WebRTC is Google’s (News - Alert) initiative to bring multimedia or real-time person-to-person communication – RTC – directly into the web browser. Although it may be a year before the IETF and W3C standards are in place, WebRTC is already available in Google’s Chrome and Mozilla’s (News - Alert) Firefox browsers and applications are emerging.
WebRTC-ready browsers are able to “talk to each other” with HiFi-quality sound and HD video – which is telepresence quality – with fallback to ordinary 3.5kHz telephony voice. However, there is no signaling protocol defined – no SIP, no phone numbers, not even a WebRTC address. The idea is that the parties already are in contact, e.g. chatting on Facebook (News - Alert) and then decide to talk or have a videoconference. Or you send an e-mail with a URL, asking your friend to click on that link at 3 p.m. when you are available for a call. Another application is click to call on the company or call center web page. Why stretch out for a phone once you’ve found a telephone number, instead of just clicking on the Support and Sales buttons?
The last application is obviously something that must integrated into an enterprise PBX (News - Alert) or UC solution. Another application is to have the web browser as the softphone for the PBX, available on any Internet-connected device with a browser. That will increase the enterprise UC usage, both on the LAN and for home workers and road warriors.
These applications imply that PBXs and UC solutions will come with WebRTC-to-SIP gateways and those may be part of the SIP trunking solution. There are also NAT/firewall traversal issues to handle with WebRTC (as with all RTC traffic as we know from SIP). WebRTC is over the Internet or OTT and competes for the bandwidth with data traffic, so there must be QoS handling in such solutions, especially considering the potential HD quality.
At the WebRTC conference in Atlanta we saw these kinds of solutions, integration of WebRTC with the enterprise PBX and UC solutions appearing. WebRTC will improve and be a necessary part of the enterprise UC and SIP trunking solution.
Edited by Stefania Viscusi