asterisk users - May 2015 - "Retransmission Timeout" results in dropped calls after 32 seconds

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

----- 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 10:50:02 AM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in
dropped calls	after 32 seconds
> 
> Andrew Martin wrote:
> > 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.
> 
Thanks for the clarification and help debugging this problem. I will work
with the phone vendor to see if they can resolve this from their end. If you
have any other ideas about how to disable re-INVITEs on the asterisk side,
beyond what I have done already, please let me know.

Thanks,

Andrew

Hi,

In my experience, all Yealink phones work just fine with Asterisk, we have
hundreds (perhaps even low-thousands) out there with customers on Asterisk
1.2, 1.6.2, 1.8 and 11.

If you are accurately representing the SIP trace on the phone and the SIP
trace on Asterisk, then I would strongly suggest a SIP ALG exists in the
network between the two devices and that SIP ALG does not understand SIP
properly. The two halves simply do not match, so something must surely be
interfering.

In my experience it is often an innocent looking Cisco router. Cisco's SIP
implementation is "SIP By Cisco" rather than "RFC compliant
SIP". If that
is the case Cisco call it a "SIP fixup" and you just need to disable
it.

Hope that helps,
Steve


On Wed, 13 May 2015 at 16:59 Andrew Martin <amartin at xes-inc.com> 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 10:50:02 AM
> > Subject: Re: [asterisk-users] "Retransmission Timeout"
results in
> dropped calls       after 32 seconds
> >
> > Andrew Martin wrote:
> > > 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.
> >
>
> Thanks for the clarification and help debugging this problem. I will work
> with the phone vendor to see if they can resolve this from their end. If
> you
> have any other ideas about how to disable re-INVITEs on the asterisk side,
> beyond what I have done already, please let me know.
>
> Thanks,
>
> Andrew
>
> --
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