WebRTC Will Drive SIP Trunking of PBXs, UC and Call Center Solutions

Ask the Expert

WebRTC Will Drive SIP Trunking of PBXs, UC and Call Center Solutions

By TMCnet Special Guest
Karl Stahl, Chairman and CTO at Ingate Systems AB
  |  September 23, 2014

WebRTC is high quality multimedia real-time communication in the web browser. Although the IETF and W3C standards still have to be completed, Google and Mozilla (News - Alert) already support WebRTC in their Chrome and Firefox web browsers.

Using a WebRTC browser with the PBX or with UC solutions requires a SIP/WebRTC gateway that can bring very important features to the enterprise SIP-based infrastructure:

• You can get a voice/video telepresence quality SIP client in the web browser without any installation.

• That SIP client can be used remotely due to the built-in NAT/firewall traversal methods. Such a client is available to anyone, anywhere they can surf.

• A WebRTC-style invitation can be sent as an http link to a person you want to call you. When clicking that link, a browser window opens and he or she will be able to talk and videoconference with you.

With the SIP/WebRTC gateway, such links will go into the PBX (News - Alert) infrastructure with forwards, auto attendants, queues, conference bridges, etc., instead of bypassing it as WebRTC by itself would do.

At the recent WebRTC conference, Avaya (News - Alert) demonstrated the call center killer application. A logged-in customer could call the right call agent within his or her SIP UC infrastructure by clicking on a web page button, while all customer information was provided to the call agent.

Such web-based click-to-call applications will be widespread when WebRTC is available in the major browsers.

WebRTC is interfacing through the SIP gateway to the PBX or UC solution both as a client and via its SIP trunking interface. Thus, two important components for SIP trunking are already in place and used: The Internet connection and the PBX SIP trunking interface, and it is thereby close to also move over from an old PSTN/TDM connection to the telephone network and instead use a SIP trunk from an ITSP – if that is not already in place. The third component required for SIP trunking, the E-SBC, may also be included with the SIP/WebRTC gateway, making your PBX ready to connect to an ITSP’s SIP trunk, with lower telephony cost and other benefits.

Ingate Systems (News - Alert) demonstrated such a SIP/WebRTC gateway at the recent ITEXPO, SIP Trunking, UC and WebRTC Seminars in Las Vegas. It allows UC vendors to add all WebRTC features with their products, also including the E-SBC for SIP trunking and the Q-TURN server for WebRTC. It can be tried out at http://webrtc.ingate.com.

Edited by Maurice Nagle