Calls unexpectedly terminated

[vocdot]

Hi,
I'm using FusionPBX with Freeswitch 1.0.2 and SipJS 0.20.
Some calls terminate unexpectedly after 2, 5, 7 or more minutes.

The BYE SIP message show a Normal clearing reason:

| sip.Transport | Received WebSocket text message:
BYE sip:21@m37bo843abrm.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WSS 104.154.17.139:7443;rport;branch=z9hG4bKgmBmcmt1yKQUQ
Max-Forwards: 70
From: <sip:+33333333333@mydomain.com>;tag=rKgrp3mmj9F0D
To: "21" <sip:21@mydomain.com>;tag=kq7hadr903
Call-ID: ejqk1mfeq40qe8i98f67
CSeq: 57322251 BYE
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

What could be the problem?
Could be a NAT issue?
FusionPBX is installed on GoogleCloud VM.
The client uses the SIP profile "internal" with "ext-rtp-ip=IP_SERVER", "ext-sip-ip=IP_SERVER" and WSS-binding enabled.
The client is behind firewall and uses "stun.l.google.com:19302" as STUN server
What are other parameters that I could change?