WTF is a jitterbuffer?
http://lmgtfy.com/?q=jitterbuffer
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: Re: [asterisk-users] JITTERBUFFER function
On Thu, Jan 29, 2015 at 4:56 AM, Torbjorn Abrahamsson
torbjorn.abrahams...@gmail.com wrote:
Hello!
I am going to use the JITTERBUFFER function in a SIP (and local channels)
only setup, but have some questions of how to use it:
1. Do I
On Thu, Jan 29, 2015 at 4:56 AM, Torbjorn Abrahamsson
torbjorn.abrahams...@gmail.com wrote:
Hello!
I am going to use the JITTERBUFFER function in a SIP (and local channels)
only setup, but have some questions of how to use it:
1. Do I need to activate jbenable in sip.conf? Or is it
Hello!
I am going to use the JITTERBUFFER function in a SIP (and local channels)
only setup, but have some questions of how to use it:
1. Do I need to activate jbenable in sip.conf? Or is it enough to call
the JITTERBUFFER function?
2. What is the preferred way to invoke
1. Do I need to activate jbenable in sip.conf? Or is it enough to
call
the JITTERBUFFER function?
You only need to use the JITTERBUFFER function.
The jbenable option will enable a jitter buffer on every channel
created for that peer (or, if global, for every peer in the system).
I thought this meant that jbenable alone was not enough, and that you
needed to set jbforce=yes. Incorrect then
Answering myself, it seems I was incorrect, as jbenable is enough to
activate the buffers. I see the options different meanings now. Sorry about
the buzz.. :)
Second, if I
- dotnetdub dotnet...@gmail.com wrote:
I would not think you'd need to worry about jitter on a normal 100mbit LAN
unless there is heavy traffic or people are running their PC's through the
phone (don't remember if the 501 has two ethernet ports...). Typically the
quality issues
Do you seperate your voice and data networks?
On 9 April 2010 14:56, Tim Nelson tnel...@rockbochs.com wrote:
- dotnetdub dotnet...@gmail.com wrote:
I would not think you'd need to worry about jitter on a normal 100mbit
LAN unless there is heavy traffic or people are running
- dotnetdub dotnet...@gmail.com wrote:
Do you seperate your voice and data networks?
Un-top-posting...
Yes, I separate voice and data. Typically this is done using separate switches
where possible, other times, using VLANs with appropriate QoS. Regardless, your
phone and PC are
On 9 April 2010 16:46, Tim Nelson tnel...@rockbochs.com wrote:
- dotnetdub dotnet...@gmail.com wrote:
Do you seperate your voice and data networks?
Un-top-posting...
Yes, I separate voice and data. Typically this is done using separate
switches where possible, other times, using
What is the consensus on using the 1.4 jitterbuffer? Do most people
enable it?
We have a PSTN server that has our RBS T1 trunks in a central location,
then have clients that connect via SIP to us for access to those trunks.
Most of them are just fine, but lately we have a handful that are
- Jeff LaCoursiere j...@jeff.net wrote:
What is the consensus on using the 1.4 jitterbuffer? Do most people
enable it?
We have a PSTN server that has our RBS T1 trunks in a central
location,
then have clients that connect via SIP to us for access to those
trunks.
Most of them are
On Thu, 8 Apr 2010, Tim Nelson wrote:
- Jeff LaCoursiere j...@jeff.net wrote:
What is the consensus on using the 1.4 jitterbuffer? Do most people
enable it?
We have a PSTN server that has our RBS T1 trunks in a central
location,
then have clients that connect via SIP to us for
- Jeff LaCoursiere j...@jeff.net wrote:
On Thu, 8 Apr 2010, Tim Nelson wrote:
- Jeff LaCoursiere j...@jeff.net wrote:
What is the consensus on using the 1.4 jitterbuffer? Do most
people
enable it?
We have a PSTN server that has our RBS T1 trunks in a central
location,
I would not think you'd need to worry about jitter on a normal 100mbit
LAN unless there is heavy traffic or people are running their PC's through
the phone (don't remember if the 501 has two ethernet ports...). Typically
the quality issues are introduced on your WAN connectivity between the
I continued trying.
Now I reached 2 results.
1.
Asterisk ver1.6 or more has bug .
When you want to use jitter and PLC and want to see packet-log , you will
set ' jblog=yes ' on 'sip.conf '.
But Asterisk can't make log-file.
In /tmp/ packet-log-file will be made, if jb-modules work
Thank you for advice.
Do you get the same results if you use:
iax2 test losspct x
Where x is the loss percent you'd like to test?
Yes, I did it.
On CLI show:
VvvvLvvvLLvv
vvLvvLvvv
What user are you running Asterisk as?
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hi.
What user are you running Asterisk as?
I tried 2 patarn.
First , I worked asterisk as 'asterisk', and tested.
But jitter and PLC didn't work correct.
So I thought it may be caused permission problem,
and made a new system working asterisk as 'root'.
Now I tested as root.
And same
On 16/01/10 12:56 AM, nak...@02.246.ne.jp wrote:
Hi, I have a question about jitterbuffer and PLC.
Do you get the same results if you use:
iax2 test losspct x
Where x is the loss percent you'd like to test?
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Hi, I have a question about jitterbuffer and PLC.
I use Asterisk 1.6.2.0 and 1.6.0.20 or older.
I use uLaw.
My system map:
=
[ asterisk 2 ] -- # LOSS # -- # A # -- [ asterisk 1 ] -- # B # -- [ X-lite ]
Hello,
From http://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer, it says
that there For Asterisk 1.2 there was no jitterbuffer in the RTP-based
channels (i.e. chan_sip).
I am using 1.2 and Ind there is no reason to upgrade. Are there any
developments on this?
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James
1 sep 2009 kl. 08.17 skrev James Mutuku:
Hello,
From http://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer,
it says that there For Asterisk 1.2 there was no jitterbuffer in the
RTP-based channels (i.e. chan_sip).
I am using 1.2 and Ind there is no reason to upgrade. Are there
The project I am working on is really big. Unless I upgrade during
christmas(by then the project will be several months overdue). Just googled
further and saw some patches. I will try them and see.
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On 1/09/09 9:43 PM, James Mutuku wrote:
The project I am working on is really big. Unless I upgrade during
christmas(by then the project will be several months overdue). Just
googled further and saw some patches. I will try them and see.
In which case you probably shouldn't be using Asterisk
I did am not the one who started the project. the client has been running
1.2 for years and they needed additional features set up
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On 1/09/09 10:02 PM, James Mutuku wrote:
I did am not the one who started the project. the client has been
running 1.2 for years and they needed additional features set up
There was an Asterisk backports site - you might want to check in google
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Cheers,
Matt Riddell
Director
Matt Riddell wrote:
On 1/09/09 10:02 PM, James Mutuku wrote:
I did am not the one who started the project. the client has been
running 1.2 for years and they needed additional features set up
There was an Asterisk backports site - you might want to check in google
Pretty sure that site is
I doubt using one of the patches is a good idea either, it will lack the
needed testing and it's all quite fragile.
Zoa
Leif Madsen wrote:
Matt Riddell wrote:
On 1/09/09 10:02 PM, James Mutuku wrote:
I did am not the one who started the project. the client has been
running 1.2
It's long gone.
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hi,
I am working on a project to perform the voip call quality.
i want to get some statistics about the call quality with asterisk.
I used the following command: iax2 show netstats
and the result changes depending on the configuration of iax.conf.
When i enable jitterbuffer=yes and
i want to get some statistics about the call quality with asterisk.
I used the following command: iax2 show netstats
and the result changes depending on the configuration of iax.conf.
When i enable jitterbuffer=yes and forcejitterbuffer=yes, i get the
following result:
voip*CLI iax2 show
Hi Tony,
please do not send HTML-only messages to mailing lists. There are a lot
of people using mail programs that do not display html. You're html
messages are annoying and that's the reason why you get only a few
answers.
Volker
On Fr, 02 Nov 2007, Tony Plack [EMAIL PROTECTED] wrote:
When initiating a call from a SIP phone to another SIP phone through Asterisk 1.4 (latest SVN), I get the following:
[Nov 2 10:08:55] WARNING[7292] abstract_jb.c: Failed to put first frame in the jitterbuffer on channel 'SIP/5001-08266108'
[Nov 2 10:08:55] WARNING[7292] abstract_jb.c: Failed to
but keep in mind, that jb for sip (generic jitterbuffer) is implemented
differently, than iax, so it works only for SIP-SIP calls, or SIP-ZAP
and, curious, eg. for SIP-ZAP call must be activated for (outgoing) ZAP
channel :-\
yusuf wrote:
[EMAIL PROTECTED] wrote:
In iax.conf there is
[EMAIL PROTECTED] wrote:
In iax.conf there is option jitterbuffer
how about sip protocol ? Are jitterbuffer can configure in sip.conf ?
In 1.2 and 1.0 there is no jitter buffer for SIP. I think 1.4 might
have a SIP jitter buffer, but I'm not sure. Check sip.conf.sample in 1.4.
In iax.conf there is option jitterbuffer
how about sip protocol ? Are jitterbuffer can configure in sip.conf ?
Thanks, for your share
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[EMAIL PROTECTED] wrote:
In iax.conf there is option jitterbuffer
how about sip protocol ? Are jitterbuffer can configure in sip.conf ?
Thanks, for your share
If you upgrade to 1.4, there is a jitterbuffer available now for the SIP
channel.
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thanks,
Yusuf
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This message has been
I know, that jitterbuffer should be set at receiving side and on
outgoing call leg,
ie. if sipphone calls to asterisk and outgoing to zap chanel, I should
set jitterbuffer on zap channel (to dejjitter audio stream from sipphone)
but what about pure voip situation (i.e. iax-iax, sip-iax,
please, can somebody tell us, if currently used jitterbuffer
implementations (iax or sip w/ jb patch) are really working/usefull if
jitter is frequently changing between 10-1000ms (on cdma connection)?
I have really big problems with using jitterbuffer between two asterisks:
- with iax2, I
I'm trying asterisk 1.2.9.1 with rtp jitterbuffer patch from
http://asterisk-backports.org
and seems, that this working only for sip-sip calls (probably also for
sip-zap),
I have jb enabled and forced in sip.conf, I can see debug log messages
from jitterbuffer, but only for sip-sip calls, not
Thank You Patrick,After some minor problems in some file paths I had success compiling.The only problem was the codec_g726 witch does an illegal call and Asterisk doesn´t come up. But I only use g729 or g723 so I´ve deleted it from the modules directory and asterisk came up.
I´m going to test it
for a link to patch.
Jan Fousek
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Komu: asterisk-users@lists.digium.com
Datum: 09.08.2006 23:26
Předmět: [asterisk-users] Jitterbuffer on SIP
Thank You Patrick,
After some minor problems in some file paths I had
Hi,Is that a way to patch a running asterisk 1.2.9.1 instalation with the experimental SIP Jitterbuffer support ?Thanks
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On Tue, 2006-08-08 at 10:45 -0300, Thierry Querette wrote:
Hi,
Is that a way to patch a running asterisk 1.2.9.1 instalation with the
experimental SIP Jitterbuffer support ?
Yes, see http://www.asterisk-backports.org
http://asterisk-backports.org/downloads/ast_jb-1.2.9.1+rtp-keep-jb+fax
On 25 May 2006, at 20:43, Dr. Michael J. Chudobiak wrote:
I've been having problems with incoming IAX2 calls - some work, but
a large fraction are answered with dead air or disconnects from
my IAX provider.
Disabling the jitterbuffer seems to eliminate the problem (so far)!
Has anyone
There isn't quite enough info in that log to tell what is going on.
What you have above is part of 2 separate conversations.
You have the tail end of a successful registration with 70.87.18.51
and the HANGUP of a call with 64.26.157.230 which your asterisk seems
to be confused about.
Could you
I've been having problems with incoming IAX2 calls - some work, but a
large fraction are answered with dead air or disconnects from my IAX
provider.
Disabling the jitterbuffer seems to eliminate the problem (so far)! Has
anyone else seen this? I'm using 1.2.6, but I'm not sure what my
Dr. Michael J. Chudobiak wrote:
Disabling the jitterbuffer seems to eliminate the problem (so far)! Has
anyone else seen this? I'm using 1.2.6, but I'm not sure what my
provider is using.
Oops, the problem still happens without the jitterbuffer - so something
else is causing it. Any ideas?
On Thu, 2006-05-25 at 16:10 -0400, Dr. Michael J. Chudobiak wrote:
Dr. Michael J. Chudobiak wrote:
Disabling the jitterbuffer seems to eliminate the problem (so far)! Has
anyone else seen this? I'm using 1.2.6, but I'm not sure what my
provider is using.
Oops, the problem still happens
I find that DTMF does not work reliably if jitterbuffer=on for certain
IAX providers. For instance, DTMF tones are missed entirely about half
the time when I dial into an exgn.net account. However, it always works
fine for an unlimitel.ca account.
Someone else has seen this too:
I find that DTMF does not work reliably if jitterbuffer=on for certain
IAX providers. For instance, DTMF tones are missed entirely about half
the time when I dial into an exgn.net account. However, it always works
fine for an unlimitel.ca account.
Someone else has seen this too:
Rich Adamson wrote:
I find that DTMF does not work reliably if jitterbuffer=on for certain
Can anyone suggest a workaround (other than jitterbuffer=off)?
Might try turning off trunking (assuming you have it turned on) and
test again. Seems a couple of parameters interact and probably has
On Feb 27, 2006, at 6:09 AM, Dr. Michael J. Chudobiak wrote:
I find that DTMF does not work reliably if jitterbuffer=on for certain
IAX providers. For instance, DTMF tones are missed entirely about half
the time when I dial into an exgn.net account. However, it always
works fine for an
this is a time issue.
change your date to older value. everything works again.
paradise dove
On 1/25/06, stevanus [EMAIL PROTECTED] wrote:
Hi guys,
I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at
the third days I activated setting jitterbuffer=yes and suddenly there
Hi guys,
I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at
the third days I activated setting jitterbuffer=yes and suddenly there
is no voice when the call is picked up. It's really weird as if asterisk
stops sending rtp packet. I've checked asterisk log and found
Hello All,
What is the advantage of jitterbuffer on
zap channel? do you have any suggestion on the setting for home usage? Is there
anydisadvantage in using it?
Cheers,
Anto
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I have users that are using IAX clients, dialling into
meetme conferences. They will be on varying connection speeds. Firstly, should
jitterbuffer be used with meetme? Secondly, I have read some posts which
indicate that jitterbuffer is not that stable. Is it stable enough to
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