----- Original Message ----- > From: "Joshua Colp" <jc...@digium.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Wednesday, May 13, 2015 10:10:25 AM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped > calls after 32 seconds > > Andrew Martin wrote: > > ----- Original Message ----- > > <snip> > > > > > > > Most noteworthy is that the phone seems to send the OK for cseq 103, but it > > seems that the asterisk server never received this OK, which is why it kept > > re-transmitting the INVITE (103). Is this OK supposed to go to the asterisk > > server, or to the other phone? If it is supposed to go to the asterisk > > server, > > I suppose the explanation could be network turbulence prevented this OK > > from > > getting back to the server - does this seem like what happened? If so, what > > should be happening differently to ensure that this call doesn't get > > dropped? > > The traffic is between the phone and Asterisk. As to why, I have no > idea. The packets aren't getting to Asterisk - that's all I can say. I > doubt it's network turbulence. Likely getting lost/blocked somewhere. > Since some packet loss is a possibility, I assume the protocol has mechanisms for dealing with it. What should be happening differently in the communication when packet loss occurs? Should the phone just be re-sending the OK, instead of printing "<0> | ERROR | receive a request with same cseq??" to its log? Or should Asterisk be starting with a new cseq on each INVITE retry?
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