Hi all,
I was wondering whether you could help us out here. We use TMG as a proxy and firewall for 3 years now, with our telephony solution, without any issues until recently.
Whenever a conversation is being estabilished I often now hear no sound from the PSTN into my CTI.
When I go and see the SIP negotiation, in the first calls I try to place, I get an invite with the IP of the TMG solution(shouldn't happen) and when this happens, no sound from PSTN to my workstation.
After two or three tries, the invite packet has the media server ip address, and now I get to hear sound in both directions correctly.
I show you the examples:
Not working:
11:35:22.498 V 7652 70 100 #default# INVITE sip:1228@192.18.0.116:5835 SIP/2.011:35:22.498 V 7652 70 100 #default# From: <sip:991234566@hostname:5060>;tag=20d06ba8-52001f0a-13d8-45026-73ab7e-470428ab-73ab7e
11:35:22.498 V 7652 70 100 #default# To: <sip:1228@hostname:5060>;tag=ada0688-740010ac-16cb-45026-2622-7c4c7110-2622
11:35:22.498 V 7652 70 100 #default# Call-ID: 1026f388-52001f0a-13d8-45026-73ab7e-1b88d43d-73ab7e
11:35:22.498 V 7652 70 100 #default# CSeq: 3 INVITE
11:35:22.498 V 7652 70 100 #default# Via: SIP/2.0/UDP :5060;received=ipaddress;branch=z9hG4bKmi!w_s!cwqGmi!w_s!cwqG0OiY4*wEqE-_WwGUYm4-Qmu*-.3-199072c8
11:35:22.498 V 7652 70 100 #default# Record-Route: <sip:19249689162017768909AOUD@sipproxyip;lr;dayaRRParam19015319001988772821>
11:35:22.498 V 7652 70 100 #default# Via: SIP/2.0/UDP b2bip:5080;branch=z9hG4bK-73ab85-c3d6033c-7bebf078
11:35:22.498 V 7652 70 100 #default# Max-Forwards: 69
11:35:22.498 V 7652 70 100 #default# Supported: timer,replaces
11:35:22.498 V 7652 70 100 #default# Contact: <sip:mychangendnumber@b2bip:5080>
11:35:22.498 V 7652 70 100 #default# Session-Expires: 21721;refresher=uac
11:35:22.498 V 7652 70 100 #default# Min-SE: 300
11:35:22.498 V 7652 70 100 #default# Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,UPDATE,MESSAGE,NOTIFY
11:35:22.498 V 7652 70 100 #default# Content-Type: application/sdp
11:35:22.498 V 7652 70 100 #default# Content-Length: 263
11:35:22.498 V 7652 70 100 #default#
11:35:22.498 V 7652 70 100 #default# v=0
11:35:22.498 V 7652 70 100 #default# o=OneMedia 1381411 3 IN IP4 192.168.60.2
11:35:22.498 V 7652 70 100 #default# s=Collab SDP
11:35:22.498 V 7652 70 100 #default# c=IN IP4 192.168.60.2
11:35:22.498 V 7652 70 100 #default# t=0 0
11:35:22.498 V 7652 70 100 #default# m=audio 16452 RTP/AVP 18 0 8 101
11:35:22.498 V 7652 70 100 #default# a=rtpmap:18 G729/8000
11:35:22.498 V 7652 70 100 #default# a=fmtp:18 annexb=no
11:35:22.498 V 7652 70 100 #default# a=rtpmap:0 PCMU/8000
11:35:22.498 V 7652 70 100 #default# a=rtpmap:8 PCMA/8000
11:35:22.498 V 7652 70 100 #default# a=rtpmap:101 telephone-event/8000
11:35:22.498 V 7652 70 100 #default# a=fmtp:101 0-15
Working example:
11:40:26.485 V 7652 69 100 #default# INVITE sip:1228@192.18.0.116:5835 SIP/2.011:40:26.485 V 7652 69 100 #default# From: <sip:mychangednumber@hostname:5060>;tag=20db60e8-52001f0a-13d8-45026-73acb0-6121f523-73acb0
11:40:26.485 V 7652 69 100 #default# To: <sip:1228@blablabla:5060>;tag=ada1f08-740010ac-16cb-45026-2753-286d9544-2753
11:40:26.485 V 7652 69 100 #default# Call-ID: 1021ba88-52001f0a-13d8-45026-73acb0-37d84935-73acb0
11:40:26.485 V 7652 69 100 #default# CSeq: 3 INVITE
11:40:26.485 V 7652 69 100 #default# Via: SIP/2.0/UDP 10.31.0.21:5060;received=10.31.0.31;branch=z9hG4bKmi!w_s!cwqGmi!w_s!cwqG0OiY4*wEqE2UWwGUYmWoYm4g8.3-1998d038
11:40:26.485 V 7652 69 100 #default# Record-Route: <sip:19249689162017768909AOUD@192.31.0.31;lr;dayaRRParam19015319001988772821>
11:40:26.485 V 7652 69 100 #default# Via: SIP/2.0/UDP b2bip:5080;branch=z9hG4bK-73acb5-c3daa6ae-13fcdcd7
11:40:26.485 V 7652 69 100 #default# Max-Forwards: 69
11:40:26.485 V 7652 69 100 #default# Supported: timer,replaces
11:40:26.485 V 7652 69 100 #default# Contact: <sip:mychangendnumber@b2bip:5080>
11:40:26.485 V 7652 69 100 #default# Session-Expires: 21721;refresher=uac
11:40:26.485 V 7652 69 100 #default# Min-SE: 300
11:40:26.485 V 7652 69 100 #default# Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,UPDATE,MESSAGE,NOTIFY
11:40:26.485 V 7652 69 100 #default# Content-Type: application/sdp
11:40:26.485 V 7652 69 100 #default# Content-Length: 261
11:40:26.485 V 7652 69 100 #default#
11:40:26.485 V 7652 69 100 #default# v=0
11:40:26.485 V 7652 69 100 #default# o=OneMedia 1381613 3 IN IP4 192.31.0.23
11:40:26.485 V 7652 69 100 #default# s=Collab SDP
11:40:26.485 V 7652 69 100 #default# c=IN IP4 192.31.0.23
11:40:26.485 V 7652 69 100 #default# t=0 0
11:40:26.485 V 7652 69 100 #default# m=audio 19730 RTP/AVP 18 0 8 101
11:40:26.485 V 7652 69 100 #default# a=rtpmap:18 G729/8000
11:40:26.485 V 7652 69 100 #default# a=fmtp:18 annexb=no
11:40:26.485 V 7652 69 100 #default# a=rtpmap:0 PCMU/8000
11:40:26.485 V 7652 69 100 #default# a=rtpmap:8 PCMA/8000
11:40:26.485 V 7652 69 100 #default# a=rtpmap:101 telephone-event/8000
11:40:26.485 V 7652 69 100 #default# a=fmtp:101 0-15
Note: I proposedly changed ip addresses here, but bolded the ips I would like you to see.
In the not working scenario the Ip that the software receives the sip packet is the TMG ip address(defaut GW, and Proxy IP address).
In the working scenario, I receive the ip address from the media server, which would be the correct scenario, and lets me hear sound.
It is clear to me that sip manipulation is being made, at TMG level, but I did no change to would make this happen.
How can I check whether it is some configuration issue, or a bug? TMG is fully updated. Maybe it may have started after the last update (it was working well at SP1 level).
the networking between TMG and the sip server, has had no configuration changes for more than a year. I see that the packet leaves the sip server ip the media server ip addres, and when arrives at the TMG, it changes the packet...
Thanks,
Nuno Silva