WebRTC Service - How It Works

The WebRTC Service offers a comprehensive suite of functions designed to seamlessly integrate WebRTC-based voice and video communication with both customer SIP systems and iotcomms.io cloud services.

Built on open standards, the service prevents vendor lock-in and ensures flexibility and interoperability for developers and application owners.

The WebRTC Service is available as a SaaS offer, in the cloud or in a hybrid cloud deployment model. The hybrid cloud deployment option lets you run the service within your own data center using the Hybrid Enabler Service to meet highest privacy standards while the iotcomms.io team manages and monitors the service for best operational efficiency.

The key functions of the WebRTC Service are:

The WebRTC Service is built on open standards, including SIP over WebSocket, to prevent vendor lock-in and ensure compatibility across diverse WebRTC and SIP environments. This standards-based approach allows for broad interoperability, giving developers flexibility in their application ecosystems.

The service bridges modern WebRTC applications with legacy SIP systems by converting SIP over WebSocket to SIP/TLS, SIP/TCP, or SIP/UDP. This function enables seamless communication between WebRTC clients and traditional SIP platforms without modifications to client applications, extending WebRTC capabilities to legacy systems.

To ensure secure interoperability between WebRTC and SIP devices, the WebRTC Service converts WebRTC's DTLS media encryption to SRTP or RTP as needed. This function supports consistent, secure media flow across platforms with different encryption requirements, enabling smooth, high-quality voice and video communication.

The WebRTC Service integrates WebRTC clients with the Public Switched Telephone Network (PSTN) via iotcomms.io's SIP Core Service trunk connectivity. This function allows WebRTC users to place and receive calls to and from traditional phone systems, expanding communication options for WebRTC applications.

The WebRTC Service integrates with existing on-premise infrastructure by providing a routing function that uses configured route headers to direct WebRTC requests to specific SIP systems within a customer's network. This feature ensures compatibility with established backend SIP configurations, allowing businesses to seamlessly extend WebRTC functionality to their internal SIP systems.

The WebRTC Service integrates with the iotcomms.io SIP Mediaserver Service, enhancing interactive voice and video capabilities to for example contact center applications. This integration supports use cases where users call directly from a web browser supporting WebRTC to engage in interactive communication flows -- such as IVR or agent-assisted sessions.

With these key functions, the iotcomms.io WebRTC Service provides a scalable, secure, and interoperable solution for applications that require WebRTC access to SIP infrastructure and iotcomms.io services, making it ideal for real-time communications in diverse business and application environments.