On 2016-08-09 10:06, Faheem Muhammad wrote:
Jacek, This might be a bug or configuration issue, but you need to understand the SIP Session Timers. With Session Timers you can control the round trip time and Call Setup time of SIP Requests.
I don't think you really mean SIP Session Timers (https://tools.ietf.org/html/rfc4028) these do not affect RTT or call setup, but provide kind of 'keepalive' and session expiration for established calls.
In case of GSM Network with high delay you need to set the T1 timer a higher value like 1000ms (500 ms default). Similarly you can reduce the Call setup time by configuring 'T2' upto you choice as per you telephony network. Configure t1min, timert1 and timerb according to your network. Also set session-type=uas.
Yes, tweaking the T1 and T2 timers may work for me. I'll try that, though the old 'qualify' magic with chan_sip was quite convenient. I wonder why it doesn't work with chan_pjsip.
Jacek -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
