Guys,
Quick question: Why is it that even though I explicitly define
codec g711ulaw under "voice register pool". When the "INVITE" is sent out SDP
always includes all the codecs and DTMF types. It is also the same for DTMF
relay. Even though I put in "rte-nte" only, it offers "sip-notify" also. Most
of the time this is not a problem. But sometimes where a transcoder needs to be
invoked it fails to do so primarily because SDP suggests that phone supports it
but in fact it only takes g711ulaw.
Thanks,
INVITE sip:5001@10.10.202.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.200.209:1024;branch=z9hG4bK5ab46861
From: "br2ph6 3006" <sip:3006@10.10.202.1>;tag=0024c4fd1064000c41ef374d-b13ef517
To: <sip:5001@10.10.202.1;user=phone>
Call-ID: 0024c4fd-10640008-ebc44099-8bcd8a93@192.168.14.20
Max-Forwards: 70
Date: Tue, 03 May 2011 18:17:29 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7962G/8.4.0
Contact: <sip:3006@192.168.200.209:1024;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: "br2ph6 3006"
<sip:3006@10.10.202.1>;party=calling;id-type=subscriber;privacy=off;screen=yes
Allow-Events: kpml,dialog
Content-Length: 326
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 717 0 IN IP4 192.168.14.20
s=SIP Call
t=0 0
m=audio 17132 RTP/AVP 0 8 18 116 101
c=IN IP4 192.168.200.209
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
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