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