several factors : - check 'show translation' from asterisk to see how long it will take for transcoding between your codecs. with your machine, should not be long.
- the h323 endpoints latency. ( a lot of times this attributes to delay.) - echo cancellation and zitter buffer ( zitter significantly improves this.) -apu On 7/27/05, Bashir Ullah <[EMAIL PROTECTED]> wrote: > > Hi All > > I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb > ram, with g729 for i686 , (fedora 1). > > my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen > otherparty realtime voice , but other party geting sip party's voice 1 sec > later (not realtime). > > please some help me to solve this issu, last one month i am tring different > different way to solve this issu. > > is it codec problem or something else. > > > thanks > > bashir > _______________________________________________ > 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 > > _______________________________________________ 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
