Checking Audio Interconnection in the Conference Bridge (moved)

Moved to: ​https://docs.pjsip.org/en/latest/specific-guides/audio-troubleshooting/checks/conf_connections.html

You will get the silly no audio problem if you don't connect the call to the sound device in the conference bridge!

Use pjsua's cl (conference list) command from the pjsua's menu to check if the connection is made between the call and the sound device in the conference bridge. If you're presented with something like the following:

>>> cl
Conference ports:
Port #00[16KHz/10ms]         Master/sound  transmitting to: #1
Port #01[16KHz/20ms]   sip:user@localhost  transmitting to: #0

then your call does have bidirectional media flow with the sound device (the cl command output above shows that the audio device is transmitting to the call and the call is transmitting to the sound device, thus bidirectional media flow between sound device and call is established).

If you don't see the bidirectional media flow between sound device and the call, you can connect them using pjsua's cc (conference connect) command as shown in the command sequence below:

>>> cl
Conference ports:
Port #00[16KHz/10ms]         Master/sound  transmitting to:
Port #01[16KHz/20ms]   sip:user@localhost  transmitting to:

The above output shows no media flow between call and sound device. The command below will establish unidirectional media flow from the sound device to the call:

>>> cc 0 1
Success

And the command below will establish another unidirectional media flow from the reverse direction, from the call to the sound device:

>>> cc 1 0
Success

Now if we check again the connection status in the conference bridge, you should see this output:

>>> cl
Conference ports:
Port #00[16KHz/10ms]         Master/sound  transmitting to: #1
Port #01[16KHz/20ms]   sip:user@localhost  transmitting to: #0

>>>