1、 ETSI TS 1Universal Mobile TelWeb Real-Timeaccess to the IP MsuProt(3GPP TS 24.3TECHNICAL SPECIFICATION124 371 V13.5.0 (2017elecommunications System (LTE; me Communications (WebRTCMultimedia (IM) Core Networksubsystem (IMS); Stage 3; rotocol specification .371 version 13.5.0 Release 1317-01) (UMTS);
2、 TC) rk (CN) 13) ETSI ETSI TS 124 371 V13.5.0 (2017-01)13GPP TS 24.371 version 13.5.0 Release 13Reference RTS/TSGC-0124371vd50 Keywords LTE,UMTS ETSI 650 Route des Lucioles F-06921 Sophia 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 As
3、sociation but non lucratif enregistre la Sous-Prfecture de Grasse (06) N 7803/88 Important notice The present document can be downloaded from: http:/www.etsi.org/standards-search The present document may be made available in electronic versions and/or in print. The content of any electronic and/or p
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9、and owned by the GSM Association. ETSI ETSI TS 124 371 V13.5.0 (2017-01)23GPP TS 24.371 version 13.5.0 Release 13Intellectual Property Rights IPRs essential or potentially essential to the present document may have been declared to ETSI. The information pertaining to these essential IPRs, if any, is
10、 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 available on the
11、 ETSI Web server (https:/ipr.etsi.org/). 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 server) which are, or may be
12、, or may become, essential to the present document. Foreword This Technical Specification (TS) has been produced by ETSI 3rd Generation Partnership Project (3GPP). The present document may refer to technical specifications or reports using their 3GPP identities, UMTS identities or GSM identities. Th
13、ese should be interpreted as being references to the corresponding ETSI deliverables. The cross reference between GSM, UMTS, 3GPP and ETSI identities can be found under http:/webapp.etsi.org/key/queryform.asp. Modal verbs terminology In the present document “shall“, “shall not“, “should“, “should no
14、t“, “may“, “need not“, “will“, “will not“, “can“ and “cannot“ are to be interpreted as described in clause 3.2 of the ETSI Drafting Rules (Verbal forms for the expression of provisions). “must“ and “must not“ are NOT allowed in ETSI deliverables except when used in direct citation. ETSI ETSI TS 124
15、371 V13.5.0 (2017-01)33GPP TS 24.371 version 13.5.0 Release 13Contents Intellectual Property Rights 2g3Foreword . 2g3Modal verbs terminology 2g3Foreword . 6g31 Scope 7g32 References 7g33 Definitions and abbreviations . 9g33.1 Definitions 9g33.2 Abbreviations . 9g34 Overview of WebRTC access to IMS 9
16、g34.1 General . 9g35 Functional entities 10g35.1 General . 10g35.2 WIC (WebRTC IMS Client) 10g35.3 WWSF (WebRTC Web Server Function) 10g35.4 WAF (WebRTC Authorisation Function) 10g35.5 eP-CSCF (P-CSCF enhanced for WebRTC) 11g35.6 eIMS-AGW (IMS Access Gateway enhanced for WebRTC) 11g35A Data transpor
17、t . 11g35A.1 General . 11g35A.2 UE 11g35A.3 WWSF (WebRTC Web Server Function) 11g35A.4 eP-CSCF (P-CSCF enhanced for WebRTC) 11g35B Data framing and securing . 12g35B.1 General . 12g35B.2 UE 12g35B.3 WWSF (WebRTC Web Server Function) 12g35B.4 eP-CSCF (P-CSCF enhanced for WebRTC) 12g35C Data formats .
18、 12g35C.1 General . 12g35C.2 UE 12g35C.3 WWSF (WebRTC Web Server Function) 13g35C.4 eP-CSCF (P-CSCF enhanced for WebRTC) 13g35D Connection management 13g35D.1 General . 13g35D.2 UE 13g35D.3 WWSF (WebRTC Web Server Function) 14g35D.4 eP-CSCF (P-CSCF enhanced for WebRTC) 14g35E Presentation and contro
19、l . 14g35E.1 General . 14g35E.2 UE 14g35E.3 WWSF (WebRTC Web Server Function) 14g35E.4 eP-CSCF (P-CSCF enhanced for WebRTC) 14g35F Local system support functions 14g35F.1 General . 14g35F.2 UE 14g35F.3 WWSF (WebRTC Web Server Function) 15g35F.4 eP-CSCF (P-CSCF enhanced for WebRTC) 15g3ETSI ETSI TS 1
20、24 371 V13.5.0 (2017-01)43GPP TS 24.371 version 13.5.0 Release 136 Registration and authentication 15g36.1 General . 15g36.2 WIC (WebRTC IMS Client) 16g36.2.1 WIC registration of individual Public User Identity using IMS authentication 16g36.2.1.1 General 16g36.2.1.2 W2 using SIP Digest credentials
21、. 16g36.2.1.3 W2 using IMS-AKA . 16g36.2.2 WIC registration of individual public user identity based on web authentication 16g36.2.3 WIC registration of individual public user identity from a pool of public user identities. 17g36.3 WWSF (WebRTC Web Server Function) and WAF (WebRTC Authorisation Func
22、tion) . 17g36.3.1 WIC registration of individual public user identity using web credentials . 17g36.3.2 WIC registration of individual public user identity from a pool of public user identities. 17g36.4 eP-CSCF (P-CSCF enhanced for WebRTC) 17g36.4.1 WIC registration of individual Public User Identit
23、y using IMS authentication 17g36.4.1.1 Determination of IMS authentication mechanism . 17g36.4.1.2 W2 using SIP Digest credentials . 17g36.4.1.3 W2 using IMS-AKA . 18g36.4.2 WIC registration of individual public user identity using web credentials . 19g36.4.3 WIC registration of individual public us
24、er identity from a pool of public user identities. 19g36A Deregistration . 20g36A.1 General . 20g36A.2 WIC (WebRTC IMS Client) 20g36A.3 eP-CSCF (P-CSCF enhanced for WebRTC) 20g37 Call origination and termination . 20g37.1 General . 20g37.2 WIC (WebRTC IMS Client) 21g37.2.1 General 21g37.2.2 WIC orig
25、inating call . 21g37.2.3 WIC terminating call 21g37.2.4 WIC emergency call . 21g37.3 WWSF (WebRTC Web Server Function) 22g37.4 eP-CSCF (P-CSCF enhanced for WebRTC) 22g37.4.1 General 22g37.4.2 WIC originating call . 22g37.4.3 WIC terminating call 23g37.4.4 WIC emergency call . 24g37.4.5 Media optimiz
26、ation procedure 24g37.4.5.1 WIC originating call 24g37.4.5.2 WIC terminating call . 26g38 Data channel open and close 28g38.1 General . 28g38.2 WIC (WebRTC IMS Client) 28g38.2.1 General 28g38.2.2 WIC originating call . 28g38.2.3 WIC terminating call 28g38.3 WWSF (WebRTC Web Server Function) 29g38.4
27、eP-CSCF (P-CSCF enhanced for WebRTC) 29g38.4.1 General 29g38.4.2 WIC originating call . 29g38.4.3 WIC terminating call 30g39 Call modification 30g310 IP multimedia application support in the IM CN subsystem using webRTC 30g310.1 General . 30g310.2 Access to MMTel and supplementary services using web
28、RTC . 30g310.2.1 General 30g310.2.2 WIC (WebRTC IMS Client) . 31g310.2.2.1 SIP based protocol used by the WIC . 31g310.2.2.2 non-SIP based protocol used by the WIC . 31g3ETSI ETSI TS 124 371 V13.5.0 (2017-01)53GPP TS 24.371 version 13.5.0 Release 1310.2.3 WWSF (WebRTC Web Server Function) 31g310.2.4
29、 eP-CSCF (P-CSCF enhanced for WebRTC) 31g3Annex A (informative): Example signalling flows . 32g3A.1 Scope of signalling flows . 32g3A.2 Void 32g3A.3 Signalling flows for registration . 32g3A.3.1 Void 32g3A.3.2 WIC registration of individual public user identity based on web authentication 32g3A.3.3
30、Void 34g3A.4 Void 34g3A.5 Void 34g3Annex B (informative): Change history . 35g3History 37g3ETSI ETSI TS 124 371 V13.5.0 (2017-01)63GPP TS 24.371 version 13.5.0 Release 13Foreword This Technical Specification has been produced by the 3rdGeneration Partnership Project (3GPP). The contents of the prese
31、nt document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: Version x.y.z
32、 where: x the first digit: 1 presented to TSG for information; 2 presented to TSG for approval; 3 or greater indicates TSG approved document under change control. y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. z the third digit
33、 is incremented when editorial only changes have been incorporated in the document. ETSI ETSI TS 124 371 V13.5.0 (2017-01)73GPP TS 24.371 version 13.5.0 Release 131 Scope The present document provides the details for allowing Web Real-Time Communication (WebRTC) IMS Clients (WIC) to access the IP Mu
34、ltimedia (IM) Core Network (CN) subsystem. The present document is applicable to WebRTC IMS client (WIC), eP-CSCF, eIMS-AGW, WebRTC Web Server Function (WWSF) and WebRTC Authorization Function (WAF). 2 References The following documents contain provisions which, through reference in this text, const
35、itute provisions of the present document. - References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. - For a specific reference, subsequent revisions do not apply. - For a non-specific reference, the latest version applies. In the case
36、 of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document in the same Release as the present document. 1 3GPP TR 21.905: “Vocabulary for 3GPP Specifications“. 2 IETF RFC 7118: “The WebSocket Protocol as a Transpor
37、t for the Session Initiation Protocol (SIP)“. 3 3GPP TS 24.229: “IP multimedia call control protocol based on Session Initiation Protocol (SIP) and Session Description Protocol (SDP); Stage 3“. 4 3GPP TS 23.228: “ IP Multimedia Subsystem (IMS); Stage 2“. 5 IETF RFC 5763: “Framework for Establishing
38、a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)“. 6 IETF RFC 5764: “Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)“. 7 3GPP TS 22.173: “IP Multimedia Core Network Sub
39、system (IMS) Multimedia Telephony Service and supplementary services; Stage 1“. 8 3GPP TS 24.173: “IMS multimedia telephony communication service and supplementary services; Stage 3“. 9 3GPP TS 33.203: “Access security for IP based services“. 10 RFC 6750 (October 2012): “The OAuth 2.0 Authorization
40、Framework: Bearer Token Usage“. 11 3GPP TS 23.292: “IP Multimedia Subsystem (IMS) Centralized Services; Stage 2“. 12 RFC 5009 (September 2007): “Private Header (P-Header) Extension to the Session Initiation Protocol (SIP) for Authorization of Early Media“. 13 3GPP TS 23.334: “IMS Application Level G
41、ateway (IMS-ALG) IMS Access Gateway (IMS-AGW) interface“. 14 RFC 4145 (September 2005): “TCP-Based Media Transport in the Session Description Protocol (SDP)“. 15 RFC 4572 (July 2006): “Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the Session Description Pro
42、tocol (SDP)“. 16 draft-ietf-rtcweb-data-channel-13 (January 2015): “WebRTC Data Channels“. ETSI ETSI TS 124 371 V13.5.0 (2017-01)83GPP TS 24.371 version 13.5.0 Release 13Editors note: The above document cannot be formally referenced until it is published as an RFC. 17 draft-ietf-rtcweb-data-protocol
43、-09 (January 2015): “WebRTC Data Channel Establishment Protocol“. Editors note: The above document cannot be formally referenced until it is published as an RFC. 18 draft-ietf-mmusic-sctp-sdp-16 (February 2016): “Stream Control Transmission Protocol (SCTP)-Based Media Transport in the Session Descri
44、ption Protocol (SDP)“. Editors note: The above document cannot be formally referenced until it is published as an RFC. 19 RFC 3261 (June 2002): “SIP: Session Initiation Protocol“. 20 RFC 3264 (June 2002): “An Offer/Answer Model with the Session Description Protocol (SDP)“. 21 RFC 7675 (October 2015)
45、: “STUN Usage for Consent Freshness“. 22 RFC 5245 (April 2010): “Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols“. 23 draft-ietf-tsvwg-sctp-dtls-encaps-09 (January 2015): “DTLS Encapsulation of SCTP Packets“. Editors
46、note (WI: IMS_WebRTC): The above document cannot be formally referenced until it is published as an RFC. 24 RFC 6455 (December 2011): “The WebSocket Protocol“. 25 draft-ietf-mmusic-sdp-bundle-negotiation-29 (April 2016): “Negotiating Media Multiplexing Using the Session Description Protocol (SDP)“.
47、Editors note: The above document cannot be formally referenced until it is published as an RFC. 26 RFC 3581 (August 2003): “An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing“. 27 draft-yusef-sipcore-sip-oauth-02 (April 2015): “The Session Initiation Protocol (SIP)
48、OAuth“. Editors note: The above document cannot be formally referenced until it is published as an RFC. 28 RFC 6544 (March 2012): “TCP Candidates with Interactive Connectivity Establishment (ICE)“. 29 Void. 30 draft-ietf-rtcweb-overview-15 (January 2016): “Overview: Real Time Protocols for Brower-ba
49、sed Applications“. Editors note: The above document cannot be formally referenced until it is published as an RFC. 31 draft-ietf-rtcweb-gateways-02 (January 2016): “WebRTC Gateways“. Editors note: The above document cannot be formally referenced until it is published as an RFC. 32 RFC 3310 (September 2002): “Hypertext Transfer Protocol (HTTP) Digest Authentication Using Authentication and Key Agreement (AKA)“. 33 RFC 4169 (November 2005): “Hypertext Transfer Protocol (HTTP) Digest Authentication Using
