asterisk Sip.conf: Allow=alaw but Sound quality might be degraded by network issues (packet loss, delay) or Asterisk server limitations (CPU) etc
Paulo On Thu, Oct 9, 2008 at 1:16 PM, Karsten Schubotz <[EMAIL PROTECTED]> wrote: > Hi all, > > I have a question concerning negotiation of audio codec. > I'm using codec_pcmapcmu included in sipXtapi. If I make a direct > connection between two softphones using sipxtapi, the negotiated codec is > PCMA. The voice quality is good! But by making a connection over an Asterisk > server, the codec PCMU was negotiated. The voice quality is really bad. It > sounds interruptedly, and the pitch of the voice seems to vary irregularly. > Something didn't match. Does someone have an Idea what the cause could be? > Perhaps I can influence the codec negotiation, so the PCMA codec could be > used-> is it possible? > Thank you very much for a solution! > > Regards > Karsten > > -- > Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: > http://www.gmx.net/de/go/multimessenger > _______________________________________________ > sipxtapi-dev mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/
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