Our whole development team works from 6 different states so we use a 24x7 conference as part of our virtual office. There is a public conference for freeswitch users as well. http://conference.freeswitch.org sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org> On Thu, Aug 27, 2009 at 7:10 AM, Raimund Sacherer <rs@runsolutions.com>wrote:
very offtopic, but, if this is not to personally, but how do you get to spend literally 8-12 hours a day on a conference? :-)
best
-- Raimund Sacherer - RunSolutions Open Source It Consulting - Email: rs@runsolutions.com tel: 625 40 32 08
Parc Bit - Centro Empresarial Son Espanyol Edificio Estel - Local 3D 07121 - Palma de Mallorca Baleares
On Aug 26, 2009, at 10:45 PM, Anthony Minessale wrote:
There is no bug, it's all dependent on your network conditions. I spend literally 8-12 hours a day on a conference and there is no delay.
The important param is <param name="rtp-autoflush-during-bridge" value="true"/>
in the sofia profile in question.
if you have delay with that in place, then it's probably not FS
On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <pbd@suspiria.net> wrote:
Hi,
I am on a quite recent version (i assume):
FreeSWITCH Version 1.0.trunk (14461)
Should the bug be fixed in this revision ? What config settings would a have to check to limit delay (even at the cost of reduction in quality).
Thanks
*Von:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *Im Auftrag von *Anthony Minessale *Gesendet:* Mittwoch, 26. August 2009 20:57 *An:* freeswitch-users@lists.freeswitch.org *Betreff:* Re: [Freeswitch-users] delay buildup in conference
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 <pbd@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
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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