[GnomeMeeting-list] Sound configuration

["rémi beghin" <beghinre gmail com>]

Hello,

I just installed ekiga-snapshot on my debian etch and opened a
PC-to-phone account.

The first call I made worked well, it was a PC-to-phone call, me and
the other person had a perfect quality.
I then tried a second PC-to-phone call, I heard the other person but
he wasn't hearing anything.

After this all my PC-to-phone calls stopped working, there is no sound
for me or the other person.

Here is the console output of a PC-to-phone call :

  from PCM-16
         Clock Rate = 8000
         Frame Time = 8
         Max Bit Rate = 13200
         Max Frame Size = 16
         Needs Jitter = 1
         Rx Frames Per Packet = 7
         Tx Frames Per Packet = 4
    to GSM-06.10
         Clock Rate = 8000
         Frame Time = 160
         Max Bit Rate = 13200
         Max Frame Size = 33
         Needs Jitter = 1
         Rx Frames Per Packet = 7
         Tx Frames Per Packet = 4

       patch.cxx(251)	Patch	Added sink
  from GSM-06.10
         Clock Rate = 8000
         Frame Time = 160
         Max Bit Rate = 13200
         Max Frame Size = 33
         Needs Jitter = 1
         Rx Frames Per Packet = 7
         Tx Frames Per Packet = 4
    to PCM-16
         Clock Rate = 8000
         Frame Time = 8
         Max Bit Rate = 13200
         Max Frame Size = 16
         Needs Jitter = 1
         Rx Frames Per Packet = 7
         Tx Frames Per Packet = 4




I tried after this to call sip:500 ekiga net, I heard the "answering
machine voice but my voice is not "working". Here is the output of my
call :

       patch.cxx(251)	Patch	Added sink
  from PCM-16
         Clock Rate = 8000
         Frame Time = 8
         Max Bit Rate = 64000
         Max Frame Size = 16
         Needs Jitter = 1
         Rx Frames Per Packet = 240
         Tx Frames Per Packet = 30
    to G.711-ALaw-64k
         Clock Rate = 8000
         Frame Time = 8
         Max Bit Rate = 64000
         Max Frame Size = 8
         Needs Jitter = 1
         Rx Frames Per Packet = 240
         Tx Frames Per Packet = 30

       patch.cxx(251)	Patch	Added sink
  from G.711-ALaw-64k
         Clock Rate = 8000
         Frame Time = 8
         Max Bit Rate = 64000
         Max Frame Size = 8
         Needs Jitter = 1
         Rx Frames Per Packet = 240
         Tx Frames Per Packet = 30
    to PCM-16
         Clock Rate = 8000
         Frame Time = 8
         Max Bit Rate = 64000
         Max Frame Size = 16
         Needs Jitter = 1
         Rx Frames Per Packet = 240
         Tx Frames Per Packet = 30

       patch.cxx(251)	Patch	Added sink
  from YUV420P
         Adaptive Packet Delay = 0
         Clock Rate = 90000
         Dynamic Video Quality = 0
         Encoding Quality = 6
         Frame Height = 144
         Frame Time = 3600
         Frame Width = 176
         Max Bit Rate = 128000
         Target Bit Rate = 64000
    to H.261(QCIF)
         Adaptive Packet Delay = 1
         Clock Rate = 90000
         Dynamic Video Quality = 1
         Encoding Quality = 6
         Frame Height = 144
         Frame Time = 9000
         Frame Width = 176
         Max Bit Rate = 128000
         Target Bit Rate = 64000

       patch.cxx(251)	Patch	Added sink
  from H.261(QCIF)
         Adaptive Packet Delay = 1
         Clock Rate = 90000
         Dynamic Video Quality = 1
         Encoding Quality = 6
         Frame Height = 144
         Frame Time = 3600
         Frame Width = 176
         Max Bit Rate = 128000
         Target Bit Rate = 64000
    to YUV420P
         Adaptive Packet Delay = 0
         Clock Rate = 90000
         Dynamic Video Quality = 0
         Encoding Quality = 6
         Frame Height = 144
         Frame Time = 3600
         Frame Width = 176
         Max Bit Rate = 128000
         Target Bit Rate = 64000



The only other phone software I have used before is skype. I will try
to make some tests with other SIP phones if you think it is needed.

Thank

bye

Rémi


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