[OpenSER-Devel] how to determine the codec used before RTP traffic was sent?

[Bogdan-Andrei Iancu]

Hi Yan,

I'm afraid you cannot determine it. Each UA sends a list of supported 
codecs and the other party will pick whatever it likes and start 
transmitting RTP.

Regards,
Bogdan

yanlin wrote:
> Hi, all
>
> i'm confused with codec choosing.
> anybody knows how to determine the codec before rtp traffic was acturally received?
>
> i used openser + xlite + asterisk, 
> when make call, i found 
> 	"m=audio xxxx RTP/AVP 107 6 0 8 3 5 101" in INVITE message,
> and
> 	"m=audio xxxx RTP/AVP 3 0 8 101" in 200 OK reply message,
> then codec PCMU(0) was used in actual call, 
> anybody know why PCMU(0) was choose? why not GSM(3) or PCMA(8)?
>   
> yan lin
> 2007.7.20
>
>
>
>
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