Comments on: How Will WebRTC Change the VoIP Industry?

By: Tsahi Levent-Levi

Tsahi Levent-Levi — Fri, 03 Aug 2012 10:42:16 +0000

John,

There is no default signaling for WebRTC, and I think this is what makes this standard/package so great. You can use whatever you feel like: XMPP, SIP (both have JavaScript packages these days) or you can use your own proprietary mechanism on top of JavaScript.
If you are looking to connect to a third party PBX or an existing ecosystem, then I suggest you try and look for a SIP implementation (http://sipml5.org) and even checkout what Asterisk are doing to support WebRTC on the server side.

By: john_conor

john_conor — Fri, 03 Aug 2012 10:03:39 +0000

Hi Tsahi,

First of all a very informative post.

Though what I would like to know is if I say download the webRtc code, will it have any ‘default’ signaling stack(say SIP) as part of its architecture?

I am not a web developer but plan to use webRtc as part of an embedded end point(a Voip phone for instance) and the signalling part of webRtc is currently a bit of a grope in the dark.

Thanks & regards,
John

By: Tsahi Levent-Levi

Tsahi Levent-Levi — Tue, 10 Apr 2012 05:09:32 +0000

In reply to Alexey.

Alexey,
That is exactly what I am talking about – companies such as your own connecting the “old” world of SIP/PSTN (or others) to the new one of WebRTC.
There are going to be many ways of doing that moving forward.

By: Tsahi Levent-Levi

Tsahi Levent-Levi — Tue, 10 Apr 2012 04:56:30 +0000

In reply to Randell Jesup.

Randell,
Thanks for the tip – I’ve seen that progress around Opus but haven’t given it much thought yet. I’ll definitely be looking to it more closely now.

By: Randell Jesup

Randell Jesup — Tue, 10 Apr 2012 03:42:52 +0000

In addition to the codecs already in WebRTC, you should also make note of Opus, which is just being moved towards RFC status, and which can handle all the way from G.729 bandwidth levels to 44KHz multi-channel audio at hundreds of Kbps, and do so while equaling or beating existing codecs (including iLBC and iSAC) at the same bitrates – and it’s free. We at Mozilla are pushing hard to get the IETF to adopt Opus as a mandatory-to-implement codec in addition to G.711.

By: Alexey

Alexey — Sun, 08 Apr 2012 17:48:41 +0000

Gateways are already there and you can make a call to SIP or PSTN using WebRTC right now http://blog.zingaya.com/2012/03/29/webrtc-version-of-zingaya/

By: Tsahi Levent-Levi

Tsahi Levent-Levi — Fri, 06 Apr 2012 19:05:55 +0000

In reply to Shimon.

I am ignoring Skype because I don’t see them connecting to anything else anytime soon – besides maybe Lync, but that is yet to be seen.
And if we are talking about Skype, then they were the first to adopt VP8 a few months ago so they are ahead of the rest in the potential to use WebRTC. If I were Skype – that’s what I’d use.

By: Tsahi Levent-Levi

Tsahi Levent-Levi — Fri, 06 Apr 2012 19:04:29 +0000

In reply to Hong-Quan Yue.

Hong-Quan,

You are correct – developers can use whatever they want.
My first selection would be either XMPP or SIP – or a proprietary thing.
H.323 would go next.
3G-324M won’t work at all as it is a circuit switching technology that isn’t really suitable for IP.

By: Shimon

Shimon — Fri, 06 Apr 2012 11:19:38 +0000

how can you ignore skype?
today it is the biggest enterprise video conferencing system in the world!!
Skype “users made 95 billion minutes of voice and video calls” during the first half of 2010, with a full 40 percent of those minutes being video.
37 percent of users surveyed say they use Skype for business purposes
so roughly we have 95B*40%*37%=14B enterprise skype video minutes during 6 months!!!!!

By: Hong-Quan Yue

Hong-Quan Yue — Thu, 05 Apr 2012 21:53:55 +0000

One thing is good for WebRTC developer is that you can add your favorite signaling. I would say that H.323 and 3G-324M over ip are more suitable for the mass deployment.