[Freeswitch-users] delay buildup in conference
Raimund Sacherer
rs at runsolutions.com
Thu Aug 27 05:10:36 PDT 2009
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 at runsolutions.com
tel: 625 40 32 08
Parc Bit - Centro Empresarial Son Espanyol
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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 at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org
> ] Im Auftrag von Anthony Minessale
> Gesendet: Mittwoch, 26. August 2009 20:57
> An: freeswitch-users at 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 at 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
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
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>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
>
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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 at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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