1、 ETSI TS 124 371 V15.0.0 (2018-06) Universal Mobile Telecommunications System (UMTS); LTE; Web Real-Time Communications (WebRTC) access to the IP Multimedia (IM) Core Network (CN) subsystem (IMS); Stage 3; Protocol specification (3GPP TS 24.371 version 15.0.0 Release 15) TECHNICAL SPECIFICATION ETSI
2、 ETSI TS 124 371 V15.0.0 (2018-06)13GPP TS 24.371 version 15.0.0 Release 15Reference RTS/TSGC-0124371vf00 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 Association but non luc
3、ratif 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 print versions of the
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9、by the GSM Association. ETSI ETSI TS 124 371 V15.0.0 (2018-06)23GPP TS 24.371 version 15.0.0 Release 15Intellectual Property Rights Essential patents IPRs essential or potentially essential to normative deliverables may have been declared to ETSI. The information pertaining to these essential IPRs,
10、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 availa
11、ble on the 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,
12、 or may be, or may become, essential to the present document. Trademarks The present document may include trademarks and/or tradenames which are asserted and/or registered by their owners. ETSI claims no ownership of these except for any which are indicated as being the property of ETSI, and conveys
13、 no right to use or reproduce any trademark and/or tradename. Mention of those trademarks in the present document does not constitute an endorsement by ETSI of products, services or organizations associated with those trademarks. Foreword This Technical Specification (TS) has been produced by ETSI 3
14、rd Generation Partnership Project (3GPP). The present document may refer to technical specifications or reports using their 3GPP identities, UMTS identities or GSM identities. These should be interpreted as being references to the corresponding ETSI deliverables. The cross reference between GSM, UMT
15、S, 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“, “need not“, “will“, “will not“, “can“ and “cannot“ are to be interpreted as described in clause 3.2 of the ETSI
16、 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 V15.0.0 (2018-06)33GPP TS 24.371 version 15.0.0 Release 15Contents Intellectual Property Rights 2g3Foreword . 2g3Modal
17、 verbs terminology 2g3Foreword . 6g31 Scope 7g32 References 7g33 Definitions and abbreviations . 9g33.1 Definitions 9g33.2 Abbreviations . 9g34 Overview of WebRTC access to IMS 10g34.1 General . 10g35 Functional entities 10g35.1 General . 10g35.2 WIC (WebRTC IMS Client) 10g35.3 WWSF (WebRTC Web Serv
18、er Function) 11g35.4 WAF (WebRTC Authorisation Function) 11g35.5 eP-CSCF (P-CSCF enhanced for WebRTC) 11g35.6 eIMS-AGW (IMS Access Gateway enhanced for WebRTC) 11g35A Data transport . 11g35A.1 General . 11g35A.2 UE 11g35A.3 WWSF (WebRTC Web Server Function) 11g35A.4 eP-CSCF (P-CSCF enhanced for WebR
19、TC) 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 General . 12g35C.2 UE 13g35C.3 WWSF (WebRTC Web Server Function) 13g35C.4 eP-CSCF (P-CSCF enhanced for WebRTC)
20、 13g35D Connection management 14g35D.1 General . 14g35D.2 UE 14g35D.3 WWSF (WebRTC Web Server Function) 14g35D.4 eP-CSCF (P-CSCF enhanced for WebRTC) 14g35E Presentation and control . 14g35E.1 General . 14g35E.2 UE 14g35E.3 WWSF (WebRTC Web Server Function) 14g35E.4 eP-CSCF (P-CSCF enhanced for WebR
21、TC) 14g35F Local system support functions 15g35F.1 General . 15g35F.2 UE 15g35F.3 WWSF (WebRTC Web Server Function) 15g35F.4 eP-CSCF (P-CSCF enhanced for WebRTC) 15g36 Registration and authentication 15g3ETSI ETSI TS 124 371 V15.0.0 (2018-06)43GPP TS 24.371 version 15.0.0 Release 156.1 General . 15g
22、36.2 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 . 16g36.2.1.3 W2 using IMS-AKA . 16g36.2.2 WIC registration of individual public user identity based on web authenticati
23、on 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) . 17g36.3.1 WIC registration of individual public user identity using web credentials . 17g36.3.2 WIC registration
24、 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 IMS authentication 17g36.4.1.1 Determination of IMS authentication mechanism . 17g36.4.1.2 W2 using SIP Digest c
25、redentials . 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 identity from a pool of public user identities. 19g36A Deregistration . 20g36A.1 General . 20g36A.2 WIC (WebRTC IMS C
26、lient) 20g36A.3 eP-CSCF (P-CSCF enhanced for WebRTC) 20g37 Call origination and termination . 20g37.1 General . 20g37.2 WIC (WebRTC IMS Client) 20g37.2.1 General 20g37.2.2 WIC originating call . 21g37.2.3 WIC terminating call 21g37.2.4 WIC emergency call . 22g37.3 WWSF (WebRTC Web Server Function) 2
27、2g37.4 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 procedure 24g37.4.5.1 WIC originating call 24g37.4.5.2 WIC terminating call . 27g38 Data channel open and close 29
28、g38.1 General . 29g38.2 WIC (WebRTC IMS Client) 29g38.2.1 General 29g38.2.2 WIC originating call . 29g38.2.3 WIC terminating call 30g38.3 WWSF (WebRTC Web Server Function) 30g38.4 eP-CSCF (P-CSCF enhanced for WebRTC) 30g38.4.1 General 30g38.4.2 WIC originating call . 30g38.4.3 WIC terminating call 3
29、1g39 Call modification 32g310 IP multimedia application support in the IM CN subsystem using webRTC 32g310.1 General . 32g310.2 Access to MMTel and supplementary services using webRTC . 32g310.2.1 General 32g310.2.2 WIC (WebRTC IMS Client) . 32g310.2.2.1 SIP based protocol used by the WIC . 32g310.2
30、.2.2 non-SIP based protocol used by the WIC . 32g310.2.3 WWSF (WebRTC Web Server Function) 32g310.2.4 eP-CSCF (P-CSCF enhanced for WebRTC) 32g3ETSI ETSI TS 124 371 V15.0.0 (2018-06)53GPP TS 24.371 version 15.0.0 Release 15Annex A (informative): Example signalling flows . 33g3A.1 Scope of signalling
31、flows . 33g3A.2 Void 33g3A.3 Signalling flows for registration . 33g3A.3.1 Void 33g3A.3.2 WIC registration of individual public user identity based on web authentication 33g3A.3.3 Void 35g3A.4 Void 35g3A.5 Void 35g3Annex B (informative): Change history . 36g3History 38g3ETSI ETSI TS 124 371 V15.0.0
32、(2018-06)63GPP TS 24.371 version 15.0.0 Release 15Foreword This Technical Specification has been produced by the 3rdGeneration Partnership Project (3GPP). The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG m
33、odify 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 the first digit: 1 presented to TSG for information; 2 presented to TSG for approval; 3 or greater indicates T
34、SG 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 incremented when editorial only changes have been incorporated in the document. ETSI ETSI TS 124 371 V15.0.0 (2018-0
35、6)73GPP TS 24.371 version 15.0.0 Release 151 Scope The present document provides the details for allowing Web Real-Time Communication (WebRTC) IMS Clients (WIC) to access the IP Multimedia (IM) Core Network (CN) subsystem. The present document is applicable to WebRTC IMS client (WIC), eP-CSCF, eIMS-
36、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 provisions of the present document. - References are either specific (identified by date of publication, edition n
37、umber, 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 reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest
38、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 Session Initiation Protocol (SIP)“. 3 3GPP TS 24.229: “IP multimedia call control protocol based on Session In
39、itiation 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 Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)“. 6 IETF RF
40、C 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 (IMS) Multimedia Telephony Service and supplementary services; Stage 1“. 8 3GPP TS 24.173: “IMS multimedia teleph
41、ony 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: Bearer Token Usage“. 11 3GPP TS 23.292: “IP Multimedia Subsystem (IMS) Centralized Services; Stage 2“. 12 RFC
42、 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 (IMS-ALG) IMS Access Gateway (IMS-AGW) interface“. 14 RFC 4145 (September 2005): “TCP-Based Media Transport in th
43、e Session Description Protocol (SDP)“. 15 RFC 8122 (March 2017): “Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the Session Description Protocol (SDP)“. 16 draft-ietf-rtcweb-data-channel-13 (January 2015): “WebRTC Data Channels“. ETSI ETSI TS 124 371 V15.0.0
44、 (2018-06)83GPP TS 24.371 version 15.0.0 Release 15Editors note: The above document cannot be formally referenced until it is published as an RFC. 17 draft-ietf-rtcweb-data-protocol-09 (January 2015): “WebRTC Data Channel Establishment Protocol“. Editors note: The above document cannot be formally r
45、eferenced until it is published as an RFC. 18 draft-ietf-mmusic-sctp-sdp-25 (March 2017): “Stream Control Transmission Protocol (SCTP)-Based Media Transport in the Session Description Protocol (SDP)“. Editors note: The above document cannot be formally referenced until it is published as an RFC. 19
46、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 Usage for Consent Freshness“. 22 RFC 5245 (April 2010): “Interactive Connectivity Establishment (ICE): A Protocol
47、for Network Address Translator (NAT) Traversal for Offer/Answer Protocols“. 23 RFC 8261 (November 2017): “Datagram Transport Layer Security (DTLS) Encapsulation of SCTP Packets“. 24 RFC 6455 (December 2011): “The WebSocket Protocol“. 25 draft-ietf-mmusic-sdp-bundle-negotiation-29 (April 2016): “Nego
48、tiating 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 draft-ietf-s
49、ipcore-sip-authn-00 (May 2017): “Third-Party Authentication for Session Initiation Protocol (SIP)“. 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-18 (March 2017): “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 Void.
