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

[Joshua Colp]

Andrew Martin wrote:
> ----- Original Message -----

<snip>

>
> 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.

Asterisk has sent a re-INVITE to have the media flow directly. The 
device (seems) to respond with the 200 OK (you can tell based on the 
CSeq) for the initial INVITE, and not for the re-INVITE. As Asterisk 
gets no response to its re-INVITE it gives up and terminates the dialog.

-- 
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