[Freeswitch-users] members audio conference

[Madovsky]

I did a lot of silly things since I learn FS   :D
but not to try to transcode on FS if I use a proxy media....
I would just to minimize the transcoding....
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, December 20, 2010 4:29 PM
  Subject: Re: [Freeswitch-users] members audio conference





  On Mon, Dec 20, 2010 at 12:02 PM, Madovsky <infos at madovsky.org> wrote:

    > FS core converts every member from their codec to L16 
    ok I understand, so is it possible to change from L16 to another codec in a conference dialplan ?
  I don't think you understood what Steven was saying. You don't need to worry about the transcoding. As long as there is a codec that FS supports then the transcoding happens without you having to do anything. Unless you've done something silly like proxy media then you shouldn't need to worry about codecs.


  -MC


      ----- Original Message ----- 
      From: Steven Ayre 
      To: FreeSWITCH Users Help 
      Sent: Monday, December 20, 2010 2:48 PM
      Subject: Re: [Freeswitch-users] members audio conference


      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.

      Steve on iPhone

      On 20 Dec 2010, at 17:59, Michael Collins <msc at freeswitch.org> wrote:


        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. 
        -MC


        On Fri, Dec 17, 2010 at 6:49 PM, Madovsky <infos at madovsky.org> wrote:

          nohgint strange on logs.
          but I guess it's a codec and rate problem.
          tried with different SIP phones and it works.
          how a conference manage the codecs  ?
          I know the rate can be set in profile,
          but how conference codec is managed
           if all members have different codec and rate ?



            ----- Original Message ----- 
            From: Michael Collins 
            To: FreeSWITCH Users Help 
            Sent: Friday, December 17, 2010 7:54 PM
            Subject: Re: [Freeswitch-users] members audio conference


            What do you see in the debug logs? Did you compare the logs for a working vs. non-working call? Anything different? 


            -MC


            On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <infos at madovsky.org> wrote:

              when the first member creates and enters in a new conference
              everything is ok. but if a new memeber enters there is no audio
              in the conference, unless the ivr.

              I have a very simple conference dialplan like this

                      <extension name="create_conference">
                              <condition field="${sip_to_uri}" expression="^000(\d{10,15})@$${domain}$">
                                      <action application="set" data="instant_ringback=true"/>
                                      <action application="ring_ready"/>
                                      <action application="sleep" data="3000"/>
                                      <action application="answer"/>
                                      <action application="conference" data="$1-${domain_name}@wideband"/>
                              </condition>
                      </extension>


              I tried to add
              <action application="export" data="absolute_codec_string=speex at 16000k"/>

              but no success

              Thanks

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