2005/11/17, Michael Graves <[EMAIL PROTECTED]>: > Call quality is ok, but it seems to add considerable latency. I suspect > that the call is fully decoded back to analogue (or maybe not quite > that far) on one of the audio devices in the OS, then encoded into SIP > for the outbound leg. That would imply additional delay in all cases.
It uses the skype api, then is the api (the skype propietary client) who decodes the sound (adding some latency). Then the sip part may be including some extra latency. It sould use a low latency codec, compression is not needed in a local machine... But there is a big problem with this program: it needs windows to run, adding failure point to the circuit... and then needing of an extra machine only for acting as gateway: bad solution. I think we will need to wait until someone hacked the skype protocol... If there is someone interested on doing it. -- Alejandro Vargas _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
