asterisk users - May 2015 - "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.

-- 
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

----- 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?

Andrew Martin wrote:
> ----- 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?
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