<div>Hi,</div><div> </div><div>Sorry to trouble you with matters that have been out there much, I'll need to do more searching.</div><div> </div><div>This helps me search for the right things and to make recommendations for my collegues, thanks! It's quite interesting that Kamailio and even clientside stuff is basically ready, only problem is the media part.</div>
<div> </div><div>Cheers,</div><div>Pirjo</div><div><br><br> </div><div class="gmail_quote">2013/1/29 Peter Dunkley <span dir="ltr"><<a href="mailto:peter.dunkley@crocodile-rcs.com" target="_blank">peter.dunkley@crocodile-rcs.com</a>></span><br>
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<div>
Hello,<br>
<br>
This topic has been covered many times on the Kamailio/SIP Router lists and the answers you require can be found by searching the lists.<br>
<br>
I will briefly recap:
<ul>
<li>SIP signalling and media are totally separate things
<li>SIP over WebSockets will allow an HTML5 based client to exchange signalling information with standard soft-phones and hard-phones. However, this does not mean that the media will interwork
<li>HTML5 media streaming uses WebRTC. WebRTC mandates the use of the RTP/SAVPF media profile which is not yet supported by many soft-phones, hard-phones, or media servers
</li></li></li></ul>
<br>
This means that, provided you have configured Kamailio and sipml5 correctly, you can get the signalling part of a call working but you will almost certainly have media issues. Kamailio is a SIP signalling device, not a media device, so fixing these media issues is outside of the scope of Kamailio.<br>
<br>
You do have a few of options with regards to the media but they are limited at the moment.
<ul>
<li>You can try and find a phone/client that supports RTP/SAVPF (the only ones I know of are the Doubango clients and they sometimes have other issues).
<li>You can use a media server to convert from RTP/SAVPF (Asterisk supports this in theory, but does have issues - I believe there are fixes in the latest Asterisk trunk if you want to compile it yourself - and there may be some non-open-source media servers available).
<li>You can use an RTP Proxy to convert from RTP/SAVPF (erlrtpproxy has this feature on the roadmap, but I don't know whether it is available yet).
</li></li></li></ul>
<br>
As for IE support, your guess is as good as mine. Microsoft has its own agenda and has recently been pushing the competing CU-Web-RTC specification. I have a personal opinion about how things will eventually evolve but no facts to share here - I don't believe anyone outside of Microsoft could tell you what will actually happen with IE.<br>
<br>
Regards,<br>
<br>
Peter<div><div class="h5"><br>
<br>
<br>
On Mon, 2013-01-28 at 20:45 +0200, Pirjo Ahvenainen wrote:
</div></div><blockquote type="CITE">
<pre>Greetings gurus!
I'm playing with an idea to create a web based softphone (html5 + no
installations for the end user) and use Kamailio's websocket module
for backend. I'd love to hear about your comments, challenges and
successes using such configuration. Is it a feasible way to construct
a softphone even today when even IE9 does not support websockets, as
such? I'm sure IE9 will end up in specs as a must-support platform.
A collegue tried using sipml5 with webrtc against a SnomONE pbx (I
know... ;)), and said there's no way it can work, but I'm not
convinced the idea itself wouldn't work.
It would help me lots if I could make a simple example using Kamailio
with SIP over websockets, can you comment on how much effort do I need
on Kamailio side to make this work? Do I need off-default config
scripting, or is it enough to just set up the module and set the
parameters? And even with the risk of stepping a little off topic, if
anyone has worked on web based softphones, I'd love to hear if you can
recommend on how to approach this.
Cheers,
Pirjo
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<pre>--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
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<br></blockquote></div><br><br clear="all"><br>-- <br><div> </div><div>Pirjo Ahvenainen</div><div>+35844 559 7729</div>