asterisk users - May 2015 - "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.

Thanks,

Andrew

----- 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

Andrew Martin wrote:

<snip>
>>
> 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?
I don't know if chan_sip will allow this, but if it does... it'll keep 
transmitting over and over... it would be better to get to the bottom of 
the problem. Do a packet capture on the machine running Asterisk and see 
where the packet goes. That's the only thing left really. It's also 
possible something got fixed in relation to directmedia between your 
version and latest 11.

-- 
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