WebRTC and Enterprise Integration

Post by Kevin Wiseman

WebRTC is a great initiative for enabling browser-to-browser communication without the need for any plug-ins. Enterprises thrive on communications and have extensive telecommunications systems in place to help employees, partners and customers collaborate effectively. WebRTC provides enterprises with innovative ways of extending communication lines in new and imaginative ways to reach and support customers as well as constrain communication costs

As designed and specified, WebRTC is intended to communication enable Web based services such as enabling members of a social network to communicate with one another. A consequence of this remit is that the signalling and who can be contacted is left to the implementing service provider to define. This allows the service provider to back the offered communications service with whatever signalling mechanism makes sense to the service being provided.

Session Initiation Protocol (SIP) is the predominant signalling protocol used by enterprises and telco service providers. For enterprises to integrate WebRTC seamlessly into their existing communication system requires the use of a WebRTC service implementation that provides a gateway into the enterprises SIP infrastructure. To achieve this there are two approaches a) SIP enable the browser typically by use of a script based SIP stack or b) use of a SIP gateway.

SIP in the browser has a number of issues namely reliance on SBCs and other network elements being able to accept SIP over web sockets, handle all the specific NAT traversal requirements of WebRTC specific SDP along with complicated and bloated browser side scripts. Ultimately you want to enable web based communications that are easy to add to any web page, don’t require web developers to be SIP developers and don’t introduce any unnecessary vulnerability.

The alternative is a SIP gateway. Thrupoint’s Fusion Web product provides a SIP gateway for WebRTC based services. It provides seamless integration of WebRTC and SIP, rendering WebRTC enabled browsers as voice and video communications devices within a SIP based infrastructure. Your web developers remain focused on building great web sites invoking simple yet powerful script functions to make and control calls. The browser to gateway communication uses HTTP based technologies thus eliminating the need for any changes to existing infrastructure beyond deployment of the gateway.

The goal of WebRTC is not to replace SIP or any other existing communications infrastructure. WebRTC deals with a very specific problem – voice and video communications to/from the browser without the use of any plug-ins. Using SIP to backend your WebRTC service provides you with capabilities that are greater than the sum of their parts.