[asterisk-dev] Maximum retries exceeded on transmission issue.

[Johansson Olle E]

25 jun 2008 kl. 03.21 skrev Stuart Elvish:

> Hi Johan,
>
> Thanks for the pointer. I missed your bug during my searching. You
> also bought my attention to the fact that I think the only time it
> happens is when there is more than 1 call in progress. Do you know
> what sort of equipment your provider uses? The SIP debug on our side
> showed up AudiocodesGW which is on the providers side of the equation.
> And, did the provider let you know what they did to fix the problem?
>
> The provider I am working with says that they have found the problem
> and they will try a fix today so we will see if that works.
>
> Please excuse the ignorance Kevin, but why can't Asterisk be modified
> to, under certain conditions and with the appropriate flag set (as
> some have suggested in sip.conf), override this condition so it is no
> longer critical?

As Kevin said, the INVITE-200OK-ACK sequence, the "three way handshake"
is essential to SIP over UDP. Without it, you would get many more  
problems
since the ACK is used just to confirm that all settings in the 200 OK  
are ok,
and that we have a signalling path that works for the rest of the SIP  
session.
If you don't get an ACK, then you won't get other important SIP messages
during the session either.

A configuration option to support software that is that broken doesn't
make sense, since it doesn't fix anything, just causes more issues.

Your connection is broken. Fix it.

/O