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Content for  TR 26.930  Word version:  18.0.0

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1  Scopep. 15

The present document extends immersive Real-time Communication for WebRTC (iRTCW) and introduces a new concept called native WebRTC signalling.
This document includes following aspects:
  1. Analysis of gaps and required enhancements of terminal device and network architectures including additional functional entities (e.g., WebRTC Signalling Server, ICE-STUN Server, IMS Interworking Gateway, NNI Gateway).
  2. Impacts and possible enhancements for the WebRTC-based U-plane components in terms of adaptation, media handling, and cross-layer optimizations over 5G systems.
  3. C-Plane signalling protocol details (e.g., based on JSON) for the common WebRTC-based immersive RTC session management.
  4. Information elements in the C/U-Plane signal (including NNI) to enhance connectivity of media sessions with carrier assistance for WebRTC-based applications (including OTT applications).
  5. Minimal functional capabilities needed to support the enhancements identified in 2, 3 and 4 (including transport, NAT-traversal, and XR conferencing), state transitions, and typical call flows.
  6. Consideration of collaboration formation with other WGs in 3GPP and SDOs including IETF and W3C.
  7. Enhancements for E2E QoS realizations over 5G systems for communications between MNOs and WebRTC clients operating over 5G access or non-5G access (e.g., Wi-Fi) using WebRTC-based transport. This also includes communication between WebRTC clients operating on tethering/tethered devices.
  8. Security aspects and rate adaptation in tethered use cases (including coordination of Uu and non-3GPP access).
One of the main scopes of this document is to study the enhancements of RTC network for supporting collaboration scenario 4 specified in TS 26.506. Collaboration scenario 4 enables inter-operable WebRTC services, which is extended collaboration scenario 3 with functions to support MNO to MNO inter-operability. This collaboration scenario 4 could include the roaming case, but this case is outside the scope of this document.
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2  Referencesp. 15

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]
TR 21.905: "Vocabulary for 3GPP Specifications".
[2]
TS 23.222: "Functional architecture and information flows to support Common API Framework for 3GPP Northbound APIs; Stage 2".
[3]
TS 23.228: "IP Multimedia Subsystem (IMS); Stage 2".
[4]
TS 23.501: "System architecture for the 5G System (5GS); Stage 2".
[5]
TS 23.502: "Procedures for the 5G System (5GS); Stage 2".
[5A]
TS 23.503: "Policy and charging control framework for the 5G System (5GS); Stage 2".
[6]
TS 23.548: "5G System Enhancements for Edge Computing; Stage 2".
[7]
TS 23.558: "Architecture for enabling Edge Applications".
[8]
TS 24.229: "IP multimedia call control protocol based on Session Initiation Protocol (SIP) and Session Description Protocol (SDP); Stage 3".
[9]
TS 24.371: "Web Real-Time Communications (WebRTC) access to the IP Multimedia (IM) Core Network (CN) subsystem (IMS); Stage 3; Protocol specification".
[10]
TS 26.113: "Real-Time Media Communication; Protocols and APIs".
→ to date, still a draft
[11]
TS 26.114: "IP Multimedia Subsystem (IMS); Multimedia Telephony; Media handling and interaction".
[12]
TS 26.506: "5G Real-time Media Communication Architecture (Stage 2)".
[13]
TS 26.510: "Media delivery; interactions and APIs for provisioning and media session handling".
→ to date, still a draft
[14]
TS 26.512: "5G Media Streaming (5GMS); Protocols".
[14A]
TR 26.806: "Study on Tethering AR Glasses - Architectures, QoS and Media Aspects".
[15]
TS 29.162: "Interworking between the IM CN subsystem and IP networks".
[16]
TS 29.165: "Inter-IMS Network to Network Interface (NNI)".
[17]
TS 29.501: "5G System; Principles and Guidelines for Services Definition; Stage 3".
[18]
TS 33.501: "Security architecture and procedures for 5G system".
[19]
RFC 768:  "User Datagram Protocol".
[20]
RFC 791:  "Internet Protocol".
[21]
RFC 3261:  "SIP: Session Initiation Protocol".
[21A]
RFC 3264:  "An Offer/Answer Model with the Session Description Protocol (SDP)".
[22]
RFC 3324:  "Short Term Requirements for Network Asserted Identity".
[23]
RFC 3325:  "Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks".
[24]
RFC 3489:  "STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)".
[24A]
RFC 3629:  "UTF-8, a transformation format of ISO 10646".
[24B]
RFC 3986:  "Uniform Resource Identifier (URI): Generic Syntax".
[24C]
RFC 5761:  "Multiplexing RTP Data and Control Packets on a Single Port".
[24D]
RFC 5764:  "Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)".
[25]
RFC 5888:  "The Session Description Protocol (SDP) Grouping Framework".
[26]
RFC 6120:  "Extensible Messaging and Presence Protocol (XMPP): Core".
[27]
RFC 6455:  "The WebSocket Protocol".
[28]
RFC 6598:  "IANA-Reserved IPv4 Prefix for Shared Address Space".
[29]
RFC 6749:  "The OAuth 2.0 Authorization Framework".
[30]
RFC 7092:  "A Taxonomy of Session Initiation Protocol (SIP) Back-to-Back User Agents".
[31]
RFC 7362:  "Latching: Hosted NAT Traversal (HNT) for Media in Real-Time Communication".
[32]
RFC 7635:  "Session Traversal Utilities for NAT (STUN) Extension for Third-Party Authorization".
[32A]
RFC 7807:  "Problem Details for HTTP APIs".
[32B]
RFC 8035:  "Session Description Protocol (SDP) Offer/Answer Clarifications for RTP/RTCP Multiplexing".
[33]
RFC 8200:  "Internet Protocol, Version 6 (IPv6) Specification".
[34]
RFC 8224:  "Authenticated Identity Management in the Session Initiation Protocol (SIP)".
[35]
RFC 8225:  "PASSporT: Personal Assertion Token".
[36]
RFC 8259:  "The JavaScript Object Notation (JSON) Data Interchange Format".
[37]
RFC 8285:  "A General Mechanism for RTP Header Extensions".
[38]
RFC 8441:  "Bootstrapping WebSockets with HTTP/2".
[39]
RFC 8445:  "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal".
[40]
RFC 8446:  "The Transport Layer Security (TLS) Protocol Version 1.3".
[41]
RFC 8489:  "Session Traversal Utilities for NAT (STUN)".
[42]
RFC 8656:  "Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN)".
[43]
RFC 8588:  "Personal Assertion Token (PaSSporT) Extension for Signature-based Handling of Asserted information using toKENs (SHAKEN)".
[44]
RFC 8825:  "Overview: Real-Time Protocols for Browser-Based Applications".
[45]
RFC 8826:  "Security Considerations for WebRTC".
[46]
RFC 8827:  "WebRTC Security Architecture".
[47]
RFC 8829:  "JavaScript Session Establishment Protocol (JSEP)"
[48]
RFC 8835:  "Transports for WebRTC".
[49]
RFC 8838:  "Trickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) Protocol".
[50]
RFC 8839:  "Negotiating Media Multiplexing Using the Session Description Protocol (SDP)".
[50A]
RFC 8841:  "Session Description Protocol (SDP) Offer/Answer Procedures for Stream Control Transmission Protocol (SCTP) over Datagram Transport Layer Security (DTLS) Transport".
[51]
RFC 8859:  "A Framework for Session Description Protocol (SDP) Attributes When Multiplexing".
[52]
RFC 8864:  "Negotiation Data Channels Using the Session Description Protocol (SDP)"
[53]
RFC 8866:  "SDP: Session Description Protocol".
[54]
RFC 9110:  "HTTP Semantics".
[55]
RFC 9111:  "HTTP Caching".
[56]
RFC 9112:  "HTTP/1.1".
[57]
RFC 9113:  "HTTP/2".
[58]
RFC 9114:  "HTTP/3".
[59]
RFC 9143:  "Negotiating Media Multiplexing Using the Session Description Protocol (SDP)".
[60]
RFC 9220:  "Bootstrapping WebSockets with HTTP/3".
[61]
RFC 9293:  "Transmission Control Protocol (TCP)".
[62]
RFC 9457:  "Problem Details for HTTP APIs".
[63]
AsyncAPI Initiative "AsyncAPI Specification v3.0.0" https://github.com/asyncapi/spec/releases/tag/v3.0.0
[64]
OpenAPI Initiative "OpenAPI Specification v3.0.0" https://spec.openapis.org/oas/v3.0.0
[65]
W3C Proposed Recommendation, "WebRTC 1.0: Real-time Communication Between Browsers", https://www.w3.org/TR/webrtc/.
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3  Definitions of terms, symbols and abbreviationsp. 18

3.1  Termsp. 18

For the purposes of the present document, the terms given in TR 21.905 and the following apply. A term defined in the present document takes precedence over the definition of the same term, if any, in TR 21.905.
Content Provider (CP):
An RTC application provider who provide RTC services partially using operator's functionalities. CP connects to the operator network via UNI (RTC-4s/4m) as a WebRTC endpoint in order to use the operator's MF and WSF for the service specific content delivery.
Originating RTC network:
A network which sends Media session set up request to another network (i.e., other operator network or SP network) over the NNI.
Originating UE:
A RESPECT client on the UE which sends Media session set up request over the UNI.
RESPECT client:
A RESPECT endpoint which acts as an WebSocket client.
RESPECT endpoint:
A UE and RTC AS functionality supporting RESPECT protocol. RTC Aware Application on the UE or CP acting as UE, RTC AS for C-Plane signalling (i.e., WSF and IWF) that comply with RESPECT protocol are RESPECT endpoints. When distinction between UE/CP (acting as UE) and RTC AS is required, RESPECT endpoint (UE) or RESPECT endpoint (AS) is used respectively in this document.
RESPECT server:
A RESPECT endpoint which acts as an WebSocket server.
RTC network:
A DN dedicated to RTC. This network is compliant with the trusted DN of the RTC architecture defined TS 26.506.
RTC resource:
A media which a media session is connected to. Media service such as conference room, media content for distribution and RESPECT endpoint are example of RTC resource.
RTC user:
An RTC service user who connects to the RTC service by using RESPECT endpoint.
Service provider (SP):
An RTC application provider who provides RTC services with its own network. SP owns resources and functionalities within its network, and SP's network is connected to operator network via NNI.
Terminating RTC network:
A network which receives Media session set up request from another network (i.e., other operator network or SP network) over the NNI.
Terminating UE:
A RESPECT client on the UE which receives Media session set up request over the UNI.
User Equipment (UE):
It indicates the user equipment and servers acting as user equipment such as a content server of a content provider. User equipment includes an RESPECT endpoint.
For the purposes of the present document, the following terms and definitions given in TS 29.165 apply:
Calling number verification using signature verification and attestation information
For the purposes of the present document, the following terms and definitions given in RFC 6455 apply:
WebSocket client
WebSocket server
For the purposes of the present document, the following terms and definitions given in RFC 8825 apply:
WebRTC Browser (also called a "WebRTC User Agent" or "WebRTC UA")
WebRTC Endpoint
WebRTC Non-Browser
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3.2  Symbolsp. 19

For the purposes of the present document, the following symbols apply:
Rs-u
Reference Point between a WSF and a UE.
Rs-i
Reference Point between a WSF and another WSF in the same network (DN) or between a WSF and a WNSGF.
Rs-a
Reference Point between a WSF and a CSF.
Rs-n
Reference Point between a WNSGF and another WNSGF in an external network.
Rm-u
Reference Point between a WMCF and a UE.
Rm-i
Reference Point between a WMCF and another WMCF in the same network (DN) or between a WMCF and a WNMGF.
Rm-n
Reference Point between a WNMGF and another WNMGF in an external network.
Mc-i
Reference Point between a WSF and a WMCF.
Mc-r
Reference Point between a WNSGF and a WNMGF.
Rh-u
Reference Point between a CSF and UE. This reference point is used for providing CSF functionalities (e.g., application usage assistance such as downloading an application) to UE.
Rh-n
Reference Point between a CSF and Application service provider. This reference point is used for interaction between CSF and Application service provider for media session set up related interaction.
N5
Reference Point between a WSF and PCF.
RTC-X
Reference Point between a ASWF and application service provider.
RTC-4
Reference Point between an RTC network and a UE.
RTC-4m
Reference Point between a MF and a UE or between as ASWF and a UE.
RTC-4s
Reference Point between a WSF and a UE.
RTC-Y
Reference Point between two different operator's RTC networks or between an operator's RTC network and service provider's network.
RTC-Ym
Reference Point between a TGF and a TGF in another RTC network or a service provider's network.
RTC-Ys
Reference Point between a IWF and a IWF in another RTC network or a service provider's network.
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3.3  Abbreviationsp. 19

For the purposes of the present document, the abbreviations given in TR 21.905 and the following apply. An abbreviation defined in the present document takes precedence over the definition of the same abbreviation, if any, in TR 21.905.
ASWF
Application Supporting Web Function
CP
Content Provider
CSF
Conference Supporting Function
IBCF
Interconnection Border Control Function
IWF
Inter-working Function
MDFC
Media Data Forwarding Control
MF
Media Function
NNI
Network to Network Interface
RESPECT
REaltime&REality media Setup Protocol, Extensible and CompacT
SEPP
Security Edge Protection Proxy
SP
Service Provider
TGF
Transport Gateway Function
UNI
User to Network Interface
WMCF
WebRTC Media Centre Function
WNMGF
WebRTC NNI Media Gateway Function
WNSGF
WebRTC NNI Signalling Gateway Function
WSF
WebRTC Signalling Function
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