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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. August 2009 22:45
An: freeswitch-users@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@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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