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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 1 Internet Draft Alan Johnston 2 Document: draft-johnston-sip-call-flows-00.txt Steve Donovan 3 Category: Informational Robert Sparks 4 Chris Cunningham 5 Kevin Summers 6 MCI WorldCom 7 October 1999 9 SIP Telephony Call Flow Examples 11 Status of this Memo 13 This document is an Internet-Draft and is in full conformance with 14 all provisions of Section 10 of RFC2026[1]. 16 Internet-Drafts are working documents of the Internet Engineering 17 Task Force (IETF), its areas, and its working groups. Note that other 18 groups may also distribute working documents as Internet-Drafts. 19 Internet-Drafts are draft documents valid for a maximum of six months 20 and may be updated, replaced, or obsoleted by other documents at any 21 time. It is inappropriate to use Internet- Drafts as reference 22 material or to cite them other than as "work in progress." 23 The list of current Internet-Drafts can be accessed at 24 http://www.ietf.org/ietf/1id-abstracts.txt 25 The list of Internet-Draft Shadow Directories can be accessed at 26 http://www.ietf.org/shadow.html. 28 Abstract 30 This document gives examples of SIP IP Telephony call flows. 31 Elements in these call flows include SIP User Agents and Clients, SIP 32 Proxy and Redirect Servers, and Gateways to the PSTN (Public Switch 33 Telephone Network). Scenarios include SIP Registration, SIP to SIP 34 calling, SIP to Gateway, Gateway to SIP, and Gateway to Gateway via 35 SIP. Call flow diagrams and message details are shown. PSTN 36 telephony protocols are illustrated using SS7 (Signaling System 7), 37 ISDN (Integrated Services Digital Network) and FGB (Feature Group B) 38 circuit associated signaling. PSTN calls are illustrated using 39 global telephone numbers from the PSTN and from private extensions 40 served on a PBX (Private Branch Exchange). 42 Table of Contents 44 1 Overview...................................................3 45 1.1 General Assumptions........................................3 46 1.2 Legend for Message Flows...................................5 47 1.3 SIP Protocol Assumptions...................................6 48 2 SIP Registration Services..................................7 49 2.1 Success Scenarios..........................................7 50 2.1.1 SIP Client New Registration................................7 51 2.1.2 User Cancels Registration..................................9 52 2.1.3 User updates contact list.................................10 53 2.1.4 User Requests Current Contact List........................12 54 2.2 Failure Scenarios.........................................13 55 2.2.1 Unsuccessful SIP registration.............................13 56 3 SIP to SIP Dialing........................................15 57 3.1 Success Scenarios.........................................15 58 3.1.1 Successful SIP to SIP through two proxies.................16 59 3.1.2 Successful SIP to SIP with Proxy failure..................26 60 3.1.3 Successful SIP to SIP through SIP Firewall Proxy..........34 61 3.1.4 Successful SIP to SIP via Redirect and Proxy..............43 62 3.2 Failure Scenarios.........................................50 63 3.2.1 Unsuccessful SIP to SIP no answer.........................50 64 3.2.2 Unsuccessful SIP to SIP busy..............................56 65 3.2.3 Unsuccessful SIP to SIP no response.......................60 66 3.2.4 Unsuccessful SIP to SIP Temporarily Unavailable...........65 67 4 SIP to Gateway Dialing....................................71 68 4.1 Success Scenarios.........................................71 69 4.1.1 Successful SIP to ISUP PSTN call..........................72 70 4.1.2 Successful SIP to ISDN PBX call...........................79 71 4.1.3 Successful SIP to ISUP PSTN call with overflow............87 72 4.2 Failure Scenarios.........................................95 73 4.2.1 Unsuccessful SIP to PSTN call: Treatment from PSTN........96 74 4.2.2 Unsuccessful SIP to PSTN: REL w/Cause from PSTN..........101 75 4.2.3 Unsuccessful SIP to PSTN: ANM Timeout....................105 76 5 Gateway to SIP Dialing...................................111 77 5.1 Success Scenarios........................................111 78 5.1.1 Successful PSTN to SIP call..............................112 79 5.1.2 Successful PSTN to SIP call, Fast Answer.................118 80 5.1.3 Successful PBX to SIP call...............................123 81 5.2 Failure Scenarios........................................128 82 5.2.1 Unsuccessful PSTN to SIP REL, SIP error mapped to REL....128 83 5.2.2 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL.....130 84 5.2.3 Unsuccessful PSTN->SIP, SIP error interworking to tones..134 85 5.2.4 Unsuccessful PSTN->SIP, ACM timeout......................139 86 5.2.5 Unsuccessful PSTN->SIP, ACM timeout, stateless SPS.......144 87 5.2.6 Unsuccessful PSTN->SIP, ANM timeout......................150 88 6 Gateway to Gateway Dialing via SIP Network...............155 89 6.1 Success Scenarios........................................155 90 6.1.1 Successful ISUP PSTN to ISUP PSTN call...................156 91 6.1.2 Successful FGB PBX to ISDN PBX call with overflow........163 93 7 Acknowledgements.........................................172 94 8 References...............................................172 96 1 Overview 98 The call flows shown in this document were developed in the design of 99 a carrier-class SIP IP Telephony network. They represent an example 100 minimum set of functionality for SIP to be used in IP Telephony 101 applications. 103 It is the hope of the authors that this document will be useful for 104 SIP implementors, designers, and protocol researchers alike and will 105 help further the goal of a standard SIP implementation for IP 106 Telephony. It is envisioned that as changes to the standard and 107 additional RFCs are added that this document will reflect those 108 changes and represent the current state of a standard interoperable 109 SIP IP Telephony implementation. 111 These call flows are based on the current version 2.0 of SIP in 112 RFC2543[2]. Additions and changes to SIP necessary for PSTN 113 interworking are referenced as IETF Internet-Drafts as they are used 114 in the call flows. 116 Various PSTN signaling protocols are illustrated in this document: 117 SS7 (Signaling System 7), ISDN (Integrated Services Digital Network), 118 and FGB (Feature Group B) circuit associated signaling. They were 119 chosen to illustrate the nature of SIP/PSTN interworking _ they are 120 not a complete or even representative set. Also, some details and 121 parameters of these PSTN protocols have been omitted. The intent of 122 this document was not to provide a complete and exact mapping of PSTN 123 protocols to SIP. Rather the emphasis is on the SIP signaling, the 124 message interaction, and the modifications to SIP currently proposed 125 to solve IP Telephony issues. 127 Finally, these call flows show a minimal implementation. Not even 128 basic telephony features such as call forwarding or call waiting are 129 included. A typical carrier-class implementation of a basic set of 130 telephony features using SIP is described in another document[3]. 132 1.1 General Assumptions 134 A number of architecture, network, and protocol assumptions underlie 135 the call flows in this document. They are outlined in this section 136 so that they may be taken into consideration. Differences in these 137 assumptions will affect the nature of the call flows. 139 The authentication of SIP User Agents in these example call flows is 140 performed using SIP Digest. 142 No authentication of Gateways is shown, since it is assumed that: 144 . Gateways will only accept calls routed through a trusted Proxy. 145 . Proxies will perform the Client authentication. 146 . The Proxy and the Gateway will authenticate each other using 147 IPSec[4]. 149 The SIP Proxy Server has access to a Location Manager and other 151 databases. Information present in the Request-URI and the context 152 (From header) is sufficient to determine to which proxy or gateway 153 the message should be routed. In most cases, a primary and secondary 154 route will be determined in case of Proxy or Gateway failure 155 downstream. 157 The Proxy Servers in these call flows insert Record-Route headers 159 into requests to ensure that they are in the signaling path for 160 future message exchanges. 162 Gateways receive enough information in the Request-URI field to 163 determine how to route a call, i.e. what trunk group or link to 164 select, what digits to outpulse, etc. 166 Gateways provide tones (ringing, busy, etc) and announcements to the 168 PSTN side based on SIP response messages, or pass along audio in-band 169 tones (ringing, busy tone, etc.) in an early media stream to the SIP 170 side. 172 Two types of Gateways are described in this document: 174 . Network Gateway. This high port count PSTN gateway originates 175 and terminates calls to the PSTN. It's use is shared by many 176 customers. Incoming calls from the PSTN have the From header 177 populated with a SIP URL containing the telephone number from 178 the calling party telephone number, if available. A Network 179 Gateway typically uses carrier protocols such as SS7. 181 . Enterprise Gateway. This low port count PBX (Private Branch 182 Exchange) gateway has trunks or lines for a single customer or 183 user. Incoming calls from the PBX have the From header 184 populated with a provisionable string which uniquely identifies 185 the customer, trunk group, or carrier. This allows private 186 numbers to be interpreted in their correct context. An 187 Enterprise Gateway typically uses SS7, ISDN, circuit associated 188 signaling, or other PBX interfaces. 190 The interactions between the Proxy and Gateway can be summarized as 192 follows: 194 . The SIP Proxy Server performs digit analysis and lookup and 195 locates the correct gateway. 197 . The SIP Proxy Server performs gateway location based on primary 198 and secondary routing. 200 Digit handling by the Gateways will be as follows: 202 . Dialed digits received from a Network or Enterprise Gateway will 203 be put in a SIP URL with a telephone number. The number will 204 either be globalized (e.g. sip:+1-314-555-1111@ngw.wcom.com 205 ;user=phone) or left as a private number (sip:555-6666,phone- 206 context=p1234@gw.wcom.com;user=phone) which will require 207 interpretation based on From header. The "phone-context=" 208 qualifier is used to interpret the private number. It is used 209 the same as the tag of the same name from the Tel URL draft[5]. 210 However, its use in the user portion of the SIP URL should not 211 require changes to parsers. All Gateways will need to be 212 provisioned to be able to parse the user portion of a Request- 213 URI to determine the customer, trunk group, or circuit 214 referenced. 216 . The From header will be populated with a SIP URL with a 217 telephone number if it is Calling Party number (CgPN) from the 218 PSTN. If it is an Enterprise Gateway, a provisionable string 219 which uniquely identifies the customer, trunk group, or carrier 220 will be used in the sip URI (e.g. From: 221 sip:ProvisionableString@gw1.wcom.com ;user=phone). 223 . Note that an alternative to using a SIP URL for telephone 224 numbers is the TEL URL[5]. The major difference between using 225 the SIP URL and the TEL URL is that the SIP URL is routable in a 226 SIP network (resolves down to an IP address) where the TEL URL 227 is not (it just represents digits). For example, a SIP URL can 228 be used in a Contact header, but a TEL URL can not. 230 These flows show UDP for transport. TCP could also be used. 232 1.2 Legend for Message Flows 234 Important Note: In this text version of this Internet Draft, figures 235 containing the message flows have been deleted. They will be present 236 in the next draft of this document. A PostScript (.ps) and Acrobat 237 (.pdf) version are available which contain the figures. 239 Dashed lines _ represent control messages that are mandatory to the 240 call scenario. These control messages can be SIP or PSTN signaling. 242 Solid lines _ represent media paths between network elements. 244 Dashed line with parenthesis around name - represent optional control 245 messages. 247 Messages are identified in the Figures as F1, F2, etc. This 248 references the message details in the table that follows the Figure. 249 Comments in the message details are shown in the following form: 251 /* Comments. */ 253 1.3 SIP Protocol Assumptions 255 Except for the following, this call flows document uses the April 256 1999 version 2.0 of SIP defined by RFC2543[2]. The following 257 changes/extensions are assumed throughout: 259 . A Contact header is included with every INVITE message. 261 . A Contact header is included in every 200 OK Response. 263 . The 183 Session Progress response message[5] is used in SIP to 264 Gateway and Gateway to Gateway via SIP calling (Sections 4 and 265 6). The 183 response with SDP will cause the User Agent to 266 immediately play the SDP media stream to hear in-band call 267 progress information. See Section 4 for more information. 269 . A Content-Length header is present in every message, set to zero 270 if there is no message body. 272 . The final entry in a Route header is always the Contact 273 information obtained from the INVITE or 200 OK messages. 275 . In the SDP message bodies, the time field is "t=0 0" It is 276 expected that an actual SDP message body would have a non-zero 277 start timestamp. 279 2 SIP Registration Services 281 2.1 Success Scenarios 283 Registration either validates or invalidates a SIP client for user 284 services provided by the SIP proxy and/or SIP server. Additionally, 285 the client provides one or more contact locations to the SIP server 286 with the registration request. 288 2.1.1 289 SIP Client New Registration 291 User B initiates a new SIP session with the SIP server (i.e. the user 292 "logs on to" the SIP server). User B sends a SIP REGISTER request to 293 the SIP server. The request includes the user's contact list. The 294 SIP server provides a challenge to User B. User B enters her/his 295 valid user ID and password. User B's SIP client encrypts the user 296 information according to the challenge issued by the SIP server and 297 sends the response to the SIP server. The SIP server validates the 298 user's credentials. It registers the user in its contact database 299 and returns a response (200 OK) to User B's SIP client. The response 300 includes the user's current contact list in Contact headers. The 301 format of the authentication shown is SIP digest as described by 302 RFC2543[2]. 304 Message Details 306 REGISTER F1 307 B->SIP server 309 REGISTER sip:ss2.wcom.com SIP/2.0 310 Via: SIP/2.0/UDP there.com:5060 311 From: TheLittleGuy 312 To: TheLittleGuy 313 Call-ID: 123456789@here.com 314 CSeq: 1 REGISTER 315 Contact: TheLittleGuy 316 Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone 317 Contact: tel:+1-972-555-2222 318 Content-Length: 0 320 Unauthorized F2 321 SIP server-> User B 323 SIP/2.0 401 Unauthorized 324 Via: SIP/2.0/UDP there.com:5060 325 From: TheLittleGuy 326 To: TheLittleGuy 327 Call-ID: 123456789@here.com 328 CSeq: 1 REGISTER 329 WWW-Authenticate: Digest realm="MCI WorldCom SIP", domain="wcom.com", 330 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", stale="FALSE", 331 algorithm="MD5" 332 Content-Length: 0 334 REGISTER F3 335 B->SIP server 337 REGISTER sip:ss2.wcom.com SIP/2.0 338 Via: SIP/2.0/UDP there.com:5060 339 From: TheLittleGuy 340 To: TheLittleGuy 341 Call-ID: 123456790@here.com 342 CSeq: 1 REGISTER 343 Contact: TheLittleGuy 344 Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone 345 Contact: tel:+1-972-555-2222 346 Authorization:Digest username="UserB", realm="MCI WorldCom SIP", 347 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 348 uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc" 349 Content-Length: 0 351 200 OK F4 352 SIP server 353 -> B 355 SIP/2.0 200 OK 356 Via: SIP/2.0/UDP there.com:5060 357 From: TheLittleGuy 358 To: TheLittleGuy 359 Call-ID: 1234567890@here.com 360 CSeq: 1 REGISTER 361 Contact: TheLittleGuy 362 Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone 363 Contact: tel:+1-972-555-2222 364 Content-Length: 0 366 2.1.2 367 User Cancels Registration 369 User B wishes to cancel her/his registration with the SIP 370 registrar/redirect server. User B sends a SIP REGISTER request to the 371 SIP server. The request has an expiration period of 0 and applies to 372 all existing contact locations. Since the user already has 373 authenticated with the server, the user supplies authentication 374 credentials with the request and is not challenged by the server. 375 The SIP server validates the user's credentials. It clears the 376 user's contact list, and returns a response (200 OK) to User B's SIP 377 client. 379 Message Details 381 REGISTER F1 382 B->SIP server 384 REGISTER sip:ss2.wcom.com SIP/2.0 385 Via: SIP/2.0/UDP there.com:5060 386 From: TheLittleGuy 387 To: TheLittleGuy 388 Call-ID: 123456791@here.com 389 CSeq: 1 REGISTER 390 Expires: 0 391 Contact: * 392 Authorization:Digest username="UserB", realm="MCI WorldCom SIP", 393 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 394 uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc" 395 Content-Length: 0 397 200 OK F2 398 SIP server 399 -> B 401 SIP/2.0 200 OK 402 Via: SIP/2.0/UDP there.com:5060 403 From: TheLittleGuy 404 To: TheLittleGuy 405 Call-ID: 1234567891@here.com 406 CSeq: 1 REGISTER 407 Content-Length: 0 409 2.1.3 410 User updates contact list 412 User B wishes to update the list of addresses where the SIP server 413 will redirect INVITE requests. Note this scenario assumes that 414 Scenario 2.1.1 has taken place, but 2.1.2 has not (i.e. User B 415 currently has 3 contacts registered with SS2.) 417 User B sends a SIP REGISTER request to the SIP server. User B's 418 request includes an updated contact list. Since the user already has 419 authenticated with the server, the user supplies authentication 420 credentials with the request and is not challenged by the server. 421 The SIP server validates the user's credentials. It registers the 422 user in its contact database, updates the user's contact list, and 423 returns a response (200 OK) to User B's SIP client. The response 424 includes the user's current contact list in Contact headers. 426 Message Details 428 REGISTER F1 429 B->SIP server 431 REGISTER sip:ss2.wcom.com SIP/2.0 432 Via: SIP/2.0/UDP there.com:5060 433 From: TheLittleGuy 434 To: TheLittleGuy 435 Call-ID: 123456791@here.com 436 CSeq: 1 REGISTER 437 Contact: mailto:UserB@there.com 438 Authorization:Digest username="UserB", realm="MCI WorldCom SIP", 439 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 440 uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc" 441 Content-Length: 0 443 200 OK F2 444 SIP server 445 -> B 447 SIP/2.0 200 OK 448 Via: SIP/2.0/UDP there.com:5060 449 From: TheLittleGuy 450 To: TheLittleGuy 451 Call-ID: 1234567891@here.com 452 CSeq: 1 REGISTER 453 Contact: TheLittleGuy 454 Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone 455 Contact: tel:+1-972-555-2222 456 Contact: mailto:UserB@there.com 457 Content-Length: 0 459 2.1.4 460 User Requests Current Contact List 462 User B sends a register request to the Proxy serverer containing no 463 Contact headers, indicating the user wishes to query the server for 464 the user's current contact list. Since the user already has 465 authenticated with the server, the user supplies authentication 466 credentials with the request and is not challenged by the server. The 467 SIP server validates the user's credentials. It registers the user 468 in its contact database and returns a response (200 OK) to User B's 469 SIP client. The response includes the user's current contact list in 470 Contact headers. 472 Message Details 474 REGISTER F1 475 B->SIP server 477 REGISTER sip:ss2.wcom.com SIP/2.0 478 Via: SIP/2.0/UDP there.com:5060 479 From: TheLittleGuy 480 To: TheLittleGuy 481 Call-ID: 123456792@here.com 482 CSeq: 1 REGISTER 483 Authorization:Digest username="UserB", realm="MCI WorldCom SIP", 484 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 485 uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc" 486 Content-Length: 0 488 200 OK F2 489 SIP server 490 -> B 492 SIP/2.0 200 OK 493 Via: SIP/2.0/UDP there.com:5060 494 From: TheLittleGuy 495 To: TheLittleGuy 496 Call-ID: 1234567892@here.com 497 CSeq: 1 REGISTER 498 Contact: TheLittleGuy 499 Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone 500 Contact: tel:+1-972-555-2222 501 Contact: mailto:UserB@there.com 502 Content-Length: 0 504 2.2 Failure Scenarios 506 2.2.1 507 Unsuccessful SIP registration 509 User B sends a SIP REGISTER request to the SIP server. The SIP 510 server provides a challenge to User B. User B enters her/his user ID 511 and password. User B's SIP client encrypts the user information 512 according to the challenge issued by the SIP server and sends the 513 response to the SIP server. The SIP server attempts to validate the 514 user's credentials, but they are not valid (the user's password does 515 not match the password established for the user's account). The 516 server returns a response (401 Unauthorized) to User B's SIP client. 518 Message Details 520 REGISTER F1 521 B->SIP server 523 REGISTER sip:ss2.wcom.com SIP/2.0 524 Via: SIP/2.0/UDP there.com:5060 525 From: TheLittleGuy 526 To: TheLittleGuy 527 Call-ID: 123456789@here.com 528 CSeq: 1 REGISTER 529 Contact: TheLittleGuy 530 Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone 531 Contact: tel:+1-972-555-2222 532 Content-Length: 0 534 Unauthorized F2 535 SIP server-> User B 537 SIP/2.0 401 Unauthorized 538 Via: SIP/2.0/UDP there.com:5060 539 From: TheLittleGuy 540 To: TheLittleGuy 541 Call-ID: 123456789@here.com 542 CSeq: 1 REGISTER 543 WWW-Authenticate: Digest realm="MCI WorldCom SIP", domain="wcom.com", 544 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", stale="FALSE", 545 algorithm="MD5" 546 Content-Length: 0 548 REGISTER F3 549 B->SIP server 551 REGISTER sip:ss2.wcom.com SIP/2.0 552 Via: SIP/2.0/UDP there.com:5060 553 From: TheLittleGuy 554 To: TheLittleGuy 555 Call-ID: 123456791@here.com 556 CSeq: 1 REGISTER 557 Contact: TheLittleGuy 558 Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone 559 Contact: tel:+1-972-555-2222 560 Authorization:Digest username="UserB", realm="MCI WorldCom SIP", 561 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 562 uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc" 563 Content-Length: 0 565 Note: The response above encodes the incorrect password _IForgot_ 566 Unauthorized F4 567 SIP server-> User B 569 SIP/2.0 401 Unauthorized 570 Via: SIP/2.0/UDP there.com:5060 571 From: TheLittleGuy 572 To: TheLittleGuy 573 Call-ID: 1234567891@here.com 574 CSeq: 1 REGISTER 575 WWW-Authenticate: Digest realm="MCI WorldCom SIP", domain="wcom.com", 576 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", stale="FALSE", 577 algorithm="MD5" 578 Content-Length: 0 580 3 SIP to SIP Dialing 582 3.1 Success Scenarios 584 This section details calls between two SIP User Agent Clients (UACs) 585 _ User A and User B. User A (TheLittleGuy sip:UserA@here.com) and 586 User B (TheBigGuy sip:UserB@there.com) are assumed to be SIP phones 587 or SIP-enabled devices. Calls route using at least one SIP Proxy 588 server. The successful calls show the initial signaling, the 589 exchange of media information in the form of SDP payloads, the 590 establishment of the media session, then finally the termination of 591 the call. 593 SIP digest authentication is used by the first Proxy Server to 594 authenticate the caller User A. It is assumed that User B has 595 registered with Proxy Server SS2 as per Section 2.1 to be able to 596 receive the calls. 598 3.1.1 599 Successful SIP to SIP through two proxies 601 In this scenario, User A completes a call to User B using two proxies 602 SS1 and SS2. The initial INVITE (F1) does not contain the 603 Authorization credentials SS1 requires, so a 407 Proxy Authorization 604 response is sent containing the challenge information. A new INVITE 605 (F4) is then sent containing the correct credentials and the call 606 proceeds. The call terminates when User B disconnects by initiating 607 a BYE message. 609 SS1 inserts a Record-Route header into the INVITE message to ensure 610 that it is present in all subsequent message exchanges. SS2 also 611 inserts itself into the Record-Route header. The ACK (F15) and BYE 612 (F18) both have a Route header. 614 A tag is inserted by User B in message F9 since the initial INVITE 615 message contains more than one Via header and may have been forked. 617 Message Details 619 INVITE F1 620 A -> Proxy 1 622 INVITE sip:UserB@ss1.wcom.com SIP/2.0 623 Via: SIP/2.0/UDP here.com:5060 624 From: TheBigGuy 625 To: TheLittleGuy 626 Call-Id: 12345600@here.com 627 CSeq: 1 INVITE 628 Contact: TheBigGuy 629 Content-Type: application/sdp 630 Content-Length: 132 632 v=0 633 o=UserA 2890844526 2890844526 IN IP4 here.com 634 t=0 0 635 c=IN IP4 100.101.102.103 636 m=audio 49170 RTP/AVP 0 637 a=rtpmap:0 PCMU/8000 638 /* Proxy 1 challenges User A for authentication */ 640 407 Proxy Authorization Required F2 641 Proxy 1 -> User A 643 SIP/2.0 407 Proxy Authorization Required 644 Via: SIP/2.0/UDP here.com:5060 645 From: TheBigGuy 646 To: TheLittleGuy 647 Call-Id: 12345600@here.com 648 CSeq: 1 INVITE 649 Proxy-Authenticate: Digest realm="MCI WorldCom SIP", 650 domain="wcom.com", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", 651 opaque="", stale="FALSE", algorithm="MD5" 652 Content-Length: 0 654 ACK F3 655 A -> Proxy 1 657 ACK sip:UserB@ss1.wcom.com SIP/2.0 658 Via: SIP/2.0/UDP here.com:5060 659 From: TheBigGuy 660 To: TheLittleGuy 661 Call-Id: 12345600@here.com 662 CSeq: 1 INVITE 663 Content-Length: 0 665 /* User A responds be re-sending the INVITE with authentication 666 credentials in it. */ 668 INVITE F4 669 A -> Proxy 1 671 INVITE sip:UserB@ss1.wcom.com SIP/2.0 672 Via: SIP/2.0/UDP here.com:5060 673 From: TheBigGuy 674 To: TheLittleGuy 675 Call-Id: 12345601@here.com 676 CSeq: 1 INVITE 677 Contact: TheBigGuy 678 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 679 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 680 uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" 681 Content-Type: application/sdp 682 Content-Length:132 683 v=0 684 o=UserA 2890844526 2890844526 IN IP4 here.com 685 t=0 0 686 c=IN IP4 100.101.102.103 687 m=audio 49170 RTP/AVP 0 688 a=rtpmap:0 PCMU/8000 690 /* Proxy 1 accepts the credentials and forwards the INVITE to Proxy 691 2. Proxy 1 is assumed to have been authenticated by Proxy 2 using 692 IPSec. Client for A prepares to receive data on port 49170 from the 693 network. */ 695 INVITE 696 F5 697 Proxy 698 1 -> 699 Proxy 700 2 702 INVITE sip:UserB@ss2.wcom.com SIP/2.0 703 Via: SIP/2.0/UDP ss1.wcom.com:5060 704 Via: SIP/2.0/UDP here.com:5060 705 Record-Route: 706 From: TheBigGuy 707 To: TheLittleGuy 708 Call-Id: 12345601@here.com 709 CSeq: 1 INVITE 710 Contact: TheBigGuy 711 Content-Type: application/sdp 712 Content-Length:132 714 v=0 715 o=UserA 2890844526 2890844526 IN IP4 here.com 716 t=0 0 717 c=IN IP4 100.101.102.103 718 m=audio 49170 RTP/AVP 0 719 a=rtpmap:0 PCMU/8000 721 (100 Trying F6 722 Proxy 1 723 -> A) 724 SIP/2.0 100 Trying 725 Via: SIP/2.0/UDP here.com:5060 726 From: TheBigGuy 727 To: TheLittleGuy 728 Call-Id: 12345601@here.com 729 CSeq: 1 INVITE 730 Content-Length: 0 732 INVITE F7 733 Proxy 2 734 ->B 736 INVITE sip:UserB@there.com SIP/2.0 737 Via: SIP/2.0/UDP ss2.wcom.com:5060 738 Via: SIP/2.0/UDP ss1.wcom.com:5060 739 Via: SIP/2.0/UDP here.com:5060 740 Record-Route: , 741 From: TheBigGuy 742 To: TheLittleGuy 743 Call-Id: 12345601@here.com 744 CSeq: 1 INVITE 745 Contact: TheBigGuy 746 Content-Type: application/sdp 747 Content-Length:132 749 v=0 750 o=UserA 2890844526 2890844526 IN IP4 here.com 751 t=0 0 752 c=IN IP4 100.101.102.103 753 m=audio 49170 RTP/AVP 0 754 a=rtpmap:0 PCMU/8000 756 (100 Trying F8 757 Proxy 2 758 -> Proxy 1) 760 SIP/2.0 100 Trying 761 Via: SIP/2.0/UDP ss1.wcom.com:5060 762 Via: SIP/2.0/UDP here.com:5060 763 From: TheBigGuy 764 To: TheLittleGuy 765 Call-Id: 12345601@here.com 766 CSeq: 1 INVITE 767 Content-Length: 0 769 180 Ringing F9 770 B -> Proxy 2 772 SIP/2.0 180 Ringing 773 Via: SIP/2.0/UDP ss2.wcom.com:5060 774 Via: SIP/2.0/UDP ss1.wcom.com:5060 775 Via: SIP/2.0/UDP here.com:5060 776 From: TheBigGuy 777 To: TheLittleGuy ;tag=314159 778 Call-Id: 12345601@here.com 779 CSeq: 1 INVITE 780 Content-Length: 0 782 180 783 Ringin 784 g F10 785 Proxy 2 786 -> 787 Proxy 788 1 790 SIP/2.0 180 Ringing 791 Via: SIP/2.0/UDP ss1.wcom.com:5060 792 Via: SIP/2.0/UDP here.com:5060 793 From: TheBigGuy 794 To: TheLittleGuy ;tag=314159 795 Call-Id: 12345601@here.com 796 CSeq: 1 INVITE 797 Content-Length: 0 799 180 Ringing F11 800 Proxy1 801 ->A 803 SIP/2.0 180 Ringing 804 Via: SIP/2.0/UDP here.com:5060 805 From: TheBigGuy 806 To: TheLittleGuy ;tag=314159 807 Call-Id: 12345601@here.com 808 CSeq: 1 INVITE 809 Content-Length: 0 811 200 OK F12 812 B -> Proxy 2 814 SIP/2.0 200 OK 815 Via: SIP/2.0/UDP ss2.wcom.com:5060 816 Via: SIP/2.0/UDP ss1.wcom.com:5060 817 Via: SIP/2.0/UDP here.com:5060 818 Record-Route: , 819 From: TheBigGuy 820 To: TheLittleGuy ;tag=314159 821 Call-Id: 12345601@here.com 822 CSeq: 1 INVITE 823 Contact: TheLittleGuy 824 Content-Type: application/sdp 825 Content-Length: 134 827 v=0 828 o=UserB 2890844527 2890844527 IN IP4 there.com 829 t=0 0 830 c=IN IP4 110.111.112.113 831 m=audio 3456 RTP/AVP 0 832 a=rtpmap:0 PCMU/8000 834 200 OK 835 F13 836 Proxy 837 2 838 -> 839 Proxy 840 1 842 SIP/2.0 200 OK 843 Via: SIP/2.0/UDP ss1.wcom.com:5060 844 Via: SIP/2.0/UDP here.com:5060 845 Record-Route: , 846 From: TheBigGuy 847 To: TheLittleGuy ;tag=314159 848 Call-Id: 12345601@here.com 849 CSeq: 1 INVITE 850 Contact: TheLittleGuy 851 Content-Type: application/sdp 852 Content-Length: 134 854 v=0 855 o=UserB 2890844527 2890844527 IN IP4 there.com 856 t=0 0 857 c=IN IP4 110.111.112.113 858 m=audio 3456 RTP/AVP 0 859 a=rtpmap:0 PCMU/8000 861 200 OK F14 862 Proxy 1 863 -> A 864 SIP/2.0 200 OK 865 Via: SIP/2.0/UDP here.com:5060 866 Record-Route: , 867 From: TheBigGuy 868 To: TheLittleGuy ;tag=314159 869 Call-Id: 12345601@here.com 870 CSeq: 1 INVITE 871 Contact: TheLittleGuy 872 Content-Type: application/sdp 873 Content-Length: 134 875 v=0 876 o=UserB 2890844527 2890844527 IN IP4 there.com 877 t=0 0 878 c=IN IP4 110.111.112.113 879 m=audio 3456 RTP/AVP 0 880 a=rtpmap:0 PCMU/8000 882 ACK F15 883 A -> Proxy 1 885 ACK sip:UserB@ss1.wcom.com SIP/2.0 886 Via: SIP/2.0/UDP here.com:5060 887 Route: , 888 From: TheBigGuy 889 To: TheLittleGuy ;tag=314159 890 Call-Id: 12345601@here.com 891 CSeq: 1 ACK 892 Content-Length: 0 894 ACK F16 895 Proxy 1 -> Proxy 2 897 ACK sip:UserB@ss2.wcom.com SIP/2.0 898 Via: SIP/2.0/UDP ss1.wcom.com:5060 899 Via: SIP/2.0/UDP here.com:5060 900 Route: 901 From: TheBigGuy 902 To: TheLittleGuy ;tag=314159 903 Call-Id: 12345601@here.com 904 CSeq: 1 ACK 905 Content-Length: 0 907 ACK F17 908 Proxy 2 ->B 910 ACK sip: UserB@there.com SIP/2.0 911 Via: SIP/2.0/UDP ss2.wcom.com:5060 912 Via: SIP/2.0/UDP ss1.wcom.com:5060 913 Via: SIP/2.0/UDP here.com:5060 914 From: TheBigGuy 915 To: TheLittleGuy ;tag=314159 916 Call-Id: 12345601@here.com 917 CSeq: 1 ACK 918 Content-Length: 0 920 /* RTP streams are established between A and B */ 922 /* User B Hangs Up with User A. */ 924 BYE F18 925 User B 926 -> Proxy 2 928 BYE sip: UserA@ss2.wcom.com SIP/2.0 929 Via: SIP/2.0/UDP there.com:5060 930 Route: , 931 From: TheLittleGuy ;tag=314159 932 To: TheBigGuy 933 Call-Id: 12345601@here.com 934 CSeq: 1 BYE 935 Content-Length: 0 937 BYE 938 F19 939 Proxy 940 2 -> 941 Proxy 942 1 944 BYE sip: UserA@ss1.wcom.com SIP/2.0 945 Via: SIP/2.0/UDP ss2.wcom.com:5060 946 Via: SIP/2.0/UDP there.com:5060 947 Route: 948 From: TheLittleGuy ;tag=314159 949 To: TheBigGuy 950 Call-Id: 12345601@here.com 951 CSeq: 1 BYE 952 Content-Length: 0 954 BYE F20 955 Proxy 1 -> User A 957 BYE sip: UserA@here.com SIP/2.0 958 Via: SIP/2.0/UDP ss1.wcom.com:5060 959 Via: SIP/2.0/UDP ss2.wcom.com:5060 960 Via: SIP/2.0/UDP there.com:5060 961 From: TheLittleGuy ;tag=314159 962 To: TheBigGuy 963 Call-Id: 12345601@here.com 964 CSeq: 1 BYE 965 Content-Length: 0 967 200 OK F21 968 User A 969 -> Proxy 1 971 SIP/2.0 200 OK 972 Via: SIP/2.0/UDP ss1.wcom.com:5060 973 Via: SIP/2.0/UDP ss2.wcom.com:5060 974 Via: SIP/2.0/UDP there.com:5060 975 From: TheLittleGuy ;tag=314159 976 To: TheBigGuy 977 Call-Id: 12345601@here.com 978 CSeq: 1 BYE 979 Content-Length: 0 981 200 OK F22 982 Proxy 1 -> Proxy 2 984 SIP/2.0 200 OK 985 Via: SIP/2.0/UDP ss2.wcom.com:5060 986 Via: SIP/2.0/UDP there.com:5060 987 From: TheLittleGuy ;tag=314159 988 To: TheBigGuy 989 Call-Id: 12345601@here.com 990 CSeq: 1 BYE 991 Content-Length: 0 992 200 OK F23 993 Proxy 2 -> User B 995 SIP/2.0 200 OK 996 Via: SIP/2.0/UDP there.com:5060 997 From: TheLittleGuy ;tag=314159 998 To: TheBigGuy 999 Call-Id: 12345601@here.com 1000 CSeq: 1 BYE 1001 Content-Length: 0 1003 3.1.2 1004 Successful SIP to SIP with Proxy failure 1006 In this scenario, User A completes a call to User B via a Proxy Server. 1007 User A is configured for a primary SIP Proxy Server SS1 and a secondary 1008 SIP Proxy Server SS2 (Or is able to use DNS SRV records to locate SS1 1009 and SS2). SS1 is out of service and does not respond to INVITEs (it is 1010 reachable, but unresponsive). After sending a CANCEL to SS1, User A 1011 then completes the call to User B using SS2. 1013 Message Details 1015 INVITE F1 1016 A -> Proxy 1 1018 INVITE sip:UserB@ss1.wcom.com SIP/2.0 1019 Via: SIP/2.0/UDP here.com:5060 1020 From: TheBigGuy 1021 To: TheLittleGuy 1022 Call-Id: 12345600@here.com 1023 CSeq: 1 INVITE 1024 Contact: TheBigGuy 1025 Content-Type: application/sdp 1026 Content-Length:132 1028 v=0 1029 o=UserA 2890844526 2890844526 IN IP4 here.com 1030 t=0 0 1031 c=IN IP4 100.101.102.103 1032 m=audio 49170 RTP/AVP 0 1033 a=rtpmap:0 PCMU/8000 1035 INVITE F2 1036 A -> Proxy 1 1038 Same as Message F1 1040 INVITE F3 1041 A -> Proxy 1 1043 Same as Message F1 1045 INVITE F4 1046 A -> Proxy 1 1048 Same as Message F1 1050 INVITE F5 1051 A -> Proxy 1 1053 Same as Message F1 1055 INVITE F6 1056 A -> Proxy 1 1058 Same as Message F1 1060 INVITE F7 1061 A -> Proxy 1 1063 Same as Message F1 1065 /* User A gives up on the unresponsive proxy and sends a CANCEL. If 1066 any 200 OK responses come back to the INVITE, User A sends an ACK, 1067 then a BYE. */ 1069 CANCEL F8 1070 A -> Proxy 1 1072 CANCEL sip:UserB@ss1.wcom.com SIP/2.0 1073 Via: SIP/2.0/UDP here.com:5060 1074 From: TheBigGuy 1075 To: TheLittleGuy 1076 Call-Id: 12345600@here.com 1077 CSeq: 1 CANCEL 1078 INVITE F9 1079 A -> Proxy 2 1081 INVITE sip:UserB@ss2.wcom.com SIP/2.0 1082 Via: SIP/2.0/UDP here.com:5060 1083 From: TheBigGuy 1084 To: TheLittleGuy 1085 Call-Id: 12345601@here.com 1086 CSeq: 1 INVITE 1087 Contact: TheBigGuy 1088 Content-Type: application/sdp 1089 Content-Length:132 1091 v=0 1092 o=UserA 2890844526 2890844526 IN IP4 here.com 1093 t=0 0 1094 c=IN IP4 100.101.102.103 1095 m=audio 49170 RTP/AVP 0 1096 a=rtpmap:0 PCMU/8000 1098 /* Proxy 2 challenges User A for authentication */ 1099 407 Proxy Authorization Required F10 1100 Proxy 2 -> User A 1102 SIP/2.0 407 Proxy Authorization Required 1103 Via: SIP/2.0/UDP here.com:5060 1104 From: TheBigGuy 1105 To: TheLittleGuy 1106 Call-Id: 12345601@here.com 1107 CSeq: 1 INVITE 1108 Proxy-Authenticate: Digest realm="MCI WorldCom SIP", 1109 domain="wcom.com", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", 1110 opaque="", stale="FALSE", algorithm="MD5" 1111 Content-Length: 0 1113 ACK F11 1114 A -> Proxy 2 1116 ACK sip:UserB@ss2.wcom.com SIP/2.0 1117 Via: SIP/2.0/UDP here.com:5060 1118 From: TheBigGuy 1119 To: TheLittleGuy 1120 Call-Id: 12345601@here.com 1121 CSeq: 1 INVITE 1122 Content-Length: 0 1124 /* User A responds be re-sending the INVITE with authentication 1125 credentials in it. */ 1127 INVITE F12 1128 A -> Proxy 2 1130 INVITE sip:UserB@ss2.wcom.com SIP/2.0 1131 Via: SIP/2.0/UDP here.com:5060 1132 From: TheBigGuy 1133 To: TheLittleGuy 1134 Call-Id: 12345602@here.com 1135 CSeq: 1 INVITE 1136 Contact: TheBigGuy 1137 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 1138 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 1139 uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc" 1140 Content-Type: application/sdp 1141 Content-Length:132 1143 v=0 1144 o=UserA 2890844526 2890844526 IN IP4 here.com 1145 t=0 0 1146 c=IN IP4 100.101.102.103 1147 m=audio 49170 RTP/AVP 0 1148 a=rtpmap:0 PCMU/8000 1150 /* Proxy 2 accepts the credentials and forwards the INVITE to User B. 1151 Client for A prepares to receive data on port 49170 from the network. 1152 */ 1154 INVITE F13 1155 Proxy 2 ->B 1157 INVITE sip:UserB@there.com SIP/2.0 1158 Via: SIP/2.0/UDP ss2.wcom.com:5060 1159 Via: SIP/2.0/UDP here.com:5060 1160 Record-Route: 1161 From: TheBigGuy 1162 To: TheLittleGuy 1163 Call-Id: 12345602@here.com 1164 CSeq: 1 INVITE 1165 Contact: TheBigGuy 1166 Content-Type: application/sdp 1167 Content-Length:132 1169 v=0 1170 o=UserA 2890844526 2890844526 IN IP4 here.com 1171 t=0 0 1172 c=IN IP4 100.101.102.103 1173 m=audio 49170 RTP/AVP 0 1174 a=rtpmap:0 PCMU/8000 1175 (100 Trying F14 1176 Proxy 2 -> User A) 1178 SIP/2.0 100 Trying 1179 Via: SIP/2.0/UDP ss2.wcom.com:5060 1180 Via: SIP/2.0/UDP here.com:5060 1181 From: TheBigGuy 1182 To: TheLittleGuy 1183 Call-Id: 12345602@here.com 1184 CSeq: 1 INVITE 1185 Content-Length: 0 1187 180 Ringing F15 1188 B -> Proxy 2 1190 SIP/2.0 180 Ringing 1191 Via: SIP/2.0/UDP ss2.wcom.com:5060 1192 Via: SIP/2.0/UDP here.com:5060 1193 From: TheBigGuy 1194 To: TheLittleGuy ;tag=314159 1195 Call-Id: 12345602@here.com 1196 CSeq: 1 INVITE 1197 Content-Length: 0 1199 180 Ringing F16 1200 Proxy 2 1201 -> A 1203 SIP/2.0 180 Ringing 1204 Via: SIP/2.0/UDP here.com:5060 1205 From: TheBigGuy 1206 To: TheLittleGuy ;tag=314159 1207 Call-Id: 12345602@here.com 1208 CSeq: 1 INVITE 1209 Content-Length: 0 1211 200 OK F17 1212 B -> Proxy 2 1214 SIP/2.0 200 OK 1215 Via: SIP/2.0/UDP ss2.wcom.com:5060 1216 Via: SIP/2.0/UDP here.com:5060 1217 Record-Route: 1218 From: TheBigGuy 1219 To: TheLittleGuy ;tag=314159 1220 Call-Id: 12345602@here.com 1221 CSeq: 1 INVITE 1222 Contact: TheLittleGuy 1223 Content-Type: application/sdp 1224 Content-Length: 134 1226 v=0 1227 o=UserB 2890844527 2890844527 IN IP4 there.com 1228 t=0 0 1229 c=IN IP4 110.111.112.113 1230 m=audio 3456 RTP/AVP 0 1231 a=rtpmap:0 PCMU/8000 1233 200 OK F18 1234 Proxy 2 ->A 1236 SIP/2.0 200 OK 1237 Via: SIP/2.0/UDP here.com:5060 1238 Record-Route: 1239 From: TheBigGuy 1240 To: TheLittleGuy ;tag=314159 1241 Call-Id: 12345602@here.com 1242 CSeq: 1 INVITE 1243 Contact: TheLittleGuy 1244 Content-Type: application/sdp 1245 Content-Length: 134 1247 v=0 1248 o=UserB 2890844527 2890844527 IN IP4 there.com 1249 t=0 0 1250 c=IN IP4 110.111.112.113 1251 m=audio 3456 RTP/AVP 0 1252 a=rtpmap:0 PCMU/8000 1254 ACK F19 1255 A -> Proxy 2 1257 ACK sip:UserB@ss2.wcom.com SIP/2.0 1258 Via: SIP/2.0/UDP here.com:5060 1259 Route: 1260 From: TheBigGuy 1261 To: TheLittleGuy ;tag=314159 1262 Call-Id: 12345602@here.com 1263 CSeq: 1 ACK 1264 Content-Length: 0 1266 ACK F20 1267 Proxy 2 ->B 1268 ACK sip: UserB@there.com SIP/2.0 1269 Via: SIP/2.0/UDP ss2.wcom.com:5060 1270 Via: SIP/2.0/UDP here.com:5060 1271 From: TheBigGuy 1272 To: TheLittleGuy ;tag=314159 1273 Call-Id: 12345602@here.com 1274 CSeq: 1 ACK 1275 Content-Length: 0 1277 /* RTP streams are established between A and B */ 1279 /* User B Hangs Up with User A. */ 1280 BYE F21 1281 User B 1282 -> Proxy 2 1284 BYE sip: UserA@ss2.wcom.com SIP/2.0 1285 Via: SIP/2.0/UDP there.com:5060 1286 Route: 1287 From: TheLittleGuy ;tag=314159 1288 To: TheBigGuy 1289 Call-Id: 12345602@here.com 1290 CSeq: 1 BYE 1291 Content-Length: 0 1293 BYE F22 1294 Proxy 2 -> User A 1296 BYE sip: UserA@here.com SIP/2.0 1297 Via: SIP/2.0/UDP ss2.wcom.com:5060 1298 Via: SIP/2.0/UDP there.com:5060 1299 From: TheLittleGuy ;tag=314159 1300 To: TheBigGuy 1301 Call-Id: 12345602@here.com 1302 CSeq: 1 BYE 1303 Content-Length: 0 1305 200 OK F23 1306 User A 1307 -> Proxy 2 1308 SIP/2.0 200 OK 1309 Via: SIP/2.0/UDP ss2.wcom.com:5060 1310 Via: SIP/2.0/UDP there.com:5060 1311 From: TheLittleGuy ;tag=314159 1312 To: TheBigGuy 1313 Call-Id: 12345602@here.com 1314 CSeq: 1 BYE 1315 Content-Length: 0 1317 200 OK F24 1318 Proxy 2 -> User B 1320 SIP/2.0 200 OK 1321 Via: SIP/2.0/UDP there.com:5060 1322 From: TheLittleGuy ;tag=314159 1323 To: TheBigGuy 1324 Call-Id: 12345602@here.com 1325 CSeq: 1 BYE 1326 Content-Length: 0 1328 3.1.3 1329 Successful SIP to SIP through SIP Firewall Proxy 1331 User A completes a call to User B through a Firewall Proxy and a SIP 1332 Proxy. The signaling message exchange is identical to 3.1.1 but the 1333 media stream setup is not end-to-end _ the Firewall proxy terminates 1334 both media streams and bridges them. This is done by the Proxy 1335 modifying the SDP in the INVITE (F1) and 200 OK (F11) messages. 1337 In addition to firewall traversal, this back-to-back User Agent 1338 Client and User Agent Server could be used as part of an anonymizer 1339 service (in which all identifying information on User A would be 1340 removed), or to perform codec media conversion, such as mu-law to A- 1341 law conversion of PCM on an international call. 1343 Message Details 1345 INVITE F1 1346 A->SIP FW 1348 INVITE sip:UserB@ fwp1.wcom.com SIP/2.0 1349 Via: SIP/2.0/UDP here.com:5060 1350 From: TheBigGuy 1351 To: TheLittleGuy 1352 Call-Id: 12345600@here.com 1353 CSeq: 1 INVITE 1354 Contact: TheBigGuy 1355 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 1356 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 1357 uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" 1358 Content-Type: application/sdp 1359 Content-Length:132 1361 v=0 1362 o=UserA 2890844526 2890844526 IN IP4 here.com 1363 t=0 0 1364 c=IN IP4 100.101.102.103 1365 m=audio 49170 RTP/AVP 0 1366 a=rtpmap:0 PCMU/8000 1367 Client for A prepares to receive data on port 49170 from the 1368 network.*/ 1370 INVITE F2 1371 SS FW -> SS1 1373 INVITE sip:UserB@ss1.wcom.com SIP/2.0 1374 Via: SIP/2.0/UDP fwp1.wcom.com:5060 1375 Via: SIP/2.0/UDP here.com:5060 1376 Record-Route: 1377 From: TheBigGuy 1378 To: TheLittleGuy 1379 Call-Id: 12345600@here.com 1380 CSeq: 1 INVITE 1381 Contact: TheBigGuy 1382 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 1383 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 1384 uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" 1385 Content-Type: application/sdp 1386 Content-Length:134 1388 v=0 1389 o=UserA 2890844526 2890844526 IN IP4 here.com 1390 t=0 0 1391 c=IN IP4 200.201.202.203 1392 m=audio 1000 RTP/AVP 0 1393 a=rtpmap:0 PCMU/8000 1395 (100 Trying F3 1396 SIP FW-> A) 1398 SIP/2.0 100 Trying 1399 Via: SIP/2.0/UDP here.com:5060 1400 From: TheBigGuy 1401 To: TheLittleGuy 1402 Call-Id: 12345600@here.com 1403 CSeq: 1 INVITE 1404 Content-Length: 0 1406 /* SIP FW prepares to proxy data from port 1000 to 1407 100.101.102.103/49170. SS1 uses a location manager function to 1408 determine where B is actually located. Based upon location analysis 1409 the call is forwarded to User B */ 1410 INVITE F4 1411 SS1->B 1413 INVITE sip:UserB@there.com SIP/2.0 1414 Via: SIP/2.0/UDP ss1.wcom.com:5060 1415 Via: SIP/2.0/UDP fwp1.wcom.com:5060 1416 Via: SIP/2.0/UDP here.com:5060 1417 Record-Route: , 1418 From: TheBigGuy 1419 To: TheLittleGuy 1420 Call-Id: 12345600@here.com 1421 CSeq: 1 INVITE 1422 Contact: TheBigGuy 1423 Content-Type: application/sdp 1424 Content-Length:134 1426 v=0 1427 o=UserA 2890844526 2890844526 IN IP4 here.com 1428 t=0 0 1429 c=IN IP4 200.201.202.203 1430 m=audio 1000 RTP/AVP 0 1431 a=rtpmap:0 PCMU/8000 1433 (100 Trying F5 1434 SS1 -> SIP FW) 1436 SIP/2.0 100 Trying 1437 Via: SIP/2.0/UDP fwp1.wcom.com:5060 1438 Via: SIP/2.0/UDP here.com:5060 1439 From: TheBigGuy 1440 To: TheLittleGuy 1441 Call-Id: 12345600@here.com 1442 CSeq: 1 INVITE 1443 Content-Length: 0 1445 (100 Trying F6 1446 B -> SS1) 1448 SIP/2.0 100 Trying 1449 Via: SIP/2.0/UDP ss1.wcom.com:5060 1450 Via: SIP/2.0/UDP fwp1.wcom.com:5060 1451 Via: SIP/2.0/UDP here.com:5060 1452 From: TheBigGuy 1453 To: TheLittleGuy 1454 Call-Id: 12345600@here.com 1455 CSeq: 1 INVITE 1456 Content-Length: 0 1457 180 Ringing F7 1458 B->SS1 1460 SIP/2.0 180 Ringing 1461 Via: SIP/2.0/UDP ss1.wcom.com:5060 1462 Via: SIP/2.0/UDP fwp1.wcom.com:5060 1463 Via: SIP/2.0/UDP here.com:5060 1464 From: TheBigGuy 1465 To: TheLittleGuy ;tag=314159 1466 Call-Id: 12345600@here.com 1467 CSeq: 1 INVITE 1468 Content-Length: 0 1470 180 Ringing F8 1471 SS1 -> SIP FW 1473 SIP/2.0 180 Ringing 1474 Via: SIP/2.0/UDP fwp1.wcom.com:5060 1475 Via: SIP/2.0/UDP here.com:5060 1476 From: TheBigGuy 1477 To: TheLittleGuy ;tag=314159 1478 Call-Id: 12345600@here.com 1479 CSeq: 1 INVITE 1480 Content-Length: 0 1482 180 Ringing F9 1483 SIP FW 1484 -> A 1486 SIP/2.0 180 Ringing 1487 Via: SIP/2.0/UDP here.com:5060 1488 From: TheBigGuy 1489 To: TheLittleGuy ;tag=314159 1490 Call-Id: 12345600@here.com 1491 CSeq: 1 INVITE 1492 Content-Length: 0 1494 200 OK F10 1495 B->SS1 1497 SIP/2.0 200 OK 1498 Via: SIP/2.0/UDP ss1.wcom.com:5060 1499 Via: SIP/2.0/UDP fwp1.wcom.com:5060 1500 Via: SIP/2.0/UDP here.com:5060 1501 Record-Route: , 1502 From: TheBigGuy 1503 To: TheLittleGuy ;tag=314159 1504 Call-Id: 12345600@here.com 1505 CSeq: 1 INVITE 1506 Contact: TheLittleGuy 1507 Content-Type: application/sdp 1508 Content-Length: 133 1510 v=0 1511 o=UserB 2890844527 2890844527 IN IP4 there.com 1512 t=0 0 1513 c=IN IP4 110.111.112.113 1514 m=audio 3456 RTP/AVP 0 1515 a=rtpmap:0 PCMU/8000 1517 200 OK F11 1518 SS1 -> SIP FW 1520 SIP/2.0 200 OK 1521 Via: SIP/2.0/UDP gw1.wcom.com:5060 1522 Via: SIP/2.0/UDP here.com:5060 1523 Record-Route: , 1524 From: TheBigGuy 1525 To: TheLittleGuy ;tag=314159 1526 Call-Id: 12345600@here.com 1527 CSeq: 1 INVITE 1528 Contact: TheLittleGuy 1529 Content-Type: application/sdp 1530 Content-Length: 134 1532 v=0 1533 o=UserB 2890844527 2890844527 IN IP4 there.com 1534 t=0 0 1535 c=IN IP4 110.111.112.113 1536 m=audio 3456 RTP/AVP 0 1537 a =rtpmap:0 PCMU/8000 1539 200 OK F12 1540 SIP FW 1541 -> A 1543 SIP/2.0 200 OK 1544 Via: SIP/2.0/UDP here.com:5060 1545 Record-Route: , 1546 From: TheBigGuy 1547 To: TheLittleGuy ;tag=314159 1548 Call-Id: 12345600@here.com 1549 CSeq: 1 INVITE 1550 Contact: TheLittleGuy 1551 Content-Type: application/sdp 1552 Content-Length: 134 1554 v=0 1555 o=UserB 2890844527 2890844527 IN IP4 there.com 1556 t=0 0 1557 c=IN IP4 200.201.202.203 1558 m=audio 1002 RTP/AVP 0 1559 a=rtpmap:0 PCMU/8000 1561 /* The gateway prepares to proxy packets from port 1001 to 1562 110.111.112.113/3456 1563 ACK F13 1564 A->SIP FW 1566 ACK sip:UserB@fwp1.wcom.com SIP/2.0 1567 Via: SIP/2.0/UDP here.com:5060 1568 Route: , 1569 From: TheBigGuy 1570 To: TheLittleGuy ;tag=314159 1571 Call-Id: 12345600@here.com 1572 CSeq: 1 ACK 1573 Content-Length: 0 1575 ACK F14 1576 SIP FW -> SS1 1578 ACK sip:UserB@ss1.wcom.com SIP/2.0 1579 Via: SIP/2.0/UDP fwp1.wcom.com:5060 1580 Via: SIP/2.0/UDP here.com:5060 1581 Route: 1582 From: TheBigGuy 1583 To: TheLittleGuy ;tag=314159 1584 Call-Id: 12345600@here.com 1585 CSeq: 1 ACK 1586 Content-Length: 0 1588 ACK F15 1589 SS1->B 1591 ACK sip: UserB@there.com SIP/2.0 1592 Via: SIP/2.0/UDP ss1.wcom.com:5060 1593 Via: SIP/2.0/UDP fwp1.wcom.com:5060 1594 Via: SIP/2.0/UDP here.com:5060 1595 From: TheBigGuy 1596 To: TheLittleGuy ;tag=314159 1597 Call-Id: 12345600@here.com 1598 CSeq: 1 ACK 1599 Content-Length: 0 1601 /* RTP streams are established between A and the SIP GW and between 1602 the SIP GW and B*/ 1604 /* User A Hangs Up with User B. */ 1605 BYE F16 1606 A->SIP FW 1608 BYE sip: UserB@fwp1.wcom.com SIP/2.0 1609 Via: SIP/2.0/UDP here.com:5060 1610 Route: , 1611 From: TheBigGuy 1612 To: TheLittleGuy ;tag=314159 1613 Call-Id: 12345600@here.com 1614 CSeq: 2 BYE 1615 Content-Length: 0 1617 BYE F17 1618 SIP FW 1619 -> SS1 1621 BYE sip: UserB@ss1.wcom.com SIP/2.0 1622 Via: SIP/2.0/UDP fwp1.wcom.com:5060 1623 Via: SIP/2.0/UDP here.com:5060 1624 Route: 1625 From: TheBigGuy 1626 To: TheLittleGuy ;tag=314159 1627 Call-Id: 12345600@here.com 1628 CSeq: 2 BYE 1629 Content-Length: 0 1631 BYE F18 1632 SS1->B 1633 BYE sip: UserB@there.com SIP/2.0 1634 Via: SIP/2.0/UDP ss1.wcom.com:5060 1635 Via: SIP/2.0/UDP fwp1.wcom.com:5060 1636 Via: SIP/2.0/UDP here.com:5060 1637 From: TheBigGuy 1638 To: TheLittleGuy ;tag=314159 1639 Call-Id: 12345600@here.com 1640 CSeq: 2 BYE 1641 Content-Length: 0 1643 200 OK F19 1644 B->SS1 1646 SIP/2.0 200 OK 1647 Via: SIP/2.0/UDP ss1.wcom.com:5060 1648 Via: SIP/2.0/UDP fwp1.wcom.com:5060 1649 Via: SIP/2.0/UDP here.com:5060 1650 From: TheBigGuy 1651 To: TheLittleGuy ;tag=314159 1652 Call-Id: 12345600@here.com 1653 CSeq: 2 BYE 1654 Content-Length: 0 1656 200 OK F20 1657 SS1 -> SIP FW 1659 SIP/2.0 200 OK 1660 Via: SIP/2.0/UDP fwp1.wcom.com:5060 1661 Via: SIP/2.0/UDP here.com:5060 1662 From: TheBigGuy 1663 To: TheLittleGuy >;tag=314159 1664 Call-Id: 12345600@here.com 1665 CSeq: 2 BYE 1666 Content-Length: 0 1668 200 OK F21 1669 SIP FW 1670 -> A 1672 SIP/2.0 200 OK 1673 Via: SIP/2.0/UDP here.com:5060 1674 From: TheBigGuy 1675 To: TheLittleGuy >;tag=314159 1676 Call-Id: 12345600@here.com 1677 CSeq: 2 BYE 1678 Content-Length: 0 1680 3.1.4 1681 Successful SIP to SIP via Redirect and Proxy 1683 In this scenario, User A places a call to User B using first a 1684 Redirect server then a Proxy Server. The INVITE message is first 1685 sent to the Redirect Server. The Server returns a 302 Moved 1686 Temporarily response (F2) containing a Contact header with User B's 1687 current SIP address. User A then generates a new INVITE and sends to 1688 User B via the Proxy Server and the call proceeds normally. 1690 The call is terminated when User B sends a BYE message. 1692 Message Details 1694 INVITE F1 1695 A->Redir Proxy 1697 INVITE sip:UserB@redirect.wcom.com SIP/2.0 1698 Via: SIP/2.0/UDP here.com:5060 1699 From: TheBigGuy 1700 To: TheLittleGuy 1701 Call-Id: 12345600@here.com 1702 CSeq: 1 INVITE 1703 Contact: TheBigGuy 1704 Content-Type: application/sdp 1705 Content-Length:132 1707 v=0 1708 o=UserA 2890844526 2890844526 IN IP4 here.com 1709 t=0 0 1710 c=IN IP4 100.101.102.103 1711 m=audio 49170 RTP/AVP 0 1712 a=rtpmap:0 PCMU/8000 1714 /*Client for A prepares to receive data on port 49170 from the 1715 network.*/ 1717 302 Moved Temporarily F2 1718 Redir Proxy 1719 ->A 1721 SIP/2.0 302 Moved Temporarily 1722 Contact: sip:UserB@ss2.wcom.com 1723 Via: SIP/2.0/UDP here.com:5060 1724 From: TheBigGuy 1725 To: TheLittleGuy 1726 Call-Id: 12345600@here.com 1727 CSeq: 1 INVITE 1728 Content-Length: 0 1730 ACK F3 1731 A->Redir Proxy 1733 ACK sip:UserB@redirect.wcom.com SIP/2.0 1734 Via: SIP/2.0/UDP here.com:5060 1735 From: TheBigGuy 1736 To: TheLittleGuy 1737 Call-Id: 12345600@here.com 1738 CSeq: 1 INVITE 1739 Content-Length: 0 1741 INVITE F4 1742 A -> Proxy 1744 INVITE sip:UserB@ss2.wcom.com SIP/2.0 1745 Via: SIP/2.0/UDP here.com:5060 1746 From: TheBigGuy 1747 To: TheLittleGuy 1748 Call-Id: 12345600@here.com 1749 CSeq: 2 INVITE 1750 Contact: TheBigGuy 1751 Content-Type: application/sdp 1752 Content-Length:132 1754 v=0 1755 o=UserA 2890844526 2890844526 IN IP4 here.com 1756 t=0 0 1757 c=IN IP4 100.101.102.103 1758 m=audio 49170 RTP/AVP 0 1759 a=rtpmap:0 PCMU/8000 1761 INVITE F5 1762 Proxy -> B 1764 INVITE sip:UserB@there.com SIP/2.0 1765 Via: SIP/2.0/UDP ss2.wcom.com:5060 1766 Via: SIP/2.0/UDP here.com:5060 1767 Record-Route: 1768 From: TheBigGuy 1769 To: TheLittleGuy 1770 Call-Id: 12345600@here.com 1771 CSeq: 2 INVITE 1772 Contact: TheBigGuy 1773 Content-Type: application/sdp 1774 Content-Length:132 1776 v=0 1777 o=UserA 2890844526 2890844526 IN IP4 here.com 1778 t=0 0 1779 c=IN IP4 100.101.102.103 1780 m=audio 49170 RTP/AVP 0 1781 a=rtpmap:0 PCMU/8000 1782 (100 Trying F6 1783 Proxy 1784 -> A) 1786 SIP/2.0 100 Trying 1787 Via: SIP/2.0/UDP here.com:5060 1788 From: TheBigGuy 1789 To: TheLittleGuy 1790 Call-Id: 12345600@here.com 1791 CSeq: 2 INVITE 1792 Content-Length: 0 1794 (100 Trying ) F7 B 1795 -> Proxy 1797 SIP/2.0 100 Trying 1798 Via: SIP/2.0/UDP ss2.wcom.com:5060 1799 Via: SIP/2.0/UDP here.com:5060 1800 From: TheBigGuy 1801 To: TheLittleGuy 1802 Call-Id: 12345600@here.com 1803 CSeq: 2 INVITE 1804 Content-Length: 0 1806 180 Ringing F8 1807 B->Proxy 1809 SIP/2.0 180 Ringing 1810 Via: SIP/2.0/UDP ss2.wcom.com:5060 1811 Via: SIP/2.0/UDP here.com:5060 1812 From: TheBigGuy 1813 To: TheLittleGuy >;tag=314159 1814 Call-Id: 12345600@here.com 1815 CSeq: 2 INVITE 1816 Content-Length: 0 1818 180 Ringing F9 1819 Proxy->A 1821 SIP/2.0 180 Ringing 1822 Via: SIP/2.0/UDP here.com:5060 1823 From: TheBigGuy 1824 To: TheLittleGuy >;tag=314159 1825 Call-Id: 12345600@here.com 1826 CSeq: 2 INVITE 1827 Content-Length: 0 1829 200 OK F10 1830 B->Proxy 1832 SIP/2.0 200 OK 1833 Via: SIP/2.0/UDP ss2.wcom.com:5060 1834 Via: SIP/2.0/UDP here.com:5060 1835 Record-Route: 1836 From: TheBigGuy 1837 To: TheLittleGuy >;tag=314159 1838 Call-Id: 12345600@here.com 1839 CSeq: 2 INVITE 1840 Contact: TheLittleGuy 1841 Content-Type: application/sdp 1842 Content-Length: 134 1844 v=0 1845 o=UserB 2890844527 2890844527 IN IP4 there.com 1846 t=0 0 1847 c=IN IP4 110.111.112.113 1848 m=audio 3456 RTP/AVP 0 1849 a=rtpmap:0 PCMU/8000 1851 200 OK F11 1852 Proxy->A 1854 SIP/2.0 200 OK 1855 Via: SIP/2.0/UDP here.com:5060 1856 Record-Route: 1857 From: TheBigGuy 1858 To: TheLittleGuy >;tag=314159 1859 Call-Id: 12345600@here.com 1860 CSeq: 2 INVITE 1861 Contact: TheLittleGuy 1862 Content-Type: application/sdp 1863 Content-Length: 134 1865 v=0 1866 o=UserB 2890844527 2890844527 IN IP4 there.com 1867 t=0 0 1868 c=IN IP4 110.111.112.113 1869 m=audio 3456 RTP/AVP 0 1870 a=rtpmap:0 PCMU/8000 1872 ACK F12 1873 A -> Proxy 1874 ACK sip:UserB@ss1.wcom.com SIP/2.0 1875 Via: SIP/2.0/UDP here.com:5060 1876 Route: 1877 From: TheBigGuy 1878 To: TheLittleGuy ;tag=314159 1879 Call-Id: 12345600@here.com 1880 CSeq: 2 ACK 1881 Content-Length: 0 1883 ACK F13 1884 Proxy -> B 1886 ACK sip: UserB@there.com SIP/2.0 1887 Via: SIP/2.0/UDP ss2.wcom.com:5060 1888 Via: SIP/2.0/UDP here.com:5060 1889 From: TheBigGuy 1890 To: TheLittleGuy ;tag=314159 1891 Call-Id: 12345600@here.com 1892 CSeq: 2 ACK 1893 Content-Length: 0 1895 /* RTP streams are established between A and B*/ 1897 /* User B Hangs Up with User A. */ 1898 BYE F14 1899 B->Proxy 1901 BYE sip: UserA@ss1.wcom.com SIP/2.0 1902 Via: SIP/2.0/UDP there.com:5060 1903 Route: 1904 From: TheLittleGuy ;tag=314159 1905 To: TheBigGuy 1906 Call-Id: 12345600@here.com 1907 CSeq: 1 BYE 1908 Content-Length: 0 1910 BYE F15 1911 Proxy->A 1912 BYE sip: UserA@here.com SIP/2.0 1913 Via: SIP/2.0/UDP ss2.wcom.com:5060 1914 Via: SIP/2.0/UDP there.com:5060 1915 From: TheLittleGuy ;tag=314159 1916 To: TheBigGuy 1917 Call-Id: 12345600@here.com 1918 CSeq: 1 BYE 1919 Content-Length: 0 1921 200 OK F16 1922 A -> Proxy 1924 SIP/2.0 200 OK 1925 Via: SIP/2.0/UDP ss2.wcom.com:5060 1926 Via: SIP/2.0/UDP there.com:5060 1927 From: TheLittleGuy ;tag=314159 1928 To: TheBigGuy 1929 Call-Id: 12345600@here.com 1930 CSeq: 1 BYE 1931 Content-Length: 0 1933 200 OK F17 1934 Proxy -> B 1936 SIP/2.0 200 OK 1937 Via: SIP/2.0/UDP there.com:5060 1938 From: TheLittleGuy ;tag=314159 1939 To: TheBigGuy 1940 Call-Id: 12345600@here.com 1941 CSeq: 1 BYE 1942 Content-Length: 0 1944 3.2 Failure Scenarios 1946 3.2.1 1947 Unsuccessful SIP to SIP no answer 1949 In this scenario, User A gives up on the call before User B answers 1950 (sends a 200 OK response). User A sends a CANCEL (F9) since no final 1951 response had been received from User B. If a 200 OK to the INVITE 1952 had crossed with the CANCEL, User A would have sent an ACK then a BYE 1953 to User B in order to properly terminate the call. 1955 Note that the CSeq of the CANCEL message (F9) is not incremented. 1956 This is so that downstream clients can match the To, From, Call-ID, 1957 and CSeq of the CANCEL to the INVITE to decide which request to 1958 terminate. 1960 Message Details 1962 INVITE F1 1963 A -> Proxy 1 1965 INVITE sip:UserB@ss1.wcom.com SIP/2.0 1966 Via: SIP/2.0/UDP here.com:5060 1967 From: TheBigGuy 1968 To: TheLittleGuy 1969 Call-Id: 12345600@here.com 1970 CSeq: 1 INVITE 1971 Contact: TheBigGuy 1972 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 1973 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 1974 uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" 1975 Content-Type: application/sdp 1976 Content-Length:132 1978 v=0 1979 o=UserA 2890844526 2890844526 IN IP4 here.com 1980 t=0 0 1981 c=IN IP4 100.101.102.103 1982 m=audio 49170 RTP/AVP 0 1983 a=rtpmap:0 PCMU/8000 1985 /*Client for A prepares to receive data on port 49170 from the 1986 network.*/ 1988 INVITE F2 1989 Proxy 1 -> Proxy 2 1991 INVITE sip:UserB@ss2.wcom.com SIP/2.0 1992 Via: SIP/2.0/UDP ss1.wcom.com:5060 1993 Via: SIP/2.0/UDP here.com:5060 1994 Record-Route: 1995 From: TheBigGuy 1996 To: TheLittleGuy 1997 Call-Id: 12345600@here.com 1998 CSeq: 1 INVITE 1999 Contact: TheBigGuy 2000 Content-Type: application/sdp 2001 Content-Length:132 2003 v=0 2004 o=UserA 2890844526 2890844526 IN IP4 here.com 2005 t=0 0 2006 c=IN IP4 100.101.102.103 2007 m=audio 49170 RTP/AVP 0 2008 a=rtpmap:0 PCMU/8000 2010 (100 Trying F3 2011 Proxy 1 -> A) 2013 SIP/2.0 100 Trying 2014 Via: SIP/2.0/UDP here.com:5060 2015 From: TheBigGuy 2016 To: TheLittleGuy 2017 Call-Id: 12345600@here.com 2018 CSeq: 1 INVITE 2019 Content-Length: 0 2021 INVITE F4 2022 Proxy 2 ->B 2024 INVITE sip:UserB@there.com SIP/2.0 2025 Via: SIP/2.0/UDP ss2.wcom.com:5060 2026 Via: SIP/2.0/UDP ss1.wcom.com:5060 2027 Via: SIP/2.0/UDP here.com:5060 2028 Record-Route: , 2029 From: TheBigGuy 2030 To: TheLittleGuy 2031 Call-Id: 12345600@here.com 2032 CSeq: 1 INVITE 2033 Contact: TheBigGuy 2034 Content-Type: application/sdp 2035 Content-Length: 132 2037 v=0 2038 o=UserA 2890844526 2890844526 IN IP4 here.com 2039 t=0 0 2040 c=IN IP4 100.101.102.103 2041 m=audio 49170 RTP/AVP 0 2042 a=rtpmap:0 PCMU/8000 2044 (100 Trying F5 2045 Proxy 2 -> Proxy 1) 2047 SIP/2.0 100 Trying 2048 Via: SIP/2.0/UDP ss1.wcom.com:5060 2049 Via: SIP/2.0/UDP here.com:5060 2050 From: TheBigGuy 2051 To: TheLittleGuy 2052 Call-Id: 12345600@here.com 2053 CSeq: 1 INVITE 2054 Content-Length: 0 2056 180 Ringing F6 2057 B -> Proxy 2 2059 SIP/2.0 180 Ringing 2060 Via: SIP/2.0/UDP ss2.wcom.com:5060 2061 Via: SIP/2.0/UDP ss1.wcom.com:5060 2062 Via: SIP/2.0/UDP here.com:5060 2063 From: TheBigGuy 2064 To: TheLittleGuy ;tag=314159 2065 Call-Id: 12345600@here.com 2066 CSeq: 1 INVITE 2067 Content-Length: 0 2069 180 Ringing F7 2070 Proxy 2 -> Proxy 1 2072 SIP/2.0 180 Ringing 2073 Via: SIP/2.0/UDP ss1.wcom.com:5060 2074 Via: SIP/2.0/UDP here.com:5060 2075 From: TheBigGuy 2076 To: TheLittleGuy ;tag=314159 2077 Call-Id: 12345600@here.com 2078 CSeq: 1 INVITE 2079 Content-Length: 0 2081 180 Ringing F8 2082 Proxy1 2083 -> A 2085 SIP/2.0 180 Ringing 2086 Via: SIP/2.0/UDP here.com:5060 2087 From: TheBigGuy 2088 To: TheLittleGuy ;tag=314159 2089 Call-Id: 12345600@here.com 2090 CSeq: 1 INVITE 2091 Content-Length: 0 2092 /* User A gives up and sends a CANCEL. If a 200 OK reply to the 2093 INVITE crossed with the CANCEL and was received by User A, User A 2094 would send an ACK then a BYE to terminate the call.*/ 2096 CANCEL F9 2097 A -> Proxy 1 2099 CANCEL sip:UserB@ss1.wcom.com SIP/2.0 2100 Via: SIP/2.0/UDP here.com:5060 2101 From: TheBigGuy 2102 To: TheLittleGuy ;tag=314159 2103 Call-Id: 12345600@here.com 2104 CSeq: 1 CANCEL 2105 Content-Length: 0 2107 CANCEL F10 2108 Proxy 1 -> Proxy 2 2110 CANCEL sip: UserA@ss2.wcom.com SIP/2.0 2111 Via: SIP/2.0/UDP ss1.wcom.com:5060 2112 Via: SIP/2.0/UDP here.com:5060 2113 From: TheBigGuy 2114 To: TheLittleGuy ;tag=314159 2115 Call-Id: 12345600@here.com 2116 CSeq: 1 CANCEL 2117 Content-Length: 0 2119 CANCEL F11 2120 Proxy 2 ->B 2122 CANCEL sip: UserB@there.com SIP/2.0 2123 Via: SIP/2.0/UDP ss2.wcom.com:5060 2124 Via: SIP/2.0/UDP ss1.wcom.com:5060 2125 Via: SIP/2.0/UDP here.com:5060 2126 From: TheBigGuy 2127 To: TheLittleGuy ;tag=314159 2128 Call-Id: 12345600@here.com 2129 CSeq: 1 CANCEL 2130 Content-Length: 0 2132 200 OK F12 2133 A -> Proxy 1 2135 SIP/2.0 200 OK 2136 Via: SIP/2.0/UDP ss2.wcom.com:5060 2137 Via: SIP/2.0/UDP ss1.wcom.com:5060 2138 Via: SIP/2.0/UDP here.com:5060 2139 From: TheBigGuy 2140 To: TheLittleGuy ;tag=314159 2141 Call-Id: 12345600@here.com 2142 CSeq: 1 CANCEL 2143 Content-Length: 0 2145 200 OK F13 2146 Proxy 1 -> Proxy 2 2148 SIP/2.0 200 OK 2149 Via: SIP/2.0/UDP ss1.wcom.com:5060 2150 Via: SIP/2.0/UDP here.com:5060 2151 From: TheBigGuy 2152 To: TheLittleGuy ;tag=314159 2153 Call-Id: 12345600@here.com 2154 CSeq: 1 CANCEL 2155 Content-Length: 0 2157 200 OK F14 2158 Proxy 2 ->B 2160 SIP/2.0 200 OK 2161 Via: SIP/2.0/UDP here.com:5060 2162 From: TheBigGuy 2163 To: TheLittleGuy ;tag=314159 2164 Call-Id: 12345600@here.com 2165 CSeq: 1 CANCEL 2166 Content-Length: 0 2168 3.2.2 2169 Unsuccessful SIP to SIP busy 2171 In this scenario, User B is busy and sends a 486 Busy Here response 2172 to User A's INVITE. The 4xx response is ACKed at each signaling leg. 2174 Message Details 2176 INVITE F1 2177 User A 2178 -> Proxy 1 2180 INVITE sip:UserB@ss1.wcom.com SIP/2.0 2181 Via: SIP/2.0/UDP here.com:5060 2182 From: TheBigGuy 2183 To: TheLittleGuy 2184 Call-Id: 12345600@here.com 2185 CSeq: 1 INVITE 2186 Contact: TheBigGuy 2187 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 2188 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 2189 uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" 2190 Content-Type: application/sdp 2191 Content-Length: 132 2193 v=0 2194 o=UserA 2890844526 2890844526 IN IP4 here.com 2195 t=0 0 2196 c=IN IP4 100.101.102.103 2197 m=audio 49170 RTP/AVP 0 2198 a=rtpmap:0 PCMU/8000 2200 /*Client for A prepares to receive data on port 49170 from the 2201 network.*/ 2202 INVITE F2 2203 Proxy 1 -> Proxy 2 2205 INVITE sip:UserB@ss2.wcom.com SIP/2.0 2206 Via: SIP/2.0/UDP ss1.wcom.com:5060 2207 Via: SIP/2.0/UDP here.com:5060 2208 Record-Route: 2209 From: TheBigGuy 2210 To: TheLittleGuy 2211 Call-Id: 12345600@here.com 2212 CSeq: 1 INVITE 2213 Contact: TheBigGuy 2214 Content-Type: application/sdp 2215 Content-Length: 132 2217 v=0 2218 o=UserA 2890844526 2890844526 IN IP4 here.com 2219 t=0 0 2220 c=IN IP4 100.101.102.103 2221 m=audio 49170 RTP/AVP 0 2222 a=rtpmap:0 PCMU/8000 2224 (100 Trying F3 2225 Proxy 1 -> User A) 2227 SIP/2.0 100 Trying 2228 Via: SIP/2.0/UDP here.com:5060 2229 From: TheBigGuy 2230 To: TheLittleGuy 2231 Call-Id: 12345600@here.com 2232 CSeq: 1 INVITE 2233 Content-Length: 0 2235 INVITE F4 2236 Proxy 2 -> 2237 User B 2239 INVITE sip:UserB@there.com SIP/2.0 2240 Via: SIP/2.0/UDP ss2.wcom.com:5060 2241 Via: SIP/2.0/UDP ss1.wcom.com:5060 2242 Via: SIP/2.0/UDP here.com:5060 2243 Record-Route: , 2244 From: TheBigGuy 2245 To: TheLittleGuy 2246 Call-Id: 12345600@here.com 2247 CSeq: 1 INVITE 2248 Contact: TheBigGuy 2249 Content-Type: application/sdp 2250 Content-Length: 132 2252 v=0 2253 o=UserA 2890844526 2890844526 IN IP4 here.com 2254 t=0 0 2255 c=IN IP4 100.101.102.103 2256 m=audio 49170 RTP/AVP 0 2257 a=rtpmap:0 PCMU/8000 2258 (100 Trying F5 2259 Proxy 2 -> Proxy 1) 2261 SIP/2.0 100 Trying 2262 Via: SIP/2.0/UDP ss1.wcom.com:5060 2263 Via: SIP/2.0/UDP here.com:5060 2264 From: TheBigGuy 2265 To: TheLittleGuy 2266 Call-Id: 12345600@here.com 2267 CSeq: 1 INVITE 2268 Content-Length: 0 2270 486 Busy Here F6 User B 2271 -> Proxy 2 2273 SIP/2.0 486 Busy Here 2274 Via: SIP/2.0/UDP ss2.wcom.com:5060 2275 Via: SIP/2.0/UDP ss1.wcom.com:5060 2276 Via: SIP/2.0/UDP here.com:5060 2277 From: TheBigGuy 2278 To: TheLittleGuy 2279 Call-Id: 12345600@here.com 2280 CSeq: 1 INVITE 2281 Content-Length: 0 2283 ACK F7 2284 Proxy 2 -> User B 2286 ACK sip: UserB@there.com SIP/2.0 2287 Via: SIP/2.0/UDP ss2.wcom.com:5060 2288 From: TheBigGuy 2289 To: TheLittleGuy ;tag=314159 2290 Call-Id: 12345600@here.com 2291 CSeq: 1 ACK 2292 Content-Length: 0 2294 486 Busy Here F8 Proxy 2 -> Proxy 1 2296 SIP/2.0 486 Busy Here 2297 Via: SIP/2.0/UDP ss1.wcom.com:5060 2298 Via: SIP/2.0/UDP here.com:5060 2299 From: TheBigGuy 2300 To: TheLittleGuy 2301 Call-Id: 12345600@here.com 2302 CSeq: 1 INVITE 2303 Content-Length: 0 2304 ACK F9 2305 Proxy 1 -> Proxy 2 2307 ACK sip:UserB@ss2.wcom.com SIP/2.0 2308 Via: SIP/2.0/UDP ss1.wcom.com:5060 2309 From: TheBigGuy 2310 To: TheLittleGuy ;tag=314159 2311 Call-Id: 12345600@here.com 2312 CSeq: 1 ACK 2313 Content-Length: 0 2315 486 Busy Here F10 2316 Proxy 1 -> User A 2318 SIP/2.0 486 Busy Here 2319 Via: SIP/2.0/UDP here.com:5060 2320 From: TheBigGuy 2321 To: TheLittleGuy 2322 Call-Id: 12345600@here.com 2323 CSeq: 1 INVITE 2324 Content-Length: 0 2326 ACK F11 2327 User A 2328 -> Proxy 1 2330 ACK sip:UserB@ss1.wcom.com SIP/2.0 2331 Via: SIP/2.0/UDP here.com:5060 2332 From: TheBigGuy 2333 To: TheLittleGuy ;tag=314159 2334 Call-Id: 12345600@here.com 2335 CSeq: 1 ACK 2336 Content-Length: 0 2338 3.2.3 2339 Unsuccessful SIP to SIP no response 2341 In this example, there is no response from User B to User A's INVITE 2342 messages being re-transmitted by Proxy 2. After the sixth re- 2343 transmission, Proxy 2 gives up and sends a CANCEL to User B and a 480 2344 No Response to User A. Note that the CANCEL would also be 2345 retransmitted six times, as governed by SIP timer T1 as in Section 2346 5.2.6. 2348 Message Details 2350 INVITE F1 2351 User A 2352 -> Proxy 1 2354 INVITE sip:UserB@ss1.wcom.com SIP/2.0 2355 Via: SIP/2.0/UDP here.com:5060 2356 From: TheBigGuy 2357 To: TheLittleGuy 2358 Call-Id: 12345600@here.com 2359 CSeq: 1 INVITE 2360 Contact: TheBigGuy 2361 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 2362 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 2363 uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" 2364 Content-Type: application/sdp 2365 Content-Length: 132 2367 v=0 2368 o=UserA 2890844526 2890844526 IN IP4 here.com 2369 t=0 0 2370 c=IN IP4 100.101.102.103 2371 m=audio 49170 RTP/AVP 0 2372 a=rtpmap:0 PCMU/8000 2373 /*Client for A prepares to receive data on port 49170 from the 2374 network.*/ 2375 INVITE F2 2376 Proxy 1 -> Proxy 2 2378 INVITE sip:UserB@ss2.wcom.com SIP/2.0 2379 Via: SIP/2.0/UDP ss1.wcom.com:5060 2380 Via: SIP/2.0/UDP here.com:5060 2381 Record-Route: 2382 From: TheBigGuy 2383 To: TheLittleGuy 2384 Call-Id: 12345600@here.com 2385 CSeq: 1 INVITE 2386 Contact: TheBigGuy 2387 Content-Type: application/sdp 2388 Content-Length: 132 2390 v=0 2391 o=UserA 2890844526 2890844526 IN IP4 here.com 2392 t=0 0 2393 c=IN IP4 100.101.102.103 2394 m=audio 49170 RTP/AVP 0 2395 a=rtpmap:0 PCMU/8000 2397 (100 Trying F3 2398 Proxy 1 -> User A) 2400 SIP/2.0 100 Trying 2401 Via: SIP/2.0/UDP here.com:5060 2402 From: TheBigGuy 2403 To: TheLittleGuy 2404 Call-Id: 12345600@here.com 2405 CSeq: 1 INVITE 2406 Content-Length: 0 2408 INVITE F4 2409 Proxy 2 -> User B 2411 INVITE sip:UserB@there.com SIP/2.0 2412 Via: SIP/2.0/UDP ss2.wcom.com:5060 2413 Via: SIP/2.0/UDP ss1.wcom.com:5060 2414 Via: SIP/2.0/UDP here.com:5060 2415 Record-Route: , 2416 From: TheBigGuy 2417 To: TheLittleGuy 2418 Call-Id: 12345600@here.com 2419 CSeq: 1 INVITE 2420 Contact: TheBigGuy 2421 Content-Type: application/sdp 2422 Content-Length: 132 2424 v=0 2425 o=UserA 2890844526 2890844526 IN IP4 here.com 2426 t=0 0 2427 c=IN IP4 100.101.102.103 2428 m=audio 49170 RTP/AVP 0 2429 a=rtpmap:0 PCMU/8000 2431 (100 Trying F5 2432 Proxy 2 -> Proxy 1) 2434 SIP/2.0 100 Trying 2435 Via: SIP/2.0/UDP ss1.wcom.com:5060 2436 Via: SIP/2.0/UDP here.com:5060 2437 From: TheBigGuy 2438 To: TheLittleGuy 2439 Call-Id: 12345600@here.com 2440 CSeq: 1 INVITE 2441 Content-Length: 0 2443 INVITE F6 2444 Proxy 2 -> User B 2446 Resend of Message F4 2447 INVITE F7 2448 Proxy 2 -> User B 2450 Resend of Message F4 2451 INVITE F8 2452 Proxy 2 -> User B 2454 Resend of Message F4 2455 INVITE F9 2456 Proxy 2 -> User B 2458 Resend of Message F4 2459 INVITE F10 2460 Proxy 2 -> User B 2462 Resend of Message F4 2463 INVITE F11 2464 Proxy 2 -> User B 2466 Resend of Message F4 2467 CANCEL F12 2468 Proxy 2 -> User B 2469 CANCEL sip:UserB@there.com SIP/2.0 2470 Via: SIP/2.0/UDP ss2.wcom.com:5060 2471 From: TheBigGuy 2472 To: TheLittleGuy 2473 Call-Id: 12345600@here.com 2474 CSeq: 1 CANCEL 2475 Content-Length: 0 2477 480 No Response F13 Proxy 2 -> Proxy 1 2479 SIP/2.0 480 No Response 2480 Via: SIP/2.0/UDP ss1.wcom.com:5060 2481 Via: SIP/2.0/UDP here.com:5060 2482 From: TheBigGuy 2483 To: TheLittleGuy 2484 Call-Id: 12345600@here.com 2485 CSeq: 1 INVITE 2486 Content-Length: 0 2488 ACK F14 2489 Proxy 1 -> Proxy 2 2491 ACK sip:UserB@ss2.wcom.com SIP/2.0 2492 Via: SIP/2.0/UDP ss1.wcom.com:5060 2493 From: TheBigGuy 2494 To: TheLittleGuy ;tag=314159 2495 Call-Id: 12345600@here.com 2496 CSeq: 1 ACK 2497 Content-Length: 0 2499 480 No Response F15 2500 Proxy 1 -> User A 2502 SIP/2.0 480 No Response 2503 Via: SIP/2.0/UDP here.com:5060 2504 From: TheBigGuy 2505 To: TheLittleGuy 2506 Call-Id: 12345600@here.com 2507 CSeq: 1 INVITE 2508 Content-Length: 0 2510 ACK F16 2511 User A 2512 -> Proxy 1 2513 ACK sip:UserB@ss1.wcom.com SIP/2.0 2514 Via: SIP/2.0/UDP here.com:5060 2515 From: TheBigGuy 2516 To: TheLittleGuy ;tag=314159 2517 Call-Id: 12345600@here.com 2518 CSeq: 1 ACK 2519 Content-Length: 0 2521 3.2.4 2522 Unsuccessful SIP to SIP Temporarily Unavailable 2524 In this scenario, User B initially sends a 180 Ringing response to 2525 User A, indicating that alerting is taking place. However, then a 2526 480 Unavailable is then sent to User A. This response is 2527 acknowledged then proxied back to User A. 2529 Message Details 2531 INVITE F1 2532 A -> Proxy 1 2534 INVITE sip:UserB@ss1.wcom.com SIP/2.0 2535 Via: SIP/2.0/UDP here.com:5060 2536 From: TheBigGuy 2537 To: TheLittleGuy 2538 Call-Id: 12345600@here.com 2539 CSeq: 1 INVITE 2540 Contact: TheBigGuy 2541 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 2542 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 2543 uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" 2544 Content-Type: application/sdp 2545 Content-Length: 132 2547 v=0 2548 o=UserA 2890844526 2890844526 IN IP4 here.com 2549 t=0 0 2550 c=IN IP4 100.101.102.103 2551 m=audio 49170 RTP/AVP 0 2552 a=rtpmap:0 PCMU/8000 2554 /*Client for A prepares to receive data on port 49170 from the 2555 network.*/ 2556 INVITE F2 2557 Proxy 1 -> Proxy 2 2559 INVITE sip:UserB@ss2.wcom.com SIP/2.0 2560 Via: SIP/2.0/UDP ss1.wcom.com:5060 2561 Via: SIP/2.0/UDP here.com:5060 2562 Record-Route: 2563 From: TheBigGuy 2564 To: TheLittleGuy 2565 Call-Id: 12345600@here.com 2566 CSeq: 1 INVITE 2567 Contact: TheBigGuy 2568 Content-Type: application/sdp 2569 Content-Length: 132 2571 v=0 2572 o=UserA 2890844526 2890844526 IN IP4 here.com 2573 t=0 0 2574 c=IN IP4 100.101.102.103 2575 m=audio 49170 RTP/AVP 0 2576 a=rtpmap:0 PCMU/8000 2578 INVITE F3 2579 Proxy 2 -> B 2581 INVITE sip:UserB@there.com SIP/2.0 2582 Via: SIP/2.0/UDP ss2.wcom.com:5060 2583 Via: SIP/2.0/UDP ss1.wcom.com:5060 2584 Via: SIP/2.0/UDP here.com:5060 2585 Record-Route: , 2586 From: TheBigGuy 2587 To: TheLittleGuy 2588 Call-Id: 12345600@here.com 2589 CSeq: 1 INVITE 2590 Contact: TheBigGuy 2591 Content-Type: application/sdp 2592 Content-Length: 132 2594 v=0 2595 o=UserA 2890844526 2890844526 IN IP4 here.com 2596 t=0 0 2597 c=IN IP4 100.101.102.103 2598 m=audio 49170 RTP/AVP 0 2599 a=rtpmap:0 PCMU/8000 2601 (100 Trying F4 2602 Proxy 1 -> A) 2604 SIP/2.0 100 Trying 2605 Via: SIP/2.0/UDP here.com:5060 2606 From: TheBigGuy 2607 To: TheLittleGuy 2608 Call-Id: 12345600@here.com 2609 CSeq: 1 INVITE 2610 Content-Length: 0 2612 (100 Trying F5 2613 Proxy 2 -> Proxy 1) 2615 SIP/2.0 100 Trying 2616 Via: SIP/2.0/UDP ss1.wcom.com:5060 2617 Via: SIP/2.0/UDP here.com:5060 2618 From: TheBigGuy 2619 To: TheLittleGuy 2620 Call-Id: 12345600@here.com 2621 CSeq: 1 INVITE 2622 Content-Length: 0 2624 (100 Trying F6 2625 User B 2626 -> Proxy 2) 2628 SIP/2.0 100 Trying 2629 Via: SIP/2.0/UDP ss2.wcom.com:5060 2630 Via: SIP/2.0/UDP ss1.wcom.com:5060 2631 Via: SIP/2.0/UDP here.com:5060 2632 From: TheBigGuy 2633 To: TheLittleGuy 2634 Call-Id: 12345600@here.com 2635 CSeq: 1 INVITE 2636 Content-Length: 0 2638 180 Ringing F7 2639 B -> Proxy 2 2641 SIP/2.0 180 Ringing 2642 Via: SIP/2.0/UDP ss2.wcom.com:5060 2643 Via: SIP/2.0/UDP ss1.wcom.com:5060 2644 Via: SIP/2.0/UDP here.com:5060 2645 From: TheBigGuy 2646 To: TheLittleGuy ;tag=314159 2647 Call-Id: 12345600@here.com 2648 CSeq: 1 INVITE 2649 Content-Length: 0 2651 180 Ringing F8 2652 Proxy 2 -> Proxy 1 2654 SIP/2.0 180 Ringing 2655 Via: SIP/2.0/UDP ss1.wcom.com:5060 2656 Via: SIP/2.0/UDP here.com:5060 2657 From: TheBigGuy 2658 To: TheLittleGuy ;tag=314159 2659 Call-Id: 12345600@here.com 2660 CSeq: 1 INVITE 2661 Content-Length: 0 2663 180 Ringing F9 2664 Proxy 1 2665 -> A 2667 SIP/2.0 180 Ringing 2668 Via: SIP/2.0/UDP here.com:5060 2669 From: TheBigGuy 2670 To: TheLittleGuy ;tag=314159 2671 Call-Id: 12345600@here.com 2672 CSeq: 1 INVITE 2673 Content-Length: 0 2675 480 Temporarily Unavailable F10 2676 B -> Proxy 2 2678 SIP/2.0 480 Temporarily Unavailable 2679 Via: SIP/2.0/UDP ss2.wcom.com:5060 2680 Via: SIP/2.0/UDP ss1.wcom.com:5060 2681 Via: SIP/2.0/UDP here.com:5060 2682 From: TheBigGuy 2683 To: TheLittleGuy ;tag=314159 2684 Call-Id: 12345600@here.com 2685 CSeq: 1 INVITE 2686 Content-Length: 0 2688 ACK F11 2689 Proxy 2 ->B 2691 ACK sip: UserB@there.com SIP/2.0 2692 Via: SIP/2.0/UDP ss2.wcom.com:5060 2693 From: TheBigGuy 2694 To: TheLittleGuy ;tag=314159 2695 Call-Id: 12345600@here.com 2696 CSeq: 1 ACK 2697 Content-Length: 0 2699 480 Temporarily Unavailable F12 2700 Proxy 2 -> Proxy 1 2702 SIP/2.0 480 Temporarily Unavailable 2703 Via: SIP/2.0/UDP ss1.wcom.com:5060 2704 Via: SIP/2.0/UDP here.com:5060 2705 From: TheBigGuy 2706 To: TheLittleGuy ;tag=314159 2707 Call-Id: 12345600@here.com 2708 CSeq: 1 INVITE 2709 Content-Length: 0 2710 ACK F13 2711 Proxy 1 -> Proxy 2 2713 ACK sip: UserB@ss2.wcom.com SIP/2.0 2714 Via: SIP/2.0/UDP ss1.wcom.com:5060 2715 From: TheBigGuy 2716 To: TheLittleGuy ;tag=314159 2717 Call-Id: 12345600@here.com 2718 CSeq: 1 ACK 2719 Content-Length: 0 2721 480 Temporarily Unavailable F14 2722 Proxy1 2723 -> A 2725 SIP/2.0 480 Temporarily Unavailable 2726 Via: SIP/2.0/UDP here.com:5060 2727 From: TheBigGuy 2728 To: TheLittleGuy ;tag=314159 2729 Call-Id: 12345600@here.com 2730 CSeq: 1 INVITE 2731 Content-Length: 0 2733 ACK F15 2734 A -> Proxy 1 2736 ACK sip:UserB@ss1.wcom.com SIP/2.0 2737 Via: SIP/2.0/UDP here.com:5060 2738 From: TheBigGuy 2739 To: TheLittleGuy ;tag=314159 2740 Call-Id: 12345600@here.com 2741 CSeq: 1 ACK 2742 Content-Length: 0 2744 4 SIP to Gateway Dialing 2746 In the following scenarios, User A (TheBigGuy sip:UserA@here.com) is 2747 a SIP phone or other SIP-enabled device. User B is reachable via the 2748 PSTN at global telephone number +1-972-555-2222. User A places a call 2749 to User B through a Proxy Server SS1 and a Network Gateway. In other 2750 scenarios, User A places calls to User C, who is served via a PBX 2751 (Private Branch Exchange) and is identified by a private extension 2752 444-3333, or global number +1-918-555-3333. Note that User A uses 2753 his/her global telephone number +1-314-555-1111 in the From header in 2754 the INVITE messages. This then gives the Gateway the option of using 2755 this header to populate the calling party identification field in 2756 subsequent signaling (CgPN in ISUP). Left open is the issue of how 2757 the Gateway can determine the accuracy of the telephone number, 2758 necessary before passing it as a valid CgPN in the PSTN. Note that 2759 User A still uses his/her SIP URL in the Contact header. 2761 There is a major SIP issue in the call flows in this section and 2762 Section 6. In-band alerting (ringing tone, busy tone, recorded 2763 announcements, etc.) is present in the PSTN speech path after the 2764 receipt of the SS7 Address Complete Message (ACM) which maps to the 2765 SIP 180 Ringing response. In a SIP to SIP call, the media path is 2766 not established until the call is answered (200 OK sent). In order 2767 for the SIP caller User A to hear this alerting, it is necessary that 2768 an early media path be established to perform this. This is the 2769 purpose of the 183 Session Progress[5] responses used throughout this 2770 document in place of the 180 Ringinig. 2772 This document will be updated as this issue is further refined, with 2773 the possible inclusion of reliable responses[6] and/or additional SIP 2774 headers. 2776 4.1 Success Scenarios 2778 In these scenarios, User A is a SIP phone or other SIP-enabled 2779 device. User A places a call to User B in the PSTN or User C on a 2780 PBX through a Proxy Server SS1 and a Gateway. 2782 4.1.1 2783 Successful SIP to ISUP PSTN call 2785 User A dials the globalized E.164 number +1-972-555-2222 to reach 2786 User B. Note that A might have only dialed the last 7 digits, or 2787 some other dialing plan. It is assumed that the SIP User Agent 2788 Client converts the digits into a global number and puts them into a 2789 SIP URL. 2791 User A could use either their SIP address (sip:UserA@here.com) or SIP 2792 telephone number (sip:+1-314-555-1111@ss1.wcom.com;user=phone) in the 2793 From header. In this example, the telephone number is included, and 2794 it is shown as being passed as calling party identification through 2795 the Network Gateway to User B (F5). Note that for this number to be 2796 passed into the SS7 network, it would have to be somehow verified for 2797 accuracy. 2799 In this scenario, User B answers the call then User A disconnects the 2800 call. Signaling between NGW1 and User B's telephone switch is SS7. 2802 Message Details 2804 INVITE F1 2805 A->SS1 2807 INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 2808 Via: SIP/2.0/UDP here.com:5060 2809 From: TheBigGuy ;user=phone 2810 To: TheLittleGuy ;user=phone 2811 Call-Id: 12345600@here.com 2812 CSeq: 1 INVITE 2813 Contact: TheBigGuy 2814 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 2815 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 2816 uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" 2817 Content-Type: application/sdp 2818 Content-Length: 132 2820 v=0 2821 o=UserA 2890844526 2890844526 IN IP4 here.com 2822 t=0 0 2823 c=IN IP4 here.com 2824 m=audio 49170 RTP/AVP 0 2825 a=rtpmap:0 PCMU/8000 2826 (100 Trying F2 2827 SS1 -> User A) 2829 SIP/2.0 100 Trying 2830 Via: SIP/2.0/UDP here.com:5060 2831 From: TheBigGuy ;user=phone 2832 To: TheLittleGuy ;user=phone 2833 Call-Id: 12345600@here.com 2834 CSeq: 1 INVITE 2835 Content-Length: 0 2837 /* SS1 uses a location manager function to determine where B is 2838 actually located. Based upon location analysis the call is forwarded 2839 to NGW1. Client for A prepares to receive data on port 49170 from 2840 the network.*/ 2842 INVITE F3 2843 SS1 2844 -> NGW1 2846 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 2847 Via: SIP/2.0/UDP ss1.wcom.com:5060 2848 Via: SIP/2.0/UDP here.com:5060 2849 Record-Route: 2850 From: TheBigGuy ;user=phone 2851 To: TheLittleGuy ;user=phone 2852 Call-Id: 12345600@here.com 2853 CSeq: 1 INVITE 2854 Contact: TheBigGuy 2855 Content-Type: application/sdp 2856 Content-Length: 132 2858 v=0 2859 o=UserA 2890844526 2890844526 IN IP4 here.com 2860 t=0 0 2861 c=IN IP4 here.com 2862 m=audio 49170 RTP/AVP 0 2863 a=rtpmap:0 PCMU/8000 2865 (100 Trying F4 2866 GW -> SS1) 2868 SIP/2.0 100 Trying 2869 Via: SIP/2.0/UDP ss1.wcom.com:5060 2870 From: TheBigGuy ;user=phone 2871 To: TheLittleGuy ;user=phone 2872 Call-Id: 12345600@here.com 2873 CSeq: 1 INVITE 2874 Content-Length: 0 2876 IAM F5 2877 GW -> User B 2879 IAM 2880 CdPN=972-555-2222,NPI=E.164,NOA=National 2881 CgPN=314-555-1111,NPI=E.164,NOA=National 2882 USI=Speech 2883 CPT=0 0 2884 C=Normal 2885 CCI =Not Required 2886 ACM F6 2887 User B 2888 -> GW 2890 ACM 2891 Charge Indicator=No Charge 2892 Called Party Status=no indication 2893 Called Party's Category=ordinary subscriber 2894 End To End Method=none available 2895 Interworking=encountered 2896 End to End Information=none available 2897 ISUP Indicator=not used all the way 2898 ISDN Access Terminating access non ISDN 2899 Echo Control=not included 2900 183 Session Progress F7 2901 GW -> SS1 2903 SIP/2.0 183 Session Progress 2904 Via: SIP/2.0/UDP ss1.wcom.com:5060 2905 Via: SIP/2.0/UDP here.com:5060 2906 From: TheBigGuy ;user=phone 2907 To: TheLittleGuy 2908 ;user=phone;tag=314159 2909 Call-Id: 12345600@here.com 2910 CSeq: 1 INVITE 2911 Content-Type: application/sdp 2912 Content-Length: 150 2914 v=0 2915 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 2916 t=0 0 2917 c=IN IP4 gatewayone.wcom.com 2918 m=audio 3456 RTP/AVP 0 2919 a=rtpmap:0 PCMU/8000 2920 /* SS1 proxies the OK to User A 2921 At this point the GW will start sendin an RTP path to the receive 2922 port on A encoding anything that is being received from B via the 2923 PSTN network (i.e. ringing) */ 2925 183 Session Progress F8 2926 SS1 2927 ->User A 2929 SIP/2.0 183 Session Progress 2930 Via: SIP/2.0/UDP here.com:5060 2931 From: TheBigGuy ;user=phone 2932 To: TheLittleGuy 2933 ;user=phone;tag=314159 2934 Call-Id: 12345600@here.com 2935 CSeq: 1 INVITE 2936 Content-Type: application/sdp 2937 Content-Length: 150 2939 v=0 2940 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 2941 t=0 0 2942 c=IN IP4 gatewayone.wcom.com 2943 m=audio 3456 RTP/AVP 0 2944 a=rtpmap:0 PCMU/8000 2946 ANM F9 2947 User B 2948 -> GW 2950 ANM 2952 200 OK F10 2953 GW -> SS1 2955 SIP/2.0 200 OK 2956 Via: SIP/2.0/UDP ss1.wcom.com:5060 2957 Via: SIP/2.0/UDP here.com:5060 2958 Record-Route: 2959 From: TheBigGuy ;user=phone 2960 To: TheLittleGuy 2961 ;user=phone;tag=314159 2962 Call-Id: 12345600@here.com 2963 CSeq: 1 INVITE 2964 Contact: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 2965 Content-Type: application/sdp 2966 Content-Length: 150 2968 v=0 2969 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 2970 t=0 0 2971 c=IN IP4 gatewayone.wcom.com 2972 m=audio 3456 RTP/AVP 0 2973 a=rtpmap:0 PCMU/8000 2975 200 OK F11 2976 SS1 2977 ->User A 2979 SIP/2.0 200 OK 2980 Via: SIP/2.0/UDP here.com:5060 2981 Record-Route: 2982 From: TheBigGuy ;user=phone 2983 To: TheLittleGuy 2984 ;user=phone;tag=314159 2985 Call-Id: 12345600@here.com 2986 CSeq: 1 INVITE 2987 Contact: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 2988 Content-Type: application/sdp 2989 Content-Length: 150 2991 v=0 2992 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 2993 t=0 0 2994 c=IN IP4 gatewayone.wcom.com 2995 m=audio 3456 RTP/AVP 0 2996 a=rtpmap:0 PCMU/8000 2998 ACK F12 2999 A->SS1 3001 ACK sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 3002 Via: SIP/2.0/UDP here.com:5060 3003 Route: 3004 From: TheBigGuy ;user=phone 3005 To: TheLittleGuy 3006 ;user=phone;tag=314159 3007 Call-Id: 12345600@here.com 3008 CSeq: 1 ACK 3009 Content-Length: 0 3010 ACK F13 3011 SS1 -> GW 3013 ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 3014 Via: SIP/2.0/UDP ss1.wcom.com:5060 3015 Via: SIP/2.0/UDP here.com:5060 3016 From: TheBigGuy ;user=phone 3017 To: TheLittleGuy 3018 ;user=phone;tag=314159 3019 Call-Id: 12345600@here.com 3020 CSeq: 1 ACK 3021 Content-Length: 0 3023 /* RTP streams are established between A and B(via the GW) */ 3025 /* User A Hangs Up with User B. */ 3026 BYE F14 3027 A->SS1 3029 BYE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 3030 Via: SIP/2.0/UDP here.com:5060 3031 Route: 3032 From: TheBigGuy ;user=phone 3033 To: TheLittleGuy 3034 ;user=phone;tag=314159 3035 Call-Id: 12345600@here.com 3036 CSeq: 2 BYE 3037 Content-Length: 0 3039 BYE F15 3040 SS1 -> GW 3042 BYE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 3043 Via: SIP/2.0/UDP ss1.wcom.com:5060 3044 Via: SIP/2.0/UDP here.com:5060 3045 From: TheBigGuy ;user=phone 3046 To: TheLittleGuy 3047 ;user=phone;tag=314159 3048 Call-Id: 12345600@here.com 3049 CSeq: 2 BYE 3050 Content-Length: 0 3052 200 OK F16 3053 GW -> SS1 3055 SIP/2.0 200 OK 3056 Via: SIP/2.0/UDP ss1.wcom.com:5060 3057 Via: SIP/2.0/UDP here.com:5060 3058 From: TheBigGuy ;user=phone 3059 To: TheLittleGuy 3060 ;user=phone;tag=314159 3061 Call-Id: 12345600@here.com 3062 CSeq: 2 BYE 3063 Content-Length: 0 3065 200 OK F17 3066 SS1->A 3068 SIP/2.0 200 OK 3069 Via: SIP/2.0/UDP here.com:5060 3070 From: TheBigGuy ;user=phone 3071 To: TheLittleGuy 3072 ;user=phone;tag=314159 3073 Call-Id: 12345600@here.com 3074 CSeq: 2 BYE 3075 Content-Length: 0 3077 REL F18 3078 GW -> B 3080 REL 3081 CauseCode=16 Normal 3082 CodingStandard=CCITT 3083 RLC F19 3084 B -> GW 3086 RLC 3088 4.1.2 3089 Successful SIP to ISDN PBX call 3091 User A is a SIP device while User C is connected via an Enterprise 3092 Gateway (GW1) to a PBX. The PBX connection is via a ISDN trunk 3093 group. User A dials User C's telephone number (918-555-3333) which 3094 is globalized and put into a SIP URL. 3096 The phone-context in the username portion of the Request-URI in 3097 message F3 is used to identify the context (customer, trunk group, or 3098 line) in which the private number 444-3333 is valid. Otherwise, this 3099 INVITE message could get forwarded and the context of the digits 3100 could become lost and the call unroutable. See section 1.1 for a 3101 discussion of phone-context. 3103 Proxy SS1 looks up the telephone number and locates the Enterprise 3104 Gateway that servers User C. User C is identified by its extension 3105 (444-3333) in the Request-URI sent to GW1. 3107 User A hears the ringing provided by the Gateway on the media path 3108 established after the 183 Session Progress response is received. 3109 Signaling between GW1 and PBX C is shown as ISDN. 3111 Message Details 3113 INVITE F1 3114 A->SS1 3116 INVITE sip:+1-918-555-3333@ss1.wcom.com;user=phone SIP/2.0 3117 Via: SIP/2.0/UDP here.com:5060 3118 From: TheBigGuy ;user=phone 3119 To: TheOtherGuy ;user=phone 3120 Call-Id: 12345600@here.com 3121 CSeq: 1 INVITE 3122 Contact: TheBigGuy 3123 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 3124 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 3125 uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" 3126 Content-Type: application/sdp 3127 Content-Length: 132 3129 v=0 3130 o=UserA 2890844526 2890844526 IN IP4 here.com 3131 t=0 0 3132 c=IN IP4 here.com 3133 m=audio 49170 RTP/AVP 0 3134 a=rtpmap:0 PCMU/8000 3135 (100 Trying F2 3136 SS1 -> User A) 3138 SIP/2.0 100 Trying 3139 Via: SIP/2.0/UDP here.com:5060 3140 From: TheBigGuy ;user=phone 3141 To: TheOtherGuy ;user=phone 3142 Call-Id: 12345600@here.com 3143 CSeq: 1 INVITE 3144 Content-Length: 0 3146 /* SS1 uses a location manager function to determine where B is 3147 actually located. Based upon location analysis the call is forwarded 3148 to GW1 with the extension determined as 444-3333. Client for A 3149 prepares to receive data on port 49170 from the network.*/ 3151 INVITE F3 3152 SS1 -> GW1 3154 INVITE sip:444-3333,phone-context=p1234@gw1.wcom.com;user=phone 3155 SIP/2.0 3156 Via: SIP/2.0/UDP ss1.wcom.com:5060 3157 Via: SIP/2.0/UDP here.com:5060 3158 Record-Route: 3159 From: TheBigGuy ;user=phone 3160 To: TheOtherGuy ;user=phone 3161 Call-Id: 12345600@here.com 3162 CSeq: 1 INVITE 3163 Contact: TheBigGuy 3164 Content-Type: application/sdp 3165 Content-Length: 132 3167 v=0 3168 o=UserA 2890844526 2890844526 IN IP4 here.com 3169 t=0 0 3170 c=IN IP4 here.com 3171 m=audio 49170 RTP/AVP 0 3172 a=rtpmap:0 PCMU/8000 3174 (100 Trying F4 3175 GW -> SS1) 3177 SIP/2.0 100 Trying 3178 Via: SIP/2.0/UDP ss1.wcom.com:5060 3179 From: TheBigGuy ;user=phone 3180 To: TheOtherGuy ;user=phone 3181 Call-Id: 12345600@here.com 3182 CSeq: 1 INVITE 3183 Content-Length: 0 3185 SETUP F5 3186 GW -> User C 3188 Protocol discriminator=Q.931 3189 Call reference: Flag=0, CR value=any valid value not in use 3190 Message type=SETUP 3191 Bearer capability: Information transfer capability=0 (Speech) or 16 3192 (3.1 kHz audio) 3193 Channel identification=Preferred or exclusive B-channel 3194 Progress indicator=1 (Call is not end-to-end ISDN;further call 3195 progress information may be available inband) 3196 Called party number: 3197 Type of number and numbering plan ID=?? (private numbering plan) 3198 Digits=444-3333 3199 CALL PROCeeding F6 3200 User C 3201 -> GW 3203 Protocol discriminator=Q.931 3204 Call reference: Flag=1, CR value=value in F5 SETUP message 3205 Message type=CALL PROC 3206 Channel identification=Exclusive B-channel 3207 PROGress F7 3208 User C 3209 -> GW 3211 Protocol discriminator=Q.931 3212 Call reference: Flag=1, CR value=value in F5 SETUP message 3213 Message type=PROG 3214 Progress indicator=1 (Call is not end-to-end ISDN;further call 3215 progress information may be available inband) 3216 183 Session Progress F8 3217 GW -> SS1 3219 SIP/2.0 183 Session Progress 3220 Via: SIP/2.0/UDP ss1.wcom.com:5060 3221 Via: SIP/2.0/UDP here.com:5060 3222 From: TheBigGuy ;user=phone 3223 To: TheOtherGuy 3224 ;user=phone;tag=314159 3225 Call-Id: 12345600@here.com 3226 CSeq: 1 INVITE 3227 Content-Type: application/sdp 3228 Content-Length: 150 3230 v=0 3231 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 3232 t=0 0 3233 c=IN IP4 gatewayone.wcom.com 3234 m=audio 3456 RTP/AVP 0 3235 a=rtpmap:0 PCMU/8000 3237 /* The GW will establish an RTP path to the receive port on A 3238 encoding anything that is being received from C via the PSTN network 3239 (i.e. ringing) 3240 183 Session Progress F9 3241 SS1 3242 ->User A 3244 SIP/2.0 183 Session Progress 3245 Via: SIP/2.0/UDP here.com:5060 3246 From: TheBigGuy ;user=phone 3247 To: TheOtherGuy 3248 ;user=phone;tag=314159 3249 Call-Id: 12345600@here.com 3250 CSeq: 1 INVITE 3251 Content-Type: application/sdp 3252 Content-Length: 150 3254 v=0 3255 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 3256 t=0 0 3257 c=IN IP4 gatewayone.wcom.com 3258 m=audio 3456 RTP/AVP 0 3259 a=rtpmap:0 PCMU/8000 3261 CONNect F10 3262 User C 3263 -> GW 3265 Protocol discriminator=Q.931 3266 Call reference: Flag=1, CR value=value in F5 SETUP message 3267 Message type=CONN 3268 CONNect ACK F11 3269 GW -> User C 3271 Protocol discriminator=Q.931 3272 Call reference: Flag=0, CR value=value in F5 SETUP message 3273 Message type=CONN ACK 3274 200 OK F12 3275 GW -> SS1 3277 SIP/2.0 200 OK 3278 Via: SIP/2.0/UDP ss1.wcom.com:5060 3279 Via: SIP/2.0/UDP here.com:5060 3280 Record-Route: 3281 From: TheBigGuy ;user=phone 3282 To: TheOtherGuy 3283 ;user=phone;tag=314159 3284 Call-Id: 12345600@here.com 3285 CSeq: 1 INVITE 3286 Contact: sip:444-3333,phone-context=p1234@gw1.wcom.com ;user=phone 3287 Content-Type: application/sdp 3288 Content-Length: 150 3290 v=0 3291 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 3292 t=0 0 3293 c=IN IP4 gatewayone.wcom.com 3294 m=audio 3456 RTP/AVP 0 3295 a=rtpmap:0 PCMU/8000 3297 200 OK F13 3298 SS1 3299 ->User A 3301 SIP/2.0 200 OK 3302 Via: SIP/2.0/UDP here.com:5060 3303 Record-Route: 3304 From: TheBigGuy ;user=phone 3305 To: TheOtherGuy 3306 ;user=phone;tag=314159 3307 Call-Id: 12345600@here.com 3308 CSeq: 1 INVITE 3309 Contact: sip:444-3333,phone-context=p1234@gw1.wcom.com ;user=phone 3310 Content-Type: application/sdp 3311 Content-Length: 150 3313 v=0 3314 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 3315 t=0 0 3316 c=IN IP4 gatewayone.wcom.com 3317 m=audio 3456 RTP/AVP 0 3318 a=rtpmap:0 PCMU/8000 3320 ACK F14 3321 A->SS1 3322 ACK sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone SIP/2.0 3323 Via: SIP/2.0/UDP here.com:5060 3324 Route: ;user=phone 3325 From: TheBigGuy ;user=phone 3326 To: TheOtherGuy 3327 ;user=phone;tag=314159 3328 Call-Id: 12345600@here.com 3329 CSeq: 1 ACK 3330 Content-Length: 0 3332 ACK F15 3333 SS1 -> GW 3335 ACK sip:444-3333,phone-context=p1234@gw1.wcom.com;user=phone SIP/2.0 3336 Via: SIP/2.0/UDP ss1.wcom.com:5060 3337 Via: SIP/2.0/UDP here.com:5060 3338 From: TheBigGuy ;user=phone 3339 To: TheOtherGuy 3340 ;user=phone;tag=314159 3341 Call-Id: 12345600@here.com 3342 CSeq: 1 ACK 3343 Content-Length: 0 3345 /* RTP streams are established between A and B(via the GW) */ 3347 /* User A Hangs Up with User B. */ 3348 BYE F16 3349 A->SS1 3351 BYE sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone SIP/2.0 3352 Via: SIP/2.0/UDP here.com:5060 3353 Route: ;user=phone 3354 From: TheBigGuy ;user=phone 3355 To: TheOtherGuy 3356 ;user=phone;tag=314159 3357 Call-Id: 12345600@here.com 3358 CSeq: 2 BYE 3359 Content-Length: 0 3360 BYE F17 3361 SS1 -> GW 3363 BYE sip:444-3333,phone-context=p1234@gw1.wcom.com;user=phone SIP/2.0 3364 Via: SIP/2.0/UDP ss1.wcom.com:5060 3365 Via: SIP/2.0/UDP here.com:5060 3366 From: TheBigGuy ;user=phone 3367 To: TheOtherGuy 3368 ;user=phone;tag=314159 3369 Call-Id: 12345600@here.com 3370 CSeq: 2 BYE 3371 Content-Length: 0 3373 200 OK F18 3374 GW -> SS1 3376 SIP/2.0 200 OK 3377 Via: SIP/2.0/UDP ss1.wcom.com:5060 3378 Via: SIP/2.0/UDP here.com:5060 3379 From: TheBigGuy ;user=phone 3380 To: TheOtherGuy 3381 ;user=phone;tag=314159 3382 Call-Id: 12345600@here.com 3383 CSeq: 2 BYE 3384 Content-Length: 0 3386 200 OK F19 3387 SS1->A 3389 SIP/2.0 200 OK 3390 Via: SIP/2.0/UDP here.com:5060 3391 From: TheBigGuy ;user=phone 3392 To: TheOtherGuy 3393 ;user=phone;tag=314159 3394 Call-Id: 12345600@here.com 3395 CSeq: 2 BYE 3396 Content-Length: 0 3398 DISConnect F20 3399 GW -> User C 3401 Protocol discriminator=Q.931 3402 Call reference: Flag=1, CR value=value in F4 SETUP message 3403 Message type=DISC 3404 Cause=16 (Normal clearing) 3405 RELease F21 3406 User C 3407 -> GW 3409 Protocol discriminator=Q.931 3410 Call reference: Flag=0, CR value=value in F4 SETUP message 3411 Message type=REL 3412 RELease COMplete F22 3413 GW -> User C 3415 Protocol discriminator=Q.931 3416 Call reference: Flag=1, CR value=value in F4 SETUP message 3417 Message type=REL COM 3419 4.1.3 3420 Successful SIP to ISUP PSTN call with overflow 3421 User A calls User B through SS1 working as a proxy server. SS1 tries 3422 an Enterprise Gateway GW1. GW1 is not available and responds with a 3423 503 Service Unavailable (F4). The call is then routed to a Network 3424 Gateway NGW2. User B answers the call. The call is terminated when 3425 User A disconnects the call. NGW2 and User B's telephone switch use 3426 SS7 signaling. 3428 Message Details 3430 INVITE F1 3431 A->SS1 3433 INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 3434 Via: SIP/2.0/UDP here.com:5060 3435 From: TheBigGuy ;user=phone 3436 To: TheLittleGuy ;user=phone 3437 Call-Id: 12345600@here.com 3438 CSeq: 1 INVITE 3439 Contact: TheBigGuy 3440 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 3441 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 3442 uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" 3443 Content-Type: application/sdp 3444 Content-Length: 132 3446 v=0 3447 o=UserA 2890844526 2890844526 IN IP4 here.com 3448 t=0 0 3449 c=IN IP4 here.com 3450 m=audio 49170 RTP/AVP 0 3451 a=rtpmap:0 PCMU/8000 3453 /* SS1 uses a location manager function to determine where B is 3454 actually located. SS1 receives a primary route NGW1 and a secondary 3455 route NGW2. NGW1 is tried first */ 3457 INVITE F2 3458 SS1 -> NGW1 3460 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 3461 Via: SIP/2.0/UDP ss1.wcom.com:5060 3462 Via: SIP/2.0/UDP here.com:5060 3463 Record-Route: 3464 From: TheBigGuy ;user=phone 3465 To: TheLittleGuy ;user=phone 3466 Call-Id: 12345600@here.com 3467 CSeq: 1 INVITE 3468 Contact: TheBigGuy 3469 Content-Type: application/sdp 3470 Content-Length: 132 3472 v=0 3473 o=UserA 2890844526 2890844526 IN IP4 here.com 3474 t=0 0 3475 c=IN IP4 here.com 3476 m=audio 49170 RTP/AVP 0 3477 a=rtpmap:0 PCMU/8000 3479 (100 Trying F3 Proxy -> User A) 3481 SIP/2.0 100 Trying 3482 Via: SIP/2.0/UDP ss1.wcom.com:5060 3483 Via: SIP/2.0/UDP here.com:5060 3484 From: TheBigGuy ;user=phone 3485 To: TheLittleGuy ;user=phone 3486 Call-Id: 12345600@here.com 3487 CSeq: 1 INVITE 3488 Content-Length: 0 3490 503 Service Unavailable F4 3491 NGW1-> SS1 3493 SIP/2.0 503 Service Unavailable 3494 Via: SIP/2.0/UDP ss1.wcom.com:5060 3495 Via: SIP/2.0/UDP here.com:5060 3496 Record-Route: 3497 From: TheBigGuy ;user=phone 3498 To: TheLittleGuy GW2 3521 ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 3522 Via: SIP/2.0/UDP ss1.wcom.com:5060 3523 From: TheBigGuy ;user=phone 3524 To: TheLittleGuy 3525 ;user=phone;tag=123456789 3526 Call-Id: 12345600@here.com 3527 CSeq: 1 INVITE 3528 Content-Length: 0 3530 /* SS1 now tries secondary route to NGW2 */ 3531 INVITE F6 SS1 3532 -> NGW2 3534 INVITE sip:+1-972-555-2222@ngw2.wcom.com;user=phone SIP/2.0 3535 Via: SIP/2.0/UDP ss1.wcom.com:5060 3536 Via: SIP/2.0/UDP here.com:5060 3537 Record-Route: 3538 From: TheBigGuy ;user=phone 3539 To: TheLittleGuy ;user=phone 3540 Call-Id: 12345600@here.com 3541 CSeq: 1 INVITE 3542 Contact: TheBigGuy 3543 Content-Type: application/sdp 3544 Content-Length: 132 3546 v=0 3547 o=UserA 2890844526 2890844526 IN IP4 here.com 3548 t=0 0 3549 c=IN IP4 here.com 3550 m=audio 49170 RTP/AVP 0 3551 a=rtpmap:0 PCMU/8000 3552 IAM F7 3553 NGW2 -> User B 3555 IAM 3556 CdPN=972-555-2222,NPI=E.164,NOA=National 3557 CgPN=314-555-1111,NPI=E.164,NOA=National 3558 USI=Speech 3559 CPT=0 0 3560 C=Normal 3561 CCI =Not Required 3562 ACM F8 3563 User B 3564 -> NGW2 3566 ACM 3567 Charge Indicator=No Charge 3568 Called Party Status=no indication 3569 Called Party's Category=ordinary subscriber 3570 End To End Method=none available 3571 Interworking=encountered 3572 End to End Information=none available 3573 ISUP Indicator=not used all the way 3574 ISDN Access Terminating access non ISDN 3575 Echo Control=not included 3576 183 Session Progress F9 3577 NGW2 3578 -> SS1 3580 SIP/2.0 183 Session Progress 3581 Via: SIP/2.0/UDP ss1.wcom.com:5060 3582 Via: SIP/2.0/UDP here.com:5060 3583 From: TheBigGuy ;user=phone 3584 To: TheLittleGuy 3585 ;user=phone;tag=314159 3586 Call-Id: 12345600@here.com 3587 CSeq: 1 INVITE 3588 Content-Type: application/sdp 3589 Content-Length: 150 3591 v=0 3592 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 3593 t=0 0 3594 c=IN IP4 gatewayone.wcom.com 3595 m=audio 3456 RTP/AVP 0 3596 a=rtpmap:0 PCMU/8000 3597 /* The GW will establish an RTP path to the receive port on A 3598 encoding anything that is being received from B via the PSTN network 3599 (i.e. ringing) */ 3601 183 Session Progress F10 3602 SS1 -> User A 3604 SIP/2.0 183 Session Progress 3605 Via: SIP/2.0/UDP here.com:5060 3606 From: TheBigGuy ;user=phone 3607 To: TheLittleGuy 3608 ;user=phone;tag=314159 3609 Call-Id: 12345600@here.com 3610 CSeq: 1 INVITE 3611 Content-Type: application/sdp 3612 Content-Length: 150 3614 v=0 3615 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 3616 t=0 0 3617 c=IN IP4 gatewayone.wcom.com 3618 m=audio 3456 RTP/AVP 0 3619 a=rtpmap:0 PCMU/8000 3621 ANM F11 3622 User B 3623 -> NGW2 3625 ANM 3627 200 OK F12 3628 NGW2 3629 -> SS1 3631 SIP/2.0 200 OK 3632 Via: SIP/2.0/UDP ss1.wcom.com:5060 3633 Via: SIP/2.0/UDP here.com:5060 3634 Record-Route: 3635 From: TheBigGuy ;user=phone 3636 To: TheLittleGuy 3637 ;user=phone;tag=314159 3638 Call-Id: 12345600@here.com 3639 CSeq: 1 INVITE 3640 Contact: sip:+1-972-555-2222@ngw2.wcom.com;user=phone 3641 Content-Type: application/sdp 3642 Content-Length: 150 3644 v=0 3645 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 3646 t=0 0 3647 c=IN IP4 gatewayone.wcom.com 3648 m=audio 3456 RTP/AVP 0 3649 a=rtpmap:0 PCMU/8000 3651 200 OK F13 3652 SS1 3653 -> User A 3655 SIP/2.0 200 OK 3656 Via: SIP/2.0/UDP here.com:5060 3657 Record-Route: 3658 From: TheBigGuy ;user=phone 3659 To: TheLittleGuy 3660 ;user=phone;tag=314159 3661 Call-Id: 12345600@here.com 3662 CSeq: 1 INVITE 3663 Contact: sip:+1-972-555-2222@ngw2.wcom.com;user=phone 3664 Content-Type: application/sdp 3665 Content-Length: 150 3667 v=0 3668 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 3669 t=0 0 3670 c=IN IP4 gatewayone.wcom.com 3671 m=audio 3456 RTP/AVP 0 3672 a=rtpmap:0 PCMU/8000 3674 ACK F14 3675 A->SS1 3677 ACK sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 3678 Via: SIP/2.0/UDP here.com:5060 3679 Route: ;user=phone 3680 From: TheBigGuy ;user=phone 3681 To: TheLittleGuy 3682 ;user=phone;tag=314159 3683 Call-Id: 12345600@here.com 3684 CSeq: 1 ACK 3685 Content-Length: 0 3687 ACK F15 3688 SS1 -> NGW2 3689 ACK sip:+1-972-555-2222@ngw2.wcom.com;user=phone SIP/2.0 3690 Via: SIP/2.0/UDP ss1.wcom.com:5060 3691 Via: SIP/2.0/UDP here.com:5060 3692 From: TheBigGuy ;user=phone 3693 To: TheLittleGuy 3694 ;user=phone;tag=314159 3695 Call-Id: 12345600@here.com 3696 CSeq: 1 ACK 3697 Content-Length: 0 3699 /* RTP streams are established between A and B(via the GW) */ 3701 /* User A Hangs Up with User B. */ 3702 BYE F16 3703 A->SS1 3705 BYE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 3706 Via: SIP/2.0/UDP here.com:5060 3707 Route: ;user=phone 3708 From: TheBigGuy ;user=phone 3709 To: TheLittleGuy 3710 ;user=phone;tag=314159 3711 Call-Id: 12345600@here.com 3712 CSeq: 2 BYE 3713 Content-Length: 0 3715 BYE F17 3716 SS1 -> NGW2 3718 BYE sip:+1-972-555-2222@ngw2.wcom.com;user=phone SIP/2.0 3719 Via: SIP/2.0/UDP ss1.wcom.com:5060 3720 Via: SIP/2.0/UDP here.com:5060 3721 From: TheBigGuy ;user=phone 3722 To: TheLittleGuy 3723 ;user=phone;tag=314159 3724 Call-Id: 12345600@here.com 3725 CSeq: 2 BYE 3726 Content-Length: 0 3728 200 OK F18 3729 NGW2 -> SS1 3730 SIP/2.0 200 OK 3731 Via: SIP/2.0/UDP ss1.wcom.com:5060 3732 Via: SIP/2.0/UDP here.com:5060 3733 From: TheBigGuy ;user=phone 3734 To: TheLittleGuy 3735 ;user=phone;tag=314159 3736 Call-Id: 12345600@here.com 3737 CSeq: 2 BYE 3738 Content-Length: 0 3740 200 OK F19 3741 SS1-> User A 3743 SIP/2.0 200 OK 3744 Via: SIP/2.0/UDP here.com:5060 3745 From: TheBigGuy ;user=phone 3746 To: TheLittleGuy 3747 ;user=phone;tag=314159 3748 Call-Id: 12345600@here.com 3749 CSeq: 2 BYE 3750 Content-Length: 0 3752 REL F20 3753 GW -> B 3755 REL 3756 CauseCode=16 Normal 3757 CodingStandard=CCITT 3758 RLC F21 3759 B -> GW 3761 RLC 3763 4.2 Failure Scenarios 3765 In these failure scenarios, the call does not complete. In most 3766 cases, however, a media stream is still setup. This is due to the 3767 fact that most failures in dialing to the PSTN result in in-band 3768 tones (busy, reorder tones) or announcements ("The number you have 3769 dialed has changed. The new number is..."). The 183 Session 3770 Progress[5] response containing SDP media information is used to 3771 setup this early media path so that the caller User A knows the final 3772 disposition of the call. 3774 The media stream is either terminated by the caller after the tone or 3775 announcement has been heard and understood, or by the Gateway after a 3776 timer expires. 3778 In other failure scenarios, a SS7 Release with Cause Code is mapped 3779 to a SIP response. In these scenarios, the early media path is not 3780 used, but the actual failure code is conveyed to the caller by the 3781 SIP User Agent Client. 3783 4.2.1 3784 Unsuccessful SIP to PSTN call: Treatment from PSTN 3786 User A calls User B in the PSTN through a proxy server SS1 and a 3787 Network Gateway NGW1. The call is rejected by the PSTN with an in- 3788 band treatment (tone or recording) played. User A hears the 3789 treatment and then issues a CANCEL (F9) to terminate the call. (A BYE 3790 is not sent since no final response was ever received by User A.) 3792 Message Details 3794 INVITE F1 3795 A->SS1 3797 INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 3798 Via: SIP/2.0/UDP here.com:5060 3799 From: TheBigGuy ;user=phone 3800 To: TheLittleGuy ;user=phone 3801 Call-Id: 12345600@here.com 3802 CSeq: 1 INVITE 3803 Contact: TheBigGuy 3804 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 3805 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 3806 uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" 3807 Content-Type: application/sdp 3808 Content-Length: 132 3810 v=0 3811 o=UserA 2890844526 2890844526 IN IP4 here.com 3812 t=0 0 3813 c=IN IP4 here.com 3814 m=audio 49170 RTP/AVP 0 3815 a=rtpmap:0 PCMU/8000 3817 (100 Trying F2 SS1 -> A) 3819 SIP/2.0 100 Trying 3820 Via: SIP/2.0/UDP here.com:5060 3821 From: TheBigGuy ;user=phone 3822 To: TheLittleGuy ;user=phone 3823 Call-Id: 12345600@here.com 3824 CSeq: 1 INVITE 3825 Content-Length: 0 3826 /* SS1 uses a location manager function to determine where B is 3827 actually located. Based upon location analysis the call is forwarded 3828 to NGW1. Client for A prepares to receive data on port 49170 from 3829 the network.*/ 3831 INVITE F3 3832 SS1 -> NGW1 3834 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 3835 Via: SIP/2.0/UDP ss1.wcom.com:5060 3836 Via: SIP/2.0/UDP here.com:5060 3837 Record-Route: 3838 From: TheBigGuy ;user=phone 3839 To: TheLittleGuy ;user=phone 3840 Call-Id: 12345600@here.com 3841 CSeq: 1 INVITE 3842 Contact: TheBigGuy 3843 Content-Type: application/sdp 3844 Content-Length: 132 3846 v=0 3847 o=UserA 2890844526 2890844526 IN IP4 here.com 3848 t=0 0 3849 c=IN IP4 here.com 3850 m=audio 49170 RTP/AVP 0 3851 a=rtpmap:0 PCMU/8000 3853 (100 Trying F4 NGW1-> SS1) 3855 SIP/2.0 100 Trying 3856 Via: SIP/2.0/UDP ss1.wcom.com:5060 3857 From: TheBigGuy ;user=phone 3858 To: TheLittleGuy ;user=phone 3859 Call-Id: 12345600@here.com 3860 CSeq: 1 INVITE 3861 Content-Length: 0 3863 IAM F5 3864 GW -> User B 3866 IAM 3867 CdPN=972-555-2222,NPI=E.164,NOA=National 3868 CgPN=314-555-1111,NPI=E.164,NOA=National 3869 USI=Speech 3870 CPT=0 0 3871 C=Normal 3872 CCI =Not Required 3873 ACM F6 3874 User B 3875 -> GW 3877 ACM 3878 Charge Indicator=No Charge 3879 Called Party Status=no indication 3880 Called Party's Category=ordinary subscriber 3881 End To End Method=none available 3882 Interworking=encountered 3883 End to End Information=none available 3884 ISUP Indicator=not used all the way 3885 ISDN Access Terminating access non ISDN 3886 Echo Control=not included 3887 183 Session Progress F7 3888 GW -> SS1 3890 SIP/2.0 183 Session Progress 3891 Via: SIP/2.0/UDP ss1.wcom.com:5060 3892 Via: SIP/2.0/UDP here.com:5060 3893 From: TheBigGuy ;user=phone 3894 To: TheLittleGuy 3895 ;user=phone;tag=314159 3896 Call-Id: 12345600@here.com 3897 CSeq: 1 INVITE 3898 Content-Type: application/sdp 3899 Content-Length: 150 3901 v=0 3902 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 3903 t=0 0 3904 c=IN IP4 gatewayone.wcom.com 3905 m=audio 3456 RTP/AVP 0 3906 a=rtpmap:0 PCMU/8000 3908 183 Session Progress F8 3909 SS1 3910 ->User A 3912 SIP/2.0 183 Session Progress 3913 Via: SIP/2.0/UDP here.com:5060 3914 From: TheBigGuy ;user=phone 3915 To: TheLittleGuy 3916 ;user=phone;tag=314159 3917 Call-Id: 12345600@here.com 3918 CSeq: 1 INVITE 3919 Content-Type: application/sdp 3920 Content-Length: 150 3921 v=0 3922 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 3923 t=0 0 3924 c=IN IP4 gatewayone.wcom.com 3925 m=audio 3456 RTP/AVP 0 3926 a=rtpmap:0 PCMU/8000 3928 /* User A listens to recorded announcement from the PSTN then hangs 3929 up */ 3931 CANCEL F9 3932 A->SS1 3934 CANCEL sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 3935 Via: SIP/2.0/UDP here.com:5060 3936 From: TheBigGuy ;user=phone 3937 To: TheLittleGuy 3938 ;user=phone;tag=314159 3939 Call-Id: 12345600@here.com 3940 CSeq: 1 CANCEL 3941 Content-Length: 0 3943 CANCEL F10 3944 SS1 -> GW 3946 CANCEL sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 3947 Via: SIP/2.0/UDP ss1.wcom.com:5060 3948 Via: SIP/2.0/UDP here.com:5060 3949 From: TheBigGuy ;user=phone 3950 To: TheLittleGuy 3951 ;user=phone;tag=314159 3952 Call-Id: 12345600@here.com 3953 CSeq: 1 CANCEL 3954 Content-Length: 0 3956 200 OK F11 3957 GW -> SS1 3959 SIP/2.0 200 OK 3960 Via: SIP/2.0/UDP ss1.wcom.com:5060 3961 Via: SIP/2.0/UDP here.com:5060 3962 From: TheBigGuy ;user=phone 3963 To: TheLittleGuy 3964 ;user=phone;tag=314159 3965 Call-Id: 12345600@here.com 3966 CSeq: 1 CANCEL 3967 Content-Length: 0 3969 200 OK F12 3970 SS1->A 3972 SIP/2.0 200 OK 3973 Via: SIP/2.0/UDP here.com:5060 3974 From: TheBigGuy ;user=phone 3975 To: TheLittleGuy 3976 ;user=phone;tag=314159 3977 Call-Id: 12345600@here.com 3978 CSeq: 1 CANCEL 3979 Content-Length: 0 3981 REL F13 3982 GW -> B 3984 REL 3985 CauseCode=16 Normal 3986 CodingStandard=CCITT 3987 RLC F14 3988 B -> GW 3990 RLC 3992 4.2.2 3993 Unsuccessful SIP to PSTN: REL w/Cause from PSTN 3994 User A calls PSTN User B through a Proxy Server SS1 and a Network 3995 Gateway NGW1. However, User A does not provide enough digits for the 3996 call to be completed. (In a real scenario, this call might have been 3997 rejected by SS1 based on incomplete address. However, especially on 3998 international calls, the number of digits in the number is not 3999 obvious, and this scenario may result.) The call is rejected by the 4000 PSTN with a SS7 Release message REL containing a specific Cause 4001 value. This cause value (28) is mapped by the Gateway to a SIP 484 4002 Address Incomplete response which is proxied back to User A. 4004 Message Details 4006 INVITE F1 4007 A->SS1 4009 INVITE sip:+1-972-555-222@ss1.wcom.com;user=phone SIP/2.0 4010 Via: SIP/2.0/UDP here.com:5060 4011 From: TheBigGuy ;user=phone 4012 To: TheLittleGuy ;user=phone 4013 Call-Id: 12345600@here.com 4014 CSeq: 1 INVITE 4015 Contact: TheBigGuy 4016 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 4017 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 4018 uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" 4019 Content-Type: application/sdp 4020 Content-Length: 132 4022 v=0 4023 o=UserA 2890844526 2890844526 IN IP4 here.com 4024 t=0 0 4025 c=IN IP4 here.com 4026 m=audio 49170 RTP/AVP 0 4027 a=rtpmap:0 PCMU/8000 4029 (100 Trying F2 SS1 -> A) 4031 SIP/2.0 100 Trying 4032 Via: SIP/2.0/UDP here.com:5060 4033 From: TheBigGuy ;user=phone 4034 To: TheLittleGuy ;user=phone 4035 Call-Id: 12345600@here.com 4036 CSeq: 1 INVITE 4037 Content-Length: 0 4038 /* SS1 uses a location manager function to determine where B is 4039 actually located. Based upon location analysis the call is forwarded 4040 to NGW1. Client for A prepares to receive data on port 49170 from 4041 the network. */ 4043 INVITE F3 4044 SS1 4045 -> NGW1 4047 INVITE sip:+1-972-555-222@ngw1.wcom.com;user=phone SIP/2.0 4048 Via: SIP/2.0/UDP ss1.wcom.com:5060 4049 Via: SIP/2.0/UDP here.com:5060 4050 Record-Route: 4051 From: TheBigGuy ;user=phone 4052 To: TheLittleGuy ;user=phone 4053 Call-Id: 12345600@here.com 4054 CSeq: 1 INVITE 4055 Contact: TheBigGuy 4056 Content-Type: application/sdp 4057 Content-Length: 132 4059 v=0 4060 o=UserA 2890844526 2890844526 IN IP4 here.com 4061 t=0 0 4062 c=IN IP4 here.com 4063 m=audio 49170 RTP/AVP 0 4064 a=rtpmap:0 PCMU/8000 4066 (100 Trying F4 NGW1-> SS1) 4068 SIP/2.0 100 Trying 4069 Via: SIP/2.0/UDP ss1.wcom.com:5060 4070 From: TheBigGuy ;user=phone 4071 To: TheLittleGuy ;user=phone 4072 Call-Id: 12345600@here.com 4073 CSeq: 1 INVITE 4074 Content-Length: 0 4076 IAM F5 4077 GW -> User B 4079 IAM 4080 CdPN=972-555-2222,NPI=E.164,NOA=National 4081 CgPN=314-555-1111,NPI=E.164,NOA=National 4082 USI=Speech 4083 CPT=0 0 4084 C=Normal 4085 CCI =Not Required 4086 REL F6 4087 User B 4088 -> GW 4090 REL 4091 CauseValue=28 Address Incomplete 4092 CodingStandard=CCITT 4093 RLC F7 4094 GW -> User B 4096 RLC 4098 /* Network Gateway maps CauseValue=28 to the SIP message 484 Address 4099 Incomplete */ 4101 484 Address Incomplete F8 4102 GW -> SS1 4104 SIP/2.0 484 Address Incomplete 4105 Via: SIP/2.0/UDP ss1.wcom.com:5060 4106 Via: SIP/2.0/UDP here.com:5060 4107 From: TheBigGuy ;user=phone 4108 To: TheLittleGuy ;user=phone; 4109 tag=314159 4110 Call-Id: 12345600@here.com 4111 CSeq: 1 INVITE 4112 Content-Length: 0 4114 ACK F9 4115 SS1 -> GW 4117 ACK sip:+1-972-555-222@ngw1.wcom.com;user=phone SIP/2.0 4118 Via: SIP/2.0/UDP ss1.wcom.com:5060 4119 From: TheBigGuy ;user=phone 4120 To: TheLittleGuy ;user=phone; 4121 tag=314159 4122 Call-Id: 12345600@here.com 4123 CSeq: 1 ACK 4124 Content-Length: 0 4126 484 Address Incomplete F10 4127 SS1 -> User A 4128 SIP/2.0 484 Address Incomplete 4129 Via: SIP/2.0/UDP here.com:5060 4130 From: TheBigGuy ;user=phone 4131 To: TheLittleGuy ;user=phone; 4132 tag=314159 4133 Call-Id: 12345600@here.com 4134 CSeq: 1 INVITE 4135 Content-Length: 0 4137 ACK F11 4138 User A 4139 -> SS1 4141 ACK sip:+1-972-555-222@ss1.wcom.com;user=phone SIP/2.0 4142 Via: SIP/2.0/UDP here.com:5060 4143 From: TheBigGuy ;user=phone 4144 To: TheLittleGuy ;user=phone; 4145 tag=314159 4146 Call-Id: 12345600@here.com 4147 CSeq: 1 ACK 4148 Content-Length: 0 4150 4.2.3 4151 Unsuccessful SIP to PSTN: ANM Timeout 4153 User A calls User B in the PSTN through a proxy server SS1 and a 4154 Gateway GW1. The call is released by the Gateway after its ISUP T9 4155 timer expires due to no ANswer Message (ANM) being received. The 4156 Gateway sends a SS7 Release REL message to the PSTN and a 480 4157 Temporarily Unavailable response to User A in the SIP network. 4159 Message Details 4161 INVITE F1 4162 A->SS1 4164 INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 4165 Via: SIP/2.0/UDP here.com:5060 4166 From: TheBigGuy ;user=phone 4167 To: TheLittleGuy ;user=phone 4168 Call-Id: 12345600@here.com 4169 CSeq: 1 INVITE 4170 Contact: TheBigGuy 4171 Authorization:Digest username="UserA", realm="MCI WorldCom SIP", 4172 nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", 4173 uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" 4174 Content-Type: application/sdp 4175 Content-Length: 132 4177 v=0 4178 o=UserA 2890844526 2890844526 IN IP4 here.com 4179 t=0 0 4180 c=IN IP4 here.com 4181 m=audio 49170 RTP/AVP 0 4182 a=rtpmap:0 PCMU/8000 4184 /* SS1 uses a location manager function to determine where B is 4185 actually located. Based upon location analysis the call is forwarded 4186 to GW1. Client for A prepares to receive data on port 49170 from the 4187 network.*/ 4189 (100 Trying F2 SS1 4190 -> A) 4192 SIP/2.0 100 Trying 4193 Via: SIP/2.0/UDP here.com:5060 4194 From: TheBigGuy ;user=phone 4195 To: TheLittleGuy ;user=phone 4196 Call-Id: 12345600@here.com 4197 CSeq: 1 INVITE 4198 Content-Length: 0 4200 INVITE F3 4201 SS1 -> GW1 4202 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 4203 Via: SIP/2.0/UDP ss1.wcom.com:5060 4204 Via: SIP/2.0/UDP here.com:5060 4205 Record-Route: 4206 From: TheBigGuy ;user=phone 4207 To: TheLittleGuy ;user=phone 4208 Call-Id: 12345600@here.com 4209 CSeq: 1 INVITE 4210 Contact: TheBigGuy 4211 Content-Type: application/sdp 4212 Content-Length: 132 4214 v=0 4215 o=UserA 2890844526 2890844526 IN IP4 here.com 4216 t=0 0 4217 c=IN IP4 here.com 4218 m=audio 49170 RTP/AVP 0 4219 a=rtpmap:0 PCMU/8000 4221 (100 Trying F4 GW1 4222 -> SS1) 4224 SIP/2.0 100 Trying 4225 Via: SIP/2.0/UDP ss1.wcom.com:5060 4226 From: TheBigGuy ;user=phone 4227 To: TheLittleGuy ;user=phone 4228 Call-Id: 12345600@here.com 4229 CSeq: 1 INVITE 4230 Content-Length: 0 4232 IAM F5 4233 GW -> User B 4235 IAM 4236 CdPN=972-555-2222,NPI=E.164,NOA=National 4237 CgPN=314-555-1111,NPI=E.164,NOA=National 4238 USI=Speech 4239 CPT=0 0 4240 C=Normal 4241 CCI =Not Required 4242 ACM F6 4243 User B 4244 -> GW 4246 ACM 4247 Charge Indicator=No Charge 4248 Called Party Status=no indication 4249 Called Party's Category=ordinary subscriber 4250 End To End Method=none available 4251 Interworking=encountered 4252 End to End Information=none available 4253 ISUP Indicator=not used all the way 4254 ISDN Access Terminating access non ISDN 4255 Echo Control=not included 4256 183 Session Progress F7 4257 GW -> SS1 4259 SIP/2.0 183 Session Progress 4260 Via: SIP/2.0/UDP ss1.wcom.com:5060 4261 Via: SIP/2.0/UDP here.com:5060 4262 From: TheBigGuy ;user=phone 4263 To: TheLittleGuy 4264 ;user=phone;tag=314159 4265 Call-Id: 12345600@here.com 4266 CSeq: 1 INVITE 4267 Content-Type: application/sdp 4268 Content-Length: 150 4270 v=0 4271 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 4272 t=0 0 4273 c=IN IP4 gatewayone.wcom.com 4274 m=audio 3456 RTP/AVP 0 4275 a=rtpmap:0 PCMU/8000 4277 183 Session Progress F8 4278 SS1 -> User A 4280 SIP/2.0 183 Session Progress 4281 Via: SIP/2.0/UDP here.com:5060 4282 From: TheBigGuy ;user=phone 4283 To: TheLittleGuy 4284 ;user=phone;tag=314159 4285 Call-Id: 12345600@here.com 4286 CSeq: 1 INVITE 4287 Content-Type: application/sdp 4288 Content-Length: 150 4290 v=0 4291 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 4292 t=0 0 4293 c=IN IP4 gatewayone.wcom.com 4294 m=audio 3456 RTP/AVP 0 4295 a=rtpmap:0 PCMU/8000 4296 /* After ISUP T9 Timer expires, Network Gateway sends REL to ISUP 4297 network and 480 to SIP network */ 4299 REL F9 GW -> User B 4301 REL 4302 CauseCode=16 Normal 4303 CodingStandard=CCITT 4304 RLC F10 User B 4305 -> GW 4307 RLC 4308 480 Temporarily Unavailable F11 4309 GW -> SS1 4311 SIP/2.0 480 Temporarily Unavailable 4312 Via: SIP/2.0/UDP ss1.wcom.com:5060 4313 Via: SIP/2.0/UDP here.com:5060 4314 From: TheBigGuy ;user=phone 4315 To: TheLittleGuy 4316 ;user=phone;tag=314159 4317 Call-Id: 12345600@here.com 4318 CSeq: 1 INVITE 4319 Content-Length: 0 4321 ACK F12 4322 SS1 -> GW 4324 ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 4325 Via: SIP/2.0/UDP ss1.wcom.com:5060 4326 From: TheBigGuy ;user=phone 4327 To: TheLittleGuy 4328 ;user=phone;tag=314159 4329 Call-Id: 12345600@here.com 4330 CSeq: 1 ACK 4331 Content-Length: 0 4333 480 Temporarily Unavailable F13 4334 SS1 -> User A 4336 SIP/2.0 480 Temporarily Unavailable 4337 Via: SIP/2.0/UDP here.com:5060 4338 From: TheBigGuy ;user=phone 4339 To: TheLittleGuy 4340 ;user=phone;tag=314159 4341 Call-Id: 12345600@here.com 4342 CSeq: 1 INVITE 4343 Content-Length: 0 4345 ACK F14 4346 User A 4347 -> SS1 4349 ACK sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 4350 Via: SIP/2.0/UDP here.com:5060 4351 From: TheBigGuy ;user=phone 4352 To: TheLittleGuy 4353 ;user=phone;tag=314159 4354 Call-Id: 12345600@here.com 4355 CSeq: 1 ACK 4356 Content-Length: 0 4358 5 Gateway to SIP Dialing 4360 5.1 Success Scenarios 4362 In these scenarios, User A is placing calls from the PSTN to User B 4363 in a SIP network. User A's telephone switch signals to a Network 4364 Gateway (NGW1) using SS7. 4366 Since the called SIP User Agent does not send in-band signaling 4367 information, no early media path needs to be established on the IP 4368 side. As a result, the 183 Session Progress response is not used. 4369 However, NGW1 will establish a one way speech path prior to call 4370 completion, and generate ringing for the PSTN caller. Any tones or 4371 recordings are generated by NGW1 and played in this speech path. 4372 When the call completes successfully, NGW1 bridges the PSTN speech 4373 path with the IP media path. 4375 5.1.1 4376 Successful PSTN to SIP call 4378 In this scenario, User A from the PSTN calls User B through a Network 4379 Gateway NGW1 and Proxy Server SS1. When User B answers the call the 4380 media path is setup end-to-end. The call terminates when User A 4381 hangs up the call, with User A's telephone switch sending a SS7 4382 Release message which is mapped to a BYE by NGW1. 4384 Message Details 4386 IAM F1 4387 User A -> GW 4389 IAM 4390 CgPN=314-555-1111,NPI=E.164,NOA=National 4391 CdPN=972-555-2222,NPI=E.164,NOA=National 4392 USI=Speech 4393 CPT=0 0 4394 C=Normal 4395 CCI =Not Required 4396 INVITE F2 4397 A->SS1 4399 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 4400 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4401 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4402 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 4403 Call-Id: 12345602@ngw1.wcom.com 4404 CSeq: 1 INVITE 4405 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4406 Content-Type: application/sdp 4407 Content-Length: 150 4409 v=0 4410 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 4411 t=0 0 4412 c=IN IP4 gatewayone.wcom.com 4413 m=audio 3456 RTP/AVP 0 4414 a=rtpmap:0 PCMU/8000 4416 /* SS1 uses a location manager function to determine where B is 4417 actually located. Based upon location analysis the call is forwarded 4418 to GW1. GW1 prepares to receive data on port 3456 from User A.*/ 4419 INVITE F3 4420 SS1 4421 ->User B 4423 INVITE sip:UserB@there.com SIP/2.0 4424 Via: SIP/2.0/UDP ss1.wcom.com:5060 4425 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4426 Record-Route: 4427 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4428 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 4429 Call-Id: 12345602@ngw1.wcom.com 4430 CSeq: 1 INVITE 4431 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4432 Content-Type: application/sdp 4433 Content-Length: 150 4435 v=0 4436 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 4437 t=0 0 4438 c=IN IP4 gatewayone.wcom.com 4439 m=audio 3456 RTP/AVP 0 4440 a=rtpmap:0 PCMU/8000 4442 (100 Trying F4 User B -> SS1) 4444 SIP/2.0 100 Trying 4445 Via: SIP/2.0/UDP ss1.wcom.com:5060 4446 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4447 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4448 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 4449 Call-Id: 12345602@ngw1.wcom.com 4450 CSeq: 1 INVITE 4451 Content-Length: 0 4453 180 Ringing F5 User B -> SS1 4455 SIP/2.0 180 Ringing 4456 Via: SIP/2.0/UDP ss1.wcom.com:5060 4457 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4458 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4459 To: sip:+1-972-555-2222@ngw1.wcom.com>;user=phone;tag=314159 4460 Call-Id: 12345602@ngw1.wcom.com 4461 CSeq: 1 INVITE 4462 Content-Length: 0 4464 180 Ringing F6 SS1 4465 -> NGW1 4467 SIP/2.0 180 Ringing 4468 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4469 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4470 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4471 Call-Id: 12345602@ngw1.wcom.com 4472 CSeq: 1 INVITE 4473 Content-Length: 0 4475 ACM F7 4476 NGW1-> User A 4478 ACM 4479 Charge Indicator=No Charge 4480 Called Party Status=no indication 4481 Called Party's Category=ordinary subscriber 4482 End To End Method=none available 4483 Interworking=encountered 4484 End to End Information=none available 4485 ISUP Indicator=not used all the way 4486 ISDN Access Terminating access non ISDN 4487 Echo Control=not included 4488 200 OK F8 User B -> SS1 4490 SIP/2.0 200 OK 4491 Via: SIP/2.0/UDP ss1.wcom.com:5060 4492 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4493 Record-Route: 4494 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4495 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4496 Call-Id: 12345602@ngw1.wcom.com 4497 Contact: TheLittleGuy 4498 CSeq: 1 INVITE 4499 Content-Type: application/sdp 4500 Content-Length: 150 4502 v=0 4503 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 4504 t=0 0 4505 c=IN IP4 110.111.112.113 4506 m=audio 3456 RTP/AVP 0 4507 a=rtpmap:0 PCMU/8000 4509 200 OK F9 SS1 4510 -> NGW1 4512 SIP/2.0 200 OK 4513 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4514 Record-Route: 4515 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4516 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4517 Call-Id: 12345602@ngw1.wcom.com 4518 CSeq: 1 INVITE 4519 Contact: TheLittleGuy 4520 Content-Type: application/sdp 4521 Content-Length: 150 4523 v=0 4524 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 4525 t=0 0 4526 c=IN IP4 110.111.112.113 4527 m=audio 3456 RTP/AVP 0 4528 a=rtpmap:0 PCMU/8000 4530 ACK F10 GW1 -> SS1 4532 ACK sip:UserB@ss1.wcom.com SIP/2.0 4533 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4534 Route: 4535 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4536 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4537 Call-Id: 12345602@ngw1.wcom.com 4538 CSeq: 1 ACK 4539 Content-Length: 0 4541 ACK F11 SS1 -> User B 4543 ACK sip:UserB@there.com SIP/2.0 4544 Via: SIP/2.0/UDP ss1.wcom.com:5060 4545 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4546 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4547 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4548 Call-Id: 12345602@ngw1.wcom.com 4549 CSeq: 1 ACK 4550 Content-Length: 0 4552 ANM F12 4553 User B 4554 -> NGW1 4556 ANM 4557 /* RTP streams are established between A and B (via the GW) */ 4559 /* User A Hangs Up with User B. */ 4560 REL F13 4561 User A 4562 -> NGW1 4564 REL 4565 CauseCode=16 Normal 4566 CodingStandard=CCITT 4567 RLC F14 4568 NGW1-> User A 4570 RLC 4571 BYE F15 4572 NGW1-> SS1 4574 BYE sip:UserB@ss1.wcom.com SIP/2.0 4575 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4576 Route: 4577 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4578 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4579 Call-Id: 12345602@ngw1.wcom.com 4580 CSeq: 2 BYE 4581 Content-Length: 0 4583 BYE F16 4584 SS1 -> User B 4586 BYE sip:UserB@there.com SIP/2.0 4587 Via: SIP/2.0/UDP ss1.wcom.com:5060 4588 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4589 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4590 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4591 Call-Id: 12345602@ngw1.wcom.com 4592 CSeq: 2 BYE 4593 Content-Length: 0 4595 200 OK F17 4596 User B 4597 -> SS1 4599 SIP/2.0 200 OK 4600 Via: SIP/2.0/UDP ss1.wcom.com:5060 4601 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4602 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4603 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4604 Call-Id: 12345602@ngw1.wcom.com 4605 CSeq: 2 BYE 4606 Content-Length: 0 4608 200 OK F18 SS1 -> NGW1 4610 SIP/2.0 200 OK 4611 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4612 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4613 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4614 Call-Id: 12345602@ngw1.wcom.com 4615 CSeq: 2 BYE 4616 Content-Length: 0 4618 5.1.2 4619 Successful PSTN to SIP call, Fast Answer 4621 This "fast answer" scenario is similar to 5.1.1 except that User B 4622 immediately accepts the call, sending a 200 OK (F5) without sending a 4623 180 Ringing response. The Gateway then sends an Answer Message (ANM) 4624 without sending an Address Complete Message (ACM). 4626 Message Details 4628 IAM F1 4629 User A -> GW 4631 IAM 4632 CgPN=314-555-1111,NPI=E.164,NOA=National 4633 CdPN=972-555-2222,NPI=E.164,NOA=National 4634 USI=Speech 4635 CPT=0 0 4636 C=Normal 4637 CCI =Not Required 4638 INVITE F2 4639 GW -> SS1 4641 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 4642 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4643 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4644 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 4645 Call-Id: 12345602@ngw1.wcom.com 4646 CSeq: 1 INVITE 4647 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4648 Content-Type: application/sdp 4649 Content-Length: 150 4651 v=0 4652 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 4653 t=0 0 4654 c=IN IP4 gatewayone.wcom.com 4655 m=audio 3456 RTP/AVP 0 4656 a=rtpmap:0 PCMU/8000 4658 /* SS1 uses a location manager function to determine where B is 4659 actually located. Based upon location analysis the call is forwarded 4660 to User B. User B prepares to receive data on port 3456 from User 4661 A.*/ 4662 INVITE F3 4663 SS1 -> User B 4665 INVITE UserB@there.com SIP/2.0 4666 Via: SIP/2.0/UDP ss1.wcom.com:5060 4667 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4668 Record-Route: 4669 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4670 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 4671 Call-Id: 12345602@ngw1.wcom.com 4672 CSeq: 1 INVITE 4673 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4674 Content-Type: application/sdp 4675 Content-Length: 150 4677 v=0 4678 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 4679 t=0 0 4680 c=IN IP4 gatewayone.wcom.com 4681 m=audio 3456 RTP/AVP 0 4682 a=rtpmap:0 PCMU/8000 4684 (100 Trying F4 4685 SS1 -> GW1) 4687 SIP/2.0 100 Trying 4688 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4689 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4690 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 4691 Call-Id: 12345602@ngw1.wcom.com 4692 CSeq: 1 INVITE 4693 Content-Length: 0 4695 200 OK F5 User B -> SS1 4697 SIP/2.0 200 OK 4698 Via: SIP/2.0/UDP ss1.wcom.com:5060 4699 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4700 Record-Route: 4701 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4702 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4703 Call-Id: 12345602@ngw1.wcom.com 4704 CSeq: 1 INVITE 4705 Contact: TheLittleGuy 4706 Content-Type: application/sdp 4707 Content-Length: 150 4708 v=0 4709 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 4710 t=0 0 4711 c=IN IP4 110.111.112.113 4712 m=audio 3456 RTP/AVP 0 4713 a=rtpmap:0 PCMU/8000 4715 200 OK F6 SS1 4716 -> GW1 4718 SIP/2.0 200 OK 4719 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4720 Record-Route: 4721 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4722 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4723 Call-Id: 12345602@ngw1.wcom.com 4724 CSeq: 1 INVITE 4725 Contact: TheLittleGuy 4726 Content-Type: application/sdp 4727 Content-Length: 150 4729 v=0 4730 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 4731 t=0 0 4732 c=IN IP4 110.111.112.113 4733 m=audio 3456 RTP/AVP 0 4734 a=rtpmap:0 PCMU/8000 4736 ACK F7 GW1 4737 ->SS1 4739 ACK UserB@ss1.wcom.com SIP/2.0 4740 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4741 Route: 4742 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4743 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4744 Call-Id: 12345602@ngw1.wcom.com 4745 CSeq: 1 ACK 4746 Content-Length: 0 4748 ACK F8 SS1 4749 ->User B 4751 ACK UserB@there.com SIP/2.0 4752 Via: SIP/2.0/UDP ss1.wcom.com:5060 4753 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4754 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4755 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4756 Call-Id: 12345602@ngw1.wcom.com 4757 CSeq: 1 ACK 4758 Content-Length: 0 4760 ANM F9 4761 User B 4762 ->GW 4764 ANM 4766 /* RTP streams are established between A and B (via the GW) */ 4768 /* User A Hangs Up with User B. */ 4769 REL F10 4770 User A 4771 ->GW 4773 REL 4774 CauseCode=16 Normal 4775 CodingStandard=CCITT 4776 RLC F11 4777 GW -> User A 4779 RLC 4780 BYE F12 4781 GW -> SS1 4783 BYE sip:UserB@ss1.wcom.com SIP/2.0 4784 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4785 Route: 4786 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4787 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4788 Call-Id: 12345602@ngw1.wcom.com 4789 CSeq: 2 BYE 4790 Content-Length: 0 4792 BYE F13 4793 SS1 -> User B 4795 BYE sip:UserB@there.com SIP/2.0 4796 Via: SIP/2.0/UDP ss1.wcom.com:5060 4797 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4798 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4799 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4800 Call-Id: 12345602@ngw1.wcom.com 4801 CSeq: 2 BYE 4802 Content-Length: 0 4804 200 OK F14 4805 User B 4806 -> SS1 4808 SIP/2.0 200 OK 4809 Via: SIP/2.0/UDP ss1.wcom.com:5060 4810 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4811 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4812 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4813 Call-Id: 12345602@ngw1.wcom.com 4814 CSeq: 2 BYE 4815 Content-Length: 0 4817 200 OK F15 SS1 ->GW 4819 SIP/2.0 200 OK 4820 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4821 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 4822 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 4823 Call-Id: 12345602@ngw1.wcom.com 4824 CSeq: 2 BYE 4825 Content-Length: 0 4827 5.1.3 4828 Successful PBX to SIP call 4830 In this scenario, User A calls from PBX A to User B through GW1 and 4831 SS1 working as a proxy server. Signaling between PBX A and GW1 is 4832 Feature Group B (FGB) circuit associated signaling (in-band mult- 4833 frequency outpulsing). After the receipt of the 180 Ringing from 4834 User B, GW1 generates ringing tone for User A. 4836 User B answers the call by sending a 200 OK. The call terminates 4837 when User A hangs up, causing GW1 to send a BYE. 4839 Message Details 4841 MF Digits F1 4842 PBX A 4843 ->GW1 4845 KP 1 972 555 2222 ST 4846 INVITE F2 4847 A->SS1 4849 INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 4850 Via: SIP/2.0/UDP gw1.wcom.com:5060 4851 From: PBX_A ;user=phone 4852 To: sip:+1-972-555-2222@ss1.wcom.com;user=phone 4853 Call-Id: 12345602@gw1.wcom.com 4854 CSeq: 1 INVITE 4855 Contact: PBX_A ;user=phone 4856 Content-Type: application/sdp 4857 Content-Length: 150 4859 v=0 4860 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 4861 t=0 0 4862 c=IN IP4 gatewayone.wcom.com 4863 m=audio 3456 RTP/AVP 0 4864 a=rtpmap:0 PCMU/8000 4866 /* SS1 uses a location manager function to determine where B is 4867 actually located. Based upon location analysis the call is forwarded 4868 to GW1.*/ 4869 INVITE F3 4870 SS1 4871 ->User B 4873 INVITE sip:UserB@there.com SIP/2.0 4874 Via: SIP/2.0/UDP ss1.wcom.com:5060 4875 Via: SIP/2.0/UDP gw1.wcom.com:5060 4876 Record-Route: 4877 From: PBX_A ;user=phone 4878 To: sip:+1-972-555-2222@ss1.wcom.com;user=phone 4879 Call-Id: 12345602@gw1.wcom.com 4880 CSeq: 1 INVITE 4881 Contact: PBX_A ;user=phone 4882 Content-Type: application/sdp 4883 Content-Length: 150 4885 v=0 4886 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 4887 t=0 0 4888 c=IN IP4 gatewayone.wcom.com 4889 m=audio 3456 RTP/AVP 0 4890 a=rtpmap:0 PCMU/8000 4892 (100 Trying F4 SS1 4893 -> GW) 4895 SIP/2.0 100 Trying 4896 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4897 From: PBX_A ;user=phone 4898 To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 4899 Call-Id: 12345602@gw1.wcom.com 4900 CSeq: 1 INVITE 4901 Content-Length: 0 4903 180 Ringing F5 User B -> SS1 4905 SIP/2.0 180 Ringing 4906 Via: SIP/2.0/UDP ss1.wcom.com:5060 4907 Via: SIP/2.0/UDP ngw1.wcom.com:5060 4908 From: PBX_A ;user=phone 4909 To: sip:+1-972-555-2222@ss1.wcom.com;user=phone 4910 ;tag=314159 4911 Call-Id: 12345602@gw1.wcom.com 4912 CSeq: 1 INVITE 4913 Content-Length: 0 4915 180 Ringing F6 SS1 4916 -> GW1 4918 SIP/2.0 180 Ringing 4919 Via: SIP/2.0/UDP gw1.wcom.com:5060 4920 From: PBX_A ;user=phone 4921 To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 4922 Call-Id: 12345602@gw1.wcom.com 4923 CSeq: 1 INVITE 4924 Content-Length: 0 4926 /* One way Voice path is established between GW and the PBX for 4927 ringing. */ 4929 200 OK F7 User B -> SS1 4931 SIP/2.0 200 OK 4932 Via: SIP/2.0/UDP ss1.wcom.com:5060 4933 Via: SIP/2.0/UDP gw1.wcom.com:5060 4934 Record-Route: 4935 From: PBX_A ;user=phone 4936 To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 4937 Call-Id: 12345602@gw1.wcom.com 4938 Contact: TheLittleGuy 4939 CSeq: 1 INVITE 4940 Content-Type: application/sdp 4941 Content-Length: 134 4943 v=0 4944 o=UserB 2890844527 2890844527 IN IP4 there.com 4945 t=0 0 4946 c=IN IP4 110.111.112.113 4947 m=audio 3456 RTP/AVP 0 4948 a=rtpmap:0 PCMU/8000 4950 200 OK F8 SS1 4951 -> GW1 4953 SIP/2.0 200 OK 4954 Via: SIP/2.0/UDP gw1.wcom.com:5060 4955 Record-Route: 4956 From: PBX_A ;user=phone 4957 To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 4958 Call-Id: 12345602@gw1.wcom.com 4959 CSeq: 1 INVITE 4960 Contact: TheLittleGuy 4961 Content-Type: application/sdp 4962 Content-Length: 134 4964 v=0 4965 o=UserB 2890844527 2890844527 IN IP4 there.com 4966 t=0 0 4967 c=IN IP4 110.111.112.113 4968 m=audio 3456 RTP/AVP 0 4969 a=rtpmap:0 PCMU/8000 4971 ACK F9 GW1 4972 ->SS1 4974 ACK sip:UserB@ss1.wcom.com SIP/2.0 4975 Via: SIP/2.0/UDP gw1.wcom.com:5060 4976 Route: 4977 From: PBX_A ;user=phone 4978 To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 4979 Call-Id: 12345602@gw1.wcom.com 4980 CSeq: 1 ACK 4981 Content-Length: 0 4983 ACK F10 SS1 4984 ->User B 4986 ACK sip:UserB@there.com SIP/2.0 4987 Via: SIP/2.0/UDP ss1.wcom.com:5060 4988 Via: SIP/2.0/UDP gw1.wcom.com:5060 4989 From: PBX_A ;user=phone 4990 To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 4991 Call-Id: 12345602@ngw1.wcom.com 4992 CSeq: 1 ACK 4993 Content-Length: 0 4995 /* RTP streams are established between A and B (via the GW) */ 4997 /* User A Hangs Up with User B. */ 4998 BYE F11 4999 GW -> SS1 5001 BYE sip:UserB@ss1.wcom.com SIP/2.0 5002 Via: SIP/2.0/UDP gw1.wcom.com:5060 5003 Route: 5004 From: PBX_A ;user=phone 5005 To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 5006 Call-Id: 12345602@gw1.wcom.com 5007 CSeq: 2 BYE 5008 Content-Length: 0 5010 BYE F12 5011 SS1 -> User B 5013 BYE sip:UserB@there.com SIP/2.0 5014 Via: SIP/2.0/UDP ss1.wcom.com:5060 5015 Via: SIP/2.0/UDP gw1.wcom.com:5060 5016 From: PBX_A ;user=phone 5017 To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 5018 Call-Id: 12345602@gw1.wcom.com 5019 CSeq: 2 BYE 5020 Content-Length: 0 5022 200 OK F13 5023 User B 5024 -> SS1 5026 SIP/2.0 200 OK 5027 Via: SIP/2.0/UDP ss1.wcom.com:5060 5028 Via: SIP/2.0/UDP gw1.wcom.com:5060 5029 From: PBX_A ;user=phone 5030 To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 5031 Call-Id: 12345602@ngw1.wcom.com 5032 CSeq: 2 BYE 5033 Content-Length: 0 5035 200 OK F14 SS1 ->GW 5037 SIP/2.0 200 OK 5038 Via: SIP/2.0/UDP gw1.wcom.com:5060 5039 From: PBX_A ;user=phone 5040 To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 5041 Call-Id: 12345602@gw1.wcom.com 5042 CSeq: 2 BYE 5043 Content-Length: 0 5045 5.2 Failure Scenarios 5047 5.2.1 5048 Unsuccessful PSTN to SIP REL, SIP error mapped to REL 5050 User A attempts to call a SIP user through SS1 working as a proxy 5051 server. SS1 is unable to find any routing for the number. The call 5052 is rejected by SS1 with a REL message containing a specific Cause 5053 value mapped by the gateway based on the SIP error. 5055 Message Details 5057 IAM F1 5058 User A 5059 -> GW 5061 IAM 5062 CgPN=314-555-1111,NPI=E.164,NOA=National 5063 CdPN=972-555-9999,NPI=E.164,NOA=National 5064 USI=Speech 5065 CPT=0 0 5066 C=Normal 5067 CCI =Not Required 5068 INVITE F2 5069 A->SS1 5071 INVITE sip:+1-972-555-9999@ngw1.wcom.com;user=phone SIP/2.0 5072 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5073 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5074 To: sip:+1-972-555-9999@ngw1.wcom.com;user=phone 5075 Call-Id: 12345602@ngw1.wcom.com 5076 CSeq: 1 INVITE 5077 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5078 Content-Type: application/sdp 5079 Content-Length: 150 5081 v=0 5082 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 5083 t=0 0 5084 c=IN IP4 gatewayone.wcom.com 5085 m=audio 3456 RTP/AVP 0 5086 a=rtpmap:0 PCMU/8000 5087 /* SS1 uses a location manager function to determine where B is 5088 actually located. Based upon location analysis the call is forwarded 5089 to GW1. GW1 prepares to receive data on port 3456 from User A.*/ 5091 604 Does Not Exist Anwhere F3 5092 SS1 -> GW 5094 SIP/2.0 604 Does Not Exist Anywhere 5095 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5096 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5097 To: sip:+1-972-555-9999@ngw1.wcom.com;user=phone 5098 Call-Id: 12345602@ngw1.wcom.com 5099 CSeq: 1 INVITE 5100 Content-Length: 0 5102 ACK F4 GW1 5103 ->SS1 5105 ACK sip:+1-972-555-9999@ss1.wcom.com;user=phone SIP/2.0 5106 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5107 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5108 To: sip:+1-972-555-9999@ngw1.wcom.com;user=phone 5109 Call-Id: 12345602@ngw1.wcom.com 5110 CSeq: 1 ACK 5111 Content-Length: 0 5113 REL F5 5114 GW -> User A 5116 REL 5117 CauseCode=1 5118 CodingStandard=CCITT 5119 RLC F6 5120 User A 5121 -> GW 5123 RLC 5125 5.2.2 5126 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL 5128 In this scenario, User A calls User B through a Gateway GW1 and SS1 5129 working as a proxy server. The call is routed to User B via the 5130 gateway. The call is rejected by the User B who sends a 600 Busy 5131 Everywhere response. The Gateway sends a REL message containing a 5132 specific Cause value mapped by the gateway based on the SIP error. 5134 Since no interworking is indicated in the IAM (F1), the busy tone is 5135 generated locally by User A's telephone switch. In scenario 5.2.3, 5136 the busy signal is generated by the Gateway since interworking is 5137 indicated. 5139 Message Details 5141 IAM F1 5142 User A 5143 -> GW 5145 IAM 5146 CgPN=314-555-1111,NPI=E.164,NOA=National 5147 CdPN=972-555-2222,NPI=E.164,NOA=National 5148 USI=Speech 5149 CPT=0 0 5150 C=Normal 5151 CCI =Not Required 5152 INVITE F2 5153 A->SS1 5155 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 5156 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5157 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5158 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5159 Call-Id: 12345602@ngw1.wcom.com 5160 CSeq: 1 INVITE 5161 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5162 Content-Type: application/sdp 5163 Content-Length: 150 5165 v=0 5166 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 5167 t=0 0 5168 c=IN IP4 gatewayone.wcom.com 5169 m=audio 3456 RTP/AVP 0 5170 a=rtpmap:0 PCMU/8000 5172 /* SS1 uses a location manager function to determine where B is 5173 actually located. Based upon location analysis the call is forwarded 5174 to GW1. GW1 prepares to receive data on port 3456 from User A.*/ 5176 INVITE F3 5177 SS1 5178 ->User B 5180 INVITE UserB@there.com SIP/2.0 5181 Via: SIP/2.0/UDP ss1.wcom.com:5060 5182 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5183 Record-Route: 5184 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5185 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5186 Call-Id: 12345602@ngw1.wcom.com 5187 CSeq: 1 INVITE 5188 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5189 Content-Type: application/sdp 5190 Content-Length: 150 5192 v=0 5193 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 5194 t=0 0 5195 c=IN IP4 gatewayone.wcom.com 5196 m=audio 3456 RTP/AVP 0 5197 a=rtpmap:0 PCMU/8000 5199 (100 Trying F4 SS1 5200 -> GW1) 5202 SIP/2.0 100 Trying 5203 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5204 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5205 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5206 Call-Id: 12345602@ngw1.wcom.com 5207 CSeq: 1 INVITE 5208 Content-Length: 0 5210 600 Busy Everywhere F5 User B 5211 -> SS1 5213 SIP/2.0 600 Busy Everywhere 5214 Via: SIP/2.0/UDP ss1.wcom.com:5060 5215 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5216 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5217 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5218 Call-Id: 12345602@ngw1.wcom.com 5219 CSeq: 1 INVITE 5220 Content-Length: 0 5222 ACK F6 SS1 5223 ->User B 5225 ACK UserB@there.com SIP/2.0 5226 Via: SIP/2.0/UDP ss1.wcom.com:5060 5227 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5228 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5229 Call-Id: 12345602@ngw1.wcom.com 5230 CSeq: 1 ACK 5231 Content-Length: 0 5233 600 Busy Everywhere F7 SS1 -> GW1 5235 SIP/2.0 600 Busy Everywhere 5236 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5237 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5238 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5239 Call-Id: 12345602@ngw1.wcom.com 5240 CSeq: 1 INVITE 5241 Content-Length: 0 5243 ACK F8 GW1 5244 ->SS1 5246 ACK UserB@ss1.wcom.com SIP/2.0 5247 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5248 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5249 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5250 Call-Id: 12345602@ngw1.wcom.com 5251 CSeq: 1 ACK 5252 Content-Length: 0 5254 REL F9 5255 User A 5256 ->GW 5258 REL 5259 CauseCode=17 Busy 5260 CodingStandard=CCITT 5261 RLC F10 5262 GW -> User A 5264 RLC 5266 5.2.3 5267 Unsuccessful PSTN->SIP, SIP error interworking to tones 5269 In this scenario, User A calls User B through SS1 working as a proxy 5270 server. The call is routed to User B via the gateway. The call is 5271 rejected by the User B client. GW1 plays busy tone, and releases 5272 call after timeout. 5274 GW1 plays the busy tone since the IAM (F1) indicates the interworking 5275 is present. In scenario 5.2.2, with no interworking, the busy 5276 indication is carried in the REL Cause value and is generated locally 5277 instead. 5279 Message Details 5281 IAM F1 5282 User A 5283 -> GW 5285 IAM 5286 CgPN=314-555-1111,NPI=E.164,NOA=National 5287 CdPN=972-555-2222,NPI=E.164,NOA=National 5288 USI=Speech 5289 CPT=0 0 5290 C=Normal 5291 CCI =Not Required 5292 Interworking=encountered 5293 INVITE F2 5294 A->SS1 5296 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 5297 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5298 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5299 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5300 Call-Id: 12345602@ngw1.wcom.com 5301 CSeq: 1 INVITE 5302 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5303 Content-Type: application/sdp 5304 Content-Length: 150 5306 v=0 5307 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 5308 t=0 0 5309 c=IN IP4 gatewayone.wcom.com 5310 m=audio 3456 RTP/AVP 0 5311 a=rtpmap:0 PCMU/8000 5313 /* SS1 uses a location manager function to determine where B is 5314 actually located. Based upon location analysis the call is forwarded 5315 to GW1. GW1 prepares to receive data on port 3456 from User A.*/ 5317 INVITE F3 5318 SS1 5319 ->User B 5321 INVITE UserB@there.com SIP/2.0 5322 Via: SIP/2.0/UDP ss1.wcom.com:5060 5323 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5324 Record-Route: 5325 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5326 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5327 Call-Id: 12345602@ngw1.wcom.com 5328 CSeq: 1 INVITE 5329 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5330 Content-Type: application/sdp 5331 Content-Length: 150 5333 v=0 5334 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 5335 t=0 0 5336 c=IN IP4 gatewayone.wcom.com 5337 m=audio 3456 RTP/AVP 0 5338 a=rtpmap:0 PCMU/8000 5340 (100 Trying F4 User B -> SS1) 5342 SIP/2.0 100 Trying 5343 Via: SIP/2.0/UDP ss1.wcom.com:5060 5344 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5345 Record-Route: 5346 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5347 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5348 Call-Id: 12345602@ngw1.wcom.com 5349 CSeq: 1 INVITE 5350 Content-Length: 0 5352 600 Busy Everywhere F5 User B 5353 -> SS1 5355 SIP/2.0 600 Busy Everywhere 5356 Via: SIP/2.0/UDP ss1.wcom.com:5060 5357 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5358 Record-Route: 5359 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5360 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5361 Call-Id: 12345602@ngw1.wcom.com 5362 CSeq: 1 INVITE 5363 Content-Length: 0 5365 ACK F6 SS1 5366 ->User B 5368 ACK UserB@ss1.wcom.com SIP/2.0 5369 Via: SIP/2.0/UDP ss1.wcom.com:5060 5370 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5371 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5372 Call-Id: 12345602@ngw1.wcom.com 5373 CSeq: 1 ACK 5374 Content-Length: 0 5376 600 Busy Everywhere F7 SS1 -> GW1 5378 SIP/2.0 600 Busy Everywhere 5379 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5380 Record-Route: 5381 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5382 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5383 Call-Id: 12345602@ngw1.wcom.com 5384 CSeq: 1 INVITE 5385 Content-Length: 0 5387 ACK F8 GW1 5388 ->SS1 5390 ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 5391 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5392 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5393 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5394 Call-Id: 12345602@ngw1.wcom.com 5395 CSeq: 1 ACK 5396 Content-Length: 0 5398 ACM F9 5399 User B 5400 ->GW 5402 ACM 5403 Charge Indicator=No Charge 5404 Called Party Status=no indication 5405 Called Party's Category=ordinary subscriber 5406 End To End Method=none available 5407 Interworking=encountered 5408 End to End Information=none available 5409 ISUP Indicator=not used all the way 5410 ISDN Access Terminating access non ISDN 5411 Echo Control=not included 5413 /* One way speech path established between GW and User A. */ 5414 /* Call Released after NGW treatment timer expires. */ 5416 REL F10 5417 User A 5418 ->GW 5420 REL 5421 CauseCode=17 5422 CodingStandard=CCITT 5423 RLC F11 5424 GW -> User A 5426 RLC 5428 5.2.4 5429 Unsuccessful PSTN->SIP, ACM timeout 5431 User A calls User B through SS1 working as a proxy server. SS1 re- 5432 sends the INVITE after the expiration of SIP timer T1. User B never 5433 responds with 180 Ringing _ it is reachable but unresponsive. After 5434 the expiration of ISUP T7 timer, User A's network disconnects the 5435 call by sending a Release message REL. The Gateway maps this to a 5436 CANCEL which is re-sent by SS1 after SIP T1 timer expires. 5438 Message Details 5440 IAM F1 5441 User A 5442 -> GW 5444 IAM 5445 CgPN=314-555-1111,NPI=E.164,NOA=National 5446 CdPN=972-555-2222,NPI=E.164,NOA=National 5447 USI=Speech 5448 CPT=0 0 5449 C=Normal 5450 CCI =Not Required 5451 INVITE F2 5452 A->SS1 5454 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 5455 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5456 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5457 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5458 Call-Id: 12345602@ngw1.wcom.com 5459 CSeq: 1 INVITE 5460 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5461 Content-Type: application/sdp 5462 Content-Length: 150 5464 v=0 5465 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 5466 t=0 0 5467 c=IN IP4 gatewayone.wcom.com 5468 m=audio 3456 RTP/AVP 0 5469 a=rtpmap:0 PCMU/8000 5470 /* SS1 uses a location manager function to determine where B is 5471 actually located. Based upon location analysis the call is forwarded 5472 to GW1. GW1 prepares to receive data on port 3456 from User A.*/ 5474 INVITE F3 5475 SS1 5476 ->User B 5478 INVITE sip:UserB@there.com SIP/2.0 5479 Via: SIP/2.0/UDP ss1.wcom.com:5060 5480 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5481 Record-Route: 5482 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5483 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5484 Call-Id: 12345602@ngw1.wcom.com 5485 CSeq: 1 INVITE 5486 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5487 Content-Type: application/sdp 5488 Content-Length: 150 5490 v=0 5491 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 5492 c t=0 0 5493 c=IN IP4 gatewayone.wcom.com 5494 m=audio 3456 RTP/AVP 0 5495 a=rtpmap:0 PCMU/8000 5497 100 Trying F4 SS1 5498 -> GW 5500 SIP/2.0 100 Trying 5501 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5502 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5503 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5504 Call-Id: 12345602@ngw1.wcom.com 5505 CSeq: 1 INVITE 5506 Content-Length: 0 5508 INVITE F5 5509 GW -> SS1 5511 Same as Message F3 5513 INVITE F6 5514 SS1 5515 ->User B 5517 Same as Message F3 5519 INVITE F7 5520 SS1 5521 ->User B 5523 Same as Message F3 5525 INVITE F8 5526 SS1 5527 ->User B 5529 Same as Message F3 5531 INVITE F9 5532 SS1 5533 ->User B 5535 Same as Message F3 5537 /* ISUP Timer T7 expires in User A's access network. */ 5539 REL F10 5540 User A 5541 ->GW 5543 REL 5544 CauseCode=16 Normal 5545 CodingStandard=CCITT 5546 RLC F11 5547 GW -> User A 5549 RLC 5550 CANCEL F12 5551 GW -> SS1 5553 CANCEL sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 5554 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5555 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5556 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5557 Call-Id: 12345602@ngw1.wcom.com 5558 CSeq: 1 CANCEL 5559 Content-Length: 0 5561 CANCEL F13 5562 SS1 -> User B 5564 CANCEL sip:UserB@there.com SIP/2.0 5565 Via: SIP/2.0/UDP ss1.wcom.com:5060 5566 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5567 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5568 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5569 Call-Id: 12345602@ngw1.wcom.com 5570 CSeq: 1 CANCEL 5571 Content-Length: 0 5573 CANCEL F14 5574 GW -> SS1 5576 Same as Message F13 5578 CANCEL F15 5579 SS1 5580 ->User B 5582 Same as Message F13 5584 CANCEL F16 5585 SS1 5586 ->User B 5588 Same as Message F13 5590 CANCEL F17 5591 SS1 5592 ->User B 5594 Same as Message F13 5596 CANCEL F18 5597 SS1 5598 ->User B 5600 Same as Message F13 5602 5.2.5 5603 Unsuccessful PSTN->SIP, ACM timeout, stateless SPS 5605 In this scenario, User A calls User B through SS1 working as a 5606 stateless proxy server. Since SS1 is stateless, GW1 re-sends the 5607 INVITE and CANCEL messages after the expiration of SIP timer T1. 5608 User B does not respond with 180 Ringing. User A's network 5609 disconnects the call with a release REL. 5611 Message Details 5613 IAM F1 5614 User A 5615 -> GW 5617 IAM 5618 CgPN=314-555-1111,NPI=E.164,NOA=National 5619 CdPN=972-555-2222,NPI=E.164,NOA=National 5620 USI=Speech 5621 CPT=0 0 5622 C=Normal 5623 CCI =Not Required 5624 INVITE F2 5625 GW -> SS1 5627 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 5628 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5629 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5630 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5631 Call-Id: 12345602@ngw1.wcom.com 5632 CSeq: 1 INVITE 5633 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5634 Content-Type: application/sdp 5635 Content-Length: 150 5637 v=0 5638 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 5639 t=0 0 5640 c=IN IP4 gatewayone.wcom.com 5641 m=audio 3456 RTP/AVP 0 5642 a=rtpmap:0 PCMU/8000 5644 /* SS1 uses a location manager function to determine where B is 5645 actually located. Based upon location analysis the call is forwarded 5646 to GW1. GW1 prepares to receive data on port 3456 from User A.*/ 5648 INVITE F3 5649 SS1-> User B 5651 INVITE sip:UserB@there.com SIP/2.0 5652 Via: SIP/2.0/UDP ss1.wcom.com:5060 5653 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5654 Record-Route: 5655 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5656 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5657 Call-Id: 12345602@ngw1.wcom.com 5658 CSeq: 1 INVITE 5659 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5660 Content-Type: application/sdp 5661 Content-Length: 150 5663 v=0 5664 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 5665 t=0 0 5666 c=IN IP4 gatewayone.wcom.com 5667 m=audio 3456 RTP/AVP 0 5668 a=rtpmap:0 PCMU/8000 5670 INVITE F4 5671 GW -> SS1 5673 Same as Message F2 5675 INVITE F5 5676 SS1 -> User B 5678 Same as Message F3 5680 INVITE F6 5681 GW -> SS1 5683 Same as Message F2 5685 INVITE F7 5686 SS1 -> User B 5688 Same as Message F3 5690 INVITE F8 5691 GW -> SS1 5693 Same as Message F2 5695 INVITE F9 5696 SS1 -> User B 5698 Same as Message F3 5699 INVITE F10 5700 GW -> SS1 5702 Same as Message F2 5704 INVITE F11 5705 SS1 -> User B 5707 Same as Message F3 5709 INVITE F12 5710 GW -> SS1 5712 Same as Message F2 5714 INVITE F13 5715 SS1 -> User B 5717 Same as Message F3 5719 /* ISUP T7 Timer expires in User A's access network. */ 5721 REL F14 5722 User A 5723 ->GW 5725 REL 5726 CauseCode=16 Normal 5727 CodingStandard=CCITT 5728 RLC F15 5729 GW -> User A 5731 RLC 5732 CANCEL F16 5733 GW -> SS1 5735 CANCEL sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 5736 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5737 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5738 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5739 Call-Id: 12345602@ngw1.wcom.com 5740 CSeq: 1 CANCEL 5741 Content-Length: 0 5743 CANCEL F17 5744 SS1 -> User B 5746 CANCEL sip:UserB@there.com SIP/2.0 5747 Via: SIP/2.0/UDP ss1.wcom.com:5060 5748 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5749 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5750 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5751 Call-Id: 12345602@ngw1.wcom.com 5752 CSeq: 1 CANCEL 5753 Content-Length: 0 5755 CANCEL F18 5756 GW -> SS1 5758 Same as Message F16 5759 CANCEL F19 5760 SS1 -> User B 5762 Same as Message F17 5763 CANCEL F20 5764 GW -> SS1 5766 Same as Message F16 5767 CANCEL F21 5768 SS1 -> User B 5770 Same as Message F17 5771 CANCEL F22 5772 GW -> SS1 5774 Same as Message F16 5775 CANCEL F23 5776 SS1 -> User B 5778 Same as Message F17 5779 CANCEL F24 5780 GW -> SS1 5782 Same as Message F16 5783 CANCEL F25 5784 SS1 -> User B 5786 Same as Message F17 5787 CANCEL F26 5788 GW -> SS1 5790 Same as Message F16 5791 CANCEL F27 5792 SS1 -> User B 5794 Same as Message F17 5796 5.2.6 5797 Unsuccessful PSTN->SIP, ANM timeout 5799 In this scenario, User A calls User B through SS1 working as a proxy 5800 server. User B does not respond with 200 OK. User A disconnects the 5801 call with a Release message REL which is mapped by GW1 to a CANCEL. 5802 Note that if User B had sent a 200 OK response after the REL, GW1 5803 would have sent an ACK then a BYE to properly terminate the call. 5805 Message Details 5807 IAM F1 5808 User A 5809 -> GW 5811 IAM 5812 CgPN=314-555-1111,NPI=E.164,NOA=National 5813 CdPN=972-555-2222,NPI=E.164,NOA=National 5814 USI=Speech 5815 CPT=0 0 5816 C=Normal 5817 CCI =Not Required 5818 INVITE F2 5819 A->SS1 5821 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 5822 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5823 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5824 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5825 Call-Id: 12345602@ngw1.wcom.com 5826 CSeq: 1 INVITE 5827 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5828 Content-Type: application/sdp 5829 Content-Length: 150 5831 v=0 5832 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 5833 t=0 0 5834 c=IN IP4 gatewayone.wcom.com 5835 m=audio 3456 RTP/AVP 0 5836 a=rtpmap:0 PCMU/8000 5838 /* SS1 uses a location manager function to determine where B is 5839 actually located. Based upon location analysis the call is forwarded 5840 to GW1. GW1 prepares to receive data on port 3456 from User A.*/ 5842 INVITE F3 5843 SS1 5844 ->User B 5846 INVITE sip:UserB@there.com SIP/2.0 5847 Via: SIP/2.0/UDP ss1.wcom.com:5060 5848 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5849 Record-Route: 5850 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5851 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5852 Call-Id: 12345602@ngw1.wcom.com 5853 CSeq: 1 INVITE 5854 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5855 Content-Type: application/sdp 5856 Content-Length: 150 5858 v=0 5859 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com 5860 t=0 0 5861 c=IN IP4 gatewayone.wcom.com 5862 m=audio 3456 RTP/AVP 0 5863 a=rtpmap:0 PCMU/8000 5865 (100 Trying F4 User B -> SS1) 5867 SIP/2.0 100 Trying 5868 Via: SIP/2.0/UDP ss1.wcom.com:5060 5869 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5870 Record-Route: 5871 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5872 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone 5873 Call-Id: 12345602@ngw1.wcom.com 5874 CSeq: 1 INVITE 5875 Content-Length: 0 5877 180 Ringing F5 User B -> SS1 5879 SIP/2.0 180 Ringing 5880 Via: SIP/2.0/UDP ss1.wcom.com:5060 5881 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5882 Record-Route: 5883 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5884 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5885 Call-Id: 12345602@ngw1.wcom.com 5886 CSeq: 1 INVITE 5887 Content-Length: 0 5889 180 Ringing F6 SS1 5890 -> GW1 5892 SIP/2.0 180 Ringing 5893 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5894 Record-Route: 5895 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5896 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5897 Call-Id: 12345602@ngw1.wcom.com 5898 CSeq: 1 INVITE 5899 Content-Length: 0 5901 ACM F7 5902 GW -> User A 5904 ACM 5905 Charge Indicator=No Charge 5906 Called Party Status=no indication 5907 Called Party's Category=ordinary subscriber 5908 End To End Method=none available 5909 Interworking=encountered 5910 End to End Information=none available 5911 ISUP Indicator=not used all the way 5912 ISDN Access Terminating access non ISDN 5913 Echo Control=not included 5915 /* ISUP Timer T9 expires in User A's access network. */ 5917 REL F8 5918 User A 5919 -> GW 5921 REL 5922 CauseCode=16 Normal 5923 CodingStandard=CCITT 5924 RLC F9 5925 GW -> User A 5927 RLC 5928 CANCEL F10 5929 GW -> SS1 5931 CANCEL sip:UserB@ss1.wcom.com SIP/2.0 5932 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5933 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5934 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5935 Call-Id: 12345602@ngw1.wcom.com 5936 CSeq: 1 CANCEL 5937 Content-Length: 0 5939 CANCEL F11 5940 SS1 -> User B 5941 CANCEL sip:UserB@there.com SIP/2.0 5942 Via: SIP/2.0/UDP ss1.wcom.com:5060 5943 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5944 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5945 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5946 Call-Id: 12345602@ngw1.wcom.com 5947 CSeq: 1 CANCEL 5948 Content-Length: 0 5950 200 OK F12 5951 User B 5952 -> SS1 5954 SIP/2.0 200 OK 5955 Via: SIP/2.0/UDP ss1.wcom.com:5060 5956 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5957 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5958 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5959 Call-Id: 12345602@ngw1.wcom.com 5960 CSeq: 1 CANCEL 5961 Content-Length: 0 5963 200 OK F13 SS1 -> GW 5965 SIP/2.0 200 OK 5966 Via: SIP/2.0/UDP ngw1.wcom.com:5060 5967 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 5968 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 5969 Call-Id: 12345602@ngw1.wcom.com 5970 CSeq: 1 CANCEL 5971 Content-Length: 0 5973 6 Gateway to Gateway Dialing via SIP Network 5975 In these scenarios, both the caller and the called party are in the 5976 telephone network, either normal PSTN subscribers or PBX extensions. 5977 The calls route through two Gateways and at least one SIP Proxy 5978 Server. The Proxy Server performs the authentication and location of 5979 the Gateways. 5981 Note that the proposed INFO method[8] is not currently included in 5982 this document. It is anticipated that a future version will include 5983 an example of this. 5985 Again it is noted that the intent of this call flows document is not 5986 to provide a detailed parameter level mapping of SIP to PSTN 5987 protocols. For information on SIP to ISUP mapping, the reader is 5988 referred to other references[9]. 5990 6.1 Success Scenarios 5992 In these scenarios, the call is successfully completed between the 5993 two Gateways allowing the PSTN or PBX users to communicate. The 183 5994 Session Progress response is used to establish a media path between 5995 the two Gateways, allowing in-band alerting to pass from the called 5996 party telephone switch to the caller. 5998 6.1.1 5999 Successful ISUP PSTN to ISUP PSTN call 6001 In this scenario, User A in the PSTN calls User C who is served as an 6002 extension on a PBX. User A's telephone switch signals via SS7 to the 6003 Network Gateway NGW1, while User C's PBX signals via SS7 with the 6004 Enterprise Gateway GW2. The CdPN and CgPN are mapped into SIP URLs 6005 and placed in the To and From headers. SS1 looks up the dialed 6006 digits in the Request-URI and maps the digits to the PBX extension of 6007 User C served by GW2. The INVITE is then forwarded to GW2 for call 6008 completion. An early media path is established end-to-end so that 6009 User A can hear the ringing tone generated by PBX C. 6011 User C answers the call and the media path is cut through in both 6012 directions. User B hangs up terminating the call. 6014 Message Details 6016 IAM F1 6017 User A 6018 -> GW 6020 IAM 6021 CgPN=314-555-1111,NPI=E.164,NOA=National 6022 CdPN=972-555-2222,NPI=E.164,NOA=National 6023 USI=Speech 6024 CPT=0 0 6025 C=Normal 6026 CCI =Not Required 6027 INVITE F2 6028 GW1 -> SS1 6030 INVITE sip:+1-918-555-3333@ss1.wcom.com;user=phone SIP/2.0 6031 Via: SIP/2.0/UDP ngw1.wcom.com:5060 6032 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6033 To: sip:+1-918-555-3333@ss1.wcom.com;user=phone 6034 Call-Id: 12345600@ngw1.wcom.com 6035 CSeq: 1 INVITE 6036 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6037 Content-Type: application/sdp 6038 Content-Length: 134 6040 v=0 6041 o=GW1 2890844526 2890844526 IN IP4 gw1.wcom.com 6042 t=0 0 6043 c=IN IP4 100.101.102.103 6044 m=audio 49170 RTP/AVP 0 6045 a=rtpmap:0 PCMU/8000 6046 /* SS uses a location manager function to determine where B is 6047 actually located. Response is returned listing onnet and offnet 6048 routes. */ 6050 INVITE F3 6051 SS1 -> GW2 6053 INVITE sip:444-3333,phone-context=p1234@gw2.wcom.com;user=phone 6054 SIP/2.0 6055 Via: SIP/2.0/UDP ss1.wcom.com:5060 6056 Via: SIP/2.0/UDP gw1.wcom.com:5060 6057 Record-Route: 6058 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6059 To: sip:+1-918-555-3333@ss1.wcom.com;user=phone 6060 Call-Id: 12345600@ngw1.wcom.com 6061 CSeq: 1 INVITE 6062 Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6063 Content-Type: application/sdp 6064 Content-Length: 134 6066 v=0 6067 o=GW1 2890844526 2890844526 IN IP4 gw1.wcom.com 6068 t=0 0 6069 c=IN IP4 100.101.102.103 6070 m=audio 49170 RTP/AVP 0 6071 a=rtpmap:0 PCMU/8000 6073 IAM F4 6074 GW2 -> User C 6076 IAM 6077 CgPN=314-555-1111,NPI=E.164,NOA=National 6078 CdPN=444-3333,NPI=Private,NOA=Subscriber 6079 USI=Speech 6080 CPT=0 0 6081 C=Normal 6082 CCI =Not Required 6083 ACM F5 6084 User C 6085 -> GW2 6087 ACM 6088 Charge Indicator=No Charge 6089 Called Party Status=no indication 6090 Called Party's Category=ordinary subscriber 6091 End To End Method=none available 6092 Interworking=encountered 6093 End to End Information=none available 6094 ISUP Indicator=not used all the way 6095 ISDN Access Terminating access non ISDN 6096 Echo Control=not included 6098 /* Based on PROGress message, GW3 returns a 183 response with SDP 6099 allowing in-band call progress indications to be sent to the 6100 originator. */ 6102 183 Session Progress F6 6103 GW2 -> SS1 6105 SIP/2.0 183 Session Progress 6106 Via: SIP/2.0/UDP ss1.wcom.com:5060 6107 Via: SIP/2.0/UDP ngw1.wcom.com:5060 6108 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6109 To: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 6110 Call-Id: 12345600@ngw1.wcom.com 6111 CSeq: 1 INVITE 6112 Content-Type: application/sdp 6113 Content-Length: 134 6115 v=0 6116 o=PBX_B 987654321 987654321 IN IP4 gw2.wcom.com 6117 t=0 0 6118 c=IN IP4 100.101.102.104 6119 m=audio 14918 RTP/AVP 0 6120 a=rtpmap:0 PCMU/8000 6122 183 Session Progress F7 6123 SS1 -> GW1 6125 SIP/2.0 183 Session Progress 6126 Via: SIP/2.0/UDP ngw1.wcom.com:5060 6127 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6128 To: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 6129 Call-Id: 12345600@ngw1.wcom.com 6130 CSeq: 1 INVITE 6131 Content-Type: application/sdp 6132 Content-Length: 134 6134 v=0 6135 o=PBX_B 987654321 987654321 IN IP4 gw2.wcom.com 6136 t=0 0 6137 c=IN IP4 100.101.102.104 6138 m=audio 14918 RTP/AVP 0 6139 a=rtpmap:0 PCMU/8000 6141 /* GW1 receives packets from GW2 with encoded ringback, tones or 6142 other audio. GW1 decodes this and places it on the originating 6143 trunk. */ 6145 ACM F8 6146 GW1 6147 ->User A 6149 ACM 6150 Charge Indicator=No Charge 6151 Called Party Status=no indication 6152 Called Party's Category=ordinary subscriber 6153 End To End Method=none available 6154 Interworking=encountered 6155 End to End Information=none available 6156 ISUP Indicator=not used all the way 6157 ISDN Access Terminating access non ISDN 6158 Echo Control=not included 6160 /* User B answers */ 6161 ANM F9 6162 User C 6163 -> GW2 6165 ANM 6167 200 OK F10 6168 GW2 -> SS1 6170 SIP/2.0 200 OK 6171 Via: SIP/2.0/UDP ss1.wcom.com:5060 6172 Via: SIP/2.0/UDP ngw1.wcom.com:5060 6173 Record-Route: 6174 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6175 To: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 6176 Call-Id: 12345600@ngw1.wcom.com 6177 CSeq: 1 INVITE 6178 Contact: sip:444-3333,phone-context=p1234@gw2.wcom.com;user=phone 6179 Content-Type: application/sdp 6180 Content-Length: 134 6182 v=0 6183 o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com 6184 t=0 0 6185 c=IN IP4 100.101.102.104 6186 m=audio 14918 RTP/AVP 0 6187 a=rtpmap:0 PCMU/8000 6189 200 OK F11 6190 SS1 -> GW1 6192 SIP/2.0 200 OK 6193 Via: SIP/2.0/UDP ngw1.wcom.com:5060 6194 Record-Route: 6195 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6196 To: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 6197 Call-Id: 12345600@ngw1.wcom.com 6198 CSeq: 1 INVITE 6199 Contact: sip:444-3333,phone-context=p1234@gw2.wcom.com;user=phone 6200 Content-Type: application/sdp 6201 Content-Length: 134 6203 v=0 6204 o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com 6205 t=0 0 6206 c=IN IP4 100.101.102.104 6207 m=audio 14918 RTP/AVP 0 6208 a=rtpmap:0 PCMU/8000 6210 ANM F12 6211 GW1 -> User A 6213 ANM 6215 ACK F13 6216 GW1 -> SS1 6218 ACK sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone SIP/2.0 6219 Via: SIP/2.0/UDP ngw1.wcom.com:5060 6220 Route: ;user=phone 6221 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6222 To: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 6223 Call-Id: 12345600@gw1.wcom.com 6224 CSeq: 1 ACK 6225 Content-Length: 0 6227 ACK F14 6228 SS1 -> GW3 6229 ACK sip:444-3333,phone-context=p1234@gw2.wcom.com;user=phone SIP/2.0 6230 Via: SIP/2.0/UDP ss1.wcom.com:5060 6231 Via: SIP/2.0/UDP ngw1.wcom.com:5060 6232 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6233 To: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 6234 Call-Id: 12345600@ngw1.wcom.com 6235 CSeq: 1 ACK 6236 Content-Length: 0 6238 /* RTP streams are established between GW1 and GW2. */ 6240 /* User B Hangs Up with User A. */ 6241 REL F15 6242 User C-> GW2 6244 REL 6245 CauseCode=16 Normal 6246 CodingStandard=CCITT 6247 BYE F16 6248 GW3 -> SS1 6250 BYE sip:+1-314-555-1111@ss1.wcom.com;user=phone SIP/2.0 6251 Via: SIP/2.0/UDP gw2.wcom.com:5060 6252 Route: ;user=phone 6253 From: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 6254 To: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6255 Call-Id: 12345600@ngw1.wcom.com 6256 CSeq: 4 BYE 6257 Content-Length: 0 6259 RLC F17 6260 GW2 6261 ->User C 6263 RLC 6264 BYE F18 6265 SS1 -> GW1 6267 BYE sip:+1-314-555-1111@gw1.wcom.com;user=phone SIP/2.0 6268 Via: SIP/2.0/UDP ss1.wcom.com:5060 6269 Via: SIP/2.0/UDP gw2.wcom.com:5060 6270 From: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 6271 To: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6272 Call-Id: 12345600@ngw1.wcom.com 6273 CSeq: 4 BYE 6274 Content-Length: 0 6276 200 OK F19 6277 GW1 -> SS1 6279 SIP/2.0 200 OK 6280 Via: SIP/2.0/UDP ss1.wcom.com:5060 6281 Via: SIP/2.0/UDP gw2.wcom.com:5060 6282 From: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 6283 To: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6284 Call-Id: 12345600@ngw1.wcom.com 6285 CSeq: 4 BYE 6286 Content-Length: 0 6288 200 OK F20 6289 SS11 6290 ->GW3 6292 SIP/2.0 200 OK 6293 Via: SIP/2.0/UDP gw2.wcom.com:5060 6294 From: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 6295 To: sip:+1-314-555-1111@ngw1.wcom.com;user=phone 6296 Call-Id: 12345600@ngw1.wcom.com 6297 CSeq: 4 BYE 6298 Content-Length: 0 6300 REL F21 6301 User C-> GW2 6303 REL 6304 CauseCode=16 Normal 6305 CodingStandard=CCITT 6306 RLC F22 6307 GW2 6308 ->User C 6310 RLC 6312 6.1.2 6313 Successful FGB PBX to ISDN PBX call with overflow 6315 PBX User A calls PBX User C via GW1 using SS1 as a Proxy Server. 6316 During the attempt to reach User C via GW2, an error is encountered _ 6317 SS1 receives a 503 Service Unavailable (F4) response to the forwarded 6318 INVITE. This could be due to all circuits being busy, or some other 6319 outage at GW2. SS1 recognizes the error and uses an alternative 6320 route via GW3 to terminate the call. From there, the call proceeds 6321 normally with User C answering the call. The call is terminated when 6322 User C hangs up. 6324 Message Details 6326 PBX A 6327 ->GW1 6329 Seizure 6331 GW1 -> PBX A 6333 Wink 6334 MF Digits F1 6335 PBX A 6336 ->GW1 6338 KP 444 3333 ST 6339 INVITE F2 6340 GW1 -> SS1 6342 INVITE sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone 6343 SIP/2.0 6344 Via: SIP/2.0/UDP gw1.wcom.com:5060 6345 From: PBX_A 6346 To: sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone 6347 Call-Id: 12345600@gw1.wcom.com 6348 CSeq: 1 INVITE 6349 Contact: PBX_A 6350 Content-Type: application/sdp 6351 Content-Length: 136 6353 v=0 6354 o=PBX_A 2890844526 2890844526 IN IP4 gw1.wcom.com 6355 t=0 0 6356 c=IN IP4 100.101.102.103 6357 m=audio 49170 RTP/AVP 0 6358 a=rtpmap:0 PCMU/8000 6360 /* SS uses a location manager function to determine where B is 6361 actually located. Response is returned listing onnet and offnet 6362 routes. */ 6364 INVITE F3 6365 SS1 -> GW2 6367 INVITE sip:444-3333,phone-context=p1234@gw2.wcom.com;user=phone 6368 SIP/2.0 6369 Via: SIP/2.0/UDP ss1.wcom.com:5060 6370 Via: SIP/2.0/UDP gw1.wcom.com:5060 6371 Record-Route: 6372 From: PBX_A 6373 To: sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone 6374 Call-Id: 12345600@gw1.wcom.com 6375 CSeq: 1 INVITE 6376 Contact: PBX_A 6377 Content-Type: application/sdp 6378 Content-Length: 136 6380 v=0 6381 o=PBX_A 2890844526 2890844526 IN IP4 gw1.wcom.com 6382 t=0 0 6383 c=IN IP4 100.101.102.103 6384 m=audio 49170 RTP/AVP 0 6385 a=rtpmap:0 PCMU/8000 6387 503 Service Unavailable F4 6388 GW2 -> SS1 6390 SIP/2.0 503 Service Unavailable 6391 Via: SIP/2.0/UDP ss1.wcom.com:5060 6392 Via: SIP/2.0/UDP gw1.wcom.com:5060 6393 From: PBX_A 6394 To: sip:444-3333,phone-context=p1234@ss1.wcom.com 6395 ;user=phone;tag=314159 6396 Call-Id: 12345600@gw1.wcom.com 6397 CSeq: 1 INVITE 6398 Content-Length: 0 6399 ACK F5 6400 SS1 -> GW2 6402 ACK sip:444-3333,phone-context=p1234@gw2.wcom.com;user=phone SIP/2.0 6403 Via: SIP/2.0/UDP ss1.wcom.com:5060 6404 Via: SIP/2.0/UDP gw1.wcom.com:5060 6405 From: PBX_A 6406 To: sip:444-3333,phone-context=p1234@ss1.wcom.com 6407 ;user=phone;tag=314159 6408 Call-Id: 12345600@gw1.wcom.com 6409 CSeq: 1 ACK 6410 Content-Length: 0 6412 INVITE F6 6413 SS1 -> GW3 6415 INVITE sip:+1-918-555-3333@gw3.wcom.com;user=phone SIP/2.0 6416 Via: SIP/2.0/UDP ss1.wcom.com:5060 6417 Via: SIP/2.0/UDP gw1.wcom.com:5060 6418 Record-Route: 6419 From: PBX_A 6420 To: sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone 6421 Call-Id: 12345600@gw1.wcom.com 6422 CSeq: 1 INVITE 6423 Contact: PBX_A 6424 Content-Type: application/sdp 6425 Content-Length: 136 6427 v=0 6428 o=PBX_A 2890844526 2890844526 IN IP4 gw1.wcom.com 6429 t=0 0 6430 c=IN IP4 100.101.102.103 6431 m=audio 49170 RTP/AVP 0 6432 a=rtpmap:0 PCMU/8000 6434 SETUP F7 6435 GW3-> PBX C 6437 Protocol discriminator=Q.931 6438 Call reference: Flag=0, CR value=any valid value not in use 6439 Message type=SETUP 6440 Bearer capability: Information transfer capability=0 (Speech) or 16 6441 (3.1 kHz audio) 6442 Channel identification=Preferred or exclusive B-channel 6443 Progress indicator=1 (Call is not end-to-end ISDN;further call 6444 progress information may be available inband) 6445 Called party number: 6447 Type of number and numbering plan ID=33 (National number in ISDN 6448 numbering plan) 6449 Digits=918-555-3333 6451 (100 Trying F8 6452 GW3 6453 ->SS1) 6455 SIP/2.0 100 Trying 6456 Via: SIP/2.0/UDP gw1.wcom.com:5060 6457 From: PBX_A 6458 To: sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone 6459 Call-Id: 12345600@gw1.wcom.com 6460 CSeq: 1 INVITE 6461 Content-Length: 0 6463 CALL PROCeeding F9 6464 PBX C 6465 -> GW3 6467 Protocol discriminator=Q.931 6468 Call reference: Flag=1, CR value=value in F9 SETUP message 6469 Message type=CALL PROC 6470 Channel identification=Exclusive B-channel 6471 PROGress F10 6472 PBX C 6473 -> GW3 6475 Protocol discriminator=Q.931 6476 Call reference: Flag=1, CR value=value in F9 SETUP message 6477 Message type=PROG 6478 Progress indicator=1 (Call is not end-to-end ISDN;further call 6479 progress information may be available inband) 6481 /* Based on PROGress message, GW3 returns a 183 response with SDP 6482 allowing in-band call progress indications to be sent to the 6483 originator. */ 6485 183 Session Progress F11 6486 GW3 -> SS1 6488 SIP/2.0 183 Session Progress 6489 Via: SIP/2.0/UDP ss1.wcom.com:5060 6490 Via: SIP/2.0/UDP gw1.wcom.com:5060 6491 From: PBX_A 6492 To: sip:444-3333,phone-context=p1234@ss1.wcom.com 6493 ;user=phone;tag=123456789 6494 Call-Id: 12345600@gw1.wcom.com 6495 CSeq: 1 INVITE 6496 Content-Type: application/sdp 6497 Content-Length: 134 6499 v=0 6500 o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com 6501 t=0 0 6502 c=IN IP4 100.101.102.104 6503 m=audio 14918 RTP/AVP 0 6504 a=rtpmap:0 PCMU/8000 6506 183 Session Progress F12 6507 SS1 -> GW1 6509 SIP/2.0 183 Session Progress 6510 Via: SIP/2.0/UDP gw1.wcom.com:5060 6511 From: PBX_A 6512 To: sip:444-3333,phone-context=p1234@ss1.wcom.com 6513 ;user=phone;tag=123456789 6514 Call-Id: 12345600@gw1.wcom.com 6515 CSeq: 1 INVITE 6516 Content-Type: application/sdp 6517 Content-Length: 134 6519 v=0 6520 o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com 6521 t=0 0 6522 c=IN IP4 100.101.102.104 6523 m=audio 14918 RTP/AVP 0 6524 a=rtpmap:0 PCMU/8000 6526 /* GW1 receives packets from GW3 with encoded ringback, tones or 6527 other audio. GW1 decodes this and places it on the originating 6528 trunk. */ 6530 CONNect F13 6531 PBX C 6532 -> GW3 6534 Protocol discriminator=Q.931 6535 Call reference: Flag=1, CR value=value in F9 SETUP message 6536 Message type=CONN 6537 200 OK F14 6538 GW3 -> SS1 6539 SIP/2.0 200 OK 6540 Via: SIP/2.0/UDP ss1.wcom.com:5060 6541 Via: SIP/2.0/UDP gw1.wcom.com:5060 6542 Record-Route: 6543 From: PBX_A 6544 To: sip:444-3333,phone-context=p1234@ss1.wcom.com 6545 ;user=phone;tag=123456789 6546 Call-Id: 12345600@gw1.wcom.com 6547 CSeq: 1 INVITE 6548 Contact: sip:+1-918-555-3333@gw3.wcom.com;user=phone 6549 Content-Type: application/sdp 6550 Content-Length: 134 6552 v=0 6553 o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com 6554 t=0 0 6555 c=IN IP4 100.101.102.104 6556 m=audio 14918 RTP/AVP 0 6557 a=rtpmap:0 PCMU/8000 6559 200 OK F15 6560 SS1 -> GW1 6562 SIP/2.0 200 OK 6563 Via: SIP/2.0/UDP gw1.wcom.com:5060 6564 Record-Route: 6565 From: PBX_A 6566 To: sip:444-3333,phone-context=p1234@ss1.wcom.com 6567 ;user=phone;tag=123456789 6568 Call-Id: 12345600@gw1.wcom.com 6569 CSeq: 1 INVITE 6570 Contact: sip:+1-918-555-3333@gw3.wcom.com;user=phone 6571 Content-Type: application/sdp 6572 Content-Length: 134 6574 v=0 6575 o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com 6576 t=0 0 6577 c=IN IP4 100.101.102.104 6578 m=audio 14918 RTP/AVP 0 6579 a=rtpmap:0 PCMU/8000 6581 GW1 -> PBX A 6583 Seizure 6584 ACK F16 6585 GW1 -> SS1 6586 ACK sip:+1-918-555-3333@ss1.wcom.com;user=phone SIP/2.0 6587 Via: SIP/2.0/UDP gw1.wcom.com:5060 6588 Route: 6589 From: PBX_A 6590 To: sip:444-3333,phone-context=p1234@ss1.wcom.com 6591 ;user=phone;tag=123456789 6592 Call-Id: 12345600@gw1.wcom.com 6593 CSeq: 1 ACK 6594 Content-Length: 0 6596 ACK F17 6597 SS1 -> GW3 6599 ACK sip:+1-918-555-3333@gw3.wcom.com;user=phone SIP/2.0 6600 Via: SIP/2.0/UDP ss1.wcom.com:5060 6601 Via: SIP/2.0/UDP gw1.wcom.com:5060 6602 From: PBX_A 6603 To: sip:444-3333,phone-context=p1234@ss1.wcom.com 6604 ;user=phone;tag=123456789 6605 Call-Id: 12345600@gw1.wcom.com 6606 CSeq: 1 ACK 6607 Content-Length: 0 6609 CONNect ACK F18 6610 GW3-> PBX C 6612 Protocol discriminator=Q.931 6613 Call reference: Flag=0, CR value=value in F9 SETUP message 6614 Message type=CONN ACK 6616 /* RTP streams are established between GW1 and GW3. */ 6618 /* User B Hangs Up with User A. */ 6619 DISConnect F19 6620 PBX C 6621 -> GW3 6623 Protocol discriminator=Q.931 6624 Call reference: Flag=1, CR value=value in F9 SETUP message 6625 Message type=DISC 6626 Cause=16 (Normal clearing) 6627 BYE F20 6628 GW3 -> SS1 6630 BYE sip:IdentifierString@ss1.wcom.com SIP/2.0 6631 Via: SIP/2.0/UDP gw3.wcom.com:5060 6632 Route: 6633 From: sip:444-3333,phone-context=p1234@ss1.wcom.com 6634 ;user=phone;tag=123456789 6635 To: PBX_A 6636 Call-Id: 12345600@gw1.wcom.com 6637 CSeq: 1 BYE 6638 Content-Length: 0 6640 BYE F21 6641 SS1 -> GW1 6643 BYE sip:IdentifierString@gw1.wcom.com SIP/2.0 6644 Via: SIP/2.0/UDP ss1.wcom.com:5060 6645 Via: SIP/2.0/UDP gw3.wcom.com:5060 6646 From: sip:444-3333,phone-context=p1234@ss1.wcom.com 6647 ;user=phone;tag=123456789 6648 To: PBX_A 6649 Call-Id: 12345600@gw1.wcom.com 6650 CSeq: 1 BYE 6651 Content-Length: 0 6653 GW1 -> PBX A 6655 Seizure removal 6656 RELease F22 6657 GW3-> PBX C 6659 Protocol discriminator=Q.931 6660 Call reference: Flag=0, CR value=value in F9 SETUP message 6661 Message type=REL 6662 200 OK F23 6663 GW1 -> SS1 6665 SIP/2.0 200 OK 6666 Via: SIP/2.0/UDP ss1.wcom.com:5060 6667 Via: SIP/2.0/UDP gw3.wcom.com:5060 6668 From: sip:444-3333,phone-context=p1234@ss1.wcom.com 6669 ;user=phone;tag=123456789 6670 To: PBX_A 6671 Call-Id: 12345600@gw1.wcom.com 6672 CSeq: 1 BYE 6673 Content-Length: 0 6674 200 OK F24 6675 SS11 6676 ->GW3 6678 SIP/2.0 200 OK 6679 Via: SIP/2.0/UDP gw3.wcom.com:5060 6680 From: sip:444-3333,phone-context=p1234@ss1.wcom.com 6681 ;user=phone;tag=123456789 6682 To: PBX_A 6683 Call-Id: 12345600@gw1.wcom.com 6684 CSeq: 1 BYE 6685 Content-Length: 0 6687 RELease COMplete F25 6688 PBX C 6689 -> GW3 6691 Protocol discriminator=Q.931 6692 Call reference: Flag=1, CR value=value in F9 SETUP message 6693 Message type=REL COM 6694 PBX A 6695 ->GW1 6697 Seizure removal 6699 7 Acknowledgements 6701 The authors wish to thank the following individuals for their 6702 assistance and review of this call flows document: Dean Willis, 6703 Henry Sinnreich, David Devanatham, Joe Pizzimenti, Matt Cannon, John 6704 Hearty, the whole MCI WorldCom IPOP Design team, and from Nortel 6705 Networks: Scott Orton, Greg Osterhout, Pat Sollee, Doug Weisenberg, 6706 Danny Mistry, Steve McKinnon, and Denise Ingram. 6708 8 References 6710 [1] S. Bradner, "The Internet Standards Process -- Revision 3", BCP 6711 9, RFC 2026, October 1996. 6713 [2] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP: 6714 Session Initiation Protocol", RFC 2543, March 1999. 6716 [3] R. Sparks, C. Cunningham, A. Johnston, S. Donovan, D. Willis, 6717 and K. Summers, "SIP Telephony Service Examples with Call 6718 Flows", Internet Draft, Internet Engineering Task Force, October 6719 1999. Work in progress. 6721 [4] S. Kent, R. Atkinson, "Security Architecture for the Internet 6722 Protocol", RFC 2401, November 1998. 6724 [5] S. Donovan, J. Hearty, M. Cannon, H. Schulzrinne, and J. 6725 Rosenberg, "SIP 183 Session Progress Message", Internet Draft, 6726 Internet Engineering Task Force, June 1999. Work in progress. 6728 [6] J. Rosenberg, and H. Schulzrinne, "Reliability of Provisional 6729 Responses in SIP", Internet Draft, Internet Engineering Task 6730 Force, May 20, 1999, Work in progress. 6732 [7] A. Vaha-Sipila, "URLs for Telephone Calls", Internet Draft, 6733 Internet Engineering Task Force, September 1999, Work in 6734 progress 6736 [8] S. Donovan, M. Cannon, "The SIP INFO Method", Internet Draft, 6737 Internet Engineering Task Force, June 1999. Work in progress. 6739 [9] G. Camarillo, "Best Current Practice for ISUP to SIP Mapping", 6740 Internet Draft, Internet Engineering Task Force, August 1999, 6741 Work in progress. 6743 Author's Addresses 6745 Alan Johnston 6746 MCI WorldCom 6747 100 S 4th Street Phone: +1-314-342-7360 6748 St. Louis, MO 63104 Email: alan.johnston@wcom.com 6750 Steve Donovan 6751 MCI WorldCom 6752 901 International Parkway Phone: +1-972-729-1621 6753 Richardson, TX 65081 Email: steven.r.donovan@wcom.com 6755 Robert Sparks 6756 MCI WorldCom 6757 2400 N Glenville Drive Phone: +1-972-729-5241 6758 Richardson, TX 75082 Email: Robert.Sparks@wcom.com 6760 Chris Cunningham 6761 MCI WorldCom 6762 400 International Parkway Phone: +1-972-729-3110 6763 Richardson, TX 75081 Email: Chris.Cunningham@wcom.com 6765 Kevin Summers 6766 MCI WorldCom 6767 2400 N Glenvile Drive Phone: +1-972-729-7976 6768 Richardson, TX 75082 Email: Kevin.Summers@wcom.com 6770 Copyright Notice 6772 "Copyright (C) The Internet Society 1999. All Rights Reserved. 6774 This document and translations of it may be copied and furnished to 6775 others, and derivative works that comment on or otherwise explain it 6776 or assist in its implementation may be prepared, copied, published 6777 and distributed, in whole or in part, without restriction of any 6778 kind, provided that the above copyright notice and this paragraph are 6779 included on all such copies and derivative works. However, this 6780 document itself may not be modified in any way, such as by removing 6781 the copyright notice or references to the Internet Society or other 6782 Internet organizations, except as needed for the purpose of 6783 developing Internet standards in which case the procedures for 6784 copyrights defined in the Internet Standards process must be 6785 followed, or as required to translate it into languages other than 6786 English. 6788 The limited permissions granted above are perpetual and will not be 6789 revoked by the Internet Society or its successors or assigns. 6791 This document and the information contained herein is provided on an 6792 "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING 6793 TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING 6794 BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION 6795 HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF 6796 MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.