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

David Thomson jdavidthomson at hotmail.com
Sat Sep 15 05:56:07 CEST 2012 

Sorry, his name is Daniel Goepp. 
-----Original Message-----
From: jdavidthomson at hotmail.com
Date: Sat, 15 Sep 2012 03:53:51 
To: Daniel-Constantin Mierla<miconda at gmail.com>
Reply-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

I got it working. The advertise parameter wasn't available in the initial versions of rtpproxy I was using. A guy named daniel geopp has a patched version that enables advertisement of the public ec2 ip. 
Thanks for your help :)
Ttyl,
Dave
-----Original Message-----
From: Daniel-Constantin Mierla <miconda at gmail.com>
Date: Tue, 11 Sep 2012 06:24:33 
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: &lt;sip:dave at 54.245.31.65&gt; <sip:dave at 54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. 
To: &lt;sip:daniel at 54.245.31.65&gt; <sip:daniel at 54.245.31.65> . 
Contact: &lt;sip:dave at 207.219.69.217:40821;ob&gt; <sip:dave at 207.219.69.217:40821;ob> . 
Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. 
CSeq: 17407 INVITE. 
Route: &lt;sip:54.245.31.65;transport=udp;lr&gt; <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: &lt;sip:dave at 54.245.31.65&gt; <sip:dave at 54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. 
To: &lt;sip:daniel at 54.245.31.65&gt; <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: &lt;sip:54.245.31.65;lr=on;nat=yes&gt; <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: &lt;sip:dave at 54.245.31.65&gt; <sip:dave at 54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. 
To: &lt;sip:daniel at 54.245.31.65&gt; <sip:daniel at 54.245.31.65> . 
Contact: &lt;sip:dave at 207.219.69.217:40821;ob&gt; <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

More information about the sr-users mailing list