ETSI TS 124 371 V13.11.0 (2020-01)

TECHNICAL SPECIFICATION
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 13.11.0 Release 13)

3GPP TS 24.371 version 13.11.0 Release 13 1 ETSI TS 124 371 V13.11.0 (2020-01)

Reference
RTS/TSGC-0124371vdb0
Keywords
LTE,UMTS
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3GPP TS 24.371 version 13.11.0 Release 13 2 ETSI TS 124 371 V13.11.0 (2020-01)
Intellectual Property Rights
Essential patents
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pertaining to these essential IPRs, if any, is publicly available for ETSI members and non-members, and can be found
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Legal Notice
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. These shall be
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The cross reference between 3GPP and ETSI identities can be found under http://webapp.etsi.org/key/queryform.asp.
Modal verbs terminology
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"must" and "must not" are NOT allowed in ETSI deliverables except when used in direct citation.
ETSI
3GPP TS 24.371 version 13.11.0 Release 13 3 ETSI TS 124 371 V13.11.0 (2020-01)
Contents
Intellectual Property Rights . 2
Legal Notice . 2
Modal verbs terminology . 2
Foreword . 6
1 Scope . 7
2 References . 7
3 Definitions and abbreviations . 9
3.1 Definitions . 9
3.2 Abbreviations . 9
4 Overview of WebRTC access to IMS . 10
4.1 General . 10
5 Functional entities . 10
5.1 General . 10
5.2 WIC (WebRTC IMS Client) . 10
5.3 WWSF (WebRTC Web Server Function) . 11
5.4 WAF (WebRTC Authorisation Function) . 11
5.5 eP-CSCF (P-CSCF enhanced for WebRTC) . 11
5.6 eIMS-AGW (IMS Access Gateway enhanced for WebRTC) . 11
5A Data transport . 11
5A.1 General . 11
5A.2 UE . 11
5A.3 WWSF (WebRTC Web Server Function) . 11
5A.4 eP-CSCF (P-CSCF enhanced for WebRTC) . 12
5B Data framing and securing . 12
5B.1 General . 12
5B.2 UE . 12
5B.3 WWSF (WebRTC Web Server Function) . 12
5B.4 eP-CSCF (P-CSCF enhanced for WebRTC) . 12
5C Data formats . 13
5C.1 General . 13
5C.2 UE . 13
5C.3 WWSF (WebRTC Web Server Function) . 13
5C.4 eP-CSCF (P-CSCF enhanced for WebRTC) . 13
5D Connection management . 14
5D.1 General . 14
5D.2 UE . 14
5D.3 WWSF (WebRTC Web Server Function) . 14
5D.4 eP-CSCF (P-CSCF enhanced for WebRTC) . 14
5E Presentation and control . 14
5E.1 General . 14
5E.2 UE . 14
5E.3 WWSF (WebRTC Web Server Function) . 15
5E.4 eP-CSCF (P-CSCF enhanced for WebRTC) . 15
5F Local system support functions . 15
5F.1 General . 15
5F.2 UE . 15
5F.3 WWSF (WebRTC Web Server Function) . 15
5F.4 eP-CSCF (P-CSCF enhanced for WebRTC) . 15
6 Registration and authentication . 15
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6.1 General . 15
6.2 WIC (WebRTC IMS Client) . 16
6.2.1 WIC registration of individual Public User Identity using IMS authentication . 16
6.2.1.1 General . 16
6.2.1.2 W2 using SIP Digest credentials . 16
6.2.1.3 W2 using IMS-AKA . 16
6.2.2 WIC registration of individual public user identity based on web authentication . 17
6.2.3 WIC registration of individual public user identity from a pool of public user identities. 17
6.3 WWSF (WebRTC Web Server Function) and WAF (WebRTC Authorisation Function) . 17
6.3.1 WIC registration of individual public user identity using web credentials . 17
6.3.2 WIC registration of individual public user identity from a pool of public user identities. 18
6.4 eP-CSCF (P-CSCF enhanced for WebRTC) . 18
6.4.1 WIC registration of individual Public User Identity using IMS authentication . 18
6.4.1.1 Determination of IMS authentication mechanism . 18
6.4.1.2 W2 using SIP Digest credentials . 18
6.4.1.3 W2 using IMS-AKA . 19
6.4.2 WIC registration of individual public user identity using web credentials . 19
6.4.3 WIC registration of individual public user identity from a pool of public user identities. 20
6A Deregistration . 20
6A.1 General . 20
6A.2 WIC (WebRTC IMS Client) . 20
6A.3 eP-CSCF (P-CSCF enhanced for WebRTC) . 21
7 Call origination and termination . 21
7.1 General . 21
7.2 WIC (WebRTC IMS Client) . 21
7.2.1 General . 21
7.2.2 WIC originating call . 21
7.2.3 WIC terminating call . 22
7.2.4 WIC emergency call . 22
7.3 WWSF (WebRTC Web Server Function) . 22
7.4 eP-CSCF (P-CSCF enhanced for WebRTC) . 23
7.4.1 General . 23
7.4.2 WIC originating call . 23
7.4.3 WIC terminating call . 24
7.4.4 WIC emergency call . 24
7.4.5 Media optimization procedure . 25
7.4.5.1 WIC originating call . 25
7.4.5.2 WIC terminating call . 26
8 Data channel open and close . 28
8.1 General . 28
8.2 WIC (WebRTC IMS Client) . 29
8.2.1 General . 29
8.2.2 WIC originating call . 29
8.2.3 WIC terminating call . 29
8.3 WWSF (WebRTC Web Server Function) . 29
8.4 eP-CSCF (P-CSCF enhanced for WebRTC) . 29
8.4.1 General . 29
8.4.2 WIC originating call . 30
8.4.3 WIC terminating call . 30
9 Call modification . 31
10 IP multimedia application support in the IM CN subsystem using webRTC . 31
10.1 General . 31
10.2 Access to MMTel and supplementary services using webRTC . 31
10.2.1 General . 31
10.2.2 WIC (WebRTC IMS Client) . 31
10.2.2.1 SIP based protocol used by the WIC . 31
10.2.2.2 non-SIP based protocol used by the WIC . 31
10.2.3 WWSF (WebRTC Web Server Function) . 31
10.2.4 eP-CSCF (P-CSCF enhanced for WebRTC) . 31
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Annex A (informative): Example signalling flows . 33
A.1 Scope of signalling flows . 33
A.2 Void . 33
A.3 Signalling flows for registration . 33
A.3.1 Void . 33
A.3.2 WIC registration of individual public user identity based on web authentication . 33
A.3.3 Void . 35
A.4 Void . 35
A.5 Void . 35
Annex B (informative): Change history . 36
History . 38

ETSI
3GPP TS 24.371 version 13.11.0 Release 13 6 ETSI TS 124 371 V13.11.0 (2020-01)
Foreword
rd
This Technical Specification has been produced by the 3 Generation 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 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 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 incremented when editorial only changes have been incorporated in the document.
ETSI
3GPP TS 24.371 version 13.11.0 Release 13 7 ETSI TS 124 371 V13.11.0 (2020-01)
1 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-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 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 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 Transport 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 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 Subsystem (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 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 Gateway (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 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".
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3GPP TS 24.371 version 13.11.0 Release 13 8 ETSI TS 124 371 V13.11.0 (2020-01)
Editor's 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".
Editor's note: The above document cannot be formally referenced 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)".
Editor's 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 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] 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): "Negotiating Media Multiplexing
Using the Session Description Protocol (SDP)".
Editor's 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-sipcore-sip-token-authnz-02 (July 2019): "Third-Party Token-based Authentication and
Authorization for Session Initiation Protocol (SIP)".
Editor's 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".
Editor's note: The above document cannot be formally referenced until it is published as an RFC.
[31] Void
[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
Authentication and Key Agreement (AKA) Version-2".
[34] 3GPP TS 26.114: "IP multimedia subsystem (IMS); Multimedia telephony, Media handling and
interaction".
[35] RFC 7519 (May 2015): "JSON Web Token (JWT)".
[36] draft-ietf-mmusic-data-channel-sdpneg-12 (March 2017): "SDP-based Data Channel Negotiation".
Editor's note [WI: eWebRTCi_CT, CR#0044]: The above document cannot be formally referenced until it is
published as an RFC.
[37] draft-ietf-mmusic-msrp-usage-data-channel-13 (August 2019): "MSRP over Data Channels".
ETSI
3GPP TS 24.371 version 13.11.0 Release 13 9 ETSI TS 124 371 V13.11.0 (2020-01)
Editor's note [WI: eWebRTCi_CT, CR#0043]: The above document cannot be formally referenced until it is
published as an RFC.
[38] RFC 5761 (April 2010): "Multiplexing RTP Data and Control Packets on a Single Port".
[39] draft-ietf-ice-trickle-09 (April 2017): "Trickle ICE: Incremental Provisioning of Candidates for the
Interactive Connectivity Establishment (ICE) Protocol".
[40] RFC 5766 (April 2010): "Traversal Using Relays around NAT (TURN): Relay Extensions to
Session Traversal Utilities for NAT (STUN)".
3 Definitions and abbreviations
3.1 Definitions
For the purposes of the present document, the terms and definitions given in 3GPP TR 21.905 [1] and the following
apply. A term defined in the present document takes precedence over the definition of the same term, if any, in
3GPP TR 21.905 [1].
For the purposes of the present document, the following terms and definitions given in 3GPP TS 23.228 [4] annex U
apply:
P-CSCF enhanced for WebRTC (eP-CSCF)
WebRTC Authorization Function (WAF)
WebRTC IMS Client (WIC)
WebRTC Web Server Function (WWSF)

For the purposes of the present document, the following terms and definitions given in RFC 5245 [22] apply:
ICE Lite
Full ICE
Host ICE candidates
Editor's note: Terminology from draft-ietf-rtcweb-overview needs to be endorsed as part of the terminology of this
document. This document uses the terms "WebRTC device" which it is understood will be changed to
"non-WebRTC browser".
3.2 Abbreviations
For the purposes of the present document, the abbreviations given in 3GPP TR 21.905 [1] and the following apply. An
abbreviation defined in the present document takes precedence over the definition of the same abbreviation, if any, in
3GPP TR 21.905 [1].
CN Core Network
CSCF Call Session Control Function
DCEP Data Channel Establishment Protocol
eP-CSCF enhanced Proxy CSCF
IM IP Multimedia
IP Internet Protocol
WAF WebRTC Authorization Function
WebRTC Web Real-Time Communication
WWSF WebRTC Web Server Function
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4 Overview of WebRTC access to IMS
4.1 General
The relationship between functional entities for the interface at the W1 reference point, between the WWSF and the UE,
the interface at the W2 reference point, between the eP-CSCF and the UE, the interface at the W3 reference point,
between the UE and the eIMS-AGW, and the interface at the W4 reference point, between the WWSF and the WAF,
are defined in annex U of 3GPP TS 23.228 [4].
The relationship between the functional entities for interface at the Mw reference point, between the eP-CSCF and the
remainder of the IP multimedia core network subsystem, is defined in 3GPP TS 23.228 [4].
A number of appropriate mechanisms exist for signalling communication between the WIC and the eP-CSCF. Sucessful
use of a mechanism other than those specified in this document will require some form of prior agreement between the
operator of the WWSF and the operator of the eP-CSCF, as to the nature of the signalling mechanism that is to be
adopted, and therefore the interworking required at the eP-CSCF. The mechanism of prior agreement and the nature of
such agreement is not defined in this document.
A signalling transport mechanism for SIP is standardised in this release of this document, i.e. SIP over websockets (see
RFC 7118 [2]), but this is not a mechanism that has to be supported by all eP-CSCFs.
When SIP over websockets is used, it can be appropriate for the SIP used to conform to the definitions for SIP on the
Gm reference point as specified in 3GPP TS 24.229 [3]. Such a requirement is not mandatory, but where other SIP
mechanisms are used:
a) the usage will require some form of prior agreement with the operator of the eP-CSCF, as to the nature of the
signalling mechanism that is to be adopted; and
b) the SIP mechanisms will have to enable the eP-CSCF to conform to the SIP requirements over the Mw reference
point to the remainder of the IP multimedia core network subsystem as specified in 3GPP TS 24.229 [3].
SDP is used for the signalling session information between the WIC and the eP-CSCF. Such SDP conforms to
requirements for SDP on the Gm reference point.
5 Functional entities
5.1 General
5.2 WIC (WebRTC IMS Client)
A WebRTC IMS Client (WIC) establishing the service control signalling path over W2 interface, that is compliant with
this specification shall implement the role of WIC capabilities defined in subclause 6.2, subclause 7.2 and
subclause 8.2.
Where SIP over websockets is used, as specified in RFC 7118 [2], and no alternative SIP profiles have been agreed
between the operator of the eP-CSCF and the operator of the WWSF, then the SIP used by the WIC over the W2
reference point shall conform to the requirements for UE over the Gm reference point as specified in
3GPP TS 24.229 [3].
When the WebSocket protocol is used, the WIC shall act as a WebSocket Client, as defined in RFC 6455 [24].
The SDP used shall conform to the requirements for UE over the Gm reference point as specified in
3GPP TS 24.229 [3] and further specified in the present document.
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5.3 WWSF (WebRTC Web Server Function)
The WebRTC Web Server Function (WWSF) is the initial point of contact in the Web that controls access to the IMS
communications services for the WIC as specified in 3GPP TS 23.228 [4].
5.4 WAF (WebRTC Authorisation Function)
The WebRTC Authorisation Function (WAF) issues authorization tokens that are provided to the WIC via the WWSF
as specified in 3GPP TS 23.228 [4] and 3GPP TS 33.203 [9].
NOTE: The WWSF and the WAF realisations can be physically co-located or physically separate.
5.5 eP-CSCF (P-CSCF enhanced for WebRTC)
For the Mw reference point, the eP-CSCF shall conform to the requirements for the P-CSCF as specified in
3GPP TS 24.229 [3].
Where SIP over websockets is used, as specified in RFC 7118 [2], and no alternative SIP profile have been agreed
between the operator of the eP-CSCF and the operator of the WWSF, then the SIP used by the eP-CSCF over the W2
reference point shall conform to the requirements for P-CSCF over the Gm reference point as specified in
3GPP TS 24.229 [3].
The SDP used by the eP-CSCF over the W2 reference point shall conform to the requirements for UE over the Gm
reference point as specified in 3GPP TS 24.229 [3] and further specified in the present document.
5.6 eIMS-AGW (IMS Access Gateway enhanced for WebRTC)
The functionality of the eIMS-AGW is specified in 3GPP TS 23.228 [4] and in 3GPP TS 23.334 [13].
5A Data transport
5A.1 General
Data transport is the support of TCP, UDP and the means to securely set up connections between entities, as well as the
functions for deciding when to send data: Congestion management, bandwidth estimation and so on.
5A.2 UE
A UE supporting WebRTC shall support the WebRTC device functionality as specified in draft-ietf-rtcweb-
overview [30] clause 4, excluding requirements, if any, relating to specific audio and video codecs that are indirectly
referenced within the draft-ietf-rtcweb-overview [30] clause 4.
Editor's note: This clause references draft-ietf-rtcweb-transports-06 which uses the terminology "WebRTC
browser", "WebRTC endpoint" and "WebRTC device" for both ends of the transport. STUN and TURN
introduce further "server" and "client" terminology that has to be allowed for.
5A.3 WWSF (WebRTC Web Server Function)
There are no data transport requirements for the WWSF.
NOTE: Any application downloaded from the WWSF that requires data transport is expected to use it in
accordance with WebRTC device support of data transport.
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5A.4 eP-CSCF (P-CSCF enhanced for WebRTC)
The eP-CSCF and eIMS-AGW in conjunction shall support the WebRTC gateway functionality as specified in draft-
ietf-rtcweb-overview [30] clause 4, excluding requirements, if any, relating to specific audio and video codecs that are
indirectly referenced within the draft-ietf-rtcweb-overview [30] clause 4.
The eP-CSCF and eIMS-AGW in conjunction which is expected to be deployed where it can be reached with a static IP
address (as seen from the client) do not need to support full ICE; and therefore the eP-CSCF and eIMS-AGW in
conjunction may implement ICE-Lite only (specified in RFC 5245 [22]). ICE-Lite implementations do not send consent
checks, so the eP-CSCF and eIMS-AGW in conjunction may choose not to send consent checks too, but shall respond
to the received consent checks. The eP-CSCF and eIMS-AGW in conjunction with a static IP address is expected to not
need to hide its location, so the eP-CSCF and eIMS-AGW in conjunction do not need to support functionality for
operating only via a TURN server (specified in RFC 5766 [40]); instead the eP-CSCF and eIMS-AGW in conjunction
may choose to produce Host ICE candidates only.
If the eP-CSCF and eIMS-AGW in conjunction serve as a media relay into another RTP domain, the eP-CSCF and
eIMS-AGW may choose to support only features available in that network. The eP-CSCF and eIMS-AGW in
conjunction do not need to support Trickle Ice (specified in draft-ietf-ice-trickle [39]). However, the eP-CSCF and
eIMS-AGW in conjunction shall support DTLS-SRTP (specified in RFC 5764 [6]), since this is required for
interworking with WebRTC endpoints.
5B Data framing and securing
5B.1 General
Data framing RTP and other data formats that serve as containers, and their functions for data confidentiality and
integrity.
5B.2 UE
A UE supporting WebRTC shall support the WebRTC endpoint functionality as specified in draft-ietf-rtcweb-
overview [30] clause 5, excluding requirements, if any, relating to specific audio and video codecs that are indirectly
referenced within the draft-ietf-rtcweb-overview [30] clause 5.
Editor's note: This clause references RFC 3550 which uses the terminology "RTP implementation" for both ends of
the RTP. This clause references draft-ietf-rtcweb-rtp-usage which uses the terminology "WebRTC
endpoint" for both ends of the RTP, but also uses other terms e.g. "RTP endpoint".
5B.3 WWSF (WebRTC Web Server Function)
There are no data framing requirements for the WWSF.
NOTE: Any application downloaded from the WWSF that requires data framing is expected to use it in
accordance with WebRTC device support of data framing.
5B.4 eP-CSCF (P-CSCF enhanced for WebRTC)
The eP-CSCF and eIMS-AGW in conjunction shall support the WebRTC gateway functionality as specified in draft-
ietf-rtcweb-overview [30] clause 5 excluding requirements, if any, relating to specific audio and video codecs that are
indirectly referenced within the draft-ietf-rtcweb-overview [30] clause 5.
The eP-CSCF and eIMS-AGW in conjunction do not need to not support Bundle (specified in draft-ietf-mmusic-sdp-
bundle-negotiation [25]) and RTCP multiplexing (specified in RFC 5761 [38]) and any of the related RTP/ RTCP
extensions.
The eP-CSCF and eIMS-AGW in conjunction may choose to not support the Datachannel (specified in draft-ietf-
rtcweb-data-channel [16]).
ETSI
3GPP TS 24.371 version 13.11.0 Release 13 13 ETSI TS 124 371 V13.11.0 (2020-01)
5C Data formats
5C.1 General
Data format is codec specifications, format specifications and functionality specifications for the data passed between
systems. audio and video codecs, as well as formats for data and document sharing, belong in this category.
5C.2 UE
A UE supporting WebRTC shall support the WebRTC device functionality as specified in draft-ietf-rtcweb-
overview [30] clause 6, excluding requirements to implement specific audio and video codecs.
A UE offering WebRTC access to the IMS via GPRS IP-CAN (as described in 3GPP TS 24.229 [3], annex B), EPS IP-
CAN (as described in 3GPP TS 24.229 [3], annex L), or EPC via WLAN IP-CAN (as described in 3GPP TS 24.229 [3],
annex R) shall support the speech codecs according to 3GPP TS 26.114 [34] clause 5 and the front-end handling as
specified in 3GPP TS 26.114 [34] clause 11.
A UE offering WebRTC access to the IMS via xDSL, Fiber or Ethernet IP-CAN (as described in 3GPP TS 24.229 [3],
annex E) shall support the speech codecs according to 3GPP TS 26.114 [34] clause 18.
A UE supporting WebRTC access to the IMS via GPRS IP-CAN (as described in 3GPP TS 24.229 [3], annex B), EPS
IP-CAN (as described in 3GPP TS 24.229 [3], annex L), or EPC via WLAN IP-CAN (as described in
3GPP TS 24.229 [3], annex R) and supporting video communication shall support the video codecs according to
3GPP TS 26.114 [34].
A UE supporting WebRTC access to the IMS via xDSL, Fiber or Ethernet IP-CAN (as described in
3GPP TS 24.229 [3], annex E) and supporting video communication shall support the video codecs according to
3GPP TS 26.114 [34] clause 18.
Editor's note: This clause references draft-ietf-rtcweb-audio which uses the terminology "WebRTC clients" for both
ends of the RTP. The terminology used here needs to be aligned to cater for these inconsistencies.
NOTE: Media related requirements related to specific codecs, if any, to be supported by a UE supporting
WebRTC access to the IMS via IP-CAN other than GPRS IP-CAN, other than EPS IP-CAN, other than
EPC via WLAN IP-CAN and other than xDSL, Fiber or Ethernet IP-CAN are out of scope of this
specification.
5C.3 WWSF (WebRTC Web Server Function)
There are no data format requirements for the WWSF.
NOTE: Any application downloaded from the WWSF that requires data formats is expected to use it in
accordance with WebRTC device support of data formats.
5C.4 eP-CSCF (P-CSCF enhanced for WebRTC)
The eP-CSCF and eIMS-AGW in conjunction shall support the WebRTC gateway functionality as specified in draft-
ietf-rtcweb-overview [30] clause 6, excluding requirement
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