Freeswitch WSS closed on INVITE

FreeSWITCH
1

Hi,
we are experiencing an issue with a web phone that uses sip.js and FreeSWITCH. The problem occurs during the exchange of INVITE packets, both initial and REINVITE, causing the socket to close and the call to terminate. This behavior does not happen all the time, but only occasionally.

Client error:

Wed Jul 03 2024 14:32:14 GMT+0200 (Ora legale dell’Europa centrale) | sip.Transport | WebSocket closed unexpectedly
Wed Jul 03 2024 14:32:14 GMT+0200 (Ora legale dell’Europa centrale) | sip.Transport | WebSocket closed wss://freeswitch.example.com:7444/ws (code: 1006)

Freeswitch error:

ws.c:458 wss_log_errors() ws_raw_read: SSL_read: 00000001:(null):(null):(null)
ws.c:458 wss_log_errors() ws_raw_read: SSL_read: 0a000126:SSL routines:(null):unexpected eof while reading
ws.c:458 wss_log_errors() ws_raw_read: SSL_read: 0a000126:SSL routines:(null):unexpected eof while reading
tport.c:2170 tport_shutdown0() tport_shutdown0(0x7f64bc72d080, 2)
tport.c:2102 tport_close() tport_close(0x7f64bc72d080): wss/200.200.200.200:1551/sips
nua_registrar.c:203 registrar_tport_error() tport error 0: Success
tport.c:4253 tport_release() tport_release(0x7f64bc72d080): (nil) by 0x7f64bc243870 with (nil)
nua_stack.c:301 nua_stack_event() nua(0x7f64bc243870): event i_media_error 500 Transport error detected

Configuration:

    sip.js: 0.21.2
    FreeSWITCH: 1.10.11-release-25-f24064f7c9~64bit (-release-25-f24064f7c9 64bit)
    libsofia-sip-ua0: 1.13.17-8116788228-50a509b72c~bookworm 500
    Operating System: Debian 12.6

What could be causing this behavior and how can we resolve it?
Thank you very much in advance for your help.

Best regards

L.

2

Does this happen on a certain length of call time? Like always at x:xx hour/minute call. Id check to see if that’s a factor

Random things can be difficult to nail down but if you can provide steps to replicate on demand it can be fixed.

3

No, it always happens at different times when a SIP RE-Invite is sent. If there are no re-INVITEs, the call socket is never closed. To disable the RE-Invite, I set enable-timer=false in the SIP profile. The problem remains that the socket can be closed even at the beginning of the call when the first SIP INVITE is sent. What could it be?

Additionally, is there a possibility to get “premium” assistance, even if it requires payment?

Thank you
Regards

L.