Description
An RTC (Real-Time Communication) Application is a software entity that provides real-time conversational services — such as voice calling, video calling, and data collaboration — built on WebRTC technologies and conforming to the WebRTC-based application profile specified in 3GPP TS 26.113. The term encompasses two distinct deployment forms: a Native WebRTC Application, which is a standalone application (e.g., a mobile app or desktop client) that embeds the WebRTC framework directly, and a Web App, which runs inside a browser environment and leverages the browser's built-in WebRTC APIs. Both forms must comply with the same 3GPP profile to guarantee interoperability with network functions and with peer endpoints.
Architecturally, the RTC Application resides at the top of the UE-side stack. It consumes the services exposed by the RTC Client, which itself comprises an RTC Access Function and an RTC Media Session Handler. The application does not interact directly with the WebRTC framework's transport and media machinery; instead, it delegates session control and media handling to the RTC Client. This separation allows the application to focus on user experience and service logic while the RTC Client manages the complexity of session establishment, media negotiation, and network interworking.
The profile defined in TS 26.113 specifies the mandatory and optional WebRTC features that an RTC Application must support, including codecs, media transport mechanisms, security (DTLS/SRTP), and signaling interfaces. Compliance with this profile ensures that an RTC Application can interoperate with an RTC AS (Application Server) in the network and with other RTC endpoints, regardless of whether they are browser-based or native. This is critical for operator-hosted real-time services, where users may join sessions from heterogeneous devices and environments.
In the broader 3GPP ecosystem, the RTC Application is the user-facing component of the IMS-based or WebRTC-based real-time communication framework introduced in Release 19. It works in conjunction with the RTC Client on the UE and the RTC AS in the network, which hosts the Media Function and other server-side capabilities. The application may also interact with IMS core functions for service control, enabling integration with operator voice and video services, emergency calling, and supplementary services.
From a deployment perspective, RTC Applications can be delivered as operator-branded clients, third-party OTT applications that comply with the 3GPP profile, or web applications accessible via a URL. The 3GPP profile ensures that all these variants can participate in the same sessions and benefit from network capabilities such as QoS, policy control, and service continuity. This flexibility is a key enabler for operators seeking to offer WebRTC-based services that compete with, or complement, traditional circuit-switched and IMS-based telephony.
Purpose & Motivation
The RTC Application concept was created to standardize how real-time communication services are delivered over 3GPP networks using WebRTC technologies. Before Release 19, WebRTC deployments were largely proprietary, with each vendor or service provider defining its own profile, codecs, and signaling mechanisms. This fragmentation led to interoperability problems, inconsistent quality of experience, and limited ability to leverage network-level QoS and policy controls. The 3GPP profile for RTC Applications addresses these issues by defining a common set of requirements that all compliant applications must meet.
Another motivation was the desire to integrate WebRTC-based services more tightly with operator networks. Traditional OTT WebRTC applications operate as best-effort services, unable to request dedicated bearers or benefit from IMS service control. By defining the RTC Application and its interaction with the RTC Client and RTC AS, 3GPP enables operators to offer WebRTC services with guaranteed QoS, charging integration, and regulatory compliance (e.g., emergency services). This creates a level playing field where operator services can compete with OTT offerings on both quality and feature richness.
The RTC Application also serves as a bridge between the web ecosystem and the telecom ecosystem. Web developers can build applications using familiar WebRTC APIs, while operators can expose network capabilities through standardized interfaces. This convergence is expected to accelerate innovation in real-time communication services, enabling new use cases such as interactive collaboration, remote assistance, and immersive communication.
What changed in its specifications
The latest release-to-release comparison of specifications that define RTC Application. Each summary describes the whole specification, not only this term.
Added dynamic traffic marking, PDU Set handling, and multiplexed media flows for 5G real-time transport. Also added media transport and capability rules for XR sessions using WebRTC data channels and TS 26.119 profiles.
Full comparison in TS 26.113 →Evolution Across Releases
An overview written from the defining specifications, not a verified change history. For the exact text, open the specifications listed below.
Defining Specifications
3GPP specifications that define or reference RTC Application, with the latest known release. Sourced from the 3GPP document catalog — see methodology.
| Specification | Title | Release |
|---|---|---|
| TS 26.113 vj20 | Real-Time Media Communication Protocols and APIs | Rel-19 read |
| TS 26.506 vk00 | 5G Real-Time Media Communication Architecture | Rel-20 read |