC 859 3608 [11] defines a means by which a UA can be informed of a route 860 that is to be used by the UA to route all outbound requests 861 associated with the AOR used in the registration. This information 862 is scoped to the AOR within the UA, and is not specific to the 863 Contact (or Contacts) in the REGISTER request. Support of service 864 route discovery is OPTIONAL in SSPs and SIP-PBXes. 866 The registrar unilaterally generates the values of the service route 867 using whatever local policy it wishes to apply. Although it is 868 common to use the Path and/or Route information in the request in 869 composing the Service-Route, registrar behavior is not constrained in 870 any way that requires it to do so. 872 In considering the interaction between these mechanisms and the 873 registration of multiple AORs in a single request, implementors of 874 proxies, registrars, and intermediaries must keep in mind the 875 following issues, which stem from the fact that GIN effectively 876 registers multiple AORs and multiple Contacts. 878 First, all location service records that result from expanding a 879 single Contact containing a "bnc" parameter will necessarily share a 880 single path. Proxies will be unable to make policy decisions on a 881 contact-by-contact basis regarding whether to include themselves in 882 the path. Second, and similarly, all AORs on the SIP-PBX that are 883 registered with a common REGISTER request will be forced to share a 884 common Service-Route. 886 One interesting technique that Path and Service-Route enable is the 887 inclusion of a token or cookie in the user portion of the Service- 888 Route or Path entries. This token or cookie may convey information 889 to proxies about the identity, capabilities, and/or policies 890 associated with the user. Since this information will be shared 891 among several AORs and several Contacts when multiple AOR 892 registration is employed, care should be taken to ensure that doing 893 so is acceptable for all AORs and all Contacts registered in a single 894 REGISTER request. 896 8. Examples 898 Note that the following examples elide any steps related to 899 authentication. This is done for the sake of clarity. Actual 900 deployments will need to provide a level of authentication 901 appropriate to their system. 903 8.1. Usage Scenario: Basic Registration 905 This example shows the message flows for a basic bulk REGISTER 906 transaction, followed by an INVITE addressed to one of the registered 907 UAs. Example messages are shown after the sequence diagram. 909 Internet SSP SIP-PBX 910 | | | 911 | |(1) REGISTER | 912 | |Contact: | 913 | |<--------------------------------| 914 | | | 915 | |(2) 200 OK | 916 | |-------------------------------->| 917 | | | 918 |(3) INVITE | | 919 |sip:+12145550105@ssp.example.com| | 920 |------------------------------->| | 921 | | | 922 | |(4) INVITE | 923 | |sip:+12145550105@198.51.100.3 | 924 | |-------------------------------->| 925 (1) The SIP-PBX registers with the SSP for a range of AORs. 927 REGISTER sip:ssp.example.com SIP/2.0 928 Via: SIP/2.0/UDP 198.51.100.3:5060;branch=z9hG4bKnashds7 929 Max-Forwards: 70 930 To: 931 From: ;tag=a23589 932 Call-ID: 843817637684230@998sdasdh09 933 CSeq: 1826 REGISTER 934 Proxy-Require: gin 935 Require: gin 936 Supported: path 937 Contact: 938 Expires: 7200 939 Content-Length: 0 941 (3) The SSP receives a request for an AOR assigned 942 to the SIP-PBX. 944 INVITE sip:+12145550105@ssp.example.com SIP/2.0 945 Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad 946 Max-Forwards: 69 947 To: 948 From: ;tag=456248 949 Call-ID: f7aecbfc374d557baf72d6352e1fbcd4 950 CSeq: 24762 INVITE 951 Contact: 952 Content-Type: application/sdp 953 Content-Length: ... 955 956 (4) The SSP retargets the incoming request according to the 957 information received from the SIP-PBX at registration time. 959 INVITE sip:+12145550105@198.51.100.3 SIP/2.0 960 Via: SIP/2.0/UDP ssp.example.com;branch=z9hG4bKa45cd5c52a6dd50 961 Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad 962 Max-Forwards: 68 963 To: 964 From: ;tag=456248 965 Call-ID: f7aecbfc374d557baf72d6352e1fbcd4 966 CSeq: 24762 INVITE 967 Contact: 968 Content-Type: application/sdp 969 Content-Length: ... 971 973 8.2. Usage Scenario: Using Path to Control Request URI 975 This example shows a bulk REGISTER transaction with the SSP making 976 use of the "Path" header field extension [10]. This allows the SSP 977 to designate a domain on the incoming Request URI that does not 978 necessarily resolve to the SIP-PBX when the SSP applies RFC 3263 979 procedures to it. 981 Internet SSP SIP-PBX 982 | | | 983 | |(1) REGISTER | 984 | |Path: | 985 | |Contact: | 986 | |<--------------------------------| 987 | | | 988 | |(2) 200 OK | 989 | |-------------------------------->| 990 | | | 991 |(3) INVITE | | 992 |sip:+12145550105@ssp.example.com| | 993 |------------------------------->| | 994 | | | 995 | |(4) INVITE | 996 | |sip:+12145550105@pbx.example | 997 | |Route: | 998 | |-------------------------------->| 999 (1) The SIP-PBX registers with the SSP for a range of AORs. 1000 It includes the form of the URI it expects to receive in the 1001 Request-URI in its "Contact" header field, and includes 1002 information that routes to the SIP-PBX in the "Path" header 1003 field. 1005 REGISTER sip:ssp.example.com SIP/2.0 1006 Via: SIP/2.0/UDP 198.51.100.3:5060;branch=z9hG4bKnashds7 1007 Max-Forwards: 70 1008 To: 1009 From: ;tag=a23589 1010 Call-ID: 326983936836068@998sdasdh09 1011 CSeq: 1826 REGISTER 1012 Proxy-Require: gin 1013 Require: gin 1014 Supported: path 1015 Path: 1016 Contact: 1017 Expires: 7200 1018 Content-Length: 0 1020 (3) The SSP receives a request for an AOR assigned 1021 to the SIP-PBX. 1023 INVITE sip:+12145550105@ssp.example.com SIP/2.0 1024 Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad 1025 Max-Forwards: 69 1026 To: 1027 From: ;tag=456248 1028 Call-ID: 7ca24b9679ffe9aff87036a105e30d9b 1029 CSeq: 24762 INVITE 1030 Contact: 1031 Content-Type: application/sdp 1032 Content-Length: ... 1034 1035 (4) The SSP retargets the incoming request according to the 1036 information received from the SIP-PBX at registration time. 1037 Per the normal processing associated with "Path," it 1038 will insert the "Path" value indicated by the SIP-PBX at 1039 registration time in a "Route" header field, and 1040 set the request URI to the registered Contact. 1042 INVITE sip:+12145550105@pbx.example SIP/2.0 1043 Via: SIP/2.0/UDP ssp.example.com;branch=z9hG4bKa45cd5c52a6dd50 1044 Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad 1045 Route: 1046 Max-Forwards: 68 1047 To: 1048 From: ;tag=456248 1049 Call-ID: 7ca24b9679ffe9aff87036a105e30d9b 1050 CSeq: 24762 INVITE 1051 Contact: 1052 Content-Type: application/sdp 1053 Content-Length: ... 1055 1057 9. IANA Considerations 1059 This document registers a new SIP option tag to indicate support for 1060 the mechanism it defines, two new SIP URI parameters, and a "Contact" 1061 header field parameter. The process governing registration of these 1062 protocol elements is outlined in RFC5727 [21]. 1064 9.1. New SIP Option Tag 1066 This section defines a new SIP option tag per the guidelines in 1067 Section 27.1 of RFC 3261 [3]. 1068 Name: gin 1069 Description: This option tag is used to identify the extension that 1070 provides Registration for Multiple Phone Numbers in SIP. When 1071 present in a Require or Proxy-Require header field of a REGISTER 1072 request, it indicates that support for this extension is required 1073 of registrars and proxies, respectively, that are a party to the 1074 registration transaction. 1075 Reference: RFCXXXX (this document) 1077 9.2. New SIP URI Parameters 1079 This specification defines two new SIP URI parameters, as per the 1080 registry created by RFC 3969 [7]. 1082 9.2.1. 'bnc' SIP URI parameter 1084 Parameter Name: bnc 1085 Predefined Values: No (no values are allowed) 1086 Reference: RFCXXXX (this document) 1088 9.2.2. 'sg' SIP URI parameter 1090 Parameter Name: sg 1091 Predefined Values: No 1092 Reference: RFCXXXX (this document) 1094 9.3. New SIP Header Field Parameter 1096 This section defines a new SIP header field parameter per the 1097 registry created by RFC3968 [6]. 1099 Header field: Contact 1100 Parameter name: temp-gruu-cookie 1101 Predefined values: none 1102 Reference: RFCXXXX (this document) 1104 10. Security Considerations 1106 The change proposed for the mechanism described in this document 1107 takes the unprecedented step of extending the previously-defined 1108 REGISTER method to apply to more than one AOR. In general, this kind 1109 of change has the potential to cause problems at intermediaries -- 1110 such as proxies -- that are party to the REGISTER transaction. In 1111 particular, such intermediaries may attempt to apply policy to the 1112 user indicated in the "To" header field (i.e. the SIP-PBX's 1113 identity), without any knowledge of the multiple AORs that are being 1114 implicitly registered. 1116 The mechanism defined by this document solves this issue by adding an 1117 option tag to a "Proxy-Require" header field in such REGISTER 1118 requests. Proxies that are unaware of this mechanism will not 1119 process the requests, preventing them from mis-applying policy. 1120 Proxies that process requests with this option tag are clearly aware 1121 of the nature of the REGISTER request, and can make reasonable policy 1122 decisions. 1124 As noted in Section 7.4, intermediaries need to take care if they use 1125 a policy token in the Path and Service-Route mechanisms, as doing so 1126 will cause them to apply the same policy to all users serviced by the 1127 same SIP-PBX. This may frequently be the correct behavior, but 1128 circumstances can arise in which differentiation of user policy is 1129 required. 1131 Section 7.4 also notes that techniques that use a token or cookie in 1132 the Path and/or Service-Route values, and that this value will be 1133 shared among all AORs associated with a single registration. Because 1134 this information will be visible to User Agents under certain 1135 conditions, proxy designers using this mechanism in conjunction with 1136 the techniques describe in this document need to take care that doing 1137 so does not leak sensitive information. 1139 One of the key properties of the outbound client connection mechanism 1140 discussed in Section 7.3 is assurances that a single connection is 1141 associated with a single user, and cannot be hijacked by other users. 1142 With the mechanism defined in this document, such connections 1143 necessarily become shared between users. However, the only entity in 1144 a position to hijack calls as a consequence is the SIP-PBX itself. 1145 Because the SIP-PBX acts as a registrar for all the potentially 1146 affected users, it already has the ability to redirect any such 1147 communications as it sees fit. In other words, the SIP-PBX must be 1148 trusted to handle calls in an appropriate fashion, and the use of the 1149 outbound connection mechanism introduces no additional 1150 vulnerabilities. 1152 The ability to learn the identity and registration state of every 1153 user on the PBX (as described in Section 7.2.1) is invaluable for 1154 diagnostic and administrative purposes. For example, this allows the 1155 SIP-PBX to determine whether all the its extensions are properly 1156 registered with the SSP. However, this information can also be 1157 highly sensitive, as many organizations may not wish to make their 1158 entire list of phone numbers available to external entities. 1159 Consequently, SSP servers are advised to use explicit (i.e. white- 1160 list) and configurable policies regarding who can access this 1161 information, with very conservative defaults (e.g., an empty access 1162 list or an access list consisting only of the PBX itself). 1164 The procedure for generation of temporary GRUUs requires the use of 1165 an HMAC to detect any tampering that external entities may attempt to 1166 perform on the contents of a temporary GRUU. The mention of HMAC- 1167 SHA256-80 in Section 7.1.2 is intended solely as an example of a 1168 suitable HMAC algorithm. Since all HMACs used in this document are 1169 generated and consumed by the same entity, the choice of an actual 1170 HMAC algorithm is entirely up to an implementation, provided that the 1171 cryptographic properties are sufficient to prevent third parties from 1172 spoofing GRUU-related information. 1174 The procedure for generation of temporary GRUUs also requires the use 1175 of RSA keys. The selection of the proper key length for such keys 1176 requires careful analysis, taking into consideration the current and 1177 foreseeable speed of processing for the period of time during which 1178 GRUUs must remain anonymous, as well as emerging cryptographic 1179 analysis methods. The most recent guidance from RSA Laboratories 1180 [25] suggests a key length of 2048 bits for data that needs 1181 protection through the year 2030, and a length of 3072 bits 1182 thereafter. 1184 Similarly, implementors are warned to take precautionary measures to 1185 prevent unauthorized disclosure of the private key used in GRUU 1186 generation. Any such disclosure would result in the ability to 1187 correlate temporary GRUUs to each other, and potentially to their 1188 associated PBXes. 1190 Further, the use of RSA decryption when processing GRUUs received 1191 from arbitrary parties can be exploited by DoS attackers to amplify 1192 the impact of an attack: because of the presence of a cryptographic 1193 operation in the processing of such messages, the CPU load may be 1194 marginally higher when the attacker uses (valid or invalid) temporary 1195 GRUUs in the messages employed by such an attack. Normal DoS 1196 mitigation techniques, such as rate-limiting processing of received 1197 messages, should help to reduce the impact of this issue as well. 1199 Finally, good security practices should be followed regarding the 1200 duration an RSA key is used. For implementors, this means that 1201 systems MUST include an easy way to update the public key provided to 1202 the SIP-PBX. To avoid immediately invalidating all currently issued 1203 temporary GRUUs, the SSP servers SHOULD keep the retired RSA key 1204 around for a grace period before discarding it. If decryption fails 1205 based on the new RSA key, then the SSP server can attempt to use the 1206 retired key instead. By contrast, the SIP-PBXes MUST discard the 1207 retired public key immediately, and exclusively use the new public 1208 key. 1210 11. Acknowledgements 1212 This document represents the hard work of many people in the IETF 1213 MARTINI working group and the IETF RAI community as a whole. 1214 Particular thanks are owed to John Elwell for his requirements 1215 analysis of the mechanism described in this document, to Dean Willis 1216 for his analysis of the interaction between this mechanism and the 1217 Path and Service-Route extensions, to Cullen Jennings for his 1218 analysis of the interaction between this mechanism and the SIP 1219 Outbound extension, and to to Richard Barnes for his detailed 1220 security analysis of the GRUU construction algorithm. Thanks to Eric 1221 Rescorla, whose text in the appendix of RFC5627 was lifted directly 1222 to provide substantial portions of Section 7.1.2. Finally, thanks to 1223 Bernard Aboba, Francois Audet, Brian Carpenter, John Elwell, David 1224 Hancock, Ted Hardie, Martien Huysmans, Cullen Jennings, Alan 1225 Johnston, Hadriel Kaplan, Paul Kyzivat, and Radia Perlman for their 1226 careful reviews of and constructive feedback on this document. 1228 12. References 1230 12.1. Normative References 1232 [1] Krawczyk, H., Bellare, M., and R. Canetti, "HMAC: Keyed-Hashing 1233 for Message Authentication", RFC 2104, February 1997. 1235 [2] Bradner, S., "Key words for use in RFCs to Indicate Requirement 1236 Levels", BCP 14, RFC 2119, March 1997. 1238 [3] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., 1239 Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: 1240 Session Initiation Protocol", RFC 3261, June 2002. 1242 [4] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol 1243 (SIP): Locating SIP Servers", RFC 3263, June 2002. 1245 [5] Roach, A., "Session Initiation Protocol (SIP)-Specific Event 1246 Notification", RFC 3265, June 2002. 1248 [6] Camarillo, G., "The Internet Assigned Number Authority (IANA) 1249 Header Field Parameter Registry for the Session Initiation 1250 Protocol (SIP)", BCP 98, RFC 3968, December 2004. 1252 [7] Camarillo, G., "The Internet Assigned Number Authority (IANA) 1253 Uniform Resource Identifier (URI) Parameter Registry for the 1254 Session Initiation Protocol (SIP)", BCP 99, RFC 3969, 1255 December 2004. 1257 [8] Josefsson, S., "The Base16, Base32, and Base64 Data Encodings", 1258 RFC 4648, October 2006. 1260 [9] Kyzivat, P., "Registration Event Package Extension for Session 1261 Initiation Protocol (SIP) Globally Routable User Agent URIs 1262 (GRUUs)", RFC 5628, October 2009. 1264 12.2. Informative References 1266 [10] Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP) 1267 Extension Header Field for Registering Non-Adjacent Contacts", 1268 RFC 3327, December 2002. 1270 [11] Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP) 1271 Extension Header Field for Service Route Discovery During 1272 Registration", RFC 3608, October 2003. 1274 [12] Rosenberg, J., "A Session Initiation Protocol (SIP) Event 1275 Package for Registrations", RFC 3680, March 2004. 1277 [13] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating 1278 User Agent Capabilities in the Session Initiation Protocol 1279 (SIP)", RFC 3840, August 2004. 1281 [14] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller 1282 Preferences for the Session Initiation Protocol (SIP)", 1283 RFC 3841, August 2004. 1285 [15] Schulzrinne, H., "The tel URI for Telephone Numbers", RFC 3966, 1286 December 2004. 1288 [16] Eastlake, D., Schiller, J., and S. Crocker, "Randomness 1289 Requirements for Security", BCP 106, RFC 4086, June 2005. 1291 [17] Sparks, R., Hawrylyshen, A., Johnston, A., Rosenberg, J., and 1292 H. Schulzrinne, "Session Initiation Protocol (SIP) Torture Test 1293 Messages", RFC 4475, May 2006. 1295 [18] Dierks, T. and E. Rescorla, "The Transport Layer Security (TLS) 1296 Protocol Version 1.2", RFC 5246, August 2008. 1298 [19] Jennings, C., Mahy, R., and F. Audet, "Managing Client- 1299 Initiated Connections in the Session Initiation Protocol 1300 (SIP)", RFC 5626, October 2009. 1302 [20] Rosenberg, J., "Obtaining and Using Globally Routable User 1303 Agent URIs (GRUUs) in the Session Initiation Protocol (SIP)", 1304 RFC 5627, October 2009. 1306 [21] Peterson, J., Jennings, C., and R. Sparks, "Change Process for 1307 the Session Initiation Protocol (SIP) and the Real-time 1308 Applications and Infrastructure Area", BCP 67, RFC 5727, 1309 March 2010. 1311 [22] Elwell, J. and H. Kaplan, "Requirements for Multiple Address of 1312 Record (AOR) Reachability Information in the Session Initiation 1313 Protocol (SIP)", RFC 5947, September 2010. 1315 [23] Kaplan, H., Enterprise, o., contact, R., URI, u., and o. URI, 1316 "GIN with Literal AoRs for SIP in SSPs (GLASS)", 1317 draft-kaplan-martini-glass-00 (work in progress), 1318 November 2010. 1320 [24] National Institute of Standards and Technology, "Secure Hash 1321 Standard (SHS)", FIPS PUB 180-3, October 2008, . 1324 [25] Kaliski, B., "TWIRL and RSA Key Size", May 2003. 1326 Appendix A. Requirements Analysis 1328 The document "Requirements for multiple address of record (AOR) 1329 reachability information in the Session Initiation Protocol (SIP)" 1330 [22] contains a list of requirements and desired properties for a 1331 mechanism to register multiple AORs with a single SIP transaction. 1332 This section evaluates those requirements against the mechanism 1333 described in this document. 1335 REQ1 - The mechanism MUST allow a SIP-PBX to enter into a trunking 1336 arrangement with an SSP whereby the two parties have agreed on a set 1337 of telephone numbers deemed to have been assigned to the SIP-PBX. 1339 The requirement is satisfied. 1341 REQ2 - The mechanism MUST allow a set of assigned telephone numbers 1342 to comprise E.164 numbers, which can be in contiguous ranges, 1343 discrete, or in any combination of the two. 1345 The requirement is satisfied; the DIDs associated with a 1346 registration is established by bilateral agreement between the SSP 1347 and the SIP-PBX, and is not part of the mechanism described in 1348 this document. 1350 REQ3 - The mechanism MUST allow a SIP-PBX to register reachability 1351 information with its SSP, in order to enable the SSP to route to the 1352 SIP-PBX inbound requests targeted at assigned telephone numbers. 1354 The requirement is satisfied. 1356 REQ4 - The mechanism MUST allow UAs attached to a SIP-PBX to register 1357 with the SIP-PBX for AORs based on assigned telephone numbers, in 1358 order to receive requests targeted at those telephone numbers, 1359 without needing to involve the SSP in the registration process. 1361 The requirement is satisfied; in the presumed architecture, SIP- 1362 PBX UAs register with the SIP-PBX, an require no interaction with 1363 the SSP. 1365 REQ5 - The mechanism MUST allow a SIP-PBX to handle requests 1366 originating at its own UAs and targeted at its assigned telephone 1367 numbers, without routing those requests to the SSP. 1369 The requirement is satisfied; SIP-PBXes may recognize their own 1370 DID and their own GRUUs, and perform on-SIP-PBX routing without 1371 sending the requests to the SSP. 1373 REQ6 - The mechanism MUST allow a SIP-PBX to receive requests to its 1374 assigned telephone numbers originating outside the SIP-PBX and 1375 arriving via the SSP, so that the SIP-PBX can route those requests 1376 onwards to its UAs, as it would for internal requests to those 1377 telephone numbers. 1379 The requirement is satisfied 1381 REQ7 - The mechanism MUST provide a means whereby a SIP-PBX knows 1382 which of its assigned telephone numbers an inbound request from its 1383 SSP is targeted at. 1385 The requirement is satisfied. For ordinary calls and calls using 1386 Public GRUUs, the DID is indicated in the user portion of the 1387 Request-URI. For calls using Temp GRUUs constructed with the 1388 mechanism described in Section 7.1.2, the "gr" parameter provides 1389 a correlation token the SIP-PBX can use to identify which UA the 1390 call should be routed to. 1392 REQ8 - The mechanism MUST provide a means of avoiding problems due to 1393 one side using the mechanism and the other side not. 1395 The requirement is satisfied through the 'gin' option tag and the 1396 'bnc' Contact parameter. 1398 REQ9 - The mechanism MUST observe SIP backwards compatibility 1399 principles. 1401 The requirement is satisfied through the 'gin' option tag. 1403 REQ10 - The mechanism MUST work in the presence of a sequence of 1404 intermediate SIP entities on the SIP-PBX-to-SSP interface (i.e., 1405 between the SIP-PBX and the SSP's domain proxy), where those 1406 intermediate SIP entities indicated during registration a need to be 1407 on the path of inbound requests to the SIP-PBX. 1409 The requirement is satisfied through the use of the Path mechanism 1410 defined in RFC 3327 [10] 1412 REQ11 - The mechanism MUST work when a SIP-PBX obtains its IP address 1413 dynamically. 1415 The requirement is satisfied by allowing the SIP-PBX to use an IP 1416 address in the Bulk Number Contact URI contained in a REGISTER 1417 Contact header field. 1419 REQ12 - The mechanism MUST work without requiring the SIP-PBX to have 1420 a domain name or the ability to publish its domain name in the DNS. 1422 The requirement is satisfied by allowing the SIP-PBX to use an IP 1423 address in the Bulk Number Contact URI contained in a REGISTER 1424 Contact header field. 1426 REQ13 - For a given SIP-PBX and its SSP, there MUST be no impact on 1427 other domains, which are expected to be able to use normal RFC 3263 1428 procedures to route requests, including requests needing to be routed 1429 via the SSP in order to reach the SIP-PBX. 1431 The requirement is satisfied by allowing the domain name in the 1432 Request URI used by external entities to resolve to the SSP's 1433 servers via normal RFC 3263 resolution procedures. 1435 REQ14 - The mechanism MUST be able to operate over a transport that 1436 provides end-to-end integrity protection and confidentiality between 1437 the SIP-PBX and the SSP, e.g., using TLS as specified in [3]. 1439 The requirement is satisfied; nothing in the proposed mechanism 1440 prevent the use of TLS between the SSP and the SIP-PBX. 1442 REQ15 - The mechanism MUST support authentication of the SIP-PBX by 1443 the SSP and vice versa, e.g., using SIP digest authentication plus 1444 TLS server authentication as specified in [3]. 1446 The requirement is satisfied; SIP-PBXes may employ either SIP 1447 digest authentication or mutually-authenticated TLS for 1448 authentication purposes. 1450 REQ16 - The mechanism MUST allow the SIP-PBX to provide its UAs with 1451 public or temporary Globally Routable UA URIs (GRUUs) [20]. 1453 The requirement is satisfied via the mechanisms detailed in 1454 Section 7.1. 1456 REQ17 - The mechanism MUST work over any existing transport specified 1457 for SIP, including UDP. 1459 The requirement is satisfied to the extent that UDP can be used 1460 for REGISTER requests in general. The application of certain 1461 extensions and/or network topologies may exceed UDP MTU sizes, but 1462 such issues arise both with and without the mechanism described in 1463 this document. This document does not exacerbate such issues. 1465 REQ18 - Documentation MUST give guidance or warnings about how 1466 authorization policies may be affected by the mechanism, to address 1467 the problems described in Section 3.3 (of RFC5947). 1469 These issues are addressed at length in Section 10, as well as 1470 summarized in Section 7.4. 1472 REQ19 - The mechanism MUST be extensible to allow a set of assigned 1473 telephone numbers to comprise local numbers as specified in RFC3966 1474 [15], which can be in contiguous ranges, discrete, or in any 1475 combination of the two. 1477 Assignment of telephone numbers to a registration is performed by 1478 the SSP's registrar, which is not precluded from assigning local 1479 numbers in any combination it desires. 1481 REQ20 - The mechanism MUST be extensible to allow a set of 1482 arbitrarily assigned SIP URI's as specified in RFC3261 [3], as 1483 opposed to just telephone numbers, without requiring a complete 1484 change of mechanism as compared to that used for telephone numbers. 1486 The mechanism is extensible in such a fashion, as demonstrated by 1487 the document "GIN with Literal AoRs for SIP in SSPs (GLASS)" [23]. 1489 DES1 - The mechanism SHOULD allow an SSP to exploit its mechanisms 1490 for providing SIP service to ordinary subscribers in order to provide 1491 a SIP trunking service to SIP-PBXes. 1493 The desired property is satisfied; the routing mechanism described 1494 in this document is identical to the routing performed for singly- 1495 registered AORs. 1497 DES2 - The mechanism SHOULD scale to SIP-PBX's of several thousand 1498 assigned telephone numbers. 1500 The desired property is satisfied; nothing in this document 1501 precludes DID pools of arbitrary size. 1503 DES3 - The mechanism SHOULD scale to support several thousand SIP- 1504 PBX's on a single SSP. 1506 The desired property is satisfied; nothing in this document 1507 precludes an arbitrary number of SIP-PBXes from attaching to a 1508 single SSP. 1510 Author's Address 1512 Adam Roach 1513 Tekelec 1514 17210 Campbell Rd. 1515 Suite 250 1516 Dallas, TX 75252 1517 US 1519 Email: adam@nostrum.com