1、 ETSI TS 1Universal Mobile TelWeb Real-Timeaccess to the IP MsuProt(3GPP TS 24.3TECHNICAL SPECIFICATION124 371 V13.3.0 (2016elecommunications System (LTE; me Communications (WebRTCMultimedia (IM) Core Networksubsystem (IMS); Stage 3; rotocol specification .371 version 13.3.0 Release 1316-04) (UMTS);
2、 TC) ork (CN) 13) ETSI ETSI TS 124 371 V13.3.0 (2016-04)13GPP TS 24.371 version 13.3.0 Release 13Reference RTS/TSGC-0124371vd30 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 A
3、ssociation 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
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9、 by the GSM Association. ETSI ETSI TS 124 371 V13.3.0 (2016-04)23GPP TS 24.371 version 13.3.0 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 publicly
10、 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 ETSI Web
11、 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, or may
12、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. These shoul
13、d 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 not“, “may“
14、, “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 124 371 V13.3
15、.0 (2016-04)33GPP TS 24.371 version 13.3.0 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 9g34.1 Gen
16、eral . 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 transport . 11g35
17、A.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 . 12g35C.1
18、 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 control . 14g35
19、E.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 124 371 V1
20、3.3.0 (2016-04)43GPP TS 24.371 version 13.3.0 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 . 16g36.2
21、.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 Function) . 1
22、7g36.3.1 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 Identity using I
23、MS 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 user identi
24、ty 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 originating c
25、all . 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 optimization pro
26、cedure 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 eP-CSCF (
27、P-CSCF enhanced for WebRTC) 29g38.4.1 General 29g38.4.2 WIC originating call . 29g38.4.3 WIC terminating call 29g39 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 webRTC . 30g
28、310.2.1 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 V13.3.0 (2016-04)53GPP TS 24.371 version 13.3.0 Release 1310.2.3 WWSF (WebRTC Web Server Function) 31g310.2.4 eP-CSCF
29、(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 Void 34g3
30、A.4 Void 34g3A.5 Void 34g3Annex B (informative): Change history . 35g3History 37g3ETSI ETSI TS 124 371 V13.3.0 (2016-04)63GPP TS 24.371 version 13.3.0 Release 13Foreword This Technical Specification has been produced by the 3rdGeneration Partnership Project (3GPP). The contents of the present docume
31、nt 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 where: x
32、 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 is incre
33、mented when editorial only changes have been incorporated in the document. ETSI ETSI TS 124 371 V13.3.0 (2016-04)73GPP TS 24.371 version 13.3.0 Release 131 Scope The present document provides the details for allowing Web Real-Time Communication (WebRTC) IMS Clients (WIC) to access the IP Multimedia
34、(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, constitute pro
35、visions 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 of a ref
36、erence 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 Transport for the
37、 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 a Secure
38、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 Subsystem (I
39、MS) 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 Framework
40、: 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 Gateway (I
41、MS-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 Protocol (SD
42、P)“. 16 draft-ietf-rtcweb-data-channel-13 (January 2015): “WebRTC Data Channels“. ETSI ETSI TS 124 371 V13.3.0 (2016-04)83GPP TS 24.371 version 13.3.0 Release 13Editors note: The above document cannot be formally referenced until it is published as an RFC. 17 draft-ietf-rtcweb-data-protocol-09 (Janu
43、ary 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-15 (September 2015): “Stream Control Transmission Protocol (SCTP)-Based Media Transport in the Session Description Pr
44、otocol (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): “STUN
45、Usage for Consent Freshness“. Editors note: The above document cannot be formally referenced until it is published as an RFC. 22 RFC 5245 (April 2010): “Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols“. 23 draft-ietf-
46、tsvwg-sctp-dtls-encaps-09 (January 2015): “DTLS Encapsulation of SCTP Packets“. Editors 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-23 (July
47、 2015): “Negotiating Media Multiplexing Using the Session Description Protocol (SDP)“. 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
48、 draft-yusef-sipcore-sip-oauth-02 (April 2015): “The Session Initiation Protocol (SIP) 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 dra
49、ft-ietf-rtcweb-overview-14 (June 2015): “Overview: Real Time Protocols for Brower-based Applications“. Editors note: The above document cannot be formally referenced until it is published as an RFC. 31 draft-ietf-rtcweb-gateways-01 (July 2015): “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 416