which revision are you on? The defaults on the latest code and examples should be configured to minimize delay. Some of the older revisions built up some delay issues from udp buffering when timers were not synced.
On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <p...@suspiria.net> wrote: > When running conferences with users dialed in from a PSTN gateway (SIP) > and directly from remote SIP endpoints there is an ever longer buildup in > delay, reaching up to multiple seconds. Is there any way to limit the delay > ? > > > > I am not 100% sure whether the delays is caused by the SIP jitter buffer of > freeswitch or directly by the conference module. > > > > Any advice? > > > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com <msn%3aanthony_miness...@hotmail.com> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org> iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org> pstn:213-799-1400
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