ETSI TS 102 392-2005 Corporate telecommunication Networks (CN) Signalling interworking between QSIG and SIP Call Transfer《共同通信网络(CN) 在QSIG和SIP间信令互通 呼叫转移(版本1 1 1)》.pdf

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1、 ETSI TS 102 392 V1.1.1 (2005-02)Technical Specification Corporate telecommunication Networks (CN);Signalling interworking between QSIG and SIP;Call TransferETSI ETSI TS 102 392 V1.1.1 (2005-02) 2 Reference DTS/ECMA-00297 Keywords interworking, QSIG, signalling ETSI 650 Route des Lucioles F-06921 So

2、phia Antipolis Cedex - FRANCE Tel.: +33 4 92 94 42 00 Fax: +33 4 93 65 47 16 Siret N 348 623 562 00017 - NAF 742 C Association but non lucratif enregistre la Sous-Prfecture de Grasse (06) N 7803/88 Important notice Individual copies of the present document can be downloaded from: http:/www.etsi.org

3、The present document may be made available in more than one electronic version or in print. In any case of existing or perceived difference in contents between such versions, the reference version is the Portable Document Format (PDF). In case of dispute, the reference shall be the printing on ETSI

4、printers of the PDF version kept on a specific network drive within ETSI Secretariat. Users of the present document should be aware that the document may be subject to revision or change of status. Information on the current status of this and other ETSI documents is available at http:/portal.etsi.o

5、rg/tb/status/status.asp If you find errors in the present document, please send your comment to one of the following services: http:/portal.etsi.org/chaircor/ETSI_support.asp Copyright Notification No part may be reproduced except as authorized by written permission. The copyright and the foregoing

6、restriction extend to reproduction in all media. European Telecommunications Standards Institute 2005. All rights reserved. DECTTM, PLUGTESTSTM and UMTSTM are Trade Marks of ETSI registered for the benefit of its Members. TIPHONTMand the TIPHON logo are Trade Marks currently being registered by ETSI

7、 for the benefit of its Members. 3GPPTM is a Trade Mark of ETSI registered for the benefit of its Members and of the 3GPP Organizational Partners. ETSI ETSI TS 102 392 V1.1.1 (2005-02) 3 Contents Intellectual Property Rights5 Foreword.5 Brief history5 1 Scope 6 2 References 6 3 Terminology.7 4 Defin

8、itions8 4.1 External definitions 8 4.2 Other definitions.8 4.2.1 User A.8 4.2.2 User B .8 4.2.3 User C .8 4.2.4 Call transfer 8 4.2.5 Single step call transfer.8 4.2.6 Call transfer by join 8 4.2.7 Call transfer by rerouteing 8 4.2.8 Corporate telecommunication Network (CN).9 4.2.9 IP network.9 4.2.

9、10 Private Integrated Services Network (PISN) 9 4.2.11 Private Integrated services Network eXchange (PINX) .9 5 Abbreviations and acronyms9 6 Background and architecture9 7 Procedures 10 7.1 Call transfers in QSIG 10 7.2 Call transfer in SIP .10 7.3 Scope of the interworking functions.10 7.3.1 QSIG

10、side .10 7.3.2 SIP side.11 7.3.3 Discussion over transfer interworking functions 11 7.4 Mapping of numbers and URIs 13 7.5 UAC Processing .14 7.5.1 Receipt of a FACILITY message with callTransferComplete invoke APDU 14 7.5.2 Receipt of a FACILITY message with callTransferUpdate invoke APDU 14 7.5.3

11、Receipt of a FACILITY message with ssctInitiate invoke APDU14 7.5.4 Receipt of a SETUP message with ssctSetup invoke APDU15 7.5.5 Receipt of a FACILITY message with subaddressTransfer invoke APDU16 7.6 UAS Processing16 7.6.1 Receipt of a SIP REFER request 16 7.6.1.1 No embedded Replaces header field

12、 in the Refer-To URI16 7.6.1.2 Matching embedded Replaces header in the Refer-To URI18 7.6.1.3 Non-matching embedded Replaces header in the Refer-To URI 19 7.6.2 Receipt of a SIP INVITE request21 7.6.2.1 Receipt of an INVITE with no Replaces header .21 7.6.2.2 Receipt of an INVITE with a Replaces he

13、ader .21 7.6.3 Receipt of a SIP request with revised identity 22 8 Example message sequences22 8.1 Call transfer by join where User B and C are SIP participants.22 8.2 Call transfer where User A is a SIP participant23 8.3 Call transfer where User A is a SIP participant and where two gateways are use

14、d24 8.4 Call transfer where User A and User B are SIP participants 25 8.5 Single step call transfer where User B is a SIP participant 26 ETSI ETSI TS 102 392 V1.1.1 (2005-02) 4 8.6 Unsuccessful Single step call transfer where User A and User C are SIP participants 27 8.7 Single step call transfer wh

15、ere User A and User B are SIP participants 28 9 Security considerations.28 History 29 ETSI ETSI TS 102 392 V1.1.1 (2005-02) 5 Intellectual Property Rights IPRs essential or potentially essential to the present document may have been declared to ETSI. The information pertaining to these essential IPR

16、s, if any, is publicly available for ETSI members and non-members, and can be found in ETSI SR 000 314: “Intellectual Property Rights (IPRs); Essential, or potentially Essential, IPRs notified to ETSI in respect of ETSI standards“, which is available from the ETSI Secretariat. Latest updates are ava

17、ilable on the ETSI Web server (http:/webapp.etsi.org/IPR/home.asp). Pursuant to the ETSI IPR Policy, no investigation, including IPR searches, has been carried out by ETSI. No guarantee can be given as to the existence of other IPRs not referenced in ETSI SR 000 314 (or the updates on the ETSI Web s

18、erver) which are, or may be, or may become, essential to the present document. Foreword This Technical Specification (TS) has been produced by ECMA on behalf of its members and those of the European Telecommunications Standards Institute (ETSI). Brief history The present document is one of a series

19、of Ecma Standards defining the interworking of services and signalling protocols deployed in Corporate telecommunication Networks (CNs) (also known as enterprise networks). The series uses telecommunication concepts as developed by ITU-T and conforms to the framework of International Standards on Op

20、en Systems Interconnection as defined by ISO/IEC. The present document specifies call transfer interworking between the Session Initiation Protocol (SIP) and QSIG within corporate telecommunication networks (also known as enterprise networks). SIP is an Internet application-layer control (signalling

21、) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include, in particular, telephone calls. The present document is based upon the practical experience of Ecma member companies and the results of their active and continuous participation in the

22、 work of ISO/IEC JTC1, ITU-T, IETF, ETSI and other international and national standardization bodies. It represents a pragmatic and widely based consensus. ETSI ETSI TS 102 392 V1.1.1 (2005-02) 6 1 Scope The present document specifies call transfer interworking between the Session Initiation Protoco

23、l (SIP) and “QSIG“ within Corporate telecommunication Networks (CN), also known as enterprise networks. “QSIG“ is a signalling protocol that operates between Private Integrated services Network eXchanges (PINX) within a Private Integrated Services Network (PISN). A PISN provides circuit-switched bas

24、ic services and supplementary services to its users. QSIG is specified in Ecma Standards, in particular 1 (call control in support of basic services), 2 (generic functional protocol for the support of supplementary services) and a number of Standards specifying individual supplementary services. Tra

25、nsfer services are specified in 3, 6 and the QSIG signalling protocol in support of these services is specified in 4 and 7. In particular, this signalling protocol signals information about call transfer to the users who are involved. NOTE: The name QSIG was derived from the fact that it is used for

26、 signalling at the Q reference point. The Q reference point is a point of demarcation between two PINXs. SIP is an application layer protocol for establishing, terminating and modifying multimedia sessions. It is typically carried over IP. Telephone calls are considered as a type of multimedia sessi

27、on where just audio is exchanged. SIP is defined in 10. As the support of telephony within corporate networks evolves from circuit-switched technology to Internet technology, the two technologies will co-exist in many networks for a period, perhaps several years. Therefore there is a need to be able

28、 to establish, modify, terminate and transfer sessions involving participants in the SIP network and participants in the QSIG network. Such calls are supported by gateways that perform interworking between SIP and QSIG. The present document specifies SIP-QSIG signalling interworking for transfer ser

29、vices between a PISN employing QSIG and a corporate IP network employing SIP. 2 References 1 ISO/IEC 11572: “Information technology - Telecommunications and information exchange between systems - Private Integrated Services Network - Circuit mode bearer services - Inter-exchange signalling procedure

30、s and protocol“ (also published by Ecma as Standard ECMA-143). 2 ISO/IEC 11582: “Information technology - Telecommunications and information exchange between systems - Private Integrated Services Network - Generic functional protocol for the support of supplementary services - Inter-exchange signall

31、ing procedures and protocol“ (also published by Ecma as Standard ECMA-165). 3 ISO/IEC 13865: “Information technology - Telecommunications and information exchange between systems - Private Integrated Services Network - Specification, functional model and information flows - Call Transfer supplementa

32、ry service“ (also published by Ecma as Standard ECMA-177). 4 ISO/IEC 13869: “Information technology - Telecommunications and information exchange between systems - Private Integrated Services Network - Inter-exchange signalling protocol - Call Transfer supplementary service“ (also published by Ecma

33、as Standard ECMA-178). 5 ISO/IEC 17343: “Information technology - Telecommunications and information exchange between systems - Corporate telecommunication networks - Signalling interworking between QSIG and SIP - Basic services“ (also published by Ecma as Standard ECMA-339). 6 ISO/IEC 19459: “Infor

34、mation technology - Telecommunications and information exchange between systems - Private Integrated Services Network - Specification, functional model and information flows - Single Step Call Transfer Supplementary Service“ (also published by Ecma as Standard ECMA-299). ETSI ETSI TS 102 392 V1.1.1

35、(2005-02) 7 7 International Standard ISO/IEC 19460: “Information technology - Telecommunications and information exchange between systems - Private Integrated Services Network - Inter-exchange signalling protocol - Single Step Call Transfer supplementary service“ (also published by Ecma as Standard

36、ECMA-300). 8 Ecma Technical Report TR/86: “Corporate Telecommunication Networks - User Identification in a SIP/QSIG Environment“. 9 IETF RFC 2119: “Key words for use in RFCs to Indicate Requirement Levels“. 10 IETF RFC 3261: “SIP: Session Initiation Protocol“. 11 IETF RFC 3323: “A Privacy Mechanism

37、for the Session Initiation Protocol (SIP)“. 12 IETF RFC 3515: “The Session Initiation Protocol (SIP) Refer Method“. 13 IETF RFC 3891: “The Session Initiation Protocol (SIP) Replaces Header“. 14 IETF RFC 3892: “The Session Initiation Protocol (SIP) Referred-By Mechanism“. 15 R. Sparks, A. Johnston, “

38、Session Initiation Protocol Call Control - Transfer“, draft-ietf-sipping-cc-transfer-02 (work in progress). 3 Terminology In the present document, the key words “MUST“, “MUST NOT“, “REQUIRED“, “SHALL“, “SHALL NOT“, “SHOULD“, “SHOULD NOT“, “RECOMMENDED“, “MAY“, and “OPTIONAL“ are to be interpreted as

39、 described in RFC 2119 9 and indicate requirement levels for compliant SIP implementations. In the interests of keeping the normative text and the diagrams as simple as possible, the QSIG messages in the present document implicitly follow QSIG signalling rules of 1 and 2. For instance, sending a QSI

40、G DISCONNECT message on a call where a QSIG DISCONNECT has already been sent is implicitly forbidden and therefore not mentioned as such in the present document. The figures in the present document are provided as examples. They are not normative. In the interests of keeping the diagrams simple, som

41、e SIP messages (ACK, PRACK, final responses to BYE and NOTIFY) are not shown. The following notation is used for call transfer information within QSIG messages: - xxx.inv - invoke application protocol data unit (APDU) of operation xxx. - xxx.res - return result APDU of operation xxx. - xxx.err - ret

42、urn error APDU of operation xxx. The following abbreviations are used: - ctActive stands for callTransferActive. - ctComplete stands for callTransferComplete. The drawings use the following conventions: - D1 and D2 are SIP dialogs. CR1 and CR2 are QSIG call references. By convention, D1 is mapped to

43、 CR1 and D2 to CR2. - A SIP message is prefixed by (Dx-y), when it belongs to SIP dialog Dx and is part of SIP transaction y. - The method or response code of the SIP messages is displayed, as well as the name of SIP header fields that play a role in the interworking functions. Some examples display

44、 an “Identity:“ information field. It indicates that the local identity information field should be mapped to a real SIP identity information field as described in clause 7.4. ETSI ETSI TS 102 392 V1.1.1 (2005-02) 8 4 Definitions For the purposes of the present document, the following definitions ap

45、ply. 4.1 External definitions The definitions in 1 and 10 apply as appropriate. 4.2 Other definitions 4.2.1 User A The served user, i.e. the user requesting Call transfer or Single step call transfer. 4.2.2 User B A user who is in communication with User A and who will be transferred to User C. NOTE

46、: This definitions differs from 3, in order to use similar conventions for QSIG Call transfer and QSIG Single step call transfer. 4.2.3 User C The user to whom the call is transferred. 4.2.4 Call transfer The act of enabling a user (User A) to transform two of that users calls (at least one of which

47、 must be answered) into a new call between the two other users (User B and User C) in the two calls. NOTE: Call transfer is very similar to the “attended transfer“ described in 15. A Call transfer before answer is a Call transfer that occurs before User C answers the call initiated by User A. 4.2.5

48、Single step call transfer The act of enabling a served user (User A) to transfer an active call (with User B) to a user (User C) that has no call established either to User A or to User B. On successful completion of Single step call transfer, User B and User C can communicate with each other and Us

49、er A are no longer involved in a call with User B or User C. NOTE: Single step call transfer is very similar to the “basic transfer“ described in 15. 4.2.6 Call transfer by join The effecting of transfer when User A is a PISN user by joining together the connections of the calls to User B and User C at User As PINX. 4.2.7 Call transfer by rerouteing The effecting of transfer by establishing a new connection to replace all or part of the connections of the calls to User B and User C. ETSI ETSI TS 102 392 V1.1.1 (2005-0