[Asterisk-Users] Re: Asterisk<->GS and codec selection

[Bill Michaelson]

> Regarding codec selection, I see a minor difference between the FWD 
> and the local * box test cases, but I know nothing about the 
> negotiation protocol...
>
> With FWD, the OK message lists 3 Media Formats:
>
> 


Bingo...GS chokes with GSM...just disallow it in your sip.conf:
disallow=all
allow=alaw
allow=ulaw


Thank you, very much.  That got it working.  Actually, I used 
disallow=gsm as suggested by someone else.

Please forgive my ignorance, but this leaves open questions which are 
nagging me...

I expected that the SIP dialog would be a negotiation such that the 
devices agree on a mutually acceptable encoding.  And I think it's 
obvious (correct me if I'm missing any key points) that such a 
negotiation would involve selecting one of the encoding formats which 
appears in both lists presented  by each side.  It doesn't seem 
reasonable that the GS should just "flake out" as it seems to do, simply 
because it is offered an option it can't accept amongst ones that it 
can.  Is this indeed what I am seeing, or am I mischaracterizing it?

Also, as I noted earlier, shouldn't * wait for the ACK before spewing 
the audio stream?  It appears to be missing the ACK because it 
retransmits the OK shortly after it begins sending the RTP data.

These loose ends make me very uncomfortable.





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