Janus v1.3.2 Ignores rtp_port_range in Configuration – Always Uses Default Ports

General
1

Environment:

Janus Version:** 1.3.2 (commit f594ef7ddbd76338b3f0727dd70bb71789fed0a6)

  • OS:** Ubuntu 22.04 LTS
  • Deployment:** Manually compiled from source under /opt/janus
  • Startup: Running via systemd and also tested via janus -f
  • Config Path:** /opt/janus/etc/janus/janus.jcfg

In the media block of janus.jcfg:

ini

CopyEdit

media: {
rtp_port_range = “10000-20000”
audio_level_ext = false
}

Also using:

nat: {
ice_lite = true
nat_1_1_mapping = “104.245.147.134”
}

Observed Behavior:

  • Janus does not bind to any ports in 10000–20000.
  • It continues using ports like 5002, 5102, 5104, etc.
  • No log output shows Using RTP port range 10000-20000
  • No config error or warning is printed — config is silently ignored
  • Confirmed using sudo ss -ulnp | grep janus

What I’ve Tried:

  • Confirmed Janus is using the correct config file
  • Verified syntax and indentation are correct
  • Restarted Janus via systemd and tested janus -f (foreground)
  • Added audio_level_ext = false (saw this workaround referenced elsewhere)
  • Used journalctl to confirm no warnings or parsing errors
  • Confirmed that 3CX is sending RTP to dynamic ports, e.g., 36924 — but Janus never binds them

My Questions:

  1. Is this a known issue with Janus 1.3.2?
  2. Is rtp_port_range now dependent on a compile-time flag or missing plugin?
  3. Should it still be set in the media block in modern builds?
  4. Is the only solution a rebuild from the latest main branch?
2

rtp_port_range is exclusively for ports Janus will use on the WebRTC side. It has nothing to do with ports that plugins (Streaming, SIP, others) may use for their own needs. Some plugins will have a port range of their own you can configure.

3

I’ve built a WebRTC dialer that connects to a SIP server using the Janus SIP Gateway.
SIP registration works fine, and calls are successfully established.

However, I’m experiencing one-way audio:

  • Audio from the WebRTC dialer to the SIP endpoint works correctly.
  • But audio from the SIP endpoint to the WebRTC dialer is not received.

I checked the SIP server and confirmed that it is sending RTP packets to the public IP of the Janus server, but the WebRTC dialer side receives no audio.