[sip-comm-dev] Speex Issues in Sip-communicator-alpha


Right now i am trying to implement speex codec in my project using Asterisk Server. i could able to register the custom codec speex in jmf ,
but while making a call from one user agent to another useragent i was not able to receive the data and i am getting NullPointer exception at the
jspeex decoder library.
  I gone through sip-communicator alpha and tested with asterisk by specifying one user agent with ulaw codec and another user agent with speex
I cud able to make a call from ulaw user agent to speex user agent but the audio is oneway only the user agent who is using speex codec can receive the
data. and i am getting an error in asterisk as "unable to find a codec translation path from unknown to speex"..
Please let me know whether speex works in sip-communicator alpha and let me know how to implement jspeex in my project using Asterisk server.



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