On 27 Aug 2004 at 2:33, Kevin Walsh wrote: > Jorge Verastegui G [EMAIL PROTECTED] wrote: > > I've been using VoIP over a not so reliable net: I usually > > get a 5% to 10% packet loss and a very high jitter. I tried > > several codecs and parameters, and the only thing left to > > test is PLC (Packet Loss Cancellement). > > > There is no packet loss concealment in Asterisk at this time. >
Why doesn't asterisk clock to the 1000 interrupts per second instead of the incoming audio? Were there no interrupts available when it started? Even if you had no card you could use the ztdummy module and even though that might be off by a bit, surely it'd sound better than a connection which is experiencing packet loss? How much work would be required to change this? I guess it couldn't really be an option because of the totally different structure... Would it be possible for one person to make those changes or would it require the authors of all modules to recode? Matt Riddell (www.sineapps.com) _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
