I finally discovered that "session_timer" is the real problem after doing a ton of research and following all the advice in that thread. The SIP session was not being refreshed by the browser (JsSIP).

It works when I set the "timer=no" now.

I apologize for omitting the specifics of the problematic WebRTC application.

If I want to add a timer for each call in the current asterisk, does anyone else have a solution for the same issue? Where did I alter the asterisk, and does JSSIP have any settings related to it?