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.
Key highlights
Ease of use
The WebRTC Interconnection solution is built by extending the existing toolkits, thus requiring only small incremental changes in the existing application code
Short time to market
New features and transports are supported automatically by the toolkits, thus reducing overall delivery time for the new applications
Feature rich
WebRTC Interconnect Solution supports a full set of WebRTC requirements, making it easy to implement best-in-class applications
Supported Standards
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)
Architecture
WebRTC Interconnect Solution Components
SIP Toolkit
Standard SIP SDK with support of WebSocket transport.
A-RTP Toolkit
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.
ICE Toolkit
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
- SBC
- Video-conferencing bridges/cloud
- Video/audio endpoint
Resources
On-Demand WebRTC Webinar
View our recorded webinar, WebRTC: Extending the Reach of IP Multimedia Communications, to learn how to implement next-generation enterprise communication solutions with WebRTC.
