Effi,

I am not sure this is such a good idea – it really depends on the problem you are trying to solve.
As far as I remember, the sipML5 one is 1.5 Mb in size, which isn’t that nice for mobile devices. In a lot of the vendors out there, I’ve seen solutions which use SIP in the backend but prefer using a proprietary protocol (or XMPP) for the client side to connect with it.
To me this seems like the better approach for a large set of problems.

I’ve looked over at sipML5, JS open source HTML5 SIP client, and its companion webRTC2SIP. for me it looks like it fills in the needed gap for the signaling part. also allowing the interconnect to legacy/IMS network.
have you had a chance to explore that solution?

I think Tsahi summarized it pretty well. There is still a fair amount of H.323 traffic in the public Internet and private networks for voice, but largely for toll bypass. Most service providers have adopted IMS as a replacement for their legacy telephone network. IMS means a variant of SIP. However, it really is nothing but a PSTN replacement. Inside the enterprise, most people are using H.323 for video.

In fact, every time I call into Cisco to join a video conference, I use H.323 from my desk. There are new services like Spranto (www.spranto.com) where you can use H.323 to make calls from home, work, Starbucks, etc. to any H.323-compatible device in the world over the Internet. You can connect to MCUs in the cloud to do conference calls using H.323, like Blue Jeans and Vidtel.

Another fun fact is that the move away from E.164 numbers is made possible. You can call me directly using my email address (at home or work)! 🙂

I don’t have percentage information.
What I can say is this:
* In most (almost all) enterprise settings, “managed” video conferencing is done using H.323. Even if the systems support SIP – they end up configuring them to do H.323.
* Almost all VoIP is done using SIP in enterprises (I include Microsoft Lync as SIP here).
* On consumer, most is proprietary, as Skype is proprietary. And so is ooVoo and Viber as far as I know. FaceTime is “SIP” but that doesn’t help anyone. Tango – I have no clue – probably proprietary as well.

I hope this helps…

Hi
Do you have any information about the traffic of these protocols in the world? for example what percentageof the whole traffic correspond to H.323 and what percentage correspond to Sip and the others….
If you have any information plese let me know.
Thanks

OrNot,

Here’s the post trying to explain a bit more about signaling and media: http://bloggeek.me/voip-signaling-101/

Hi Tsahi,
So SIP plays the same or similar role as STUN, ICE, TURN?

I have read WebRTC ‘s doc, but can’t get a clear big picture from it. Hopefully your post can help on it.

Thanks again

OrNot,

WebRTC is a media engine. You tell it where to send media, from where to receive and in what codecs and it works. The question then is, how do you know where to send the media. That’s where signaling comes along.
I’ll do a post on the difference in a few weeks time – hope it will help.

Tsahi