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

[Andrew Martin]

----- Original Message -----
> From: "Andrew Martin" <amartin at xes-inc.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
> Sent: Monday, May 11, 2015 4:18:58 PM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls	after 32 seconds
> 
> ----- Original Message -----
> > From: "Andrew Martin" <amartin at xes-inc.com>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > <asterisk-users at lists.digium.com>
> > Sent: Monday, May 11, 2015 1:35:07 PM
> > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> > calls	after 32 seconds
> >
> > > That should be all that is required. If that were broken I'd expect
> > > issue reports to implode - what's the configuration?
> > > 
> > 
> > Here's the sip.conf (only showing a single extension since they're all the
> > same):
> > [general]
> > directmedia=no
> > directrtpsetup=no
> > dtmfmode=rfc2833
> > context=asterisk-internal
> > allowsubscribe=no
> > qualify=no
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=gsm
> > localnet=10.10.32.0/255.255.248.0
> > localnet=192.168.32.0/255.255.255.0
> > 
> > [146]
> > secret=
> > host=dynamic
> > type=friend
> > 
> > From the aforementioned sip debug capture, 146 is on the 10.10.32.0/21
> > network
> > and 113 is on the 192.168.32.0/24 network (these are directly route-able so
> > no
> > NAT is involved). However, I have now been able to reproduce the problem
> > between
> > two devices directly on the 10.10.32.0/21 network as well.
> > 
> 
> I've gathered the log for this dialog from the SIP phone:
> http://pastebin.com/aAWs4j6i
> 
> What I see is that there's an INVITE for CSeq 103, then an OK for CSeq 103,
> then another INVITE is received for CSeq 103, at which point the phone
> reports an error:
> <0> | ERROR | receive a request with same cseq??
> 
> From the asterisk side, it never seems to receive this OK for CSeq 103, hence
> the reason it sends out the INVITE again.
> 
Joshua,

As a mitigation for this problem, could I increase the "timerb" option in sip.conf
to a large value, say 1 hour (instead of the default 32 seconds)? What other
consequences would there be from this change?

Thanks,

Andrew