A couple of weeks back Lifesize announced the availability of an experimental WebRTC gateway service. They made the announcement during a webinar on WebRTC. Of course, I went to try the service shortly thereafter. While I could make the test call to the address offered, I couldn’t reach anything else, nor was it exactly obvious how that should be done.
At the appointed time I visited the the WebRTC test page using Chrome on my laptop and entered her H.323 address, which was just an IP address. The WebRTC gateway immediately connected us. We chatted briefly. She was able to explain how the gateway should accept typical H.323 dialing strings or a SIP URI.
The gateway worked reasonably well for this short call. The call quality was limited by my use of a laptop with it’s questionable built-in camera. Also by the fact that the laptop was online over my local Wifi. Wifi and high-bandwidth streaming media are not always a happy pair.