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Hello,<br>
<br>
<div class="moz-cite-prefix">On 12/19/12 2:28 AM, Raj Roy Ghandhi
wrote:<br>
</div>
<blockquote
cite="mid:CA+5JO-+7VfE6ueYjM8Vksw=67gzEXGZSs2_rpp9iTJXurY0J5A@mail.gmail.com"
type="cite">Hi,
<div>I want to do web (HTML5 + WebRTC) Sip client which can do the
video conference with multiple users.</div>
<div>Current release of <a moz-do-not-send="true"
href="http://www.sipml5.org/">SIPML</a> does 1 to 1 call.</div>
<div><br>
</div>
<div>
I have no idea of conference with many users.</div>
<div>Is it the client that we need to modify to accept call and
join the conference ?</div>
<div>Do I need to send INVITE with extra parameters ?</div>
</blockquote>
the client can do the mixing and act as a conference bridge -- many
classic sip phones do 3-way conferencing. You have to update yours
if you want the same. There is nothing that has to be in the INVITE,
the user will decide when to bridge.<br>
<br>
Alternative is to use a dedicated software as conference bridge
(like asterisk, freeswitch or sems), where each participant has to
dial a specific number for the conference room.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<br>
<blockquote
cite="mid:CA+5JO-+7VfE6ueYjM8Vksw=67gzEXGZSs2_rpp9iTJXurY0J5A@mail.gmail.com"
type="cite">
<div><br>
</div>
<div>
Please advice me.</div>
<div><br>
</div>
<div>Best Regards,</div>
<div>Roy.</div>
<div><br>
</div>
<div><br>
</div>
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<br>
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</pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a></pre>
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