Where to dig to find the cause of disconnections during outgoing calls through Multifon on FreeSwitch?

Question

Telephony

The situation is unclear, as always - everything worked perfectly, did not touch anything, at some point it stopped working.
What is:
- Freeswitch, spinning in Google Cloud
- several providers are connected to it, including Multifon
What went wrong:
- when making outgoing calls through Multifon, we have a normally established connection, a conversation that is stably interrupted after about 20 seconds
At the same time:
- all incoming calls work excellent, including the multiphone
- outgoing calls through other providers, all other things being equal, work perfectly
- outgoing calls from the same Multiphone account, but made through the desktop sip client (Telephone) keep without problems for as long as you like
- siptrace shows that at some point BYE just comes from the multiphone side, without any justification

|Time     | 10.240.0.0                            |
|         |                   | 193.201.229.35    |                   
|10.631417|         INVITE SDP (g711U g7          |SIP From: "John Doe" <sip:[email protected] To:<sip:[email protected]
|         |(48797)  ------------------>  (5060)   |
|10.678184|         100 Trying|                   |SIP Status
|         |(48797)  <------------------  (5060)   |
|10.724222|         407 Proxy Authentica          |SIP Status
|         |(48797)  <------------------  (5060)   |
|10.724376|         ACK       |                   |SIP Request
|         |(48797)  ------------------>  (5060)   |
|10.724617|         INVITE SDP (g711U g7          |SIP From: "John Doe" <sip:[email protected] To:<sip:[email protected]
|         |(48797)  ------------------>  (5060)   |
|10.771149|         100 Trying|                   |SIP Status
|         |(48797)  <------------------  (5060)   |
|11.577973|         181 Call Is Being Fo          |SIP Status
|         |(48797)  <------------------  (5060)   |
|11.776913|         183 Session Progress          |SIP Status
|         |(48797)  <------------------  (5060)   |
|12.007814|         183 Session Progress          |SIP Status
|         |(48797)  <------------------  (5060)   |
|12.187801|         180 Ringing                   |SIP Status
|         |(48797)  <------------------  (5060)   |
|13.436777|         200 OK SDP (g711A)            |SIP Status
|         |(48797)  <------------------  (5060)   |
|13.438568|         ACK       |                   |SIP Request
|         |(48797)  ------------------>  (5060)   |
|31.147090|         BYE       |                   |SIP Request
|         |(5080)   <------------------  (60246)  |
|31.158739|         200 OK    |                   |SIP Status
|         |(5080)   ------------------>  (60246)  |

BYE sip:[email protected]:5080;tport=tcp;transport=tcp;gw=multifon-79251111111 SIP/2.0
Via: SIP/2.0/TCP 193.201.229.35:5060;branch=z9hG4bKtfcpjj202oujbj56p7q0sd0000g00.1
Max-Forwards: 68
Content-Length: 0
To: "John Doe" <sip:[email protected]>;tag=amy3p0r6mtNHQ
From: <sip:[email protected]>;tag=SDq58g599-C5B5324631353641149E6B05
Call-ID: 3245e2fb-bb77-1233-9fb1-42010af00ef5
CSeq: 1 BYE

There is not a single idea in my head where to dig and where to look for the root of evil. Maybe there are some ideas?

Answer 1

in sofia-profile (looking at multiphone) add

<param name="rtcp-audio-interval-msec" value="5000"/>

Answer 2

were the updates from the client side or from the server side?