[asterisk-users] No rtpmap codec info in 200 OK

[William Scott]

Hi,

My VSP uses Asterisk to which I'm connected with an ATA.

When I receive an inbound call the invite includes the following...

v=0
o=root 32218 32218 IN IP4 202.52.129.50
s=session
c=IN IP4 202.52.129.50
t=0 0
m=audio 16864 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off – - – -
a=ptime:20
a=sendrecv

My ATA's 200 OK reply after call setup has the following...

v=0
o=CMI-SIPUA 13369 0 IN IP4 211.30.XXX.XXX
s=SIP CALL
c=IN IP4 211.30.XXX.XXX
t=0 0
m=audio 20216 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=rtcp:20217
a=silenceSupp:off – - – -
a=sendrecv


Notice there is no "rtpmap:18 G729/8000" in the reply.

The call continues fine.

Is it right that there is no codec info in the reply and the call continues?