21 Jan
2008
21 Jan
'08
1:19 a.m.
El Lunes, 21 de Enero de 2008, VoIP Forums www.Go4Calls.com escribió:
i tired with the following configuration but still
result is same. calls
disconnect in 30 - 32 sec
modparam("nathelper", "natping_interval", 20)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "rtpproxy_sock",
"unix:/var/run/rtpproxy.sock")
modparam("nathelper", "rtpproxy_disable", 0)
modparam("nathelper", "rtpproxy_disable_tout", 60)
modparam("nathelper", "rtpproxy_tout", 1)
modparam("nathelper", "rtpproxy_retr", 5)
modparam("nathelper", "sipping_method", "OPTIONS")
modparam("nathelper", "received_avp", "$avp(i:801)")
Please advise me if i need more modification?
Which kind of calls are disconnected after 30 seconds? PSTN calls or user to
user call?
In any case, you could do a "tcpdump -n port UAC_RTP_PORT" in a PC using a
softphone that uses UAC_RTP_PORT for audio. Call to PSTN (or other user) from
this softphone and monitorize with tcpdump when the audio is disconnected.
Some gateways (as Asterisk) disconnect a call by default if they don't receive
RTP during 30 seconds.
Since I don't know which kind of gateway you use I don't know if it uses
Session Timers as call monitorization way, so if your router blocks the port
after 30 seconds, then the periodic ire-INVITE or UPDATE from gateway to UAC
will not arrive so they won't be replied with "200 OK", and gateway will
discconect the call.
To test this, do a "ngrep" in a computer using a softphone registered behind
NAT (no STUN). After REGISTER you should receive a OPTIONS from proxy as keep
alive.
Another possible problem is the existence of painful ALG routers, have you
tested if your router implements SIP ALG?
--
Iñaki Baz Castillo