Janus Unable to Detect Call Hangup on SRTP and SIPS

[zidinberu]

Using Janus with freeswitch via sip plugin. Call (initiated from react native app) is over SIPS and SRTP with TLS.

Call gets connected with audio playing properly. However, when remote party hangs up, Janus thinks call is still active even though freeswitch has terminated the media and sent BYE to Janus. Janus realizes the call has ended after over 80 seconds.

The issue occurs only if freeswitch is running on tls and not running on non tls port. If freeswitch is running on both tls and non tls, then the issue doesnt occur. So that suggests janus is somehow using non tls port to communicate with freeswitch for certain messages i guess....

^_^

[lminiero]

We use Sofia SIP, which is the same library FreeSWITCH uses, and deals with everything SIP on our behalf. You should check the SIP traffic to ensure the BYE is actually getting to Janus, and that the Sofia SIP plugin is getting it and passing it to us. If the library doesn't notify us, we won't be aware of it. You can probably enable Sofia SIP debugging to see what's happening internally.