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Ingate Shows How WebRTC Will Drive SIP Trunking at the SIP Trunking, UC & WebRTC Seminars at ITEXPO Las Vegas, August 12-14

SIP Trunking Featured Article

August 11, 2014


Ingate Shows How WebRTC Will Drive SIP Trunking at the SIP Trunking, UC & WebRTC Seminars at ITEXPO Las Vegas, August 12-14


By TMCnet Special Guest
Karl Stahl, Chairman and CTO, Ingate Systems


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When the standards are completed within IETF and W3C, and Microsoft (News - Alert) and Apple join the trend by adding WebRTC support to Internet Explorer and Safari, web applications using voice, video, data and screen sharing directly between web browsers will grow rapidly. Today, Google’s Chrome and Mozilla’s (News - Alert) Firefox browsers support pre-versions of the WebRTC protocols.


WebRTC can also be integrated with existing SIP-based services and systems. Integration with PBX (News - Alert), Unified Communications or call center 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 put on a website or 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 video conference with you.

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

At the recent WebRTC IV conference, Avaya demonstrated the call center killer application. A logged-in customer could call the right call agent within their 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 (enterprise session border controller), may also be included with the SIP/WebRTC gateway, making your PBX ready to connect to an ITSP’s SIP trunk, with lower telephony costs and other benefits.

Ingate Systems (News - Alert) will demonstrate such SIP/WebRTC gateway at the ITEXPO, SIP Trunking, UC and WebRTC Seminars. Visit Ingate in room Amazon L at the show. For your free VIP pass, click here. 




Edited by Maurice Nagle

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