[SR-Users] Kamailio without sound, what to do?

[Klaus Darilion]

Am 26.07.2011 09:39, schrieb Henrik Aagaard Sørensen:
> I'm trying to setup a proxy and have installed rtpproxy.
> 
> Trying to start it via: rtpproxy -l _your_public_ip_ -s
> udp:localhost:7722 throws the warning:
> 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.
> 
> Will Kamailio accept WITH_NAT etc. if I run rtpproxy with -u as
> an unprivileged user?

Sure.

Klaus

> 
> On Tue, Jul 26, 2011 at 9:32 AM, Klaus Darilion
> <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists at pernau.at>> wrote:
> 
> 
> 
>     Am 26.07.2011 09 <tel:26.07.2011%2009>:23, schrieb Henrik Aagaard
>     Sørensen:
>     > I'm a newbee in the world of Kamailio.
>     >
>     > I've managed to setup a fresh installation of Kamailio, with
>     > authentication etc. (with help from some great guys on this
>     mailing-list).
>     >
>     > Everything seems to work with registers, calls etc. except that
>     there is
>     > no sound on my calls.
>     >
>     > How do I figure out what the problem is?
> 
>     If you want to figure out the problems yourself then a packet sniffer
>     (tcpdump, wireshark, ngrep) is your friend. Basically you watch out for
>     certain kind of packets which should be there, watch where they are sent
>     to, and if this is correct (compare with IP adresses signaled in SIP
>     payload).
> 
>     "No sound" is usually a NAT problem which can be solved by activating a
>     media relay (e.g. rtpproxy) and rewriting the SDP to route the media
>     stream via the rtpproxy.
> 
>     For message inspection I prefer ngrep:
> 
>     Just for the SIP traffic:
>     ngrep -d any -P "" -t -q -Wbyline port 5060
> 
>     For SIP traffic and RTP (usually both use UDP):
> 
>     ngrep -d any -P "" -t -q -Wbyline "" udp
> 
> 
>     Check out the SDP (body of INVITE and 200 OK) and verify if the IP
>     addresses signaled in c= line and port in m= line are correct (public
>     vs. private IP).
> 
>     Verify also if you see UDP/RTP packets sent by the SIP clients.
> 
>     If the clients are behind NAT, activate NAT traversal in the config:
>     define WITH_NAT (or similar)
> 
>     regards
>     Klaus
> 
> 
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> 
> 
> 
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