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

[yanlin]

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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