WebRTC is the latest wave of the communication technologies, allowing users to engage into audio and video communication session directly from the web browser, without the need to download a specialized client application. Various aspects of WebRTC technology are standardized in W3C and IETF drafts and recommendations.
As enterprise communications start taking advantage of WebRTC in-browser capability, there is a need to connect WebRTC clients to the enterprise communication solutions which are largely SIP based. Softil WebRTC Interconnect Solution was created specifically to address this need by utilizing best-of-breed protocol stacks and extending them to support required WebRTC transport and media capabilities.
To facilitate WebRTC interconnectivity, the following main RFCs have to be supported:
- WebSocket Transport (RFC 6455) (optional if WebRTC client doesn’t use SIP signaling)
- SIP over WebSocket (RFC 7118) (optional if WebRTC client doesn’t use SIP signaling)
- DTLS Transport (RFC 6347)
- RTP/SRTP with support for single port multiplexing (RFC 5761)
- ICE with support for single port multiplexing (RFC 5761)
Below you can see solution architecture for WebRTC Interconnect:
WebRTC Interconnect Solution Components
Standard SIP SDK with support of WebSocket transport.
Standard A-RTP SDK with SRTP add-on Module (required as all WebRTC media streams are encrypted), with support for DTLS transport and single port multiplexing.
Standard Firewall/NAT Traversal SDK with support for single port multiplexing.
Common Core WebRTC Transport Extensions Module
New module which extends Common Core functionality with support for WebRTC transports – WebSocket, DTLS and single port multiplexing.
For developers of:
- Media Servers/Media Gateways
- SIP-Based call centers
- SIP PBX
- Video-conferencing bridges/cloud
- Video/audio endpoints