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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 VIPR WG M. Barnes, Ed. 3 Internet-Draft Polycom 4 Intended status: Standards Track C. Jennings 5 Expires: September 14, 2012 Cisco 6 J. Rosenberg 7 jdrosen.net 8 M. Petit-Huguenin 9 Unaffiliated 10 March 13, 2012 12 Verification Involving PSTN Reachability: Requirements and Architecture 13 Overview 14 draft-jennings-vipr-overview-03 16 Abstract 18 The Session Initiation Protocol (SIP) has seen widespread deployment 19 within individual domains, typically supporting voice and video 20 communications. Though it was designed from the outset to support 21 inter-domain federation over the public Internet, such federation has 22 not materialized. The primary reasons for this are the complexities 23 of inter-domain phone number routing and concerns over security. 24 This document reviews this problem space, outlines requirements, and 25 then describes a new model and technique for inter-domain federation 26 with SIP involving the Public Switched Telephone Network (PSTN), 27 called Verification Involving PSTN Reachability (VIPR). VIPR 28 addresses the problems that have prevented inter-domain federation 29 over the Internet. It provides fully distributed inter-domain 30 routing for phone numbers, authorized mappings from phone numbers to 31 domains, a new technique for automated SIP anti-spam, and privacy of 32 number ownership, all while preserving the trapezoidal model of SIP. 34 Status of this Memo 36 This Internet-Draft is submitted in full conformance with the 37 provisions of BCP 78 and BCP 79. 39 Internet-Drafts are working documents of the Internet Engineering 40 Task Force (IETF). Note that other groups may also distribute 41 working documents as Internet-Drafts. The list of current Internet- 42 Drafts is at http://datatracker.ietf.org/drafts/current/. 44 Internet-Drafts are draft documents valid for a maximum of six months 45 and may be updated, replaced, or obsoleted by other documents at any 46 time. It is inappropriate to use Internet-Drafts as reference 47 material or to cite them other than as "work in progress." 48 This Internet-Draft will expire on September 14, 2012. 50 Copyright Notice 52 Copyright (c) 2012 IETF Trust and the persons identified as the 53 document authors. All rights reserved. 55 This document is subject to BCP 78 and the IETF Trust's Legal 56 Provisions Relating to IETF Documents 57 (http://trustee.ietf.org/license-info) in effect on the date of 58 publication of this document. Please review these documents 59 carefully, as they describe your rights and restrictions with respect 60 to this document. Code Components extracted from this document must 61 include Simplified BSD License text as described in Section 4.e of 62 the Trust Legal Provisions and are provided without warranty as 63 described in the Simplified BSD License. 65 Table of Contents 67 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 68 2. Conventions and Terminology . . . . . . . . . . . . . . . . . 4 69 3. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 5 70 3.1. The Phone Number Routing Problem . . . . . . . . . . . . . 5 71 3.2. The Open Pinhole Problem . . . . . . . . . . . . . . . . . 6 72 3.3. Quality of Service Problem . . . . . . . . . . . . . . . . 7 73 3.4. Troubleshooting Problem . . . . . . . . . . . . . . . . . 7 74 4. Summary of Existing Solutions . . . . . . . . . . . . . . . . 8 75 4.1. Domain Routing . . . . . . . . . . . . . . . . . . . . . . 8 76 4.2. Public ENUM . . . . . . . . . . . . . . . . . . . . . . . 8 77 4.3. Private Federations . . . . . . . . . . . . . . . . . . . 9 78 5. Key Requirements . . . . . . . . . . . . . . . . . . . . . . . 10 79 6. Executive Overview . . . . . . . . . . . . . . . . . . . . . . 10 80 6.1. Key Properties . . . . . . . . . . . . . . . . . . . . . . 11 81 6.2. Challenging Past Assumptions . . . . . . . . . . . . . . . 12 82 6.3. Technical Overview . . . . . . . . . . . . . . . . . . . . 13 83 6.3.1. Storage of Phone Numbers . . . . . . . . . . . . . . . 14 84 6.3.2. PSTN First Call . . . . . . . . . . . . . . . . . . . 15 85 6.3.3. Validation and Caching . . . . . . . . . . . . . . . . 16 86 6.3.4. SIP Call . . . . . . . . . . . . . . . . . . . . . . . 20 87 7. Security Considerations . . . . . . . . . . . . . . . . . . . 21 88 7.1. Attacks on the DHT . . . . . . . . . . . . . . . . . . . . 21 89 7.2. Theft of Phone Numbers . . . . . . . . . . . . . . . . . . 21 90 7.3. Spam . . . . . . . . . . . . . . . . . . . . . . . . . . . 22 91 7.4. Eavesdropping . . . . . . . . . . . . . . . . . . . . . . 23 92 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 23 93 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23 94 10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 24 95 10.1. Normative References . . . . . . . . . . . . . . . . . . . 24 96 10.2. Informative References . . . . . . . . . . . . . . . . . . 25 97 Appendix A. Changes since last version . . . . . . . . . . . . . 26 98 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 27 100 1. Introduction 102 The Session Initiation Protocol (SIP) was originally published as 103 [RFC2543] in May of 1999. This was followed by subsequent 104 publication of [RFC3261], which brought the protocol to sufficient 105 maturity to enable large scale market adoption. 107 SIP has achieved large scale market adoption with hundreds of 108 implementations, spanning consumer products, enterprise servers, and 109 large scale carrier equipment. It carries billions and billions of 110 minutes of calls, and has become the standard for interconnection 111 between products from different vendors. If one measures success in 112 deployment, then clearly SIP is a success. 114 SIP was designed from the ground up to enable communications between 115 users in different domains, all over the public Internet. The 116 intention was that real-time communications should be no different 117 than email or the web, with the same any-to-any connectivity that has 118 fueled the successes of those technologies. However, when SIP is 119 used between domains, it is typically through private federation 120 agreements. While such agreements are positive, they have typically 121 been limited to voice, which has limited the use of video and the 122 growth of advanced SIP features, thus preventing the innovation that 123 SIP was expected to drive. Thus, the any-to-any Internet federation 124 model envisioned by SIP has not materialized at scale. 126 This document introduces a new technology, called Verification 127 Involving PSTN Reachability (VIPR), that breaks down the barriers 128 that have prevented inter-domain voice, video and other multimedia 129 services. By stepping back and changing some of the most fundamental 130 assumptions about federation, VIPR is able to address the key 131 problems preventing its deployment. VIPR focuses on incremental 132 deployability. At the same time, VIPR ensures that SIP's trapezoidal 133 model of direct federation between domains without any intermediate 134 processing beyond IP transport is realized. That model is required 135 in order to allow innovative new services to be deployed. 137 2. Conventions and Terminology 139 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 140 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and 141 "OPTIONAL" in this document are to be interpreted as described in 142 [RFC2119]. 144 Call Agent: An entity in a SIP enabled domain that supports VIPR. 145 The Call Agent performs call processing on behalf of one or more 146 user agents represented by E.164 numbers within the domain. 147 Ticket: A shared secret that is generated after a PSTN call to 148 enable secure call setup on a subsequent inter-domain IP call 149 enabled by VIPR. 150 User Agent: As defined in [RFC3261], with the restriction that the 151 user agent must have an associated E.164 number. 153 3. Problem Statement 155 The first question that must be asked is this - why haven't we seen 156 widespread adoption of inter-domain SIP federation? The reason for 157 this is due to problems with the following - summarized in order of 158 importance/impact: 160 1. Phone number routing 161 2. Open pinhole 162 3. Quality of service 163 4. Troubleshooting 165 The first two are the most significant. 167 3.1. The Phone Number Routing Problem 169 Inter-domain federation requires that the sending domain determine 170 the address of the receiving domain, in the form of a DNS name 171 (example.com) or one or more IP addresses that can be used to reach 172 the domain. In email and in the web, this is easy. The identifiers 173 used by those services - the email address and web URL respectively - 174 embed the address of the receiving domain. A simple DNS lookup is 175 all that is required to route the connection. SIP was designed to 176 use the same email-style identifiers. 178 However, most SIP deployments utilize phone numbers in the form of 179 E.164 numbers [E.164], and not email-style SIP URIs. This is due to 180 the huge installed base of users that continue to exist solely on the 181 PSTN. In order to be reached by users on the PSTN, and in order to 182 reach them, users in SIP deployments need to be assigned a PSTN phone 183 number. Users in SIP deployments need to place that phone number on 184 business cards, use it in their email signatures, and in general, 185 give it out to their friends and colleagues, in order to be reached. 186 While those users could additionally have an email style SIP URI, the 187 phone number serves as a single, global identifier that works for 188 receiving calls from users on the PSTN as well as users within the 189 same SIP domain. 191 There are several reasons why two identifiers are used when one will 192 suffice. The universality of PSTN phone numbers is the reason why 193 most SIP deployments continue to use them - often exclusively. 195 Another reason is that many SIP deployments utilize hardphones or 196 telephony adaptors, and the user interfaces on these devices - 197 patterned after existing phones - only allow phone number based 198 dialing. Consequently, these users are only allocated PSTN phone 199 numbers, and not email-style SIP URI. 201 Finally, a large number of SIP deployments are in domains where the 202 endpoints are not IP. Rather, they are circuit based devices, 203 connected to a SIP network through a gateway. SIP is used within the 204 core of the network, providing lower cost transit, or providing 205 add-on services. Clearly, in these deployments, only phone numbers 206 are used. 208 Consequently, to make inter-domain federation incrementally 209 deployable and widely applicable, it needs to work with PSTN phone 210 numbers rather than email-style SIP URIs. Telephone numbers, unlike 211 email addresses, do not provide any indication of the address of the 212 domain which "owns" the phone number. Indeed, the notion of phone 213 number ownership is somewhat cloudy. Phone numbers can be ported 214 between carriers. They can be assigned to a user or enterprise, and 215 then later re-assigned to someone else. Phone numbers are granted to 216 users and enterprises through a complex delegation process involving 217 the ITU, governments, and telecommunications carriers, often 218 involving local regulations that vary from country to country. 220 Therefore, in order to deploy inter-domain federation, domains are 221 required to utilize some kind of mechanism to map phone numbers to 222 the address of the domain to which calls should be routed. Though 223 several techniques have been developed to address this issue, none 224 have achieved large-scale Internet deployments. 226 3.2. The Open Pinhole Problem 228 The inter-domain federation mechanism built into SIP borrows heavily 229 from email. Each domain runs a SIP server on an open port. When one 230 domain wishes to contact another, it looks up the domain name in the 231 DNS, and connects to that server on the open port. Here, "open" 232 means that the server is reachable from anywhere on the public 233 Internet, and is not blocked by firewalls. 235 This simple design worked well in the early days of email. However, 236 the email system has now become plagued with spam. This has resulted 237 in administrators spending a significant amount of time maintaining 238 spam filters. This does not always benefit the end users as in some 239 cases valid emails are dropped without the user being notified. 240 Thus, administrators of SIP domains are rightfully concerned that if 241 they make a SIP server available for anyone on the Internet to 242 contact, it will open the floodgates for SIP spam, which is far more 243 disruptive than email-based spam [RFC5039]. Administrators are also 244 concerned that an open server will create a back-door for denial-of- 245 service and other attacks that can potentially disrupt their voice 246 and video services. Administrators are often not willing to take 247 that risk since voice deployments demand higher uptimes and better 248 levels of reliability than email, especially for enterprises. 250 Fears around spam and denial-of-service attacks, when put together, 251 form the "open pinhole problem" - that domains are not willing to 252 enable SIP on an open port facing the Internet. 254 To fix this, a new model for federation is needed - a model where 255 these problems are addressed as part of the fundamental design rather 256 than after the functionality has been deployed. 258 3.3. Quality of Service Problem 260 The Internet does not provide any Quality of Service (QoS) 261 guarantees. All traffic is best effort. This is not an issue for 262 data transaction services, like web and email. It is, however, a 263 concern when using real-time services, such as voice and video. 265 That said, there are a large number of existing SIP deployments that 266 run over the Internet. Though the lack of QoS is a concern, it has 267 not proven a barrier to deployment. It is believed that if if the 268 more fundamental issues - the phone number routing and open pinhole 269 problems - can be addressed, the QoS problem will be a non-issue. As 270 such, QoS is not discussed further in this or other VIPR 271 specifications. 273 3.4. Troubleshooting Problem 275 The final problem that is prohibing large scale inter-domain 276 federation is troubleshooting. When connecting calls between 277 domains, problems can occur. Calls can be blocked. Calls can be 278 misdelivered. Features sometimes don't work. There can be one-way 279 media or no media at all. The video may not start. Call quality can 280 be poor. 282 These problems are common in SIP deployments, and they are tough to 283 troubleshoot even within a single administrative domain. When real- 284 time services extend inter-domain, the problem becomes worse. 286 Fortunately, work is underway to improve the ability for network 287 administrators to diagnose SIP problems. Common log formats 288 [I-D.ietf-sipclf-format] and consistent session IDs 289 [I-D.jones-ipmc-session-id-reqts], for example, can help troubleshoot 290 interdomain calls. 292 In addition to the above, any new technology that facilitates inter- 293 domain federation needs to have troubleshooting built-in, so that it 294 is not a barrier to deployment. Further consideration of necessary 295 built-in techniques for troubleshooting is required for successful 296 deployment of VIPR. 298 4. Summary of Existing Solutions 300 Given the value of inter-domain SIP federation, there are existing 301 deployed solutions summarized below. However, each solution approach 302 has fundamental limitations that have inhibited widespread 303 deployment. 305 4.1. Domain Routing 307 The first solution for SIP inter-domain federation is built into SIP 308 itself - domain routing. In this technique, users utilize email- 309 style SIP URIs as identifiers. By utilizing the DNS lookup mechanism 310 defined in [RFC3263], SIP enables calls to be routed between domains 311 in much the same way email is routed between domains. 313 This technique works well in theory, but it has two limitations which 314 have limited its deployment: 316 1. The majority of SIP deployments utilize phone numbers, often 317 exclusively. In such a case, domain routing cannot be used. 318 2. Domain federation brings with it the possibility (and strong 319 likelihood) of the same levels of spam and DoS attacks that have 320 plagued the email system. 322 These issues have already been discussed in sections Section 3.1 and 323 Section 3.2 respectively. 325 4.2. Public ENUM 327 Public ENUM, defined in [RFC6116] addresses the phone number routing 328 problem by placing phone numbers into the public DNS. Clients can 329 then perform a simple DNS lookup on a phone number, and retrieve a 330 SIP URI which can be used to route to that phone number. 332 Unfortunately, public ENUM requires that the entries placed into the 333 DNS be populated following a chain of responsibility that mirrors the 334 ownership of the numbers themselves. This means that, in order for a 335 number to be placed into the DNS, authorization to do so must start 336 with the ITU, and from there, move to the country, telecom regulator, 337 and ultimately the end user. The number of layers of bureaucracy 338 required to accomplish this is non-trivial. In addition, the telecom 339 operators - that would be partly responsible for populating the 340 numbers into the DNS - have little incentive to do so. As a 341 consequence, public ENUM is largely empty, and is likely to remain so 342 for the foreseeable future. 344 Instead, ENUM has evolved into a technique for federation amongst 345 closed peering partners, called private ENUM or infrastructure ENUM 346 [RFC5067]. While there is value in this technology, it does not 347 enable the open federation that public ENUM was designed to solve. 349 4.3. Private Federations 351 Private federations are a cooperative formed amongst a small number 352 of participating domains. The cooperative agrees to use a common 353 technique for federation, and through it, is able to connect to each 354 other. There are many such federations in use today. 356 Some of these federations rely on a central database, typically run 357 by the federation provider, that can be queried by participating 358 domains. The database contains mappings from phone numbers to 359 domains, and is populated by each of the participating domains, often 360 manually. Each domain implements an agreed-upon query interface that 361 can be used to access the database when a number is called. 362 Sometimes ENUM is used for this interface (called private ENUM), 363 other times, a SIP redirection is used. Some federations also 364 utilize private IP networks in order to address QoS problems. 366 Private federations work, but they have one major limitation: scale. 367 As the number of participating domains grows, several problems arise. 368 Firstly, the size of the databases become difficult to manage. 369 Secondly, the correctness of the database becomes an issue, since the 370 odds of misconfigured numbers (either intentionally or accidentally) 371 increases. As the membership grows further, the odds increase that 372 malicious domains will be let in, introducing a source of spam and 373 further problems. The owner of the federation can - and often does - 374 assume responsibility for this, and can attempt to identify and shut 375 down misbehaving participants. Indeed, as the size of the 376 federations grow, the owner of the federation needs to spend 377 increasing levels of capital on maintaining it. This often results 378 in the owners charging for membership, which can be a barrier to 379 entry. 381 5. Key Requirements 383 From the discussion on the problems of inter-domain federation and 384 the solutions that have been attempted so far, several key 385 requirements emerge: 387 REQ-1: The solution must allow for federation between any number of 388 domains. 389 REQ-2: The solution must enable users in one domain to identify 390 users in another domain through the use of their existing E.164 391 based phone numbers. 392 REQ-3: The solution must work with deployments that utilize any kind 393 of endpoint, including non-IP phones connected through gateways, 394 IP softphones and hardphones. 395 REQ-4: The solution must not require any change in user behavior. 396 The devices and techniques that users have been using previously 397 to make inter-domain calls must continue to work, but now result 398 in inter-domain calls using IP. 399 REQ-5: The solution must work worldwide, for any domain anywhere. 400 REQ-6: The solution must not require any new services from any kind 401 of centralized provider. A domain should be able to deploy 402 equipment and connect to the federation without any interaction 403 with or authorization from a centralized provider. 404 REQ-7: The solution must not require any prior arrangement between 405 domains in order to facilitate federation between those domains. 406 Federation must occur opportunistically - connections established 407 when they can be. 408 REQ-8: The solution must work for domains of any size - starting 409 with a single phone up to the largest telecom operator with tens 410 of millions of numbers. 411 REQ-9: The solution must have built-in mechanisms for preventing 412 spam and DoS attacks. These mechanisms must be fully automated. 413 REQ-10: The solution must not require any processing whatsoever by 414 SIP or RTP intermediaries. It must be possible for a direct SIP 415 connection to be established between participating domains. 416 REQ-11: The solution should adapt to VIPR call failures. The 417 solution should allow the user to make calls using the inter- 418 domain calling mechanism used prior to the initial VIPR-enabled 419 call. 421 6. Executive Overview 423 Verification Involving PSTN Reachability (VIPR) is aimed at solving 424 the problems that have prevented large-scale Internet-based SIP 425 federation of voice and video. VIPR solves these problems by 426 creating a hybrid of three technologies - the PSTN itself, a Peer to 427 Peer (P2P) network, and SIP. By using these three technologies 428 together, VIPR enables an incrementally deployable solution to 429 federation. 431 6.1. Key Properties 433 VIPR has several important properties that enable it to solve the 434 federation problem: 436 Works With Numbers: VIPR enables federation for existing PSTN phone 437 numbers. It does not require users or administrators to know or 438 configure email-style identifiers. It does not require the 439 allocation of new numbers. It does not require a change in user 440 behaviors. 441 Works with Existing Endpoints: VIPR does not require any changes to 442 endpoints. Consequently, it works with existing SIP endpoints and 443 with non-IP endpoints connected through gateways. 444 Verified Mappings: VIPR ensures that phone calls cannot be misrouted 445 or numbers stolen. The biggest issue in mapping from a phone 446 number to a domain or IP address, is determining whether the 447 mapping is correct - i.e., does the domain really own the given 448 phone number? While solutions like ENUM have solved this problem 449 by relying on centralized delegations of authorization, VIPR 450 provides a secure mapping in a fully distributed way. 451 Worldwide: VIPR works worldwide. Any domain that is connected to 452 both the PSTN and the Internet can participate. Since VIPR does 453 not depend on availability of any regional services beyond IP and 454 PSTN access - both of which are already available globally - VIPR 455 itself is globally available. 456 Scalibility: VIPR is scaleable. Any number of domains can 457 participate. 458 Self-Scale: VIPR self-scales. This means that the amount of 459 computation, memory, and bandwidth that a domain must deploy 460 scales in direct proportion to the size of their own user base. 461 Self-Learning: VIPR is completely automated. A domain does not 462 require configuration of any information about another domain. It 463 does not require provisioning of IP addresses, domain names, 464 certificates, phone number prefixes or routing rules. 465 Automated Anti-Spam VIPR has a built-in mechanism for preventing SIP 466 spam, which is specific to SIP. It is fundamentally different 467 from existing SIP anti-spam techniques which borrow from email 468 [RFC5039]. This new technique is fully automated, and requires no 469 configuration by administrators and no participation from end 470 users. 471 Feature Velocity: VIPR enables direct SIP connections between two 472 domains seeking to federate. There are no SIP intermediaries of 473 any sort between the two. This means that domains have no 474 dependencies on intermediaries for deployment of new features. 476 Secure: Security is a fundamental part of VIPR and cannot be 477 disabled. 478 Reliable: VIPR is reliable. Through its hybridization of the PSTN 479 and the Internet, it ensures that calls always go through, even in 480 cases of network failure or limited IP connectivity. 482 In order to achieve a solution with these properties, past 483 assumptions about how federations should work must be challenged. 485 6.2. Challenging Past Assumptions 487 Two unstated assumptions of SIP federation are challenged by VIPR. 489 The first assumption that federation solutions have made is this: 490 The purpose of SIP federation is to eliminate the PSTN, and 491 consequently, we cannot assume the PSTN itself as part of the 492 solution. 493 Though unstated, this assumption has clearly been part of the design 494 of existing solutions. SIP federation based on email-style URIs, as 495 defined in RFC 3261, doesn't utilize nor make mention of the PSTN. 496 Solutions like ENUM, or private registries, also do not utilize nor 497 make mention of the PSTN. However, such approaches ignore an 498 incremental solution - a solution which utilizes the PSTN itself to 499 solve the hard problems in SIP federation. 501 There are many advantages to leveraging the PSTN. It reaches 502 worldwide. It provides a global numbering translation service that 503 maps phone numbers to circuits. It is highly reliable, and provides 504 QoS. It has been built up over decades to achieve these goals. 505 Thus, building upon rather than replacing the PSTN, can provide the 506 necessary functionality once another assumption is challenged. 508 This second assumption is: 509 A federation solution must be the same as the final target 510 federation architecture, and not just a step towards it. 511 SIP's email-style federation was a pure 'target architecture'. ENUM 512 was the same - a worldwide global DNS database with everyone's phone 513 numbers providing open connectivity. 515 Historically, technologies are more successful when they are 516 incrementally deployable. As such, VIPR is very much focused on 517 incremental deployability. It discards the notion of perfect IP 518 federation for a solution that federates most, but not all calls, by 519 relying on the PSTN to fill in the gaps. 521 6.3. Technical Overview 523 A high level view of the VIPR architecture with an example is shown 524 in Figure 1. The figure shows four different domains, example.com, 525 example.net, example.org and example.edu, federated using VIPR 526 technology. Each domain is connected to both the public Internet and 527 to the traditional PSTN. For simplicity, the connection for the call 528 agents in example.org and example.edu to the PSTN is not indicated in 529 the diagram as that interface is not relevant to the subsequent 530 examples. 532 +-------+ +-------+ 533 | Call | | Call | 534 example.org | Agent | | Agent | example.edu 535 | | | | 536 +-------+ +-------+ 537 \ / 538 \ / 539 \ / 540 \ / 541 | 542 //--------\\ 543 |// \\| 544 | Internet | 545 +-------+ |\\ //| +-------+ 546 | Call |------ \\ _______//------| Call | 547 //\\ | Agent | | Agent | //\\ 548 \ / | | | | \ / 549 \/ ---| | +-----------+ | |---- \/ 550 User | |======| |======| | User 551 Agent +-------+ | PSTN | +-------+ Agent 552 example.com | | example.net 553 +-----------+ 555 Figure 1: High Level Architecture 557 For purposes of explanation, it is easiest to think of each domain as 558 having a single call agent which participates in the federation 559 solution. The functionality is decomposed into several sub- 560 components, and this is discussed in more detail below. The call 561 agent is connected to one or more user agents in the domain, and is 562 responsible for routing calls, handling features, and processing call 563 state. The call agent is stateful, and is aware of when calls start 564 and stop. Additional detail for the functional components of this 565 architecture are provided in [I-D.petithuguenin-vipr-framework]. 567 Assume that all four domains have a 'fresh' installation of VIPR, and 568 that domain example.net 'owns' +1 408 555 5xxx, a block of 1000 569 numbers allocated by its PSTN provider. 571 The VIPR mechanism can be broken into four basic steps: storage of 572 phone numbers, PSTN first call, validation and caching, and 573 subsequent SIP call(s). 575 6.3.1. Storage of Phone Numbers 577 The first step is that the call agents form a single, worldwide P2P 578 network, using a VIPR specific usage 579 [I-D.petithuguenin-vipr-reload-usage] of RELOAD 580 [I-D.ietf-p2psip-base] with a variant of the Chord algorithm. This 581 P2P network forms a distributed hash table (DHT) running amongst all 582 participating domains. A distributed hash table is like a simple 583 database, allowing storage of key-value pairs, and lookup of objects 584 by key. Unlike a normal hash table, which resides in the memory of a 585 single computer, a distributed hash table is spread across all of the 586 servers which make up the P2P network. In this case, it is spread 587 across all of the domains participating in the VIPR federation. 589 The problem solved by the variant of the Chord algorithm (and by 590 other DHT algorithms), is an answer to the following: given that the 591 desired operation is to read or write an object with key K, which 592 node in the DHT is the box that currently stores the object with that 593 key? The P2P SIP variant of the Chord algorithm provides an 594 algorithm which routes read and write operations through nodes in the 595 DHT until they eventually arrive at the right place. With Chord, 596 this will take no more than log2N hops, where N is the number of 597 nodes in the DHT. Consequently, for a DHT with 1024 nodes, 10 hops 598 are required in the worst case. For 2048, 11 hops. And so on. The 599 logarithmic factor allows DHTs to achieve efficient scale and to 600 provide a large amount of storage summed across all of the nodes that 601 make up the DHT. 603 This logarithmic hopping behavior also means that each node in the 604 DHT does not need to establish a TCP/TLS connection to every other 605 node. Rather, connections are established to a smaller subset - just 606 log(N) of the nodes. 608 In DHTs, each participating entity is identified by a Node-ID. The 609 Node-ID is a 128 bit number, assigned randomly to each entity. They 610 have no inherent semantic meaning; they are not like domain names or 611 IP addresses. 613 In the case of VIPR, each call agent is identified by one or more 614 Node-IDs. For purposes of discussion, consider the case where the 615 call agent has just one Node-ID. Each participating domain, 616 including example.net in our example, uses the DHT to store a mapping 617 from each phone number that it owns, to the domain's Node-ID. In the 618 case of example.net, it would store 1000 entries into the DHT, each 619 one being a mapping from one of its phone numbers, to the domain's 620 Node-ID. Furthermore, when the mappings are stored, the mapping is 621 actually from the SHA-1 hash of the phone number, to the Node-ID of 622 the call agent which claims ownership of that number. 624 For example, if the Node-ID of the call agent in domain example.net 625 is 0x1234 (a shorter 16 bit value to simplify discussion), the 626 entries stored into the DHT by example.net would be: 628 Key | Value 629 ---------------------------------- 630 SHA1(+14085555000) | 0x1234 631 SHA1(+14085555001) | 0x1234 632 SHA1(+14085555002) | 0x1234 633 ..... 634 SHA1(+14085555999) | 0x1234 636 Figure 2: DHT Contents 638 It is important to note that the DHT does not contain phone numbers 639 (it contains hashes of them), nor does it contain IP addresses or 640 domain names. Instead, it is a mapping from the hash of a phone 641 number (in E.164 format) to a Node-ID. 643 example.net will store this mapping when it starts up, or when a new 644 number is provisioned. The information is refreshed periodically by 645 example.net. The actual server on which these mappings are stored 646 depends on the variant of the Chord algorithm. Typically, the 647 entries will be uniformly distributed amongst all of the call agents 648 participating in the network. 650 6.3.2. PSTN First Call 652 At some point, a user agent (Alice) in example.com makes a call to +1 653 408 555 5432, which is her colleague Bob. Even though both sides have 654 VIPR, the call takes place over the plain old PSTN, per Figure 3. 655 Alice talks to Bob for a bit, and they hang up. 657 +-------+ +-------+ 658 | Call | | Call | 659 //\\ | Agent | | Agent | //\\ 660 \ / | | | | \ / 661 \/ ---| | +-----------+ | |---- \/ 662 Alice | |<=======<========>======>| | Bob 663 +-------+ | PSTN | +-------+ 664 example.com | | example.net 665 +-----------+ 667 Figure 3: PSTN First Call 669 At a random point in time after the call has completed, the call 670 agent in example.com "wakes up" and says to itself, "that's 671 interesting, someone in my domain called +1 408 555 5432, and it went 672 over the PSTN. I wonder if that number is reachable over IP 673 instead?". To make this determination, it hashes the called phone 674 number, and looks it up in the DHT. It is important to note that 675 this lookup is not at the time of an actual phone call - this lookup 676 process happens outside of any phone call, and is a background 677 process. 679 The query for +1 408 555 5432 will traverse the DHT, and eventually 680 arrive at the node that is responsible for storing the mapping for 681 that number. Typically, that node will not be example.net, but 682 rather one of the other nodes in the network (e.g., example.org). In 683 many cases, the called number will not find a matching mapping in the 684 DHT. This happens when the number that was dialed is not owned by a 685 domain participating in VIPR. When that happens, example.com takes 686 no further action. Next time there is another call to the same 687 number, it will repeat the process and check once more whether the 688 dialed number is in the DHT. 690 In this case, there is a match in the DHT, and example.com learns the 691 Node-ID of example.net. It then proceeds to the validation step per 692 Section 6.3.3. It is also possible that there are multiple matches 693 in the DHT. This can happen if another domain - example.edu for 694 example - also claims ownership of that number. When there are 695 multiple matching results, example.com learns all of them, and 696 performs the validation step with each. 698 6.3.3. Validation and Caching 700 Why not just store the domain in the DHT, instead of the Node-ID? If 701 the domain was stored in the DHT, once example.com performed the 702 lookup, it would immediately learn that the number maps to 703 example.net, and could then make a direct SIP call next time. 705 The main reason this doesn't work is security. The information in 706 the DHT is completely untrusted. There is nothing so far that 707 enables example.com to know that example.net does, in fact, own the 708 phone number in question. Indeed, if multiple domains make a claim 709 on the number, it has no way to know which one (if any) actually owns 710 it. 712 To address this critical problem, VIPR requires a mechanism called 713 phone number validation. Phone number validation is a key concept in 714 VIPR. There are several models for this validation as detailed in 715 [I-D.petithuguenin-vipr-pvp]. The essential idea is that example.com 716 will connect to the example.net server, by asking the DHT to form a 717 connection to example.net's Node-ID. Once connected, example.com 718 demands proof of ownership of the phone number. This proof comes in 719 the form of demonstrated knowledge of the previous PSTN call. When a 720 call was placed from example.com to +1 408 555 5432, the details of 721 that call - including its caller ID, start time, and stop time, 722 create a shared secret referred to as a "ticket", - information that 723 is only known to entities that participated in the call. Thus, to 724 obtain proof that example.net really owns the number in question, 725 example.com will demand a knowledge proof - that example.net is aware 726 of the details of the call. A consequence of this is that the 727 following property is maintained: 729 A domain can only call a specific number over SIP, if it had 730 previously called that exact same number over the PSTN. 732 This property is key in fighting spam and denial-of-service attacks. 733 Because calling numbers on the PSTN costs money - especially 734 international calls - VIPR creates a financial disincentive for 735 spammers. For a spammer to ring every phone in a domain with a SIP 736 call, it must have previously called every number in the domain with 737 a PSTN call, and had a successfully completed call to each and every 738 one of them. [I-D.petithuguenin-vipr-sip-antispam] provides an 739 overview and further details on the security mechanisms for VIPR for 740 mitigation of SPAM. 742 There are a great many details required for this validation protocol 743 to be secured. For example, the mechanism needs to handle the fact 744 that call start and stop times won't exactly match on both sides. It 745 needs to deal with the fact that many calls start on the top of the 746 hour. It needs to deal with the fact that caller ID is not often 747 delivered, and when it is delivered, is not reliable. It needs to 748 deal with the fact that example.com may in fact be the attacker, 749 trying to use the validation protocol to extract the shared secret 750 from example.net. All of this is, in fact, handled by the protocol. 751 The protocol is based on the Secure Remote Password for TLS 752 Authentication (SRP-TLS) [RFC5054], and is described more fully in 753 [I-D.petithuguenin-vipr-pvp]. 755 Towards the end of the validation process, domains example.com and 756 example.net had determined that each was, in fact in possession of 757 the shared secret information about the prior PSTN call. However, 758 neither side has any information about the domain names of the other 759 side. 761 At the end of the validation process, both example.com and 762 example.net have been able to ascertain that the other side did in 763 fact participate in the previous PSTN call. At that point, 764 example.com sends its domain name to example.net as shown in 765 Figure 4. 767 +-------+ +-------+ 768 | Call | | Call | 769 example.org | Agent | | Agent | example.edu 770 | | | | 771 +-------+ +-------+ 772 \ / 773 +----------------------+ \ / 774 | Hi, I am example.com.| \ / 775 | How do I reach you? | \ / 776 +--------------\-------+ //-------\\ 777 \ // \\ 778 +===\======>========>========>=====+ 779 ^ | Internet | | 780 | | | v 781 +-------+ |\\ //| +-------+ 782 | Call |------ \\ _______//------| Call | 783 //\\ | Agent | | Agent | //\\ 784 \ / | | | | \ / 785 \/ ---| | | |---- \/ 786 Alice | | | | Bob 787 +-------+ +-------+ 788 example.com example.net 790 Figure 4: Ticket Validation Step 1 792 Next, the example.net domain generates the ticket. The ticket has 793 three fundamental parts to it: 795 1. The phone number that was just validated - in this case, +1 408 796 555 5432. 797 2. The domain name that the originating side claims it has - 798 example.com in this case. 799 3. A signature generated by example.net, using a key known to itself 800 only, over the other two pieces of information. 802 Then, example.net sends to example.com - all over a secured channel - 803 a SIP URI to use for routing calls to this number, and a ticket, as 804 shown in Figure 5. The ticket is a cryptographic object, opaque to 805 example.com, but used by example.net to allow incoming SIP calls. It 806 is similar in concept to kerberos tickets - it is a grant of access. 807 In this case, it is a grant of access for example.com to call +1 408 808 555 5432, and only +1 408 555 5432. 810 +-------+ +-------+ 811 | Call | | Call | 812 example.org | Agent | | Agent | example.edu 813 | | | | 814 +-------+ +-------+ 815 \ / 816 \ / +------------------------+ 817 \ / | Here is your ticket | 818 \ / | & SIP URI to reach Bob | 819 //-------\\ +----/-------------------+ 820 // \\ / 821 +==========<========<========<===/=+ 822 | | Internet | ^ 823 v | | | 824 +-------+ |\\ //| +-------+ 825 | Call |------ \\ _______//------| Call | 826 //\\ | Agent | | Agent | //\\ 827 \ / | | | | \ / 828 \/ ---| | | |---- \/ 829 Alice | | | | Bob 830 +-------+ +-------+ 831 example.com example.net 833 Figure 5: Ticket Validation Step 2 835 The example.com call agent receives the SIP URI and ticket, and 836 stores both of them in an internal cache. This cache builds up 837 slowly over time, containing the phone number, SIP URI, and ticket, 838 for those numbers which are called by example.com and validated using 839 VIPR. Because the cache entries are only built for numbers which 840 have actually been called by users in the enterprise, the size of the 841 cache self-scales. A call agent supporting only ten users will build 842 up a cache proportional to the volume of numbers called by ten 843 people, whereas a call agent supporting ten thousand users will build 844 up a cache which is typically a thousand times larger. 846 This cache, containing the phone number, SIP URI and ticket will be 847 accessed later when Alice (or another caller from the same call 848 agent) makes another call to Bob, as detailed in Section 6.3.4. 850 6.3.4. SIP Call 852 At some point in the future, another call is made to +1 408 555 5432. 853 The caller could be Alice, or it could be any other user attached to 854 the same call agent. This time, the call agent notes that it has a 855 cached entry (including the SIP URI and ticket) for the number in 856 question. It is possible that there are multiple entries for a given 857 number. For example, both an Enterprise and Service Provider may 858 register the same number in the RELOAD distributed database. It may 859 also be possible to fork a call using the multiple entries . 860 [Editor's note: this requires further discussion as to whether we 861 want to allow multiple entries.] 863 The example.com call agent attempts to contact the SIP URI by 864 establishing a TCP/TLS connection to the SIP URI it learned. If a 865 connection cannot be made and there are no other cached entries for 866 the number in question, the call agent proceeds with the call over 867 the PSTN. This ensures that, in the event of an Internet failure or 868 server failure, the call can still proceed. Assuming the connection 869 is established, the example.com call agent sends a SIP INVITE to the 870 terminating call agent, over this newly formed secure connection. 871 The SIP INVITE request also contains the ticket, placed into a new 872 SIP header field in the message. 874 When the SIP INVITE arrives at the example.net call agent, the call 875 agent can extract the ticket from the new SIP header field. This 876 ticket is an object, opaque to example.com, that was previously 877 generated by the example.net call agent as described in 878 Section 6.3.3. example.net first verifies the signature over the 879 ticket. Remember that the example.net agent is the one that 880 generated the ticket in the first place; as such, it is in possession 881 of the key required to validate the signature. Once validated, it 882 performs two checks: 884 1. It compares the phone number in the call setup request (the 885 Request URI) against the phone number stored in the ticket. 886 2. It compares the domain name of the calling domain, learned from 887 the certificates in the mutual TLS exchange, against the domain 888 name stored in the ticket. 890 If both match, the example.net call agent knows that the calling 891 party is in fact the domain they claimed previously, and that they 892 had in fact gone through the validation process successfully for the 893 number in question. At this time, the call is now completed per 894 normal SIP processing. 896 7. Security Considerations 898 This section provides an overview of some of the key threats and how 899 they are handled at a high level. Note that the detailed security 900 solutions to handle the threats are detailed in the other relevant 901 VIPR documents as referenced in the sections below. 903 7.1. Attacks on the DHT 905 Attackers could attempt to disrupt service through a variety of 906 attacks on the DHT. 908 Firstly, it must be noted that the DHT is never used at call setup 909 time. It is accessed as a background task, solely to learn NEW 910 numbers and SIP URIs that are not already known. If an attacker was 911 able to completely destroy the P2P network, it would not result in a 912 single call to fail. Furthermore, it would not cause calls to revert 913 to the PSTN - calls to SIP URIs learned previously would still go 914 over the IP network. The only impact to such a devastating attack is 915 that a domain could not learn SIP URIs for new numbers, until the DHT 916 is restored to service. This service failure is hard for users and 917 administrators to even notice. 919 That said, VIPR prevents many of these attacks. The DHT itself is 920 secured using TLS - its usage is mandatory. Quota mechanisms are put 921 into place that prevent an attacker from storing large amounts of 922 data in the DHT as described in 923 [I-D.petithuguenin-vipr-proportional-quota]. Other attacks are 924 prevented by mechanisms defined by RELOAD [I-D.ietf-p2psip-base] 925 itself, and are not VIPR specific. 927 7.2. Theft of Phone Numbers 929 A key security threat that VIPR is trying to address is the theft of 930 phone numbers. In particular, a malicious domain could store, in the 931 DHT, phone numbers that it does not own, in an attempt to steal calls 932 targeted to those numbers. This attack is prevented by the core 933 validation mechanism as described in [I-D.petithuguenin-vipr-pvp] , 934 which performs a proof of knowledge check to verify ownership of 935 numbers. 937 An attacker could try to claim numbers it doesn't own, which are 938 claimed legitimately by other domains in the VIPR network. This 939 attack is prevented as well. Each domain storing information into 940 the DHT can never overwrite information stored by another domain. As 941 a consequence, if two domains claim the same number, two records are 942 stored in the DHT. An originating domain will validate against both, 943 and only one will validate - the real owner. 945 An attacker could actually own a phone number, use it for a while, 946 validate with it, and build up a cache of routes at other domains. 947 Then, it gives back the phone number to the PSTN provider, who 948 allocates it to someone else. However, the attacker still claims 949 ownership of the number, even though they no longer have it. This 950 attack is prevented by expiring the learned routes after a while. 951 Typically, operators do not re-assign a number for a few months, to 952 allow out-of-service messages to be played to people that still have 953 the old number. Thus, the TTL for cached routes is set to match the 954 duration that carriers typically hold numbers. 956 An attacker could advertise a lot of numbers, most of which are 957 correct, some of which are not. VIPR prevents this by requiring each 958 number to be validated individually. 960 An attacker could make a call so they know the call details of the 961 call they made and use this to forge a validation for that call. 962 They could then try to convince other users, which would have to be 963 in the same domain as the attacker, to trust this validation. This 964 is mitigated by not sharing validations inside of domains where the 965 users that can originate call from that domain are not trusted by the 966 domain. 968 7.3. Spam 970 Another serious concern is that attackers may try to launch SIP spam 971 (also known as SPIT) calls into a domain. As described in 972 Section 6.3.3 and as detailed in 973 [I-D.petithuguenin-vipr-sip-antispam], VIPR prevents this by 974 requiring that a domain make a PSTN call to a number before it will 975 allow a SIP call to be accepted to that same number. This provides a 976 financial disincentive to spammers. The current relatively high cost 977 of international calling, and the presence of national do-not-call 978 regulations, have prevented spam on the PSTN to a large degree. VIPR 979 applies those same protections to SIP connections. 981 VIPR still lowers the cost of communications, but it does so by 982 amortizing that savings over a large number of calls. The costs of 983 communications remain high for infrequent calls to many numbers, and 984 become low for frequent calls to a smaller set of numbers. Since the 985 former is more interesting to spammers, VIPR gears its cost 986 incentives away from the spammers, and towards domains which 987 collaborate frequently. 989 It is important to note that VIPR does not completely address the 990 spam problem. A large spamming clearing house organization could 991 actually incur the costs of launching the PSTN calls to numbers, and 992 then, in turn, act as a conduit allowing other spammers to launch 993 their calls to those numbers for a fee. The clearinghouse would 994 actually need to transit the signaling traffic (or, divulge the 995 private keys to their domain name), which would incur some cost. As 996 such, while this is not an impossible situation, the barrier is set 997 reasonably high to start with - high enough that it is likely to 998 deter spammers until it becomes a highly attractive target, at which 999 point other mechanisms can be brought to bear. 1001 7.4. Eavesdropping 1003 Another class of attacks involves outsiders attempting to listen in 1004 on the calls that run over the Internet, or obtain information about 1005 the call through observation of signaling. 1007 All of these attacks are prevented by requiring the usage of SIP over 1008 TLS and SRTP. These are mandatory to use. 1010 8. IANA Considerations 1012 This specification does not require any actions from IANA. 1014 9. Acknowledgements 1016 Thanks for review comments from Ken Fischer, Rob Maidhof, Michael 1017 Procter, Eric Burger, Richard Barnes and others. Thanks to Theo 1018 Zourzouvillys for pointing out the 5th theft of phone numbers attack 1019 as described in Section 7.2 . 1021 10. References 1022 10.1. Normative References 1024 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1025 Requirement Levels", BCP 14, RFC 2119, March 1997. 1027 [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, 1028 A., Peterson, J., Sparks, R., Handley, M., and E. 1029 Schooler, "SIP: Session Initiation Protocol", RFC 3261, 1030 June 2002. 1032 [I-D.ietf-p2psip-base] 1033 Jennings, C., Lowekamp, B., Rescorla, E., Baset, S., and 1034 H. Schulzrinne, "REsource LOcation And Discovery (RELOAD) 1035 Base Protocol", draft-ietf-p2psip-base-20 (work in 1036 progress), January 2012. 1038 [I-D.petithuguenin-vipr-reload-usage] 1039 Petit-Huguenin, M., Rosenberg, J., and C. Jennings, "A 1040 Usage of Resource Location and Discovery (RELOAD) for 1041 Public Switched Telephone Network (PSTN) Verification", 1042 draft-petithuguenin-vipr-reload-usage-03 (work in 1043 progress), October 2011. 1045 [I-D.petithuguenin-vipr-framework] 1046 Petit-Huguenin, M., Jennings, C., and J. Rosenberg, 1047 "Verification Involving PSTN Reachability (VIPR): 1048 Framework", draft-petithuguenin-vipr-framework-00 (work in 1049 progress), October 2011. 1051 [I-D.petithuguenin-vipr-sip-antispam] 1052 Petit-Huguenin, M., Rosenberg, J., and C. Jennings, 1053 "Session Initiation Protocol (SIP) Extensions for Blocking 1054 VoIP Spam Using PSTN Validation", 1055 draft-petithuguenin-vipr-sip-antispam-03 (work in 1056 progress), January 2012. 1058 [I-D.jennings-vipr-vap] 1059 Jennings, C., Rosenberg, J., and M. Petit-Huguenin, 1060 "Verification Involving PSTN Reachability: The ViPR Access 1061 Protocol (VAP)", draft-jennings-vipr-vap-02 (work in 1062 progress), March 2012. 1064 [I-D.petithuguenin-vipr-pvp] 1065 Petit-Huguenin, M., Rosenberg, J., and C. Jennings, "The 1066 Public Switched Telephone Network (PSTN) Validation 1067 Protocol (PVP)", draft-petithuguenin-vipr-pvp-03 (work in 1068 progress), February 2012. 1070 [I-D.petithuguenin-vipr-proportional-quota] 1071 Petit-Huguenin, M., Rosenberg, J., and C. Jennings, 1072 "Proportional Quota in REsource LOcation And Discovery 1073 (RELOAD)", draft-petithuguenin-vipr-proportional-quota-00 1074 (work in progress), October 2011. 1076 10.2. Informative References 1078 [RFC2543] Handley, M., Schulzrinne, H., Schooler, E., and J. 1079 Rosenberg, "SIP: Session Initiation Protocol", RFC 2543, 1080 March 1999. 1082 [RFC3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation 1083 Protocol (SIP): Locating SIP Servers", RFC 3263, 1084 June 2002. 1086 [E.164] ITU-T, "The International Public Telecommunication Number 1087 Plan", Recommendation E.164, May 1997. 1089 [RFC5039] Rosenberg, J. and C. Jennings, "The Session Initiation 1090 Protocol (SIP) and Spam", RFC 5039, January 2008. 1092 [RFC6116] Bradner, S., Conroy, L., and K. Fujiwara, "The E.164 to 1093 Uniform Resource Identifiers (URI) Dynamic Delegation 1094 Discovery System (DDDS) Application (ENUM)", RFC 6116, 1095 March 2011. 1097 [RFC5067] Lind, S. and P. Pfautz, "Infrastructure ENUM 1098 Requirements", RFC 5067, November 2007. 1100 [RFC5054] Taylor, D., Wu, T., Mavrogiannopoulos, N., and T. Perrin, 1101 "Using the Secure Remote Password (SRP) Protocol for TLS 1102 Authentication", RFC 5054, November 2007. 1104 [I-D.ietf-sipclf-format] 1105 Salgueiro, G., Gurbani, V., and A. Roach, "Format for the 1106 Session Initiation Protocol (SIP) Common Log Format 1107 (CLF)", draft-ietf-sipclf-format-06 (work in progress), 1108 March 2012. 1110 [I-D.jones-ipmc-session-id-reqts] 1111 Salgueiro, G., Kaplan, H., Polk, J., Liess, L., R, P., 1112 Jones, P., Jesske, R., and S. Loreto, "Requirements for an 1113 End-to-End Session Identification in IP-Based Multimedia 1114 Communication Networks", 1115 draft-jones-ipmc-session-id-reqts-01 (work in progress), 1116 January 2012. 1118 Appendix A. Changes since last version 1120 This section must be removed before publication as an RFC. 1122 Modifications between jennings-03 and jennings-02: 1124 1. Reworded REQ -11 to clarify that in the case of call failures 1125 (i.e., IP calls), the system should fallback to inter-domain 1126 calling prior to VIPR. 1127 2. Deleted REQ-12 (Handover) since it's really not specific 1128 functionality provided by VIPR. 1129 3. Moved some text from the -01 version in the Technical Overview 1130 section back into the doc (not sure why it was removed 1131 previously). 1132 4. Other editorial changes: 1134 - Added a Terminology section. 1135 - Clarified the use of the term "Call Agent". 1136 - Reworded discussion of email in section 2.2 (i.e., it's not 1137 useless). 1138 - Either changed or removed altogether terms like "neat", 1139 "clever", "incredible", "enormous" and any text that read like 1140 marketing literature as much as possible. 1141 - Removed some of the more subjective and superfluous language - 1142 i.e., condensed the text to be more concise (Section 5.2 and many 1143 others per the previous change) 1144 - Deleted explicit reference to "SIP Trunking" as the statement 1145 didn't introduce additional information in that paragraph and the 1146 term is not defined in this document. 1147 - and other minor editorial fixes. 1149 Modifications between jennings-02 and jennings-01: 1151 1. Sections 6,7,8 moved to new VIPR framework document. 1152 2. Editorial changes. 1153 3. Clarifications to re-enforce that the primary objective is not 1154 PSTN bypass but rather to enable enhanced services such as video 1155 between domains. Changed "VoIP" to "SIP" since the focus is not 1156 specifically voice. 1157 4. Added reference for new framework document. 1158 5. Section 5.3: Added references to other documents as appropriate 1159 - e.g., -pvp, -spam, etc. 1160 6. Moved validation diagrams and text (from 5.3.4) into Validation 1161 and caching section (5.3.3). 1162 7. Condensed discussion of spam in section 5.3.3 and updated SPAM 1163 section in security section. 1165 Modifications between jennings-01 and rosenberg-04: 1167 o Not specified. 1169 Modifications between rosenberg-04 and rosenberg-03 1171 o Nits. 1172 o Shorter I-Ds references. 1173 o Changed phone numbers to follow E.123 presentation. 1174 o Expanded P2P initialisms. 1175 o Uses +1 408 555 prefix for phone numbers in examples. 1177 Authors' Addresses 1179 Mary Barnes 1180 Polycom 1181 TX 1182 US 1184 Email: mary.ietf.barnes@gmail.com 1186 Cullen Jennings 1187 Cisco 1188 170 West Tasman Drive 1189 MS: SJC-21/2 1190 San Jose, CA 95134 1191 USA 1193 Phone: +1 408 421-9990 1194 Email: fluffy@cisco.com 1196 Jonathan Rosenberg 1197 jdrosen.net 1198 Monmouth, NJ 1199 US 1201 Email: jdrosen@jdrosen.net 1202 URI: http://www.jdrosen.net 1204 Marc Petit-Huguenin 1205 Unaffiliated 1207 Email: petithug@acm.org