SIP phones and VPN channel - no registration, but there is ping, nttr and access to the server's web face. Why don't UDP protocols work?

Question

SIP

There are two routers connected by a PPTP channel, behind one server, behind another phone.
The soft phone cannot register on Asterisk, while the routes are transmitted, the server is pinged, only the SIP phone does not work.
The web face of that Elastic, like I open other services, works in the VPN channel and CRM too.
What and why cuts SIP?
The second server (test new with FreePbx), which also works perfectly with internal clients, is absolutely not visible from VPN for phones.
We go further: there is also a hardware phone, it connects via OpenVPN, it has a client.
The session lasts 6 seconds, during this time Asterisk's "Talking Clock" is spoken, but once, for 6 seconds, even a call goes outside, but no more, but the phone registration and voice are transmitted, but for 6 seconds.
You can’t sin on the serviceability of the phone, because the software ZSH, through the win Open VPN channel, behaves identically and also provides communication for 6 seconds :) :(
But that’s not all: the phone works with server 2 (new) via OpenVPN and does not 6 seconds, until you hang up, but you need something with the server 1.
Through PPTP, obf servers are not available for Sip protocol devices.Who
will tell you where to dig, what kind of shamanism?

Answer 1

Sorry, this is how it flies,
1. flies it registers, returns OK
2. flies sends everything to dial the number, too OK
3 flies and receives the sound of "talking hours", for the test, and external mobile, also OK, from the limit of 6 seconds, for some reason dialing time is excluded - it's strange.
And then suddenly nat appears and after 6 seconds a kirdyk comes, and after that it flies again and the phone is registered, a scoundrel is waiting to make fun :)

Answer 2

Look at the SIP packet size and MTU of your tunnel.
Perhaps in the dump you will see message retransmits that will indicate the problem.
Another extremely common case is when SIP ALG is enabled on the router, if you disable it and enable nat support in the asterisk, then some of these bugs will go away.