Real-Time Communication Endpoint

Description

An RTC endpoint is a functional entity that implements the WebRTC Framework and is capable of participating in a real-time communication session. It serves as the media and data termination point for WebRTC-based sessions, handling the transport, encryption, and processing of media streams and data channels. RTC endpoints are deployed in two locations: within the RTC Access Function of a UE (the client-side endpoint) and within the Media Function of an RTC AS (the network-side endpoint). This dual deployment enables peer-to-peer or network-relayed media paths, depending on the session architecture and network policies.

The RTC endpoint incorporates an instance of the WebRTC Framework, which includes the necessary protocols and APIs for real-time communication. This includes ICE for NAT traversal, DTLS for secure key exchange, SRTP for media encryption, and SCTP over DTLS for data channels. The endpoint also implements the media codecs and processing functions required by the 3GPP WebRTC profile, such as Opus for audio and VP8/VP9/H.264 for video. By standardizing these capabilities, 3GPP ensures that endpoints from different vendors can interoperate seamlessly.

In a typical session, the UE-side RTC endpoint (in the RTC Access Function) establishes a media path with the network-side RTC endpoint (in the Media Function of the RTC AS). This path may be direct (peer-to-peer) or relayed through a TURN server, depending on network topology and NAT conditions. The network-side endpoint can also serve as a media gateway or mixer, enabling multiparty conferences, media transcoding, and interworking with non-WebRTC endpoints (e.g., SIP-based UEs). This flexibility is a key advantage of the RTC endpoint architecture.

The RTC endpoint interacts with the RTC Media Session Handler for session control. While the endpoint handles media transport, the Media Session Handler manages signaling (e.g., SIP, HTTP) to establish, modify, and tear down sessions. This separation of concerns allows the endpoint to focus on media processing while the session handler deals with service logic and network signaling. The endpoint may also expose APIs to the RTC Application (via the RTC Client) for media control, such as muting audio or switching video sources.

From a deployment perspective, the RTC endpoint is implemented as software within the UE's RTC Access Function and within the network's Media Function. It must comply with the 3GPP profile defined in TS 26.113 and the procedures in TS 26.506. The endpoint's capabilities, such as supported codecs and transport protocols, are negotiated during session setup to ensure compatibility between peers. This negotiation is critical for interoperability and for delivering a consistent user experience across diverse devices and networks.

Purpose & Motivation

The RTC endpoint concept was created to standardize the media and data termination points for WebRTC-based sessions in 3GPP networks. Before Release 19, WebRTC endpoints were defined by the W3C and IETF, but there was no 3GPP-specific profile to ensure interoperability with operator networks and services. The RTC endpoint fills this gap by defining a 3GPP-compliant WebRTC endpoint that can be deployed in both UEs and network servers, enabling seamless interworking and consistent service delivery.

A primary motivation was the need for network-side media processing. In many scenarios, peer-to-peer media paths are not feasible due to NAT/firewall traversal issues, or they are undesirable due to policy or charging requirements. The network-side RTC endpoint, deployed in the Media Function of the RTC AS, provides a reliable anchor point for media, enabling features such as lawful intercept, transcoding, and multiparty conferencing. It also allows operators to apply QoS and policy controls to the media stream, ensuring carrier-grade quality.

Another motivation was interoperability with legacy services. The RTC endpoint can interwork with non-WebRTC endpoints (e.g., SIP-based UEs) through media gateways, enabling operators to bridge their traditional IMS services with new WebRTC services. This is essential for a smooth transition and for protecting existing investments. The RTC endpoint thus serves as a key enabler for the convergence of web and telecom real-time communication.