[Users] Audio problem in NAT

Stephen Varga varga at zbzoom.net
Wed Feb 21 17:51:38 CET 2007


Raviprakash,

Your SDP messages are using private IPs for the RTP stream which carries 
the voice traffic. This causing the media stream to be sent to the wrong 
IP address, thus no audio.

Have you read through and tried to follow the directions in these documents?

    *
      */OpenSER & NAT/*
          o
            Run RTPproxy on a remote host
            <http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy>
          o
                http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy
            <http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy>
            <http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy>
          o
            OpenSER and RTPProxy
            <http://voip-info.org/wiki/view/OpenSER+And+RTPProxy>
          o
                __
            <http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy>_http://voip-info.org/wiki/view/OpenSER+And+RTPProxy
            <http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy>_
          o
            OpenSER and Mediaproxy
            <http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy>
          o
                http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy

A lot of good documentation has been written up on the OpenSER 
application which can be found at this website.

    http://openser.org/dokuwiki/doku.php

raviprakash sunkara wrote:
> Hello Users,
>
> I posted  so many mail to users but no one reply my issue please  
> help  me
>
> openSER proxy is mysipdomain.com , and its private_ ip is 192.168.2.60 
> and
> SIP server and Proxy is also in Behind
> UAC's are Behind the NATs
>
> --------------------- INVITE ---------------
> U 59.144.88.7:5060 -> 192.168.2.60:5060
> INVITE sip:9002 at mysipdomain.com;user=phone SIP/2.0.
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKe2a540a8170eb12a.
> From: Indian-2 <sip:8002 at mysipdomain.com;user=phone>;tag=4240982537.
> To: <sip:9002 at mysipdomain.com;user=phone>.
> Call-ID: 1685867393 at 192.168.1.2.
> CSeq: 1 INVITE.
> Contact: Indian-2 <sip:8002 at 192.168.1.2:5060;user=phone;transport=udp>.
> User-Agent: Cisco ATA 188  v3.2.1 atasip (050616A).
> Expires: 300.
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
> UPDATE.
> Supported: 100rel,replaces.
> Content-Length: 245.
> Content-Type: application/sdp.
> v=0.
> *o=8002 14279 14279 IN IP4 _192.168.1.2._*_ _
> s=ATA186 Call.
> c=IN IP4 192.168.1.2.
> t=0 0.
> m=audio 16386 RTP/AVP 0 4 8 101.
> a=rtpmap:0 PCMU/8000/1.
> a=rtpmap:4 G723/8000/1.
> a=rtpmap:8 PCMA/8000/1.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
>
> #
> U 192.168.2.60:5060 -> 61.17.248.68:3186
> INVITE sip:9002 at 192.168.2.7:5060;user=phone;transport=udp SIP/2.0.
> Max-Forwards: 10.
> Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
> Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0.
> Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
> ;branch=z9hG4bKbd027751c869b9ff.
> From: Indian-2 <sip:8002 at mysipdomain.com;user=phone>;tag=4240982537.
> To: <sip:9002 at mysipdomain.com;user=phone>.
> Call-ID: 1685867393 at 192.168.1.2.
> CSeq: 2 INVITE.
> Contact: Indian-2 <sip:8002 at 59.144.88.7:5060;user=phone;transport=udp>.
> User-Agent: Cisco ATA 188  v3.2.1 atasip (050616A).
> Expires: 300.
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
> UPDATE.
> Supported: 100rel,replaces.
> Content-Length: 265.
> Content-Type: application/sdp.
> v=0.
> o=8002 14329 14329 IN IP4 192.168.1.2.
> s=ATA186 Call.
> c=IN IP4 192.168.1.2.
> t=0 0.
> m=audio 16386 RTP/AVP 0 4 8 101.
> a=rtpmap:0 PCMU/8000/1.
> a=rtpmap:4 G723/8000/1.
> a=rtpmap:8 PCMA/8000/1.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=direction:active.
>
> ------------------- RINGING------------
> U 61.17.248.68:3186 -> 192.168.2.60:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0.
> Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
> ;branch=z9hG4bKbd027751c869b9ff.
> Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
> From: Indian-2 <sip:8002 at mysipdomain.com;user=phone>;tag=4240982537.
> To: <sip:9002 at mysipdomain.com;user=phone>;tag=1332822912.
> Call-ID: 1685867393 at 192.168.1.2.
> CSeq: 2 INVITE.
> Require: 100rel.
> RSeq: 1.
> Contact: 9002 <sip:9002 at 192.168.2.7:5060;user=phone;transport=udp>.
> Server: Cisco ATA 188  v3.2.1 atasip (050616A).
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
> UPDATE.
> Content-Length: 0.
> .
>
> #
> U 192.168.2.60:5060 -> 59.144.88.7:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
> ;branch=z9hG4bKbd027751c869b9ff.
> Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
> From: Indian-2 <sip:8002 at mysipdomain.com;user=phone>;tag=4240982537.
> To: <sip:9002 at mysipdomain.com;user=phone>;tag=1332822912.
> Call-ID: 1685867393 at 192.168.1.2.
> CSeq: 2 INVITE.
> Require: 100rel.
> RSeq: 1.
> Contact: 9002 <sip:9002 at 61.17.248.68:3186;user=phone;transport=udp>.
> Server: Cisco ATA 188  v3.2.1 atasip (050616A).
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
> UPDATE.
> Content-Length: 0.
>
> U 61.17.248.68:3186 -> 192.168.2.60:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0.
> Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
> ;branch=z9hG4bKbd027751c869b9ff.
> Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
> From: Indian-2 <sip:8002 at mysipdomain.com;user=phone>;tag=4240982537.
> To: <sip:9002 at mysipdomain.com;user=phone>;tag=1332822912.
> Call-ID: 1685867393 at 192.168.1.2.
> CSeq: 2 INVITE.
> Contact: 9002 <sip:9002 at 192.168.2.7:5060;user=phone;transport=udp>.
> Server: Cisco ATA 188  v3.2.1 atasip (050616A).
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
> UPDATE.
> Supported: replaces.
> Content-Length: 193.
> Content-Type: application/sdp.
> .
> v=0.
> *o=9002 27865 27865 IN IP4 _192.168.2.7_*_. _
> s=ATA186 Call.
> c=IN IP4 192.168.2.7.
> t=0 0.
> m=audio 16386 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000/1.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
>
> #
> U 192.168.2.60:5060 -> 59.144.88.7:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
> ;branch=z9hG4bKbd027751c869b9ff.
> Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
> From: Indian-2 <sip:8002 at mysipdomain.com;user=phone>;tag=4240982537.
> To: <sip:9002 at mysipdomain.com;user=phone>;tag=1332822912.
> Call-ID: 1685867393 at 192.168.1.2.
> CSeq: 2 INVITE.
> Contact: 9002 <sip:9002 at 61.17.248.68:3186;user=phone;transport=udp>.
> Server: Cisco ATA 188  v3.2.1 atasip (050616A).
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
> UPDATE.
> Supported: replaces.
> Content-Length: 193.
> Content-Type: application/sdp.
> .
> v=0.
> o=9002 27865 27865 IN IP4 192.168.2.7.
> s=ATA186 Call.
> c=IN IP4 192.168.2.7.
> t=0 0.
> m=audio 16386 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000/1.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
>
> #
> U 61.17.248.68:3186 -> 192.168.2.60:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0.
> Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
> ;branch=z9hG4bKbd027751c869b9ff.
> Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
> From: Indian-2 <sip:8002 at mysipdomain.com;user=phone>;tag=4240982537.
> To: <sip:9002 at mysipdomain.com;user=phone>;tag=1332822912.
> Call-ID: 1685867393 at 192.168.1.2.
> CSeq: 2 INVITE.
> Contact: 9002 <sip:9002 at 192.168.2.7:5060;user=phone;transport=udp>.
> Server: Cisco ATA 188  v3.2.1 atasip (050616A).
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
> UPDATE.
> Supported: replaces.
> Content-Length: 193.
> Content-Type: application/sdp.
> .
> v=0.
> o=9002 27865 27865 IN IP4 192.168.2.7.
> s=ATA186 Call.
> c=IN IP4 192.168.2.7.
> t=0 0.
> m=audio 16386 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000/1.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
>
> #
> U 192.168.2.60:5060 -> 59.144.88.7:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
> ;branch=z9hG4bKbd027751c869b9ff.
> Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
> From: Indian-2 <sip:8002 at mysipdomain.com;user=phone>;tag=4240982537.
> To: <sip:9002 at mysipdomain.com;user=phone>;tag=1332822912.
> Call-ID: 1685867393 at 192.168.1.2.
> CSeq: 2 INVITE.
> Contact: 9002 <sip:9002 at 61.17.248.68:3186;user=phone;transport=udp>.
> Server: Cisco ATA 188  v3.2.1 atasip (050616A).
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
> UPDATE.
> Supported: replaces.
> Content-Length: 193.
> Content-Type: application/sdp.
> v=0.
> o=9002 27865 27865 IN IP4 192.168.2.7.
> s=ATA186 Call.
> c=IN IP4 192.168.2.7.
> t=0 0.
> m=audio 16386 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000/1.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
>
> ------------------------------------------------------------------------
>
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> Users at openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>   

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