Last week LifeSize had a webinar on the topic of WebRTC. I took an hour to listen to what they had to say and pose a couple of questions. Their target audience appeared to be people who might have heard some of the hype about WebRTC, but were not otherwise familiar with this new phenomenon. Suffice it to say that the material covered was introductory.
The webinar started with a pre-recorded video of Casey King, LifeSize CTO and Simon Dudley, who is described as LifeSize video evangelist. Their pre-recorded conversation was followed by an audio-only live segment where they answered questions arising from the audience, which was reported to be over 1000 people.
If you care to view the event after the fact you’ll find a recording of the webinar here.
During the live event I posed a couple of questions in the text chat. I asked if they had any plans to support the Opus audio codec and VP8 video codec. These are core aspects of WebRTC, although the debate about whether VP8 or H.264 should be “mandatory” rages on.
In the text chat someone responded that LifeSize would in fact support Opus and VP8, although no such statement was made by the hosts.
Later in the conversation it became clear that LifeSize views WebRTC as a realm that will be accessed by some kind of gateway. It’s unclear if support for WebRTC will be extended to their end-points or only infrastructure products. They could create a new product that was a dedicated gateway, which might be necessary if they envision a lot of transcoding of video streams between H.264 and VP8.
Perhaps the most interesting thing that I learned from the webinar was that LifeSize has launched an experimental WebRTC calling service. A brief call from my laptop reveals that the service initially requires an email address. Once registered the service allows a SIP URI to be entered. It defaults to making a test call to a server that is essentially a video echo test.
I noted that while my laptop worked for the purposes of the test call, my desktop did not. With multiple video sources available on the desktop the WebRTC app selected the BlackMagic Intensity Pro, which had no input, resulting in a full-frame green video. The app did not allow me to select the appropriate source, which would have been the Logitech C920 camera.
While that issue was interesting the service is clearly stated as a beta, so I can’t fault them for such inadequacies at the moment. Clearly the use-case for this kind of client is more geared to a laptop or mobile user, where diversity of video sources is less likely.
On a more positive note, the test page seemed to work on my Nexus 7 tablet. Not that the call worked flawlessly. The process seemed to suffer limited processing power and bandwidth. However, I was pleased that it didn’t fail outright.
I think that was the first time that I’ve tapped into the WebRTC capability of Chrome in the mobile form factor.
It’s actually not clear that the address entered is a SIP URI or something else. The fact that they state “all calls are limited to 60 minutes” suggests that calling something beyond the test server is possible.
I’ve tried calling to a SIP URI without success. I’ve also tried to call using an email address that was registered to a second PC also logged into the service. No joy there either.
On that basis I posed a question in the comment stream of the blog post where the service was announced. I’ve awaited a couple of days to see if they responded. Thus far they have not. Nor have they even approved the comment for display. I’ll update this if/when they finally react.