RTC Access Function

Description

The RTC Access Function (RTC AF) is a key component of the 3GPP Real-Time Communication (RTC) architecture, defined in specifications such as TS 26.113 and TS 26.506. It serves as the gateway between the 3GPP network and external RTC endpoints, enabling real-time media and signaling exchange. The RTC AF includes an instance of the WebRTC Framework, leveraging WebRTC technology to facilitate interoperable real-time communication. It interacts with RTC endpoints via reference points RTC-4m (for media) and RTC-12 (for media as well), and exchanges signaling messages with the WebRTC Signalling Function via RTC-4s. Additionally, it exposes client APIs to the RTC Application (via RTC-7) and to the RTC Media Session Handler (via RTC-11), allowing applications to utilize its functions.

Architecturally, the RTC AF is situated at the edge of the 3GPP network, acting as an application function (AF) that can interact with the core network to request QoS and policy treatment for media flows. It may be deployed as part of the operator's network or as a third-party service. The RTC AF is responsible for establishing, modifying, and terminating media sessions with RTC endpoints, which can be UEs or other RTC clients. It handles media negotiation, codec selection, and media transport (e.g., RTP/RTCP over UDP). The signaling with the WebRTC Signalling Function ensures that session control messages (e.g., SDP offers/answers) are properly routed and processed.

The RTC AF exposes APIs that allow RTC applications to create and manage media sessions without dealing with the underlying complexity of WebRTC and network signaling. These APIs may include methods for session initiation, media control, and event subscription. The RTC Media Session Handler, which interacts via RTC-11, is responsible for managing media sessions on the client side, possibly handling media adaptation and QoS feedback. The division of labor between the RTC AF and the Media Session Handler ensures a clear separation between network-side functions and client-side functions.

In terms of protocols, the RTC AF may use SIP or other signaling protocols internally, but the external interfaces are based on WebRTC standards. The reference points RTC-4m and RTC-12 are likely used for media transport, possibly using SRTP for security. The RTC AF may also interact with the 5G core network's PCF and SMF to establish QoS flows for media, ensuring adequate quality of service. This integration with the 5G QoS framework is essential for delivering carrier-grade real-time communication.

The RTC AF is a critical enabler for operators to offer RTC services, such as voice and video calling, that are integrated with 5G network capabilities. It allows for a standardized approach to real-time communication over 5G, promoting interoperability and enabling new use cases like cloud gaming, AR/VR communication, and mission-critical push-to-talk.

Purpose & Motivation

The RTC Access Function was introduced to address the need for a standardized, interoperable real-time communication framework within 3GPP networks. Prior to its definition, real-time communication services (e.g., VoLTE, VoNR) were tightly coupled to IMS, which, while robust, was complex and not easily extensible to web-based or third-party applications. The rise of WebRTC as a de facto standard for real-time communication in web browsers and mobile apps created an opportunity for 3GPP to leverage WebRTC technology to enable a new generation of RTC services that are integrated with 5G capabilities. The RTC AF fills the gap by providing a network function that bridges WebRTC endpoints with the 5G core, ensuring that media can be transported with appropriate QoS and policy control.

The primary problem solved by the RTC AF is the lack of a unified architecture for real-time communication that can be used by both operators and third-party service providers. By exposing APIs and using WebRTC, it lowers the barrier to entry for developers to create RTC applications that can take advantage of 5G features like network slicing, edge computing, and QoS. It also allows operators to monetize their network assets by offering enhanced RTC services.

Moreover, the RTC AF enables better integration with IMS and other 3GPP services. For instance, it can interwork with IMS to provide seamless voice calling between WebRTC clients and traditional UEs. This interworking is facilitated by the WebRTC Signalling Function, which translates between WebRTC signaling and SIP. The RTC AF thus serves as a key enabler for the convergence of web and telecom technologies, allowing operators to evolve their service offerings in line with industry trends.