Intermittent connectivity issues of the SIP client, lead to calls, which last forever

We have the following issue, happening, which is quite annoying and costly some times:

we have the following setup:

Operator(softphone jitsi) --(SIP)–> Asterisk → SIP provider → (the world).

The issues, follows this workflow:

  1. The operator makes a call.
  2. During the call, the operator experiences intermittent connectivity problem and for some time (lets say seconds to few minutes), does not reach Asterisk and does not hear any voice.
  3. The operator decides to hit ‘Hang Up’ button, which as far as jitsi is concernet “terminates” the call.
  4. In the reality, the call is NOT terminated, because the “hang up” signal does not reach the Asterisk, because of the mentioned connectivity problem.
  5. We are now at point, where the asterisk thinks, that he has open channels to the Operator device and to the dialed number.
  6. The connectivity is restored after short disruption but from Jitsi standpoint, we are no longer in a call.
  7. This situation stays for hours, unless the call/channels are manually terminated.

Is there anything I can do to avoid this situation, like in-call (in-channel) keepalives, which check if the Operator’s phone is unreachable and terminates both channels?

We are using Asterisk 11.25.0

Thank you.

There are RTP timers[1] and also SIP session timers[2].

[1] asterisk/sip.conf.sample at 11 · asterisk/asterisk · GitHub
[2] asterisk/sip.conf.sample at 11 · asterisk/asterisk · GitHub

1 Like

Working as expected.
Thank you.

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