On Wednesday 12 November 2003 09:47, Mark Spencer wrote: > it's implemented on the zap side (which is now configurable with > "jitterbuffers=foo" in zapata.conf. Will this work on a SIP to SIP call?
What does the parameter jitterbuffers=XXX represent? Is it memory allocation or milliseconds of voice? Thanks, Andres > > Mark > > On Wed, 12 Nov 2003, Matteo Brancaleoni wrote: > > mmmh... I'm not sure ig chan_sip has jitter buffer. > > I think that there isn't a jb in sip, > > but correct me if I'm wrong. > > > > Matteo. > > > > Il lun, 2003-11-10 alle 16:14, Andres ha scritto: > > > Hi, > > > > > > I would like to test chan_sip with a bigger jitter buffer. Does > > > anybody know where in the code this is defined? I looked through it > > > but could not find where. > > > > > > If anybody else can find it please let me know. > > > > > > Regards, > > > Andres > > > _______________________________________________ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > Matteo Brancaleoni > > Espia System Administrator > > Email : [EMAIL PROTECTED] > > Web : http://www.espia.it > > Phone : +39 02 70633354 - ext 911 > > IAX(2): [EMAIL PROTECTED] - ext 911 > > Iaxtel: 1-700-56-62458 - ext 911 > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
