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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