26 Jul
2011
26 Jul
'11
3:32 a.m.
Am 26.07.2011 09:23, 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