WebRTC video call drops after ~28 seconds (Asterisk + JsSIP / SIP.js) – detailed troubleshooting summary

FreePBX Endpoints
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I am experiencing a consistent issue with WebRTC video calls using Asterisk + SIP over WebSocket, where the call is established successfully, media flows in both directions, but the call drops after ~28–30 seconds.

Below is a full step-by-step summary of what was tested and observed.

Asterisk 16.30.0, FreePBX 16.0.45.

  1. SIP over WebSocket (WSS): wss://example.com:8089/ws
  2. WebRTC clients tested:

    • JsSIP 3.10.0

    • SIP.js 0.21.2

  3. Browser: Chrome (latest)
  4. Transport: WebSocket + ICE
  5. STUN:

    • stun:stun.l.google.com:19302

    • also tested local STUN and TURN (I understand that this is not necessary in a local network, I did this for testing)

Network: local LAN (10.77.50.0/24), no NAT between browser and Asterisk

SIP registration is successful

  1. INVITE / 200 OK / ACK flow is correct
  2. Session is confirmed successfully
  3. No SIP errors or retransmissions
  4. SIP session timers are disabled on client side (session_timers: false)

Call termination is not caused by BYE from the browser
Media (RTP / ICE) behavior

We enabled RTP debugging on Asterisk:

rtp set debug on

Observed output during the call:

  • RTP packets are sent and received continuously

  • Both directions are active

  • ICE is in use (via ICE)

  • Example log lines:

Sent RTP packet to 10.77.50.95:10868 (via ICE)
Got  RTP packet from 10.77.50.95:33436

This confirms: RTP is flowing both ways, ICE negotiation is successful, no one-way audio/video before disconnect.

What happens after ~28–30 seconds?

  • Call is established and works correctly at first, audio and video are present

  • After ~28–30 seconds - сall is dropped, Asterisk ends the session. No visible SIP error or explicit timeout message. Behavior is 100% reproducible

Any ideas on how to fix this?

P.S.
Additionally, I noticed a very clear correlation with the RTP timeout settings in FreePBX (for my WebRTC Extension).

When I explicitly set RTP timeout values, the call was always terminated exactly after the configured timeout, regardless of signaling or media activity:

rtp timeout = 120 → call drops after 120 seconds

rtp timeout = 600 → call drops after 600 seconds

This happens consistently and reproducibly.

This behavior strongly suggests that Asterisk stops receiving RTP packets (or considers them inactive) after a fixed interval, even though:

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