[SR-Users] using kamailio with clients in a nat environment

Wed Sep 12 04:55:06 CEST 2012

Hi Daniel,
I do have the second param set to the public ip of the ec2 instance.  Thanks for that. :)
I also tried the debugger but that didn't show me anything specific but there were some errors in the output.  
I'm a little green on this subject so it might be out of my reach to put this together.  
I did see some notes about the zrtp hash so at least I know that the endpionts are trying to negotiate.  And I know that it is a one way audio thing so it is probably nat related.
Am going to do some more reading over the next few days to see if I missed something in the config.  I used this document to configure the nat traversal:http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-and-rtpproxy-server 
Is it still relevant with version 3.3?
thanks for your help!!
ttyl,Dave
Date: Tue, 11 Sep 2012 08:24:33 +0200
From: miconda at gmail.com
To: jdavidthomson at hotmail.com
CC: sr-users at lists.sip-router.org
Subject: Re: [SR-Users] using kamailio with clients in a nat environment

    Hello,

    the sdp does not show that rtpproxy was engaged. Check your config,
    you can use debugger module with cfgtrace on to see what actions are
    executed.

    Also, probably you have to advertise the public ip address of your
    ec2 instance -- see second parameter for rtpproxy module functions.

    Cheers,

    Daniel

    On 9/11/12 3:52 AM, David Thomson
      wrote:

        Hi,

        I'm using rtpproxy and per the documentation:

          rtpproxy -l public_ip -s udp:localhost:22222 -F

          Attached is the following:
          Dave registering
          Daniel registering
          Dave calling Daniel, where Dave has a public IP and
            Daniel is behind a nat.  

          Please let me know what you think is up.  

          ttyl,
          Dave

          [...]

          #
          U 207.219.69.217:40821 -> 10.248.96.110:5060
          INVITE sip:daniel at 54.245.31.65 SIP/2.0.
          Via: SIP/2.0/UDP
10.207.158.89:51362;rport;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF.
          Max-Forwards: 70.
          From:
            <sip:dave at 54.245.31.65>;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S.
          To: <sip:daniel at 54.245.31.65>.
          Contact: <sip:dave at 207.219.69.217:40821;ob>.
          Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl.
          CSeq: 17407 INVITE.
          Route: <sip:54.245.31.65;transport=udp;lr>.
          Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,
            SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS.
          Supported: replaces, 100rel, timer, norefersub.
          Session-Expires: 1800.
          Min-SE: 90.
          User-Agent: CSipSimple_SGH-T989D-15/r1841.
          Content-Type: application/sdp.
          Content-Length:   425.
          .
          v=0.
          o=- 3556317015 3556317015 IN IP4 10.207.158.89.
          s=pjmedia.
          t=0 0.
          m=audio 4008 RTP/AVP 96 3 0 8 101.
          c=IN IP4 10.207.158.89.
          a=rtcp:4009 IN IP4 10.207.158.89.
          a=sendrecv.
          a=rtpmap:96 SILK/8000.
          a=fmtp:96 useinbandfec=0.
          a=rtpmap:3 GSM/8000.
          a=rtpmap:0 PCMU/8000.
          a=rtpmap:8 PCMA/8000.
          a=rtpmap:101 telephone-event/8000.
          a=fmtp:101 0-15.
          a=zrtp-hash:1.10
            0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289.

          #
          U 10.248.96.110:5060 -> 207.219.69.217:40821
          SIP/2.0 100 trying -- your call is important to us.
          Via: SIP/2.0/UDP
10.207.158.89:51362;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF;received=207.219.69.217.
          From:
            <sip:dave at 54.245.31.65>;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S.
          To: <sip:daniel at 54.245.31.65>.
          Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl.
          CSeq: 17407 INVITE.
          Server: kamailio (3.3.0 (x86_64/linux)).
          Content-Length: 0.
          .

          #
          U 10.248.96.110:5060 -> 75.119.228.57:5060
          INVITE
            sip:daniel at 192.168.1.102:5060;transport=udp;registering_acc=54_245_31_65
            SIP/2.0.
          Record-Route: <sip:54.245.31.65;lr=on;nat=yes>.
          Via: SIP/2.0/UDP
            54.245.31.65:5060;branch=z9hG4bKd6f6.b6db86e5.0.
          Via: SIP/2.0/UDP
10.207.158.89:51362;received=207.219.69.217;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF.
          Max-Forwards: 69.
          From:
            <sip:dave at 54.245.31.65>;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S.
          To: <sip:daniel at 54.245.31.65>.
          Contact: <sip:dave at 207.219.69.217:40821;ob>.
          Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl.
          CSeq: 17407 INVITE.
          Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,
            SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS.
          Supported: replaces, 100rel, timer, norefersub.
          Session-Expires: 1800.
          Min-SE: 90.
          User-Agent: CSipSimple_SGH-T989D-15/r1841.
          Content-Type: application/sdp.
          Content-Length:   425.
          .
          v=0.
          o=- 3556317015 3556317015 IN IP4 10.207.158.89.
          s=pjmedia.
          t=0 0.
          m=audio 4008 RTP/AVP 96 3 0 8 101.
          c=IN IP4 10.207.158.89.
          a=rtcp:4009 IN IP4 10.207.158.89.
          a=sendrecv.
          a=rtpmap:96 SILK/8000.
          a=fmtp:96 useinbandfec=0.
          a=rtpmap:3 GSM/8000.
          a=rtpmap:0 PCMU/8000.
          a=rtpmap:8 PCMA/8000.
          a=rtpmap:101 telephone-event/8000.
          a=fmtp:101 0-15.
          a=zrtp-hash:1.10
            0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289.

    -- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu 		 	   		  
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