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....

^_^