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-a... 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@gmail.com To: jdavidthomson@hotmail.com CC: sr-users@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@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@54.245.31.65;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: sip:daniel@54.245.31.65. Contact: sip:dave@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@54.245.31.65;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: sip:daniel@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@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@54.245.31.65;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: sip:daniel@54.245.31.65. Contact: sip:dave@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