Hi Joshua,
Looking at traces, there are packets being sent with length 4. These show as PSH, ACK in a Wireshark. However, in TLS I don't see any such packets which I believe is the cause of the TLS connection is being closed. Can you confirm if Asterisk does send these packets on TLS transports? Kind regards, Ross ________________________________ From: [email protected] <[email protected]> on behalf of Joshua Colp <[email protected]> Sent: 09 November 2016 17:48 To: [email protected] Subject: Re: [asterisk-dev] Chan_pjsip keep_alive_interval On Wed, Nov 9, 2016, at 01:25 PM, Ross Beer wrote: > Hi, > > > I'm investigating an issue where TLS connections close with a 'RST' after > a random period of time. > > > I can see that PJSIP sets 'PJSIP_TRANSPORT_IDLE_TIME=600', with the > option in pjsip.conf 'keep_alive_internal' set, does this set both > 'PJSIP_TCP_KEEP_ALIVE_INTERVAL' and 'PJSIP_TLS_KEEP_ALIVE_INTERVAL'? The keep_alive_interval option doesn't set those in PJSIP. It controls the interval at which code in Asterisk (not PJSIP) will send a keepalive. There is no expectation that a response is received, as it does not generate a SIP request itself. It allows runtime control instead of compile time control. > > > Does a keep-alive packet actually reset 'PJSIP_TRANSPORT_IDLE_TIME' if a > response is received? If no response received, how many attempts are made > before asterisk disconnects the session? It does not reset the timer locally. Its purpose is to ensure the remote side does not disconnect us for being idle. If we received a message then our local idle timer would be reset. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com<http://www.digium.com> & www.asterisk.org<http://www.asterisk.org> Asterisk custom communications - PBX, VoIP gateways, IVRs ...<http://www.asterisk.org/> www.asterisk.org Asterisk: an open source framework that lets you build communications applications for IP PBX, VoIP gateways, conference servers and custom phone apps [https://www.digium.com/sites/digium/themes/digium/logo.png]<http://www.digium.com/> Business Phone Systems | Unified Communications | Digium<http://www.digium.com/> www.digium.com Digium offers full Unified Communications solutions with on-premises and hosted business phone systems, IP phones, and Asterisk hardware. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- api digital - problem solved.<http://www.api-digital.com/> www.api-digital.com API Digital Website asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
