WebRTC Web Server Function

Description

The WebRTC Web Server Function (WWSF) is a core network element defined in 3GPP specifications, starting from Release 12, to bridge Web Real-Time Communication (WebRTC) clients with traditional 3GPP IP Multimedia Subsystem (IMS) networks. WebRTC is a collection of APIs and protocols that enable real-time audio, video, and data communication directly within web browsers without requiring plugins. However, WebRTC uses different signaling and media protocols compared to IMS, which relies on SIP (Session Initiation Protocol) and related standards. The WWSF resolves this incompatibility by acting as a gateway or adaptation function, translating WebRTC signaling (often over WebSocket or HTTP) into SIP messages and vice versa, and handling media interworking if necessary.

Architecturally, the WWSF is positioned within the service layer of the 3GPP network, often interfacing with the IMS core components such as the Proxy-CSCF (P-CSCF), Serving-CSCF (S-CSCF), and Application Servers (AS). It is specified across multiple 3GPP technical specifications (e.g., TS 23.334, TS 29.334) that detail its interfaces, procedures, and integration points. The WWSF typically includes functional modules for signaling conversion, security handling (e.g., authentication and encryption), and media processing. It may interact with other functions like the WebRTC Interworking Function (WIF) for deeper protocol adaptation. The WWSF enables WebRTC clients—running in browsers or native apps—to register with the IMS network, initiate sessions, and use telecom services as if they were traditional IMS endpoints.

In operation, when a WebRTC client attempts to access a service (e.g., make a voice call), it connects to the WWSF via secure web protocols (HTTPS/WebSocket). The WWSF authenticates the user, often leveraging 3GPP credentials like IMSI or GBA (Generic Bootstrapping Architecture), and converts the WebRTC signaling (using JavaScript Session Establishment Protocol or similar) into SIP messages forwarded to the IMS core. For media, the WWSF may act as a back-to-back user agent (B2BUA), transcoding between WebRTC's SRTP (Secure Real-time Transport Protocol) and IMS media streams, or it may facilitate direct peer-to-peer media paths after negotiation. This allows seamless communication between WebRTC users and legacy IMS subscribers, supporting services like VoLTE, video calls, and messaging.

Key components of the WWSF include the signaling gateway, media gateway controller, and security modules. Its role is critical for extending 3GPP services to the web ecosystem, enabling operators to offer rich communication services without requiring users to install dedicated apps. By standardizing the WWSF, 3GPP ensures interoperability between web-based real-time communication and mobile networks, supporting innovation in areas like WebRTC-enabled customer service, collaborative applications, and IoT communications. The function is designed to be scalable and secure, adhering to 3GPP's regulatory and privacy requirements.

Purpose & Motivation

The WWSF was introduced in 3GPP Release 12 to address the convergence of web technologies and telecom networks, specifically the rise of WebRTC as a standard for browser-based real-time communication. Before its definition, WebRTC applications operated in isolation from IMS networks, limiting operators' ability to integrate web services with their core telecom offerings. This created silos where web users could not easily access features like carrier-grade voice, SMS, or network-based authentication. The WWSF solves this by providing a standardized interworking function, allowing operators to extend IMS services to any WebRTC-capable device, thereby increasing service reach and reducing dependency on native apps.

Historically, real-time communication over the web relied on proprietary plugins or external software, which posed security risks and compatibility issues. WebRTC emerged as an open standard to enable plugin-free communication, but its adoption in telecom was hindered by protocol mismatches with IMS. The WWSF bridges this gap, motivated by operator demands to leverage existing IMS investments while embracing web innovation. It addresses limitations such as disparate signaling protocols (WebRTC uses HTTP/WebSocket-based signaling versus SIP in IMS) and different security models (e.g., DTLS-SRTP in WebRTC vs. IPSec in IMS).

By defining the WWSF, 3GPP enables new business models, such as web-based calling services, enriched customer interactions, and seamless mobility between web and mobile clients. It also supports regulatory compliance by integrating with 3GPP's lawful interception and emergency service frameworks. In essence, the WWSF exists to future-proof mobile networks, ensuring they can interoperate with the evolving web ecosystem and provide consistent, secure real-time communication services across all access types, from 4G to 5G and beyond.

Key Features

• Interworks WebRTC signaling with IMS/SIP protocols for seamless session establishment
• Supports authentication of WebRTC clients using 3GPP credentials like IMSI or GBA
• Enables media interworking between WebRTC's SRTP and IMS media streams when required
• Provides secure communication via TLS/DTLS and integration with 3GPP security frameworks
• Facilitates service exposure, allowing WebRTC applications to access network APIs and capabilities
• Standardized across multiple 3GPP specs to ensure interoperability and consistent implementation

Evolution Across Releases

Defining Specifications