[OpenSER-Users] Calls disconnect automatically

[Iñaki Baz Castillo]

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