4K & WebRTC: A Tangential Observation

A short while ago Chris Koehncke posed the question, “Is 4K video viable for a WebRTC web application?” He also offered a well-reasoned opinion. While there’s technical support for 4K in browsers, and 4K webcams are starting to appear, in various ways bandwidth remains a constraint. As a purely practical matter, and in the most common use case, he’s perfectly correct.

The folks over at the Red5Pro blog offer a mildly dissenting opinion. They note that for use-cases beyond video conferencing, most especially one-way streams, 4K is quite practical. People stream Netflix at 4K. Heck, I’ve done it myself at least once or twice.

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Kranky & Krankier?

“Statler and Waldorf” by Source. Licensed under Fair use via Wikipedia

Chris Koehncke (aka Chris Kranky) recently posed a question in a blog post. He asked, “How good is your laptop microphone?” He then laid out an experimental series of recordings using different hardware. As an executive summary he offers, “Your current internal laptop mic is probably fine.” As you might imagine, I disagree…but there’s more to it than that.

In truth, it’s not that he’s wrong, but I think that he was asking the wrong question! The question he should have asked is, “How do I best convey my voice when using this laptop?”

The answer to that question is quite simple…use a high-quality headset, preferably one with a boom-mounted microphone. When participating in any kind of conference call, or video conference nothing can touch the quality of sound delivered by using a good headset. This has long been my belief, although I accept that it may not be a widespread.
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The Kranky Geeks Had Fun in SF

Last week marked the second Kranky Geek Conference in San Francisco. This time the entire event was streamed live, and they’ve put both the recordings and slides online quickly. Kudos to those who managed to get that organized. Great work!

The lineup of presenters was first-rate, with solid representation from WebRTC leadership at Google, Microsoft & Mozilla. Of course, Emil Ivov was there talking about SFUs for Atlassian. Tim Panton build an app live on stage. Clearly, it’s worth watching the recording if you could not be in attendance.

I was especially interested in something Nils Ohlmeier, from Mozilla mentioned. He described rendering a video stream locally to a canvas, then using that canvas as a video source for the outbound stream. He further described using this capability to create an ad hoc MCU, with the browser compositing multiple video streams into one outbound stream.

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Oh, the things that this could do! The most obvious thing that comes to mind is composite the camera stream along with the screen share, just like I was asking for back in December 2014.

That’s just a starting point. I can image folks like Vidpresso implementing a small video production switcher right in the browser. It’s exciting stuff. Very exciting indeed.

MOS, Podcasting, SIP Reinvites and WebRTC

podcast mic & logosAccording to popular legend, in the early days of talking movies there was a German director working in Hollywood whose pronounced accent skewed his use of English. He would call for another take of a scene, this time without recording sound. He’d yell out “Mit Out Sound!” Over the years industry professionals came to use the acronym MOS as a shorthand for recording a silent take.

Operating MOS may be occasionally useful in film, but it can be disastrous for a podcaster. When producing a podcast reliable audio is a must. Achieving this goal can be complicated when trying to connect to a distributed array of co-hosts & guests via the internet.

Using a SIP service like SIP2SIP.Info allows the use of high-performance audio codecs, like Opus, which makes for superior podcast audio. This is something that I’ve advocated for along time in my series called Making Use of HDVoice Right Now!

This week I had a Twitter exchange with veteran broadcaster and podcaster Mike Phillips about a problem with audio over a SIP connection.

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Improving Online Presentations With Reflect For Chrome

Reflect Example   IconWhen giving an online presentation, whether via Hangout, Webex, GotoMeeting or Jitsi Video Bridge, a screen sharing function is how the slides are presented. Screen sharing goes beyond the common death-by-PowerPoint scenario, allowing software demonstrations in real-time.

In most cases the use of screen sharing defeats a view of the presenter. This can make for a frightfully dull audience experience. I’ve long wished for a convenient way to show slides or demo software without completely losing sight of the presenter. I’ve done this myself using tools like Wirecast or vMix, but these are costly and complicated things to setup. That’s beyond the reach of most people.

In a post late last year I challenged the WebRTC community to take up this idea. WebRTC remains a battle with many fronts. They may yet bring forward such capability, but the wheels of progress turn slowly.

Happily, I’ve recently discovered an extension for Chrome called “Reflect.” Authored by Bryce Thorup, Reflect grabs your webcam stream, displaying it as an overlay on your monitor. Thus when sharing your screen, your audience can see both the slides (or software) and your webcam.

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Interconnecting Jitsi Video Bridge, ZipDX & YouTube Live

In the production of over 530 VUC sessions we’ve undertaken some odd and occasionally rather complicated arrangements. Quite possibly the most complex is when we interconnect the WebRTC-based Jitsi Video Bridge with YouTube Live and the ZipDX conference bridge. I set about described aspects of this process a year ago, but stopped short of describing how the entire arrangement worked. Well, worked most of the time. This article will bring you current with my various attempts to make this process robust and repeatable.

Preface: When we use Jitsi Video Bridge we lose a couple of the conveniences that come with a Hangout-On-Air. Where a Hangout-On-Air has an automatic link to a YouTube Live event, we must do this manually when we use JVB.
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