[sip-comm-dev] Need your help on SPEEX and iLBC implementation


  As for now we are using Sip-Communicator1.0 for our project
we would like to upgrade our sip-communicator project according to
  sip-communicator alpha version for improving the audio quality .
  we have analyzed the sip-communicator alpha coding and according to the
  Mediacontrol class and net.java.sip.communicator.impl.media.codec package we have updated the code in our sip-communicator project .
  But we are getting a problem like while using ilbc we are getting more
  noise and there is delay in transmission .
while using speex codec we dont get audio at all .we are using asterisk1.4

  plz we will be more benefited if u could tell us the classes to be refered
  and the changes has to be made for implementing these 2 code


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I am potentially working on one of these issues.

But we are getting a problem like while using ilbc we are getting more
noise and there is delay in transmission .

Do you mean that as time goes by, you get more and more noise, or do you
mean that you get more noise than actual voice ? The ilbc implementation
has been tested on a few minutes long samples (like 3/5 minutes) without
issues. Can you provide more details ?

Also, as said earlier, I am potentially working on this : in an effort
to reduce latency within Sip Communicator, we are trying to provide a
native linux capture interface to cut latency from the recording part
(please note that this package is a linux-specific one because it uses
native code).

This work is nearly finished, although I could not deal with it lately
(sorry Emil :wink: ).



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