[Freeswitch-users] members audio conference

[Steven Ayre]

L16 is raw audio, not a codec.

It can't be changed. The only way to merge audio from multiple channels is to merge the raw audio samples data.

Steve on iPhone

On 20 Dec 2010, at 20:02, "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 ?
> ----- 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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> 
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