[Freeswitch-users] delay buildup in conference

[Anthony Minessale]

being unreachable does not cause delay, sending the audio too fast does.
the timer in the rtp stack will slow down the audio stream to match the
clocked rate and if there is a difference
it can translate to delay over time.

This is been especially prevalent in cisco phones but other things have
caused a similar problem.

you can also try disabling rtp timer by setting rtp_timer_name=none before
you answer the call.

the keys are controlled by the mappings in the conference profile


it's a little unnerving talking to someone named public dump, do you have an
identity?



On Wed, Aug 26, 2009 at 4:36 PM, Public Dump <pbd at suspiria.net> wrote:

>  I am quite sure that it’s not a problem with the network connection,
> since the delay builds up gradually over time, while the host is reachable
> without any problems.
>
>
>
> I have added the suggested config lines and will check the results.
>
>
>
> By the way, can I remap the hash key, so that is does not end the
> conference ? I’ve tried remapping it, but freeswitch seems to ignore the
> setting (the console shows: hangup mapped to ‘*’ … but # still ends the
> call).
>
> It is irritating for users, since most users try to enter the conference
> pin the #-key at the end.
>
>
>
> *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 22:45
>
> *An:* freeswitch-users at lists.freeswitch.org
> *Betreff:* Re: [Freeswitch-users] delay buildup in conference
>
>
>
> 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
>
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>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
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> Twitter: http://twitter.com/FreeSWITCH_wire
>
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>
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>
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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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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