[asterisk-users] Re: [asterisk-dev] Questions regarding g.729 and g.711 in Asterisk

[Olle E Johansson]

7 aug 2006 kl. 14.37 skrev Rich Adamson:

> Chan Kwang Mien wrote:
>> Thanks. By setting allow=g729, sip1 was able to connect to sip2.  
>> Does that mean that Asterisk chooses the codec to be used between the
>> Caller and Callee ? In this case, since sip1 informs Asterisk that it
>> supports g.711 and g.729, Asterisk chooses g.729 since the Callee  
>> also
>> supports g.729. From the SIP Messages exchange, it doesn't seem that
>> Asterisk chooses the codec.
>> My previous setting was "allow=all". I was expecting "allow=all"  
>> to work
>> since that would also imply "allow=g729", isn't it ?
>
> This really belongs on the -users list since it doesn't deal with  
> developing code. Moving it there now.
>
> Each sip phone essentially negotiates a codec independently with  
> asterisk, and not as an end-to-end conversation.
Not "a" codec - many codecs.

>
> When sip1 initiates a call, it exchanges sip packets with asterisk  
> to select a compatible codec.  When asterisk places the call to  
> sip2, it exchanges sip packets with sip2 to select a compatible  
> codec and it has nothing to do with what sip1 negotiated.
Well, this is changing. With new RFCs they have something to do with  
what SIP1 offered. Note that
* More than one codec can be "approved" for a call. Each UA can then  
freely switch between the
   codecs during a call without a re-invite.
* Different codecs can be used in each direction

>
> You have two choices to correct the behavior. One, change the  
> asterisk definitions so as to show a preference (disallow=all,  
> allow=g729,ulaw), or, two, change the sip phone's definition to  
> prefer g729 as its first choice.
Or use the SIP_CODEC variable in the dialplan to set a prefered codec  
for the call.

>
> Even though many sip phones support multiple codecs, the  
> negotiation is very simple in that it offers up its first choice  
> codec (only), and if asterisk supports that first choice, that's  
> what is used. Highly dependent on the sip phone manufacturers coding.
Yes. Asterisk has some interesting behaviour too, that we will have  
to change to comply with recent standards.

/O


---
* Olle E. Johansson - oej at edvina.net
* Asterisk Training http://edvina.net/training/