Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)
Hi, is there anything new since august? I have the same problem with latest FritzBox firmenware and latest FreeSwitch installed (via Quick & Dirty install). FritzBox always sends 30ms in one paket, FS starts with 20 and then switches to 30 (maybe thats the problem?) Calling the Demo IVR at 5000 is (with default options and latest FS) no problem, both uses 30ms ptime, have no idea why. Calling the tetris-demo at 9998 is chopped up, FS using 20ms ptime, FBox 30. Conferences are also choppy. Adding <param name="rtp-autofix-timing" value="false" /> to 1000.xml (account used at the FBox) does not help. Anyone got an idea? Nico
Do you understand that the problem is with your device and not FS? So why do you think something would have changed? here is what you can do to work around it: look for this param in you sip profile config: <param name="inbound-codec-negotiation" value="generous"/> change it to the following: <param name="inbound-codec-negotiation" value="scrooge"/> restart and retry if this is not enough also edit vars.xml look for <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> replace it with: <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@30i,PCMA@30i"/> retry On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden <nico@vthadden.de>wrote:
Hi,
is there anything new since august? I have the same problem with latest FritzBox firmenware and latest FreeSwitch installed (via Quick & Dirty install).
FritzBox always sends 30ms in one paket, FS starts with 20 and then switches to 30 (maybe thats the problem?)
Calling the Demo IVR at 5000 is (with default options and latest FS) no problem, both uses 30ms ptime, have no idea why.
Calling the tetris-demo at 9998 is chopped up, FS using 20ms ptime, FBox 30.
Conferences are also choppy.
Adding <param name="rtp-autofix-timing" value="false" /> to 1000.xml (account used at the FBox) does not help.
Anyone got an idea?
Nico
_______________________________________________ FreeSWITCH-users mailing list 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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Changing those values didn't help I wonder why then it works with all other servers. Also, if I call the 9888 (so 888@conference.freeswitch.org) it works without problems, i wonder what they have changed in the config O.o On 28.04.2010 23:04, Anthony Minessale wrote:
Do you understand that the problem is with your device and not FS? So why do you think something would have changed?
here is what you can do to work around it:
look for this param in you sip profile config: <param name="inbound-codec-negotiation" value="generous"/>
change it to the following: <param name="inbound-codec-negotiation" value="scrooge"/>
restart and retry
if this is not enough also edit vars.xml
look for <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
replace it with:
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@30i,PCMA@30i"/>
retry
On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden <nico@vthadden.de <mailto:nico@vthadden.de>> wrote:
Hi,
is there anything new since august? I have the same problem with latest FritzBox firmenware and latest FreeSwitch installed (via Quick & Dirty install).
FritzBox always sends 30ms in one paket, FS starts with 20 and then switches to 30 (maybe thats the problem?)
Calling the Demo IVR at 5000 is (with default options and latest FS) no problem, both uses 30ms ptime, have no idea why.
Calling the tetris-demo at 9998 is chopped up, FS using 20ms ptime, FBox 30.
Conferences are also choppy.
Adding <param name="rtp-autofix-timing" value="false" /> to 1000.xml (account used at the FBox) does not help.
Anyone got an idea?
Nico
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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 <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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nothing it's default config on latest git HEAD On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico@vthadden.de>wrote:
Changing those values didn't help
I wonder why then it works with all other servers. Also, if I call the 9888 (so 888@conference.freeswitch.org) it works without problems, i wonder what they have changed in the config O.o
On 28.04.2010 23:04, Anthony Minessale wrote:
Do you understand that the problem is with your device and not FS? So why do you think something would have changed?
here is what you can do to work around it:
look for this param in you sip profile config: <param name="inbound-codec-negotiation" value="generous"/>
change it to the following: <param name="inbound-codec-negotiation" value="scrooge"/>
restart and retry
if this is not enough also edit vars.xml
look for <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
replace it with:
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@30i,PCMA@30i"/>
retry
On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden <nico@vthadden.de <mailto:nico@vthadden.de>> wrote:
Hi,
is there anything new since august? I have the same problem with latest FritzBox firmenware and latest FreeSwitch installed (via Quick & Dirty install).
FritzBox always sends 30ms in one paket, FS starts with 20 and then switches to 30 (maybe thats the problem?)
Calling the Demo IVR at 5000 is (with default options and latest FS) no problem, both uses 30ms ptime, have no idea why.
Calling the tetris-demo at 9998 is chopped up, FS using 20ms ptime, FBox 30.
Conferences are also choppy.
Adding <param name="rtp-autofix-timing" value="false" /> to 1000.xml (account used at the FBox) does not help.
Anyone got an idea?
Nico
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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 <MSN%3Aanthony_minessale@hotmail.com> <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
pstn:+19193869900
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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 FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Hello Nicola, we had the same problem with the FritzBoxes. Someweher in the internet we found an advice to patch Freeswitch. See my previous comment on this patch here: http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.ht... Best regards Peter Anthony Minessale schrieb:
nothing it's default config on latest git HEAD
On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico@vthadden.de <mailto:nico@vthadden.de>> wrote:
Changing those values didn't help
I wonder why then it works with all other servers. Also, if I call the 9888 (so 888@conference.freeswitch.org <mailto:888@conference.freeswitch.org>) it works without problems, i wonder what they have changed in the config O.o
On 28.04.2010 23:04, Anthony Minessale wrote: > Do you understand that the problem is with your device and not FS? > So why do you think something would have changed? > > here is what you can do to work around it: > > look for this param in you sip profile config: > <param name="inbound-codec-negotiation" value="generous"/> > > change it to the following: > <param name="inbound-codec-negotiation" value="scrooge"/> > > restart and retry > > if this is not enough also edit vars.xml > > look for > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> > > replace it with: > > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@30i,PCMA@30i"/> > > retry > > > > > On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden <nico@vthadden.de <mailto:nico@vthadden.de> > <mailto:nico@vthadden.de <mailto:nico@vthadden.de>>> wrote: > > Hi, > > is there anything new since august? > I have the same problem with latest FritzBox firmenware and latest > FreeSwitch installed (via Quick & Dirty install). > > FritzBox always sends 30ms in one paket, FS starts with 20 and then > switches to 30 (maybe thats the problem?) > > Calling the Demo IVR at 5000 is (with default options and latest FS) no > problem, both uses 30ms ptime, have no idea why. > > Calling the tetris-demo at 9998 is chopped up, FS using 20ms ptime, > FBox 30. > > Conferences are also choppy. > > Adding <param name="rtp-autofix-timing" value="false" /> > to 1000.xml (account used at the FBox) does not help. > > Anyone got an idea? > > Nico > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > <mailto:FreeSWITCH-users@lists.freeswitch.org <mailto: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 <mailto:MSN%3Aanthony_minessale@hotmail.com> > <mailto:MSN%3Aanthony_minessale@hotmail.com <mailto:MSN%253Aanthony_minessale@hotmail.com>> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> > <mailto:PAYPAL%3Aanthony.minessale@gmail.com <mailto:PAYPAL%253Aanthony.minessale@gmail.com>> > IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> > <mailto:sip%3A888@conference.freeswitch.org <mailto:sip%253A888@conference.freeswitch.org>> > googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>> > pstn:+19193869900 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto: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
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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 <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 ------------------------------------------------------------------------
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If you look at the code a bit further, you'll notice an if (....sofia_test_pflag(tech_pvt->profile, PFLAG_AUTOFIX_TIMING)) that flag is set depending on the value of the "rtp-autofix-timing" parameter in the sip profile, no need to patch anything :) Perhaps didn't you rescan the sofia profile after your config change? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca On 2010-04-29, at 9:44 AM, Peter P GMX wrote:
Hello Nicola,
we had the same problem with the FritzBoxes. Someweher in the internet we found an advice to patch Freeswitch. See my previous comment on this patch here: http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.ht...
Best regards Peter
Anthony Minessale schrieb:
nothing it's default config on latest git HEAD
On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico@vthadden.de <mailto:nico@vthadden.de>> wrote:
Changing those values didn't help
I wonder why then it works with all other servers. Also, if I call the 9888 (so 888@conference.freeswitch.org <mailto:888@conference.freeswitch.org>) it works without problems, i wonder what they have changed in the config O.o
On 28.04.2010 23:04, Anthony Minessale wrote:
Do you understand that the problem is with your device and not FS? So why do you think something would have changed?
here is what you can do to work around it:
look for this param in you sip profile config: <param name="inbound-codec-negotiation" value="generous"/>
change it to the following: <param name="inbound-codec-negotiation" value="scrooge"/>
restart and retry
if this is not enough also edit vars.xml
look for <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
replace it with:
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@30i,PCMA@30i"/>
retry
On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden <nico@vthadden.de <mailto:nico@vthadden.de> <mailto:nico@vthadden.de <mailto:nico@vthadden.de>>> wrote:
Hi,
is there anything new since august? I have the same problem with latest FritzBox firmenware and latest FreeSwitch installed (via Quick & Dirty install).
FritzBox always sends 30ms in one paket, FS starts with 20 and then switches to 30 (maybe thats the problem?)
Calling the Demo IVR at 5000 is (with default options and latest FS) no problem, both uses 30ms ptime, have no idea why.
Calling the tetris-demo at 9998 is chopped up, FS using 20ms ptime, FBox 30.
Conferences are also choppy.
Adding <param name="rtp-autofix-timing" value="false" /> to 1000.xml (account used at the FBox) does not help.
Anyone got an idea?
Nico
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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 <mailto:MSN%3Aanthony_minessale@hotmail.com> <mailto:MSN%3Aanthony_minessale@hotmail.com <mailto:MSN%253Aanthony_minessale@hotmail.com>> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> <mailto:PAYPAL%3Aanthony.minessale@gmail.com <mailto:PAYPAL%253Aanthony.minessale@gmail.com>> IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.freeswitch.org <mailto:sip%253A888@conference.freeswitch.org>> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>> pstn:+19193869900
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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 <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 ------------------------------------------------------------------------
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Thanks for the hint, should "rtp-autofix-timing" be set to true or false in that case? Also: What is the difference to "rtp auto adjust"? I would like to update the wiki on that, when this is clearified. The Fritzbox is the most common ATA here in Europe. Best rgeards Peter Mathieu Rene schrieb:
If you look at the code a bit further, you'll notice an if (....sofia_test_pflag(tech_pvt->profile, PFLAG_AUTOFIX_TIMING))
that flag is set depending on the value of the "rtp-autofix-timing" parameter in the sip profile, no need to patch anything :)
Perhaps didn't you rescan the sofia profile after your config change?
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 2010-04-29, at 9:44 AM, Peter P GMX wrote:
Hello Nicola,
we had the same problem with the FritzBoxes. Someweher in the internet we found an advice to patch Freeswitch. See my previous comment on this patch here: http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.ht...
Best regards Peter
Anthony Minessale schrieb:
nothing it's default config on latest git HEAD
On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico@vthadden.de <mailto:nico@vthadden.de>> wrote:
Changing those values didn't help
I wonder why then it works with all other servers. Also, if I call the 9888 (so 888@conference.freeswitch.org <mailto:888@conference.freeswitch.org>) it works without problems, i wonder what they have changed in the config O.o
On 28.04.2010 23:04, Anthony Minessale wrote:
Do you understand that the problem is with your device and not FS? So why do you think something would have changed?
here is what you can do to work around it:
look for this param in you sip profile config: <param name="inbound-codec-negotiation" value="generous"/>
change it to the following: <param name="inbound-codec-negotiation" value="scrooge"/>
restart and retry
if this is not enough also edit vars.xml
look for <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
replace it with:
<X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=PCMU@30i,PCMA@30i"/>
retry
On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden
<nico@vthadden.de <mailto:nico@vthadden.de>
<mailto:nico@vthadden.de <mailto:nico@vthadden.de>>> wrote:
Hi,
is there anything new since august? I have the same problem with latest FritzBox firmenware and
latest
FreeSwitch installed (via Quick & Dirty install).
FritzBox always sends 30ms in one paket, FS starts with 20
and then
switches to 30 (maybe thats the problem?)
Calling the Demo IVR at 5000 is (with default options and
latest FS) no
problem, both uses 30ms ptime, have no idea why.
Calling the tetris-demo at 9998 is chopped up, FS using 20ms
ptime,
FBox 30.
Conferences are also choppy.
Adding <param name="rtp-autofix-timing" value="false" /> to 1000.xml (account used at the FBox) does not help.
Anyone got an idea?
Nico
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Peter, I think Math's advice was to set rtp-autofix-timing to false, because in some situations, the smartness in FS can cause issues. rtp-auto-adjust has perhaps a misleading name. It's sometimes called auto-NAT or NAT autolearn. it's the ability for FreeSWITCH to adjust dynamically itself to the received RTP stream on a port, instead of what it was expecting from the SDP sent by the remote party. it does that by learning the source IP/port of the 20 first packets. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 29/04/2010 à 16:18, Peter P GMX a écrit :
Thanks for the hint,
should "rtp-autofix-timing" be set to true or false in that case?
Also: What is the difference to "rtp auto adjust"?
I would like to update the wiki on that, when this is clearified. The Fritzbox is the most common ATA here in Europe.
Best rgeards Peter
Mathieu Rene schrieb:
If you look at the code a bit further, you'll notice an if (....sofia_test_pflag(tech_pvt->profile, PFLAG_AUTOFIX_TIMING))
that flag is set depending on the value of the "rtp-autofix-timing" parameter in the sip profile, no need to patch anything :)
Perhaps didn't you rescan the sofia profile after your config change?
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 2010-04-29, at 9:44 AM, Peter P GMX wrote:
Hello Nicola,
we had the same problem with the FritzBoxes. Someweher in the internet we found an advice to patch Freeswitch. See my previous comment on this patch here: http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.ht...
Best regards Peter
Anthony Minessale schrieb:
nothing it's default config on latest git HEAD
On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico@vthadden.de <mailto:nico@vthadden.de>> wrote:
Changing those values didn't help
I wonder why then it works with all other servers. Also, if I call the 9888 (so 888@conference.freeswitch.org <mailto:888@conference.freeswitch.org>) it works without problems, i wonder what they have changed in the config O.o
On 28.04.2010 23:04, Anthony Minessale wrote:
Do you understand that the problem is with your device and not FS? So why do you think something would have changed?
here is what you can do to work around it:
look for this param in you sip profile config: <param name="inbound-codec-negotiation" value="generous"/>
change it to the following: <param name="inbound-codec-negotiation" value="scrooge"/>
restart and retry
if this is not enough also edit vars.xml
look for <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
replace it with:
<X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=PCMU@30i,PCMA@30i"/>
retry
On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden
<nico@vthadden.de <mailto:nico@vthadden.de>
<mailto:nico@vthadden.de <mailto:nico@vthadden.de>>> wrote:
Hi,
is there anything new since august? I have the same problem with latest FritzBox firmenware and
latest
FreeSwitch installed (via Quick & Dirty install).
FritzBox always sends 30ms in one paket, FS starts with 20
and then
switches to 30 (maybe thats the problem?)
Calling the Demo IVR at 5000 is (with default options and
latest FS) no
problem, both uses 30ms ptime, have no idea why.
Calling the tetris-demo at 9998 is chopped up, FS using 20ms
ptime,
FBox 30.
Conferences are also choppy.
Adding <param name="rtp-autofix-timing" value="false" /> to 1000.xml (account used at the FBox) does not help.
Anyone got an idea?
Nico
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auto-adjust will automatically switch the IP the RTP is being sent to based on the received packet's IP header. autofix timing mesures the received payload size and compares it with how much "<ptime> ms " is supposed to take. If there is a difference it thins the device is broken (a lot of UAs lie about their ptime in the SDP) and adjusts the codec's ptime in order to interop. To be the equivalent of your patch, it would have to be set to false. Out of curiosity, what codec is being used by fritzbox when the problem occurs? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca On 2010-04-29, at 10:18 AM, Peter P GMX wrote:
Thanks for the hint,
should "rtp-autofix-timing" be set to true or false in that case?
Also: What is the difference to "rtp auto adjust"?
I would like to update the wiki on that, when this is clearified. The Fritzbox is the most common ATA here in Europe.
Best rgeards Peter
Mathieu Rene schrieb:
If you look at the code a bit further, you'll notice an if (....sofia_test_pflag(tech_pvt->profile, PFLAG_AUTOFIX_TIMING))
that flag is set depending on the value of the "rtp-autofix-timing" parameter in the sip profile, no need to patch anything :)
Perhaps didn't you rescan the sofia profile after your config change?
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 2010-04-29, at 9:44 AM, Peter P GMX wrote:
Hello Nicola,
we had the same problem with the FritzBoxes. Someweher in the internet we found an advice to patch Freeswitch. See my previous comment on this patch here: http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.ht...
Best regards Peter
Anthony Minessale schrieb:
nothing it's default config on latest git HEAD
On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico@vthadden.de <mailto:nico@vthadden.de>> wrote:
Changing those values didn't help
I wonder why then it works with all other servers. Also, if I call the 9888 (so 888@conference.freeswitch.org <mailto:888@conference.freeswitch.org>) it works without problems, i wonder what they have changed in the config O.o
On 28.04.2010 23:04, Anthony Minessale wrote:
Do you understand that the problem is with your device and not FS? So why do you think something would have changed?
here is what you can do to work around it:
look for this param in you sip profile config: <param name="inbound-codec-negotiation" value="generous"/>
change it to the following: <param name="inbound-codec-negotiation" value="scrooge"/>
restart and retry
if this is not enough also edit vars.xml
look for <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
replace it with:
<X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=PCMU@30i,PCMA@30i"/>
retry
On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden
<nico@vthadden.de <mailto:nico@vthadden.de>
<mailto:nico@vthadden.de <mailto:nico@vthadden.de>>> wrote:
Hi,
is there anything new since august? I have the same problem with latest FritzBox firmenware and
latest
FreeSwitch installed (via Quick & Dirty install).
FritzBox always sends 30ms in one paket, FS starts with 20
and then
switches to 30 (maybe thats the problem?)
Calling the Demo IVR at 5000 is (with default options and
latest FS) no
problem, both uses 30ms ptime, have no idea why.
Calling the tetris-demo at 9998 is chopped up, FS using 20ms
ptime,
FBox 30.
Conferences are also choppy.
Adding <param name="rtp-autofix-timing" value="false" /> to 1000.xml (account used at the FBox) does not help.
Anyone got an idea?
Nico
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<mailto:sip%253A888@conference.freeswitch.org>>
googletalk:conf+888@conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org>
<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org
<mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>>
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Does anyone ever report this issue to Fritzbox or is too hard to conceive that they have the bug in this situation? On Thu, Apr 29, 2010 at 9:53 AM, Mathieu Rene <mrene_lists@avgs.ca> wrote:
auto-adjust will automatically switch the IP the RTP is being sent to based on the received packet's IP header. autofix timing mesures the received payload size and compares it with how much "<ptime> ms " is supposed to take. If there is a difference it thins the device is broken (a lot of UAs lie about their ptime in the SDP) and adjusts the codec's ptime in order to interop.
To be the equivalent of your patch, it would have to be set to false.
Out of curiosity, what codec is being used by fritzbox when the problem occurs?
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 2010-04-29, at 10:18 AM, Peter P GMX wrote:
Thanks for the hint,
should "rtp-autofix-timing" be set to true or false in that case?
Also: What is the difference to "rtp auto adjust"?
I would like to update the wiki on that, when this is clearified. The Fritzbox is the most common ATA here in Europe.
Best rgeards Peter
Mathieu Rene schrieb:
If you look at the code a bit further, you'll notice an if (....sofia_test_pflag(tech_pvt->profile, PFLAG_AUTOFIX_TIMING))
that flag is set depending on the value of the "rtp-autofix-timing" parameter in the sip profile, no need to patch anything :)
Perhaps didn't you rescan the sofia profile after your config change?
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 2010-04-29, at 9:44 AM, Peter P GMX wrote:
Hello Nicola,
we had the same problem with the FritzBoxes. Someweher in the internet we found an advice to patch Freeswitch. See my previous comment on this patch here:
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.ht...
Best regards Peter
Anthony Minessale schrieb:
nothing it's default config on latest git HEAD
On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <nico@vthadden.de <mailto:nico@vthadden.de>> wrote:
Changing those values didn't help
I wonder why then it works with all other servers. Also, if I call the 9888 (so 888@conference.freeswitch.org <mailto:888@conference.freeswitch.org>) it works without problems, i wonder what they have changed in the config O.o
On 28.04.2010 23:04, Anthony Minessale wrote:
Do you understand that the problem is with your device and not FS? So why do you think something would have changed?
here is what you can do to work around it:
look for this param in you sip profile config: <param name="inbound-codec-negotiation" value="generous"/>
change it to the following: <param name="inbound-codec-negotiation" value="scrooge"/>
restart and retry
if this is not enough also edit vars.xml
look for <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
replace it with:
<X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=PCMU@30i,PCMA@30i"/>
retry
On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden
<nico@vthadden.de <mailto:nico@vthadden.de>
<mailto:nico@vthadden.de <mailto:nico@vthadden.de>>> wrote:
Hi,
is there anything new since august? I have the same problem with latest FritzBox firmenware and
latest
FreeSwitch installed (via Quick & Dirty install).
FritzBox always sends 30ms in one paket, FS starts with 20
and then
switches to 30 (maybe thats the problem?)
Calling the Demo IVR at 5000 is (with default options and
latest FS) no
problem, both uses 30ms ptime, have no idea why.
Calling the tetris-demo at 9998 is chopped up, FS using 20ms
ptime,
FBox 30.
Conferences are also choppy.
Adding <param name="rtp-autofix-timing" value="false" /> to 1000.xml (account used at the FBox) does not help.
Anyone got an idea?
Nico
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org
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i think usually not, because with FS sometimes you get the patch within minutes, which you'll never get somewhere else :-) On Thu, Apr 29, 2010 at 5:01 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Does anyone ever report this issue to Fritzbox or is too hard to conceive that they have the bug in this situation?
On Thu, Apr 29, 2010 at 9:53 AM, Mathieu Rene <mrene_lists@avgs.ca> wrote:
auto-adjust will automatically switch the IP the RTP is being sent to based on the received packet's IP header. autofix timing mesures the received payload size and compares it with how much "<ptime> ms " is supposed to take. If there is a difference it thins the device is broken (a lot of UAs lie about their ptime in the SDP) and adjusts the codec's ptime in order to interop.
To be the equivalent of your patch, it would have to be set to false.
Out of curiosity, what codec is being used by fritzbox when the problem occurs?
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 2010-04-29, at 10:18 AM, Peter P GMX wrote:
Thanks for the hint,
should "rtp-autofix-timing" be set to true or false in that case?
Also: What is the difference to "rtp auto adjust"?
I would like to update the wiki on that, when this is clearified. The Fritzbox is the most common ATA here in Europe.
Best rgeards Peter
Mathieu Rene schrieb:
If you look at the code a bit further, you'll notice an if (....sofia_test_pflag(tech_pvt->profile, PFLAG_AUTOFIX_TIMING))
that flag is set depending on the value of the "rtp-autofix-timing" parameter in the sip profile, no need to patch anything :)
Perhaps didn't you rescan the sofia profile after your config change?
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 2010-04-29, at 9:44 AM, Peter P GMX wrote:
Hello Nicola,
we had the same problem with the FritzBoxes. Someweher in the internet we found an advice to patch Freeswitch. See my previous comment on this patch here:
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/045888.ht...
Best regards Peter
Anthony Minessale schrieb:
nothing it's default config on latest git HEAD
On Wed, Apr 28, 2010 at 4:17 PM, Nicola von Thadden <
nico@vthadden.de
<mailto:nico@vthadden.de>> wrote:
Changing those values didn't help
I wonder why then it works with all other servers. Also, if I call the 9888 (so 888@conference.freeswitch.org <mailto:888@conference.freeswitch.org>) it works without problems, i wonder what they have changed in the config O.o
On 28.04.2010 23:04, Anthony Minessale wrote:
> Do you understand that the problem is with your device and not FS? > So why do you think something would have changed? > > here is what you can do to work around it: > > look for this param in you sip profile config: > <param name="inbound-codec-negotiation" value="generous"/> > > change it to the following: > <param name="inbound-codec-negotiation" value="scrooge"/> > > restart and retry > > if this is not enough also edit vars.xml > > look for > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> > > replace it with: > > <X-PRE-PROCESS cmd="set" > data="outbound_codec_prefs=PCMU@30i,PCMA@30i"/>
> retry > > > > > On Wed, Apr 28, 2010 at 11:55 AM, Nicola von Thadden > <nico@vthadden.de <mailto:nico@vthadden.de>
> <mailto:nico@vthadden.de <mailto:nico@vthadden.de>>> wrote: > > Hi, > > is there anything new since august? > I have the same problem with latest FritzBox firmenware and > latest
> FreeSwitch installed (via Quick & Dirty install). > > FritzBox always sends 30ms in one paket, FS starts with 20 > and then
> switches to 30 (maybe thats the problem?) > > Calling the Demo IVR at 5000 is (with default options and > latest FS) no
> problem, both uses 30ms ptime, have no idea why. > > Calling the tetris-demo at 9998 is chopped up, FS using 20ms > ptime,
> FBox 30. > > Conferences are also choppy. > > Adding <param name="rtp-autofix-timing" value="false" /> > to 1000.xml (account used at the FBox) does not help. > > Anyone got an idea? > > Nico > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org>
> <mailto:FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org>>
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> 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<MSN%3Aanthony_minessale@hotmail.com> > <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com>
> <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com> > <mailto:MSN%253Aanthony_minessale@hotmail.com<MSN%25253Aanthony_minessale@hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> > <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
> <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com> > <mailto:PAYPAL%253Aanthony.minessale@gmail.com<PAYPAL%25253Aanthony.minessale@gmail.com>
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> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org> > <mailto:sip%253A888@conference.freeswitch.org<sip%25253A888@conference.freeswitch.org>
> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org> > <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org<googletalk%25253Aconf%25252B888@conference.freeswitch.org>
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OK how many of these devices are we talking about that you have? /b On Apr 29, 2010, at 10:34 AM, Saeed Ahmed wrote:
i think usually not, because with FS sometimes you get the patch within minutes, which you'll never get somewhere else :-)
On Thu, Apr 29, 2010 at 5:01 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: Does anyone ever report this issue to Fritzbox or is too hard to conceive that they have the bug in this situation?
I have 3 of them, in Germany I think there are about 10 Mio. Plus France, Austria,Switzerland etc. I would say some dozens of Millions. Best regards Peter Brian West schrieb:
OK how many of these devices are we talking about that you have?
/b
On Apr 29, 2010, at 10:34 AM, Saeed Ahmed wrote:
i think usually not, because with FS sometimes you get the patch within minutes, which you'll never get somewhere else :-)
On Thu, Apr 29, 2010 at 5:01 PM, Anthony Minessale <anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com>> wrote:
Does anyone ever report this issue to Fritzbox or is too hard to conceive that they have the bug in this situation?
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But you have exactly three of them. Its cheaper to go buy something thats not broken vs trying to work around this. Just saying! /b On Apr 29, 2010, at 11:05 AM, Peter P GMX wrote:
I have 3 of them, in Germany I think there are about 10 Mio. Plus France, Austria,Switzerland etc. I would say some dozens of Millions.
Best regards Peter
It's not so easy to find something with the same or comparable features: ADSL2+ (16mbit), ISDN and normal landline, VOIP, DECT, WLAN (a/b/g/n), etc Additionally they are sponsored by many ISPs, so you get them for free (or with a lot of discount). I have just recompiled from latest git version (done with the Makefile given here: http://www.freeswitch.org/eg/Makefile) and have the same issues as always. Setting rtp-autofix-timing to false removed the choppiness but added an delay in conferences (5 to 10 seconds)... CPU is between 40 and 170% (on a quadcore) and ram is just 20mb used by freeswitch, so this shouldn't be the problem. Nico On 29.04.2010 18:13, Brian West wrote:
But you have exactly three of them. Its cheaper to go buy something thats not broken vs trying to work around this.
Just saying!
/b
On Apr 29, 2010, at 11:05 AM, Peter P GMX wrote:
I have 3 of them, in Germany I think there are about 10 Mio. Plus France, Austria,Switzerland etc. I would say some dozens of Millions.
Best regards Peter
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every time you repost saying it's still not working I will reply telling you that its because the fritzbox is broken. I can assure you some combination or another of the advice we gave you probably can work around the problem and you missed it. In short, we will never have some magic patch to make it start working right because it's the other side who is broken not us. Maybe you should find another user who owns one and managed to hack it into working. On Sat, May 1, 2010 at 5:29 PM, Nicola von Thadden <nico@vthadden.de> wrote:
It's not so easy to find something with the same or comparable features: ADSL2+ (16mbit), ISDN and normal landline, VOIP, DECT, WLAN (a/b/g/n), etc
Additionally they are sponsored by many ISPs, so you get them for free (or with a lot of discount).
I have just recompiled from latest git version (done with the Makefile given here: http://www.freeswitch.org/eg/Makefile) and have the same issues as always. Setting rtp-autofix-timing to false removed the choppiness but added an delay in conferences (5 to 10 seconds)... CPU is between 40 and 170% (on a quadcore) and ram is just 20mb used by freeswitch, so this shouldn't be the problem.
Nico
On 29.04.2010 18:13, Brian West wrote:
But you have exactly three of them. Its cheaper to go buy something thats not broken vs trying to work around this.
Just saying!
/b
On Apr 29, 2010, at 11:05 AM, Peter P GMX wrote:
I have 3 of them, in Germany I think there are about 10 Mio. Plus France, Austria,Switzerland etc. I would say some dozens of Millions.
Best regards Peter
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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 FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Peter, just for the record, this box is absolutely unknown in France. The french market is saturated with triple-play boxes as they are provided for free by ISPs and the leading ISPs probably have around 95% of the market. I would not speak for the other countries, though. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 29/04/2010 à 18:05, Peter P GMX a écrit :
I have 3 of them, in Germany I think there are about 10 Mio. Plus France, Austria,Switzerland etc. I would say some dozens of Millions.
Best regards Peter
Brian West schrieb:
OK how many of these devices are we talking about that you have?
/b
On Apr 29, 2010, at 10:34 AM, Saeed Ahmed wrote:
i think usually not, because with FS sometimes you get the patch within minutes, which you'll never get somewhere else :-)
On Thu, Apr 29, 2010 at 5:01 PM, Anthony Minessale <anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com>> wrote:
Does anyone ever report this issue to Fritzbox or is too hard to conceive that they have the bug in this situation?
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Can I ship you a hammer already? /b On May 1, 2010, at 6:21 PM, David Ponzone wrote:
Peter,
just for the record, this box is absolutely unknown in France. The french market is saturated with triple-play boxes as they are provided for free by ISPs and the leading ISPs probably have around 95% of the market.
I would not speak for the other countries, though.
participants (7)
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Anthony Minessale
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Brian West
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David Ponzone
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Mathieu Rene
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Nicola von Thadden
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Peter P GMX
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Saeed Ahmed