Re: [asterisk-users] Issue with PJSIP contacts being "unavailable"

On Tue, Jun 27, 2023 at 6:57 PM <[email protected]> wrote:

> On 6/27/2023 7:29 AM, Joshua C. Colp wrote:
> > On Tue, Jun 27, 2023 at 8:22 AM <[email protected]
> > <mailto:[email protected]>> wrote:
> >
> > <snip>
> >
> >
> >     Trace from an "unavailable" ATA (not working correctly):
> >     https://paste.interlinked.us/iz07sapwrb.txt
> >
> >     Trace from an "available" ATA (working correctly):
> >     https://paste.interlinked.us/ocutyjslmg.txt
> >
> >
> > The "unavailable" ATA no longer has a working TLS connection to
> > Asterisk, resulting in OPTIONS failing, and going unreachable, and
> > eventually the Contact going away. Actually examining the TLS side
> > would be needed.
>
> Thanks, Josh. Further troubleshooting supports that as being the problem
> as well. I'll have to figure out what's changed with that.
>
> Replying to the dev list since this is not directly related, but it
> reminds me of a previous conversation we had about chan_pjsip
> automatically using the transport used for registration. This is not
> currently done; what would be your thoughts on perhaps adding an option
> to do this automatically? Currently, the provisioning system directs
> devices to the proper port based on the security options in the system
> and the TLS capabilities of the device. When something registers, I keep
> track of the port on which a device registers using AMI, logging it to a
> database. We have one port for UDP, one for TLS 1.0, and one for TLS 1.2
> (none for plain TCP at the moment). chan_sip isn't as flexible so the
> process is more straightforward there: just use the TLS 1.0 port if TLS
> is enabled, but for PJSIP, the transpiler assigns a transport based on
> the registration port. In theory, a client can toggle the transport for
> registration (TLS vs UDP), but that alone doesn't really work since
> pjsip.conf needs to be in agreement with that. It would be slightly more
> seamless if it could just sync up somehow, as right now I have to
> manually retranspile any time this occurs, and it seems kind of clunky
> to have to do all this for transports to work properly.
>
> Would there be any consideration or problem with potentially doing
> something like this? After all, it seems like there's a 1:1 mapping and
> it should be fairly straightforward. Kind of like the "line" option for
> registrations, it would help in making things "just work" more of the
> time...
>

I'm not sure what exactly you are referring to with "using the transport
used for registration". If "rewrite_contact" is set to yes then the
existing active connection should get used. If you are referring to
Asterisk establishing a new outgoing connection to the registered Contact,
then as long as it is optional and doesn't break other behavior fine.

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org

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