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.
Later that day I had a call scheduled to catch up with blogalyst extraordinaire Dave Michels. Dave is usually open to experimentation so I suggested that we meet using the LifeSize WebRTC gateway. In the past we’ve used Polycom VVX Series devices and the Avaya/Radvision Scopia Desktop.
The WebRTC gateway is just a means of using a WebRTC-capable browser as one-end point on an otherwise normal call. Since we wanted to experiment a bit I gave Dave the SIP URI for a meeting room on a Polycom MCU that ZipDX has in a West Coast colocation facility. Then I called that same meeting room via the LifeSize gateway. So that call path involved a LifeSize gateway connecting to a Polycom MCU…just to make things interesting.
After an initial hiccup the call was established and Dave and I were able to chat for a while. Twice during the conversation we lost media to/from the WebRTC leg of the call. Since the MCU held the call ongoing it was a simple matter to refresh the browser and connect anew.
Since that day I’ve made a few test calls via the LifeSize gateway. In most cases it connects me to my intended target. Most of my dialing is done using SIP URI as call addresses. That’s very convenient since all of my end-points are connected to either OnSIP or ZipDX via SIP.
So what does this prove? LifeSize has implemented an experiment gateway to bridge WebRTC into the legacy SIP and H.323 realms. I applaud their effort to interoperate with the newly emerging WebRTC standard.
I suspect that the established vendors are not going to be so easily cast aside by the newest technology on the block. At least not on the merits of video calling and conferencing alone. If the old guard embrace the new, and use it to add value to the existing, then perhaps there’s a chance for coexistence.
The question that I am left with involves the creative impact of the developer community in wielding their shiny, new WebRTC blade. Will they use it to fight new fights? Will they carve out new kingdoms? Or will they merely clash with the existing hardware makers? Surely all that can come of that is a slow death spiral as the cost of using video follows the trajectory already clearly observed in voice/minutes.