WebRTC Service - Features & Benefits
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The WebRTC Service streamlines WebRTC-based communication, providing seamless integration with legacy SIP systems, the PSTN, and additional iotcomms.io services -- all without the need for complex infrastructure management.
Seamless WebRTC-to-SIP Connectivity
Legacy Compatibility
The WebRTC Service bridges modern WebRTC applications with SIP systems, including legacy infrastructures, allowing businesses to expand capabilities without costly overhauls.
Signaling Protocol Conversion
Converts SIP over WebSocket to SIP/TLS, SIP/TCP, or SIP/UDP, ensuring compatibility across protocols.
Scalable Media Handling
Media Conversion
Translates WebRTC DTLS media encryption to SRTP or RTP, providing cross-platform media compatibility for consistent audio and video quality.
Broad Codec Support
With codec options like OPUS, G.711, and H.264, the service ensures high-quality media interoperability across devices.
Integration with PSTN and On-Premise Systems
PSTN Connectivity
Connects WebRTC clients to the Public Switched Telephone Network (PSTN) using the iotcomms.io SIP Core Service, supporting calls to and from traditional telephony networks.
On-Premise System Support
Routes WebRTC requests to specific SIP systems using configurable route headers, extending WebRTC functionality to internal networks and established SIP setups.
Enhanced Interactive Applications
Integration with SIP Mediaserver Service
Provides interactive voice and video services ideal for contact center applications, such as IVR and agent assistance directly from a browser.
Interactive Features
Supports real-time customer engagement, including voice and video call management, for users on web-based applications.
Open Standards for Flexibility
No Vendor Lock-In
By supporting open standards like SIP over WebSocket, the WebRTC Service allows businesses to avoid vendor lock-in and ensure flexibility for future integration needs.
Signaling Protocol Conversion
Converts WebRTC SIP over WebSocket to SIP/TLS, SIP/TCP, or SIP/UDP, bridging WebRTC with traditional SIP systems.
Media Conversion for Cross-Platform Compatibility
Translates DTLS media encryption to SRTP or RTP, enabling secure and compatible media flows between WebRTC and SIP endpoints.
Intuitive Log View
Provides a user-friendly log view for tracking SIP signaling and WebRTC events in real time. Designed for massive traffic volume environments, this feature supports quick troubleshooting and monitoring without deep SIP expertise.
On-Premise Integration
Routes WebRTC requests to specified on-premise SIP systems using configured route headers, ensuring compatibility with internal SIP networks.
APIs and Web UI for Device and Service Provisioning
Includes comprehensive APIs and a web interface for easy setup, management, and provisioning of WebRTC and SIP devices, simplifying configuration and scaling for new deployments.
Verified with Open Source Leading sip.js Client SDK
Fully compatible with the sip.js client SDK, a leading open-source solution for WebRTC applications, ensuring smooth integration with WebRTC applications and flexibility for future development.
JSON Web Token (JWT) Authentication Support
Supports JWT authentication to eliminate the need for managing individual SIP credentials, providing a secure and scalable method of authenticating WebRTC clients in SIP environments.
With these Key Benefits and Key Features, the WebRTC Service offers a versatile, secure, and scalable communication solution, enabling enhanced browser-based communication with minimal complexity.