Hi all,
For a project on which I'm currently working, I am having some problems
figuring out how to correctly configure Kamailio to communicate with RTP
Proxy in order to send media into and out of a network with private IP
address ranges.
I have a proxy set up to send the SIP traffic, and all of this is
working fine. However, I'm having some trouble getting the RTP Proxy
set up. Currently, when the call is connected, the offer/answer is made
and RTP Proxy seems to be taking over, but I'm having trouble getting my
audio to flow in both directions.
Examination of the traffic coming into and out of this machine seems to
indicate that the IP addresses aren't being mangled correctly.
Specifically, it appears the internal IP address isn't being changed to
reflect the IP address of the machine on which RTP Proxy is running, so
that when the caller tries to send audio back, the IP it's given to
reply to is 10.10.x.x, which obviously won't work.
I have tried experimenting with specifically setting IP addresses in the
rtpproxy_offer() and _answer() methods to no avail, as well as setting
various flags in those methods. However, I must admit that I'm not
entirely sure what's happening under the hood with these methods, or
what rtpproxy is doing with that information when it gets it. Rather
than continue to hack at this by trial and error, I'm hoping someone
here can point me in the right direction.
Any advice, example code or pep talks would be greatly appreciated.
Thanks in advance,
--
Joe Hart
Voice Systems Integrator
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0673
Hi,
I'm getting started with kamailio and siremis. I've just finished getting
the server ready and the software installed.
Once in the siremis admin menu, i've added a domian and a subscriber and I
have my softphone registered with kamailio. My doubt is, where do I add
peers so I can set up the routing rules?
Are there any tutorials?
Thanks in advanced.
Best regards.
--
Joel Smith
Cell: +34 639 03 13 53
E-Mail: joel(a)vozelia.com
<joel(a)vozelia.com>
<http://www.vozelia.com> <http://twitter.com/vozelia>
<http://www.facebook.com/pages/Vozelia-Operador-de-telefonia-IP-para-empresa…>
Sorry, his name is Daniel Goepp.
-----Original Message-----
From: jdavidthomson(a)hotmail.com
Date: Sat, 15 Sep 2012 03:53:51
To: Daniel-Constantin Mierla<miconda(a)gmail.com>
Reply-To: jdavidthomson(a)hotmail.com
Cc: <sr-users(a)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(a)gmail.com>
Date: Tue, 11 Sep 2012 06:24:33
To: <jdavidthomson(a)hotmail.com>
Cc: <sr-users(a)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> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S.
To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> .
Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> .
Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl.
CSeq: 17407 INVITE.
Route: <sip:54.245.31.65;transport=udp;lr> <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> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S.
To: <sip:daniel@54.245.31.65> <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> <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> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S.
To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> .
Contact: <sip:dave@207.219.69.217:40821;ob> <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.comhttp://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
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(a)gmail.com>
Date: Tue, 11 Sep 2012 06:24:33
To: <jdavidthomson(a)hotmail.com>
Cc: <sr-users(a)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> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S.
To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> .
Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> .
Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl.
CSeq: 17407 INVITE.
Route: <sip:54.245.31.65;transport=udp;lr> <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> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S.
To: <sip:daniel@54.245.31.65> <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> <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> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S.
To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> .
Contact: <sip:dave@207.219.69.217:40821;ob> <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.comhttp://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
Hi,
Is there a module that can be used to collect number of MESSAGEs sent
or received by a subscriber;
number of registrations by a subscriber? These will be per subscriber
as oppose to statistics collected for
entire system.
Thanks
Krish Kura