[Asterisk-Users] Codec negociation (2) -Workaround

Tue Mar 22 11:27:45 MST 2005

For OH323 there is a workaround
Before dialing out, do in your dialplan :

exten => XXX,1,SetGloabalVar(OH323_OUTCODEC=g729)

We are also preparing a version that has endpoint configuration
like in sip.conf. It will be ready soon.

George

Mike Tkachuk wrote:

>Hello,
>
>I fixed this problem for me with some asterisk patching.
>You can download patches at b2bua.berlios.de.
>Short explanation: new option 'O' in Dial application will send only 1
>codec (same as incoming) in outgoing invite. Curently only SIP channel
>patched.
>
>P.S. I'm not really good in asterisk internals, so, I'm sorry if this
>'ugly' hack, but it works for me fine.
>
>
>On Sat, 19 Mar 2005 17:01:50 -0700, Kevin P. Fleming
><kpfleming at starnetworks.us> wrote:
>  
>
>>Yves wrote:
>>
>>    
>>
>>>I receive G729 & G723 calls that I send to a provider who can handle
>>>both too, is it impossible to tell Asterisk to keep using the same codec
>>>for in & out ? It seems that he only follows the codec list in order.
>>>      
>>>
>>You are correct. Since Asterisk is a UAS/UAC and not a proxy, it
>>negotiates both sides of the call independently. Given that, when you
>>define a SIP user as allowing both G.729 and G.723, and the phone offers
>>both, Asterisk will pick one and move into the dialplan.
>>
>>However, if your provider's peer entry is also configured to allow both
>>G.729 and G.723, things should work fine. By default, Asterisk will
>>force a preference to the codec being used by the existing channel, to
>>try to avoid transcoding. That means that if you call in using G.729,
>>and the peer is set to prefer G.723 normally, Asterisk will still list
>>G.729 first in the outgoing INVITE to the peer. As long as the peer
>>respects that request, the call should go through without transcoding
>>needed.
>>
>>If you are not experiencing this behavior, then you'll need to post more
>> details and a 'sip debug' trace so we can figure out what's happening.
>>    
>>
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