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?