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

[Steve Davies]

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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