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David, Goliath and WebRTC

Lenovo X-1-Carbon Frontal-GotoMeetingFreeThe WebRTC movement is certainly gathering steam. World+Dog are now running along side that train, hoping to jump on and play with all the cool kids. It’s a fast moving train. Late comers may have a tough time getting on-board. When they do get aboard they may find that the best seats are already taken.

This opinion arises from a stream of chatter surrounding larger companies that are starting to play in WebRTC space. While it’s nice to see larger players starting to engage, I have a difficulty getting enthused about what they’ve actually done.

For example, Citrix having launched into beta a free video chat application under its GotoMeeting brand. I was a GotoMeeting user from their earliest days. I argued with my employer about the value that was to be found in the $49/mo they asked for the service. He initially balked at $600/yr, but eventually succumb when it became evident that I could save considerable travel expense by leveraging remote access for ad hoc customer support. Some years later that company actually adopted GotoAssist, which is an outstanding service.

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Talky.io Offers A Great WebRTC Experience

In the post-roll after last week’s VUC call I was able to invite Andy to join me in trying Talky a relatively new WebRTC– based video calling service from &Yet. The service is very interesting as is the company itself. I heartily recommend their blog.

TalkyIO-Screenshot

Andy and I had a good, lengthy chat using Talky. We came to the conclusion that it was the best experience with WebRTC that either of us has encountered. Admittedly, neither of us are WebRTC experts.

Talky supports screen sharing from within Chrome, given that a configuration change is made prior to starting the session. What it lacks is any kind of text chat.

 

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Remote Conference Presentation via WebRTC

BlogGeekMe-ScreenshotCredit is due to Tsahi Levent-Levi of Amdocs, not only is he a leading voice in the evolution of WebRTC, but he’s eating that very dog food himself. It’s so great to see people walk-the-walk while giving-the-talk. That is in fact what he’s planning to do.

In a blog post today he put out a call for suggestion of WebRTC-based services to highlight in giving a presentation to a conference. His plans include giving the presentation remotely via a WebRTC connection. That’s awesome.

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LifeSize on WebRTC Revisited

LifeSize WebRTC gateway LaptopA 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.

A couple of days later Emily G. from LifeSize PR responded to my inquiry about this. She offered to be the other end of a an initial test call, giving me her H.323 dialing string as a calling target.

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.

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LifeSize On WebRTC

LifeSize_Icon_SeriesLast 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.

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Audio Codes Offer First Signs Of Hardware Support for Opus Codec

AUDIO-CODES-420HD-300pxOne of the first interesting things to arise out of this weeks WebRTC conference in Atlanta is an announcement from Audio Codes that they will be adding support for the Opus codec to their 400HD series of SIP desk phones. Here’s the press release.

There seems to be a little confusion about what’s actually going to be offered.  The title of the release states, “AudioCodes Announces WebRTC Phone.” The copy goes on to describe that the hardware will be extended to provide native support the Opus codec. This capability is projected to be deliverable in Q4-2013.

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