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[pbd at suspiria.net]

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

[anthony.minessale at g...]

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 (pbd@suspiria.net)> wrote:

[pbd at suspiria.net]

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 (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



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

[anthony.minessale at g...]

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 (pbd@suspiria.net)> wrote:

[pbd at suspiria.net]

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 (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 (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. August 2009 20:57
An: freeswitch-users@lists.freeswitch.org (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 (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



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400






_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

[anthony.minessale at g...]

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@suspiria.net (pbd@suspiria.net)> wrote:

[rs at runsolutions.com]

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 (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:

[anthony.minessale at g...]

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.

sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])



On Thu, Aug 27, 2009 at 7:10 AM, Raimund Sacherer <rs@runsolutions.com (rs@runsolutions.com)> wrote:

[pbd at suspiria.net]

I have tested the suggested setting:
<settings>
<param name="rtp-autoflush-during-bridge" value="true"/>

But the problem remains. L

Von: Public Dump
Gesendet: Mittwoch, 26. August 2009 23:37
An: 'freeswitch-users@lists.freeswitch.org'
Betreff: AW: [Freeswitch-users] delay buildup in conference



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 (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 (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. August 2009 20:57
An: freeswitch-users@lists.freeswitch.org (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 (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



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400






_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400