2 Jul
2010
2 Jul
'10
1:17 a.m.
On 1 July 2010 23:03, Dmitri Korotkov <dmitri.korotkov(a)festart.ee> wrote:
Hi,
default kamailio config file(its routing part) already has rtpproxy support
in case if WITH_NAT is defined.
And there is no problems when NATed subscribers calls one to other...
I have problem only with PSTN and only with incoming call.
BR,
Dmitri
I don't think rtpproxy is being engaged in your inbound route.
02.07.2010 0:51, dotnetdub пишет:
On 1 July 2010 22:41, Dmitri Korotkov <dmitri.korotkov(a)festart.ee> wrote:
Hi,
voice:/# ps auxf |grep rtpproxy |grep -v grep
rtpproxy 1291 0.0 0.0 26800 876 ? Ssl Jun18 0:10
/usr/sbin/rtpproxy -u rtpproxy rtpproxy -l my.public.ip.here -s
udp:localhost 7722
voice:/#
kamailio.cfg:
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_ACCDB
#!define WITH_NAT
#!define WITH_PSTN
#!ifdef WITH_NAT
loadmodule "nathelper.so"
#!endif
# ----- nathelper -----
#!ifdef WITH_NAT
modparam("nathelper", "rtpproxy_sock",
"udp:127.0.0.1:7722")
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", 7)
modparam("nathelper", "sipping_from",
"sip:pinger@kamailio.org"<sip:pinger@kamailio.org>
)
modparam("registrar|nathelper", "received_avp",
"$avp(i:80)")
modparam("usrloc", "nat_bflag", 6)
#!endif
02.07.2010 0:32, dotnetdub пишет:
I'm not overly familiar with rtpproxy as we use mediaproxy but you
will
need to engage it somewhere in your script, are you doing that?
Take a look at
http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy
Can you see any rtpproxy messages in syslog?
On 1 July 2010 21:53, Dmitri Korotkov
<dmitri.korotkov(a)festart.ee>wrote;wrote:
Hello,
I have kamailio installation WITH_PSTN, WITH_NAT and rtpproxy.
Using following scenario:
[kamailio]<-sip trunk ->[asterisk gw] <->sip trunk <-> [PSTN provider]
All kamailio sip subscribers are behind nat in different networks.
1. OK. Local kamailio users can call one to other even they are on
different networks behind nat.
2. OK. Outgoing calls from kamailio users to PSTN work also very well.
3. Not OK. Incoming from PSTN side calls have only one way audio.
I tcpdump'ed kamailio box and found, that pstn provider sends RTP packets
to kamailio IP in case of answered call.
I guess that rtpproxy is not active in case of pstn call. Is it true ?
I am using more less default kamailio config
Could you please suggest solution ?
BR,
Dmitri
Hi Dmitri,
Check out the nathelper module.
Regards,
Brian