<html><body bgcolor="#FFFFFF"><div>Each member will connect to the codec with whichever codec they want to use. Conference always operates at raw audio level (L16). Core handles transcoding to convert between codecs. You can have members in the same conference using different codecs. You can control which codec a caller connects to the server with by adjusting the codec preferences on the sip profile.<br><br>Steve on iPhone</div><div><br>On 20 Dec 2010, at 20:02, "Madovsky" <<a href="mailto:infos@madovsky.org">infos@madovsky.org</a>> wrote:<br><br></div><div></div><blockquote type="cite"><div>
<div><font size="2">> </font><font size="3">FS core converts every member from
their codec to L16 </font></div>
<div>ok I understand, so is it possible to change from L16 to another
codec in a conference dialplan ?</div>
<blockquote style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<div style="FONT: 10pt arial">----- Original Message ----- </div>
<div style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><b>From:</b>
<a title="steveayre@gmail.com" href="mailto:steveayre@gmail.com">Steven Ayre</a>
</div>
<div style="FONT: 10pt arial"><b>To:</b> <a title="freeswitch-users@lists.freeswitch.org" href="mailto:freeswitch-users@lists.freeswitch.org">FreeSWITCH Users Help</a>
</div>
<div style="FONT: 10pt arial"><b>Sent:</b> Monday, December 20, 2010 2:48
PM</div>
<div style="FONT: 10pt arial"><b>Subject:</b> Re: [Freeswitch-users] members
audio conference</div>
<div><br></div>
<div>More specifically FS core converts every member from their codec to L16
which is given to the conference. The conference combines all speaking
channels and the resulting L16 is given back to the core to send to the
members, the core converting to the correct codec for each
member.<br><br>Steve on iPhone</div>
<div><br>On 20 Dec 2010, at 17:59, Michael Collins <<a href="mailto:msc@freeswitch.org"><a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a></a>>
wrote:<br><br></div>
<div></div>
<blockquote type="cite">
<div>The conference itself doesn't do any codec stuff - FreeSWITCH core
does. All I can say is compare the working versus non-working logs and look
for clues.
<div>-MC<br><br>
<div class="gmail_quote">On Fri, Dec 17, 2010 at 6:49 PM, Madovsky <span dir="ltr"><<a href="mailto:infos@madovsky.org"></a><a href="mailto:infos@madovsky.org"><a href="mailto:infos@madovsky.org">infos@madovsky.org</a></a>></span>
wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div bgcolor="#ffffff">
<div><font size="2">nohgint strange on logs.</font></div>
<div><font size="2">but I guess it's a codec and rate problem.</font></div>
<div><font size="2">tried with different SIP phones and it
works.</font></div>
<div><font size="2">how a conference manage the codecs ?</font></div>
<div><font size="2">I know the rate can be set in profile,</font></div>
<div><font size="2">but how conference codec is managed</font></div>
<div><font size="2"> </font><font size="2">if all members have different
codec and rate ?</font></div>
<div><font size="2"></font> </div>
<div><font size="2"></font> </div>
<div><font size="2"></font> </div>
<blockquote style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<div>
<div></div>
<div class="h5">
<div style="FONT: 10pt arial">----- Original Message ----- </div>
<div style="FONT: 10pt arial; BACKGROUND: #e4e4e4"><b>From:</b> <a title="msc@freeswitch.org" href="mailto:msc@freeswitch.org" target="_blank">Michael Collins</a> </div>
<div style="FONT: 10pt arial"><b>To:</b> <a title="freeswitch-users@lists.freeswitch.org" href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">FreeSWITCH Users Help</a> </div>
<div style="FONT: 10pt arial"><b>Sent:</b> Friday, December 17, 2010
7:54 PM</div>
<div style="FONT: 10pt arial"><b>Subject:</b> Re: [Freeswitch-users]
members audio conference</div>
<div><br></div>What do you see in the debug logs? Did you compare the
logs for a working vs. non-working call? Anything different?
<div><br></div>
<div>-MC<br><br>
<div class="gmail_quote">On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <span dir="ltr"><<a href="mailto:infos@madovsky.org" target="_blank"></a><a href="mailto:infos@madovsky.org"><a href="mailto:infos@madovsky.org">infos@madovsky.org</a></a>></span>
wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div bgcolor="#ffffff">
<div><font size="2">when the first member creates and enters in a new
conference</font></div>
<div><font size="2">everything is ok. but if a new memeber enters there
is no audio</font></div>
<div><font size="2">in the conference, unless the ivr.</font></div>
<div><font size="2"></font> </div>
<div><font size="2">I have a very simple conference dialplan like
this</font></div>
<div><font size="2"></font> </div>
<div><font size="2">
<extension
name="create_conference"><br>
<condition field="${sip_to_uri}"
expression="^000(\d{10,15})@$${domain}$"><br>
<action application="set"
data="instant_ringback=true"/><br>
<action
application="ring_ready"/><br>
<action application="sleep"
data="3000"/><br>
<action
application="answer"/><br>
<action application="conference" data="<a href="mailto:$1-$%7Bdomain_name%7D@wideband%22/" target="_blank">$1-${domain_name}@wideband"/</a>><br>
</condition><br>
</extension><br></font></div>
<div><font size="2"></font> </div>
<div><font size="2">I tried to add</font></div>
<div><font size="2"><action application="export" data="<a href="mailto:absolute_codec_string=speex@16000k%22/" target="_blank">absolute_codec_string=speex@16000k"/</a>></font></div>
<div><font size="2"></font> </div>
<div><font size="2">but no success</font></div>
<div><font size="2"></font> </div>
<div><font size="2">Thanks</font></div></div><br>_______________________________________________<br>FreeSWITCH-users
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