Andrew Martin wrote:
----- 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?
The 200 OK should be retransmitted until an ACK is received. It honestly
looks like the phone can't talk to Asterisk and it's just generally
screwing up signaling.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
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