I've seen G729E being implemented elsewhere. Anyone know what it adds to the table?
> -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Brian West > Sent: Wednesday, March 02, 2005 5:41 PM > To: Jacky; Asterisk Developers Mailing List > Subject: Re: [Asterisk-Dev] Digium's G.729A codec problem > > G729A vs G729B ... they are stream compatible. B on the > other hand has VAD and a few other things. B is more complex > than A but they are 100% compatible. > > /b > > On Mar 2, 2005, at 5:14 AM, Jacky wrote: > > > Hi, all, > > > > I have buy 5 Digium's G.729A codec(it just support G.729A license) > > When I calll with 2 SIP UA that support G.729A and G.729B, its rtp > > frame have some problem when softswitch with Asterisk. > > > > The voice frame have been drop, so sometime I can't hear voice. > > > > If I want to fix the problem when softswitch G.729A and > G.729B codec. > > What source code I must to modify ? > > Or some people have finished the issue, Could you show me how to do? > > > > > > -- > > Jacky > > _______________________________________________ > > Asterisk-Dev mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > _______________________________________________ > Asterisk-Dev mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev _______________________________________________ Asterisk-Dev mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
