WebRTC Interconnect Solution

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 DTBU 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.

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 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

Below you can see solution architecture for DTBU WebRTC Interconnect:

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 endpoints

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.
View webinar

White Paper: Better customer service and easier business communcations
This white paper explains how software vendors can help their business customers deliver a better service while collaborating more efficiently by adding WebRTC support to their unified communication products.
Download White Paper