Call audio drops after 15 minutes, ended calls stay open

[ZorgNed - JDooge]

Since there have been some changes to our firewall/routing (I'm not exactly sure what changes; provisions were made to allow a 2nd internet connection with automatic failover) we have the following weirdness:

• Calls (doesn't matter if incoming or outgoing) will have audio dropped after 15 minutes, but stay active to our side. The other (external) party just has the call drop.
• Calls that are ended by the other (external) party will stay open for our users.

I've read about something like a 'SIP Refresh' or 'SIP Update' that checks every X time if a call is still active, and if not it will drop - but I can't find this in the protocol information, and our SIP trunk provider (Signet.nl) claims no such thing exists.

Internally there are no issues, only with external calls.

I hope to gain (at least read-only) access to the firewalls/routers soon (in maintenance by an external company) so I can have some insight in what is configured there. They are Sonicwall boxes and I've had trouble with that in combination with 3CX before (Sonicwall's smart SIP stuff can be interfering) - but until the recent change all worked without issues.

If someone has a helpful insight - please let me know!

 

[BrenttG]

Just going off of @JohnS_3CX comments, one other option, tho a bit costly, is SD-WAN. If you use a virtualized IP scheme such that your WAN External IP lives at the other end of the SD-WAN connection, then your IP would remain unchanged when a link dropped and another took over, and there would likely not even be any packetloss or VERY little because of how SD-WAN operates, but again, this is something for the most mission critical of environments, not something your average small to med enterprise or business is going to deploy.