On Fri, November 18, 2005 11:14, Alejandro Vargas said: > 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. > >
Which is going to be a pain, as it is encrypted... :-( Reverse-engineering may be the best option, and that is: 1) Not trivial 2) Not always legal /me sighs... I'll try getting my friends on to VoipBuster instead! ;-) -- Francesco Peeters ---- GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. _______________________________________________ --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