I place a call from IP 82.68.150.81 using one of your SIP outbound trunks to a PSTN number (01275774114, also with you) destined for sip:01275774114@81.187.145.52. A summary of the SIP conversation:
82.68.150.81 -> 193.111.200.56 SIP INVITE with
SDP asking for RTP media to be sent to
82.68.150.81:14846 +++
193.111.201.32 -> 81.187.145.52 SIP INVITE with
SDP asking for RTP media to be sent to
193.111.201.32:14846 ***
81.187.145.52 -> 193.111.201.32 SIP OK with
SDP asking for RTP media to be sent to
81.187.145.52:15802
193.111.200.56 -> 82.68.150.81 SIP OK with
SDP asking for RTP media to be sent to
81.187.145.52:15802
[A complete dump of the SIP messages is below.]
The effect of this is that there is no audio in the 81.187.145.52 -> 82.68.150.81 direction, because the media address has somehow been changed in between the places marked +++ and ***.
Interestingly enough, doing the same in reverse (a call originated at 81.187.145.52, through a SIP outbound trunk to a PSTN number that terminates at sip:<blah>@82.68.150.81) works fine: the media address is not changed.
There's no NAT involved here, and (as best I can tell) no SIP ALGs. A direct call placed between the two endpoints (81.187.145.52 and 82.68.150.81) works fine.
I know it's in the nature of these problems that it's probably something to do with my setup, but I'm stumped! Can you think of anything in the set up of a Gradwell SIP outbound -> Gradwell PSTN call that would translate the SDP Media Description in the way shown above?
Thanks
Stephen
82.68.150.81 -> 193.111.200.56
INVITE sip:01275774114@sip.aisper.com SIP/2.0
Via: SIP/2.0/UDP 82.68.150.81:5060;branch=z9hG4bK13046c6e;rport
From: "02071832190" <sip:02071832190@82.68.150.81>;tag=as1250e114
To: <sip:01275774114@sip.aisper.com>
Contact: <sip:02071832190@82.68.150.81>
Call-ID: 0070abe913e931d5150fb0f20d5f60b7@82.68.150.81
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 20 Nov 2007 13:13:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 4372 4372 IN IP4 82.68.150.81
s=session
c=IN IP4 82.68.150.81
t=0 0
m=audio 14846 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
----------------------
193.111.201.32 -> 81.187.145.52
INVITE sip:01275774114@big.sock.org.uk SIP/2.0
Record-Route: <sip:193.111.201.32;lr=on;ftag=as1250e114>
Record-Route: <sip:193.111.200.56;ftag=as1250e114;lr=on>
Via: SIP/2.0/UDP 193.111.201.32:5060;branch=z9hG4bK691.86e82381.0
Via: SIP/2.0/UDP 193.111.200.56;branch=z9hG4bK691.33339d55.0
Via: SIP/2.0/UDP 82.68.150.81:5060;branch=z9hG4bK13046c6e;rport=5060
From: "02071832190" <sip:02071832190@82.68.150.81>;tag=as1250e114
To: <sip:01275774114@sip.aisper.com>
Contact: <sip:02071832190@193.111.201.32:5060>
Call-ID: 0070abe913e931d5150fb0f20d5f60b7@82.68.150.81
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 15
Date: Tue, 20 Nov 2007 13:13:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 4372 4372 IN IP4 82.68.150.81
s=session
c=IN IP4 193.111.201.32
t=0 0
m=audio 14846 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
----------------------
81.187.145.52 -> 193.111.201.32
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.111.201.32:5060;branch=z9hG4bK691.86e82381.0;received=193.111.201.32
Via: SIP/2.0/UDP 193.111.200.56;branch=z9hG4bK691.33339d55.0
Via: SIP/2.0/UDP 82.68.150.81:5060;branch=z9hG4bK13046c6e;rport=5060
Record-Route: <sip:193.111.201.32;lr=on;ftag=as1250e114>
Record-Route: <sip:193.111.200.56;ftag=as1250e114;lr=on>
From: "02071832190" <sip:02071832190@82.68.150.81>;tag=as1250e114
To: <sip:01275774114@sip.aisper.com>;tag=as74393e62
Call-ID: 0070abe913e931d5150fb0f20d5f60b7@82.68.150.81
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:01275774114@81.187.145.52>
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 31001 31001 IN IP4 81.187.145.52
s=session
c=IN IP4 81.187.145.52
t=0 0
m=audio 15802 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
----------------------
193.111.200.56 -> 82.68.150.81
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.68.150.81:5060;branch=z9hG4bK13046c6e;rport=5060
Record-Route: <sip:193.111.201.32;lr=on;ftag=as1250e114>
Record-Route: <sip:193.111.200.56;ftag=as1250e114;lr=on>
From: "02071832190" <sip:02071832190@82.68.150.81>;tag=as1250e114
To: <sip:01275774114@sip.aisper.com>;tag=as74393e62
Call-ID: 0070abe913e931d5150fb0f20d5f60b7@82.68.150.81
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:01275774114@81.187.145.52>
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 31001 31001 IN IP4 81.187.145.52
s=session
c=IN IP4 81.187.145.52
t=0 0
m=audio 15802 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -