Changes between Initial Version and Version 1 of audio-check-conf-connection

Timestamp:

Dec 5, 2006 2:44:45 PM (17 years ago)

Author:

bennylp

Comment:

--

audio-check-conf-connection

@@ +1 @@
+= Checking Audio Interconnection in the Conference Bridge =
+
+You will get the silly [wiki:audio-problem-local-no-audio "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:
+
+>>> cc 0 1
+Success
+>>> cl
+Conference ports:
+Port #00[16KHz/10ms]         Master/sound  transmitting to: #1
+Port #01[16KHz/20ms]   sip:user@localhost  transmitting to:
+
+>>> cc 1 0
+Success
+
+>>> cl
+Conference ports:
+Port #00[16KHz/10ms]         Master/sound  transmitting to: #1
+Port #01[16KHz/20ms]   sip:user@localhost  transmitting to: #0
+
+>>>
+}}}
+