[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: Friday, May 8, 2015 5:12:28 PM
> Subject: [asterisk-users] "Retransmission Timeout" results in dropped calls	after 32 seconds
> 
> Hello,
> 
> I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All
> the SIP clients are on a LAN, so no NAT is involved. I have been experiencing
> an intermittent problem where a call will be successfully answered, but then
> dropped by Asterisk 32 seconds after it is answered (with a "Retransmission
> timeout reached on transmission" error). Here is an example of this happening
> in the asterisk console:
> http://pastebin.com/7LDwHAJe
> 
> This problem only happens a fraction of the time, so I have been unable to
> enable SIP debugging before it happens to get a capture. However, usually the
> caller will just call back immediately and then the call will work without a
> problem. It sounds like SIP Timer B is what causes the call to be dropped if
> an
> ACK to the INVITE is not received within 32 seconds. How can I determine if
> this is the case and how can I resolve this "Retransmission timeout" problem?
> 
> Here is my sip.conf:
> general]
> directmedia=no
> directrtpsetup=no
> dtmfmode=rfc2833
> context=internal
> allowsubscribe=no
> qualify=no
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> localnet=10.10.32.0/255.255.248.0
> 
> 
> [123]
> secret=111111
> host=dynamic
> type=friend
> 

By doing a number of test calls today, I have managed to reproduce this while
sip debugging was on, so I have that information available now as well:
http://pastebin.com/ZJqzdvY3

This was a call from 113 to 146 via a queue. Note that the asterisk server is
at 10.10.32.251. I see the following:
INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
SIP/2.0 180 Ringing
SIP/2.0 180 Ringing
SIP/2.0 200 OK
ACK sip:146 at 10.10.32.96:5062 SIP/2.0
INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
SIP/2.0 200 OK
ACK sip:146 at 10.10.32.96:5062 SIP/2.0
INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
INVITE sip:146 at 10.10.32.96:5062 SIP/2.0

This appears to start out with a successful SIP conversation (ending with the
first ACK), so it is unclear to me why we have two new sets of INVITEs sent
afterwards. 

Also in case it is relevant, the asterisk server has two NICs set up in a bond
with bond-mode 1 (active/backup).

Does this additional debug information provide any clues to why this 
intermittent "retransmission timeout" error is occurring?

Thanks,

Andrew