I'm trying to setup a proxy and have installed rtpproxy.<div><br></div><div>Trying to start it via: rtpproxy -l _your_public_ip_ -s udp:localhost:7722 throws the warning:</div><div>rtpproxy: running this program as superuser in a remote control mode is strongly not recommended, as it poses serious security threat to your system. Use -u option to run as an unprivileged user or -F is you want to run as a superuser anyway.</div>
<div><br></div><div>Will Kamailio accept WITH_NAT etc. if I run rtpproxy with -u as an <meta http-equiv="content-type" content="text/html; charset=utf-8">unprivileged user?<br><br><div class="gmail_quote">On Tue, Jul 26, 2011 at 9:32 AM, Klaus Darilion <span dir="ltr"><<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><br>
<br>
Am <a href="tel:26.07.2011%2009" value="+12607201109">26.07.2011 09</a>:23, schrieb Henrik Aagaard Sørensen:<br>
<div><div></div><div class="h5">> I'm a newbee in the world of Kamailio.<br>
><br>
> I've managed to setup a fresh installation of Kamailio, with<br>
> authentication etc. (with help from some great guys on this mailing-list).<br>
><br>
> Everything seems to work with registers, calls etc. except that there is<br>
> no sound on my calls.<br>
><br>
> How do I figure out what the problem is?<br>
<br>
</div></div>If you want to figure out the problems yourself then a packet sniffer<br>
(tcpdump, wireshark, ngrep) is your friend. Basically you watch out for<br>
certain kind of packets which should be there, watch where they are sent<br>
to, and if this is correct (compare with IP adresses signaled in SIP<br>
payload).<br>
<br>
"No sound" is usually a NAT problem which can be solved by activating a<br>
media relay (e.g. rtpproxy) and rewriting the SDP to route the media<br>
stream via the rtpproxy.<br>
<br>
For message inspection I prefer ngrep:<br>
<br>
Just for the SIP traffic:<br>
ngrep -d any -P "" -t -q -Wbyline port 5060<br>
<br>
For SIP traffic and RTP (usually both use UDP):<br>
<br>
ngrep -d any -P "" -t -q -Wbyline "" udp<br>
<br>
<br>
Check out the SDP (body of INVITE and 200 OK) and verify if the IP<br>
addresses signaled in c= line and port in m= line are correct (public<br>
vs. private IP).<br>
<br>
Verify also if you see UDP/RTP packets sent by the SIP clients.<br>
<br>
If the clients are behind NAT, activate NAT traversal in the config:<br>
define WITH_NAT (or similar)<br>
<br>
regards<br>
Klaus<br>
<br>
<br>
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</blockquote></div><br></div>