Or use the Sipxtapi function:
sipxConfigSetAudioCodecByName( sipxInstance, « PCMA » ) ; rgds, stipus De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Paulo Vicentini Envoyé : jeudi 9 octobre 2008 18:34 À : [email protected] Objet : Re: [sipxtapi-dev] negotiation of audio codec 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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