Answered TMG performing SIP manipulation?

  • Wednesday, January 23, 2013 12:00 PM
     
     

    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/2.0
    11:35:22.498 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/2.0
    11:40:26.485 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



All Replies

  • Thursday, January 24, 2013 6:48 AM
    Moderator
     
     

    Hi,

    Thank you for the post.

    As far as I understand, one-way audio needs to be discussed with your ITSP provider.

    Regards,


    Nick Gu - MSFT

  • Monday, January 28, 2013 10:18 AM
     
     Answered

    Hi Nick,

    Actually this was not solved with the ITSP provider. After I have had applied latest software updates on TMG, for some reason it started perfoming SIP manipulation, and the purpose that TMG is in our IS is not for SIP manipulation. I had SIP Access filter enabled, and after having disabled it, our problem got solved.

    Thanks anyway.


    Nuno Silva