[asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds

[Andrew Martin]

----- Original Message -----
> From: "Joshua Colp" <jcolp at digium.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at 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?