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New Skype Alternatives for Internet Broadcasting

Skype-alternativesI have an issue with “meta“ things. I blog, but I’m not engaged with the broader realm of bloggers. Blogging about blogging baffles me. Similarly, although I’ve been involved in the VUC since 2008, I’m not really engaged in the world of podcasters/internet broadcasters. I’m trying to work on this by sharing some of the techniques that I’ve discovered in doing VUC calls.

Last fall I was advising podcaster Mike Phillips with some issues of audio quality with respect to remote participants in podcasts.  He appears to be a frequent contributor to the blog of the IAIB. It was there that I stumbled upon a post recommending Skype Alternatives For Internet Broadcasting.

This post implies that Skype is tremendously popular in this space, and yet there is some desire to seek out functional alternatives. The author, Andrew Zarian, offers the following list of alternatives; Google+ Hangouts, Zoom, Apple’s FaceTime and Cisco’s Jabber. All are certainly worthy of consideration.

However, time moves on and in just six months there are a range of new options also worthy of consideration, some arising from the fast-evolving world of WebRTC. I’d like to add a few additional services to the list;

1. Talky.IO

Talky.io is a WebRTC-based service built by &Yet largely as a proof of concept project. It was, as far as I know, one of the first of a handful of such services. It works well, providing multi-party video chat to anyone with a WebRTC-capable browser (Chrome or Firefox.)

All such services deliver audio using the Opus codec, which can provide excellent audio quality. The service allows you to name the meeting room as you like, then pass that address to participants as a simple URL. You can lock the room and share your screen.

At around the time of Mr. Zarian’s original set of recommendations WebRTC-based services were not yet sufficiently flexible to meet the needs of many people. In particular, the browsers lacked any way to select the desired microphone and camera. That problem was addressed some time early in Q4/2013. Since then I’ve had no trouble at all using such services, even though I have a diversity of audio and video I/O on my desktop computer.

What Talky.io lacks is any kind of recording capability. That you must provide locally to the producer, which isn’t entirely unexpected.

Finally, none of the current crop of web-RTC based services presently support central, producer control of participant mute state. That means each person has local control of their microphone mute. It’s should be said that diligent self-muting is next to godliness. Just be sure that all your guests know to do the right thing.

P.S. – There are a handful of free WebRTC-based chat services out there. I like Talky.io because &Yet were early to the party. They have been supportive of the community, hosting a very well-received west coast conference on real time communications. They seem like a genuinely nice bunch of people.

2. Voxeet

If you’re a regular VUC listener you may recall Voxeet from episode 471 back on January 10th. On that occasion company co-founder Stéphane Giraudie provided a good introduction to the service.

In February Voxeet transitioned the desktop clients for their free voice conference service to a WebRTC-based core. Accessible from Android and iOS mobile clients, with both Windows and a newly released Mac client, it offers both high-quality audio and a novel ability to place participants at various locations in a stereo soundscape.

I maintain that what they deliver is not the “3D audio” that they claim, but that’s really a matter of linguistics. It’s free and very good quality, so what’s not to like?

The availability of mobile and desktop clients provides a low-cost/complexity way to achieve high quality audio from remote participants.

You’ll need to make a local recording, but a stereo recording made locally will retain the stereo image, which could dramatically improve the podcast delivered to your audience.

3. GetOnSIP

GetOnSIP by Junction Networks is another free service. Initially launched as a SIP service it’s evolved into a hybrid  of SIP and WebRTC that should be very useful to podcasters. There exists an entire universe of SIP clients, both hardware and software, making such services eminently accessible.

Using SIP-based infrastructure for a podcast makes a lot of sense to me. It allows you to use SIP URI’s as addresses to bring guests into your call. They look a lot like email addresses, which makes them easy to understand for non-technical folks. The fact that you can pass someone a link that will work in a plain vanilla SIP client, without requiring registration to a SIP server, makes things fairly easy.

There are a diversity of free and cheap SIP clients for any platform you care to mention. Some SIP clients, most commonly soft phones like Jitsi, Blink or Bria, include call recording, a useful feature for podcasters.

The great thing about SIP services is that they allow you to leverage SIP hardware if desired. I still like my Polycom VVX desk phones for their great sound, convenient on-phone call recording and ability to conference between SIP services and plain old PSTN callers.

Further, by staying to IP-based services you can leverage the family of audio codecs that give your HDVoice. Whether baseline G.722, G.722.1, SPEEX, CELT, G.719 or Opus, you’ll have audio quality that rivals traditional broadcast radio.

4. Jitsi Video Bridge

Those more interested in a DIY approach may take a little more control of their own destiny by building their own infrastructure using Jitsi Video Bridge. Jitsi Video Bridge is the evolving WebRTC-based, open source video conferencing platform from the Jitsi team.

As was profiled in #VUC482, Jitsi Video Bridge is an impressive piece of work. Jisti eschews the heavy duty signal processing common to traditional MCUs, instead relying upon a stream switching approach know as selective forwarding. This strategy dramatically reduced the workload of the server, such that relatively modest hardware, like an Intel NUC, can satisfy some very real applications.

The Jitsi Video Bridge that was used for #VUC482 was a virtual machine instance with two virtual CPUs and a nominal amount of memory. It had about the same resource compliment as a VPS that once hosted this site. The most important requirement is high-quality bandwidth.

Of course, any WebRTC-based solution will feature encryption on all media handling. The fact that you can run the service yourself, whether on hardware or VM, allows to lock down all facets of its operation.

I’m seriously considering launching a local instance of JVB myself just to see what kind of trouble I can get into with such a tool in my arsenal.

Close

There you have it, from the extremely simple to a self-hosted geek-heaven, four free Skype alternatives that should serve an internet broadcaster well. Each of them are tried and true, even if they are cutting edge in some way.

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