I have seen a copy of the Invites that are leaving my carrier’s proxy, and I have seen the same invite from a TCP dump on my PC. Somewhere between the proxy and my PC, the SDP info is being rewritten and I have no idea where. The problem is, it’s rewriting the SDP info to use the proxy IP as the IP for the media stream, which won’t work. I have no ALG device between the two that could be re-writing the SDP info. Just a dd-wrt router.
Sent from carrier:
[code]INVITE sip:+1XXXXXX5023@XXX.XXX.6.246:5060;transport=udp SIP/2.0
Record-Route:sip:216.82.224.202;lr;ftag=VPSF506071629460
Record-Route:sip:4.79.212.229;lr;ftag=VPSF506071629460
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKd74e.39bf3af1.0
Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bKd74e.27f9503.0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1206617507028
From:"XXXXXX"sip:+1XXXXXX1101@4.68.250.148;tag=VPSF506071629460
To:sip:+1XXXXXX5023@4.79.212.229:5060
Call-ID: PHXMGC0120080508140604008488@209.244.63.36
CSeq: 1 INVITE
Contact:sip:+1XXXXXX1101@4.68.250.148:5060;transport=udp
Max-Forwards: 67
Content-Type: application/sdp
Content-Length: 173
Remote-Party-ID:"XXXXXX"sip:+1XXXXXX1101@4.68.250.148;party=calling;screen=yes;privacy=off
v=0
o=- 1210255564 1210255565 IN IP4 63.215.26.215
s=-
c=IN IP4 63.215.26.215
t=0 0
m=audio 61754 RTP/AVP 0 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15[/code]
Received on PC:
INVITE sip:+1XXXXXX5023@XXX.XXX.6.246:5060;transport=udp SIP/2.0
Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>
Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460>
Via: SIP/2.0/UDP 216.82.224.202:5060;branch=z9hG4bKd74e.39bf3af1.0
Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bKd74e.27f9503.0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1206617507028
From: "XXXXXXX" <sip:+1XXXXXX1101@4.68.250.148>;tag=VPSF506071629460
To: <sip:+1XXXXXX5023@4.79.212.229:5060>
Call-ID: PHXMGC0120080508140604008488@209.244.63.36
CSeq: 1 INVITE
Contact: <sip:+1XXXXXX1101@216.82.224.202:5060;transport=udp>
Max-Forwards: 67
Content-Type: application/sdp
Content-Length: 174
Remote-Party-ID: "XXXXXX" <sip:+1XXXXXX1101@4.68.250.148>;party=calling;screen=yes;privacy=off
v=0
o=- 1210255564 1210255565 IN IP4 63.215.26.215
s=-
c=IN IP4 216.82.224.202
t=0 0
m=audio 61754 RTP/AVP 0 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Here is my sip.conf:
[code][general]
externhost=XXXXXXXXX
localnet=192.168.1.0/255.255.255.0
[bandwidth]
type=peer
host=216.82.224.202
context=incoming_calls
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=g729
deny=0.0.0.0/0
permit=216.82.224.202
insecure=invite
canreinvite=no
reinvite=no
nat=no[/code]