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'1') (Obsoleted by RFC 3261, RFC 3262, RFC 3263, RFC 3264, RFC 3265) -- Possible downref: Non-RFC (?) normative reference: ref. '2' == Outdated reference: A later version (-03) exists of draft-ietf-sip-isup-01 -- Possible downref: Normative reference to a draft: ref. '3' Summary: 7 errors (**), 0 flaws (~~), 4 warnings (==), 4 comments (--). Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Internet Engineering Task Force Gonzalo Camarillo 3 Internet draft Adam Roach 4 Ericsson 5 August 2001 6 Expires: February 2002 Jon Peterson 7 NeuStar 9 Lyndon Ong 10 Ciena 12 Mapping of ISUP Overlap Signalling to SIP 14 Status of this Memo 16 This document is an Internet-Draft and is in full conformance with 17 all provisions of Section 10 of RFC2026. 19 Internet-Drafts are working documents of the Internet Engineering 20 Task Force (IETF), its areas, and its working groups. Note that 21 other groups may also distribute working documents as Internet- 22 Drafts. Internet-Drafts are draft documents valid for a maximum of 23 six months and may be updated, replaced, or obsoleted by other 24 documents at any time. It is inappropriate to use Internet- Drafts 25 as reference material or to cite them other than as "work in 26 progress." 28 The list of current Internet-Drafts can be accessed at 29 http://www.ietf.org/ietf/1id-abstracts.txt 30 The list of Internet-Draft Shadow Directories can be accessed at 31 http://www.ietf.org/shadow.html. 33 Abstract 35 This document describes a way to map ISUP overlap signalling to SIP. 37 Camarillo/Roach/Peterson/Ong 1 38 Mapping of ISUP Overlap Signalling to SIP 40 TABLE OF CONTENTS 42 1 Introduction.................................................2 43 2 Overlap signalling in SIP....................................2 44 3 ISUP to SIP..................................................3 45 3.1 Waiting for the minimum amount of digits.....................3 46 3.2 Sending the first INVITE.....................................3 47 3.3 Sending overlap signalling to the SIP network................4 48 3.4 Applicability of this mechanism..............................5 49 3.5 Receiving multiple responses.................................5 50 3.6 Canceling pending INVITE transactions........................6 51 3.7 INVITEs reaching multiple gateways...........................6 52 4 SIP to ISUP..................................................6 53 4.1 Receiving subsequent INVITEs.................................6 54 5 Conclusions..................................................6 55 6 Acknoledgements..............................................7 56 7 References...................................................7 57 8 Authors� addresses...........................................7 59 1. Introduction 61 A mapping between the Session Initiation Protocol (SIP) [1] and the 62 ISDN User Part (ISUP) [2] of SS7 is described in [3]. However, [3] 63 just takes into consideration ISUP en-bloc signalling. En-bloc 64 signalling consists of sending the complete telephone number of the 65 callee in the first signalling message. Although modern switches 66 always use en-bloc signalling, some parts of the PSTN still use 67 overlap signalling. Overlap signalling consists of sending just some 68 digits of the callee�s number in the first signalling message. 69 Further digits are sent in subsequent signalling messages. 71 2. Overlap signalling in SIP 73 SIP uses en-bloc signalling. The Request-URI of an INVITE message 74 contains the whole address of the callee. Even if the Request-URI 75 contains a tel URI instead of a SIP URI, the INVITE contains the 76 whole number. Breaking this principle would just bring undesirable 77 problems to network designers. Therefore, it is strongly recommended 78 not to use any kind of overlap signalling in a SIP network. The 79 recommended behavior is to convert overlap signalling to en-bloc at 80 the edge of the network and then use en-bloc signalling in SIP. A 81 gateway connected to a part of the PSTN where overlap signalling is 82 used can perform this conversion through the use of timers. 84 However, although its use is discouraged, some applications need to 85 use overlap signalling in order to meet service requirements (i.e. 86 establishment time). Such applications should use the mechanism 87 described in this document. This document also describes in which 88 scenarios is acceptable to use such a mechanism and when, on the 89 other hand, it is completely unacceptable to use overlap. 91 Camarillo/Roach/Peterson/Ong 2 92 Mapping of ISUP Overlap Signalling to SIP 94 3. ISUP to SIP 96 In this scenario the gateway receives an IAM (Initial Address 97 Message) that contains just a portion of the called number. The rest 98 of the digits dialed arrive later in one or more SAMs (Subsequent 99 Address Message). 101 3.1 Waiting for the minimum amount of digits 103 If the IAM contain less than the minimum amount of digits to route a 104 call, the gateway starts T35 and waits until the minimum amount of 105 digits that can represent a telephone number is received (or a stop 106 digit is received). If T35 expires before the minimum amount of 107 digits (or a stop digit) has been received a REL with cause value 28 108 is sent to the ISUP side. 110 If a stop digit is received the INVITE message generated by the 111 gateway will contain the complete called number. Therefore, the call 112 proceeds as usual - no overlap signalling in the SIP network. 114 3.2 Sending the first INVITE 116 There are cases when the gateway, after having received the 117 minimum amount of digits, cannot know whether the number received is 118 a complete number or not. Since supporting overlap signalling in the 119 SIP network is an option that may be deemed undesirable, the gateway 120 may elect to collect digits until a timer (T10) expires or a stop 121 digit (such as #) is entered by the user (note that T10 is 122 refreshed every time a new digit is received). 124 In this case, when T10 expires and an INVITE with the digits 125 collected so far is sent to the SIP side. After this, any SAM 126 received is ignored. 128 PSTN MGC/MG SIP 129 | | | 130 |-----------IAM----------->| Starts T10 | 131 | | | 132 |-----------SAM----------->| Starts T10 | 133 | | | 134 |-----------SAM----------->| Starts T10 | 135 | | | 136 | | | 137 | T10 expires |---------INVITE---------->| 138 | | | 140 Note that T10 is defined for conversion between CAS signalling and 141 en-bloc ISUP. PSTN switches usually implement an equivalent 142 proprietary timer to convert overlap ISUP to en-bloc ISUP. This 144 Camarillo/Roach/Peterson/Ong 3 145 Mapping of ISUP Overlap Signalling to SIP 147 document uses T10 and does not define a new timer because T10 seems 148 suitable for overlap to SIP conversion. 150 3.3 Sending overlap signalling to the SIP network 152 Although the behavior just described is recommended by this 153 document, a gateway might still decide to send overlap signalling in 154 the SIP network. In this case, the gateway should proceed as 155 follows. 157 As soon as the minimum amount of digits is received an INVITE is 158 sent and T10 is started. This INVITE is built following the 159 procedures described in [3]. 161 If a SAM arrives T10 is refreshed and a new INVITE with the new 162 digits received is sent. The new INVITE has the same Call-ID and the 163 same From tag as the first INVITE sent, but has an updated Request- 164 URI field. The new INVITE contains no To tag. 166 Note that it is possible to receive a response to the first 167 INVITE before having sent the second INVITE. In this case, the 168 response received would contain a To tag and information 169 (Record-Route and Contact) to build a Route header. The new 170 INVITE to be sent (containing new digits) should not use any of 171 these headers. That is, the new INVITE does not contain neither 172 To tag nor Route header. This way this new INVITE can be routed 173 dynamically by the network providing services (see Section 174 3.7). 176 The new INVITE should, of course, contain a Cseq field. It is 177 recommended that the Cseq of the new INVITE is higher than any of 178 the previous Cseq that the gateway has generated for this Call-ID 179 (no matter for which call-leg the Cseq was generated). 181 When an INVITE forks responses from different locations might 182 be received. New requests such as PRACK, COMET or INVITEs 183 negotiating early media can be sent to every particular remote 184 location. This implies that the Cseq number spaces of different 185 call-legs within the same call are different. Sending a new 186 INVITE with a Cseq that is still unused by any of the remote 187 destinations avoids confusion at the destination. 189 If the gateway is encapsulating ISUP messages as SIP bodies, it 190 should place the IAM and all the SAMs received so far in this 191 INVITE. 193 PSTN MGC/MG SIP 194 | | | 195 |-----------IAM----------->| Starts T10 | 196 | |---------INVITE---------->| 197 | | | 198 |-----------SAM----------->| Starts T10 | 199 | |---------INVITE---------->| 201 Camarillo/Roach/Peterson/Ong 4 202 Mapping of ISUP Overlap Signalling to SIP 204 | | | 205 |-----------SAM----------->| Starts T10 | 206 | |---------INVITE---------->| 207 | | | 209 If class 4, 5 or 6 final responses arrives (e.g. 484 address 210 incomplete) for the pending INVITE transactions before T10 has 211 expired the gateway should not send any REL. A REL is sent just if 212 no more SAMs arrive, T10 expires and all the INVITEs sent have been 213 answered with a final response (different than 200 OK). 215 PSTN MGC/MG SIP 216 | | | 217 |-----------IAM----------->| Starts T10 | 218 | |---------INVITE---------->| 219 | |<---------484-------------| 220 | |----------ACK------------>| 221 | | | 222 | | | 223 | T10 expires | | 224 |<----------REL------------| | 226 The status code of the response to the last INVITE sent by the 227 gateway (the one that contained more digits) is used to calculate 228 the cause value of the REL as described in [3]. 230 3.4 Applicability of this mechanism 232 This mechanism is applicable only under certain circumstances. A 233 ingress gateway may use overlap signalling in SIP only if an 234 analysis of the called party number shows that it belongs to a part 235 of the PSTN where overlap signalling is used. This ensures that a 236 particular prefix of the number does not identify any other user. 238 When en-bloc signalling is used in the PSTN a phone number might be 239 a prefix of another one. This situation is not common, but it can 240 certainly occur. If overlap signalling was used in this situation a 241 different user than the one the caller intended to call might be 242 contacted. 244 3.5 Receiving multiple responses 246 When overlap signalling in SIP is used the ingress gateway sends 247 multiple INVITEs. Accordingly, it will receive multiple responses. 248 The responses to all the INVITEs sent except for the last one are 249 typically 400 class responses (e.g. 484 Address Incomplete or 490 250 Request Updated) that terminate the INVITE transaction. 252 However, a 183 Session Progress response with a media description 253 can also be received. The media stream will typically contain a 254 message such as "The number you have just dialed does not exist". 256 Camarillo/Roach/Peterson/Ong 5 257 Mapping of ISUP Overlap Signalling to SIP 259 The issue of receiving different 183 Session Progress responses with 260 media descriptions does not only apply to overlap signalling. When 261 vanilla SIP is used, several responses can also arrive to a gateway 262 if the INVITE forked. It is then up to the gateway to decide which 263 media stream should be played to the user. 265 However, overlap signalling adds a requirement to this process. A 266 media stream corresponding to the response to an INVITE with a 267 greater number of digits should be given more priority than media 268 streams from responses with less digits. 270 3.6 Canceling pending INVITE transactions 272 When a gateway sends a new INVITE containing new digits, it should 273 not CANCEL the previous INVITE transaction. This CANCEL could arrive 274 before the new INVITE to an egress gateway and trigger a REL before 275 the new INVITE arrived. INVITE transactions are typically terminated 276 by the reception of 400 class responses. 278 However, once a 200 OK response has been received, the gateway 279 should CANCEL all the previous INVITE transactions that did not 280 contain enough digits. A particular gateway might implement a timer 281 to wait for some time before sending any CANCEL. This gives time to 282 all the previous INVITE transactions to terminate smoothly without 283 generating more signalling traffic (CANCEL messages). 285 3.7 INVITEs reaching multiple gateways 287 Since every new INVITE sent by a gateway represents a new 288 transaction they can be routed in different ways. For instance, the 289 first INVITE might be routed to a particular gateway and a 290 subsequent INVITE to another. The result is that both gateways 291 generate an IAM. Since one of the IAMs (or both) has an incomplete 292 number, it would fail, having already consumed PSTN resources. 294 It has been proposed to make all the INVITEs follow the same path as 295 the first one. This proposal would resolve the problem of having 296 INVITEs hitting different gateways, but would restrict the number of 297 services the SIP network can provide. It would not be possible to 298 route a subsequent INVITE to an application server just because the 299 previous one was routed in a different way. 301 This issue should be taken into consideration before using overlap 302 signalling in SIP. If sending multiple IAMs to the PSTN is not 303 acceptable in a particular domain, overlap signalling should not be 304 used. 306 4. SIP to ISUP 308 In this scenario the gateway receives multiple INVITEs that belong 309 to the same call (same Call-ID) but have different Request-URIs. 311 4.1 Receiving subsequent INVITEs 313 Camarillo/Roach/Peterson/Ong 6 314 Mapping of ISUP Overlap Signalling to SIP 316 An egress gateway does not have any means to know whether SIP 317 overlap signalling is being used or not. So, upon reception of an 318 INVITE, the gateway generates an IAM following the procedures 319 described in [3]. 321 If a gateway receives a subsequent INVITE with the same Call-ID and 322 From tag as the previous one and an updated Request-URI, a SAM 323 should be generated as opposed to a new IAM. Upon reception of a 324 subsequent INVITE, the INVITE received previously is answered with 325 490 Request Updated. 327 If the gateway is attached to the PSTN in an area where en-bloc 328 signalling is used, a REL for the previous IAM and a new IAM should 329 be generated. 331 5. Conclusions 333 The mechanism described in this document is intended to be used in a 334 close environment. Using it in an open network such as the Internet 335 would cause problems such as multiple IAMs generated. If this 336 mechanism was used with telephone numbers that belong to an en-bloc 337 zone, calls could end up reaching a different callee than the one 338 who was supposed to receive the call. 340 Due to these problems, it is strongly recommended that this 341 mechanism is only used if a particular application must fulfil 342 strong requirements regarding establishment delay. Otherwise, the 343 ingress gateway should always perform overlap to en-bloc conversion. 345 6. Acknowledgments 347 The authors would like to thank Jonathan Rosenberg and Olli Hynonen 348 for their feedback on this document. 350 7. References 352 [1] M. Handley, H. Schulzrinne, E. Schooler, J. Rosenberg, "SIP: 353 Session Initiation Protocol", RFC 2543, IETF; March 1999. 355 [2] "Application of the ISDN user part of CCITT signaling system No. 356 7 for international ISDN interconnections" ITU-T Q.767 357 recommendation, February 1991. 359 [3] G. Camarillo, A. Roach, J. Peterson, L. Ong, "ISUP to SIP 360 Mapping", draft-ietf-sip-isup-01.txt, IETF; May 2001. Work in 361 progress. 363 8. Authors� Addresses 365 Gonzalo Camarillo 366 Ericsson 367 Advanced Signalling Research Lab 369 Camarillo/Roach/Peterson/Ong 7 370 Mapping of ISUP Overlap Signalling to SIP 372 FIN-02420 Jorvas 373 Finland 374 Phone: +358 9 299 3371 375 Fax: +358 9 299 3052 376 Email: Gonzalo.Camarillo@ericsson.com 378 Adam Roach 379 Ericsson Inc. 380 Mailstop L-04 381 851 International Pkwy. 382 Richardson, TX 75081 383 USA 384 Phone: +1 972-583-7594 385 Fax: +1 972-669-0154 386 E-Mail: Adam.Roach@ericsson.com 388 Jon Peterson 389 NeuStar, Inc 390 1800 Sutter St Suite 570 391 Concord, CA 94520 392 USA 393 E-Mail: jon.peterson@neustar.com 395 Lyndon Ong 396 Ciena 397 10480 Ridgeview Court 398 Cupertino, CA 95014 399 E-Mail: lyOng@ciena.com 401 Full Copyright Statement 403 Copyright (c) The Internet Society (2001). All Rights Reserved. 405 This document and translations of it may be copied and furnished to 406 others, and derivative works that comment on or otherwise explain it 407 or assist in its implementation may be prepared, copied, published 408 and distributed, in whole or in part, without restriction of any 409 kind, provided that the above copyright notice and this paragraph 410 are included on all such copies and derivative works. However, this 411 document itself may not be modified in any way, such as by removing 412 the copyright notice or references to the Internet Society or other 413 Internet organizations, except as needed for the purpose of 414 developing Internet standards in which case the procedures for 415 copyrights defined in the Internet Standards process must be 416 followed, or as required to translate it into languages other than 417 English. 419 The limited permissions granted above are perpetual and will not be 420 revoked by the Internet Society or its successors or assigns. 422 Camarillo/Roach/Peterson/Ong 8 423 Mapping of ISUP Overlap Signalling to SIP 425 This document and the information contained herein is provided on an 426 "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING 427 TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING 428 BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION 429 HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF 430 MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. 432 Camarillo/Roach/Peterson/Ong 9