Displaying 14 results from an estimated 14 matches for "dejitt".
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dewitt
2004 Dec 21
2
Jitter buffer
[sorry for the loss of proper attributions, this is from two messages]:
[Me]
>This is something I've encountered in trying to make a particular
> asterisk application handle properly IAX2 frames which contain either
> 20ms of 40ms of speex data. For a CBR case, where the bitrate is
> known, this is fairly easy to do, especially if the frames _do_ always
> end on byte
2005 Feb 12
2
Intermediary jitter buffering
Hello,
I understand that only the destination of a call should do jitter
buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no
transfers), PhoneA and PhoneB need to perform their own jitter buffering,
and Asterisk will just forward the frames, correct?
What happens if the peer does not support jitter buffering, but is
close by so there's no need for jitter buffering? My
2008 Sep 22
2
Newbie: Get echo cancellation level
Hi:
I'm using speex to perform echo cancellation in Windows. I'm aware of the problem about out of sync clocks in record and play sample rates in usual sound cards . In order to have an idea of how good is my echo cancelation working I would like to know if there is any #define thing i can pass to speex_echo_ctl to get the actual level of echo cancellation. If not, how can i extract that
2004 Nov 17
1
Jitter buffer
...r
speaker's frame (transcoded if needed), and we mix and recode the
summation of each speaker for all others.
In the application we're using, there can be a _lot_ of jitter (not just
the 200ms worth that your jitterbuffer seems to account for, but 1
second or more), and if we don't dejitter first, we can easily end up
with cases where:
a) We send out subsequent frames for different speakers with overlapping
timestamps.
b) Different speakers have different clock skews, and over time, these
will be very significant. In this case, as speakers change, listeners
will see this as a...
2010 Apr 08
3
jitterbuffer
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 having
latency and jitter issues. I am hesitant to just turn on the jitter
buffer in
2007 Feb 14
6
Fax with T.38
Hi all,
I install the last version of Asterisk and I tried to send faxes, but
nothing works.
Here is my configuration:
Analog Fax <----> IP <----> Asterisk <----> IP <----> Patton M-ATA
<----> Analog Fax 2
I tried Analog Fax 2 -> Analog Fax but nothing works!!
In the Patton configuration I put G711 and no silence suppression.
In asterisk I have
2008 Aug 28
0
meetme + jitter buffer
Hi,
I was wondering if there's any sense in increasing audiobuffer above the minimal '2' in meetme, if every channel is already dejittered before (Local/.../nj - as described at: http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/)
Will it help in anything, or just increase delay?
Thanks,
Stan
2004 Nov 16
0
Jitter buffer
...based on the return value of the _get
> method: dropping frames, interpolating, etc.
I guess...
> I can see how you'd do that, but I don't think that would work for me.
> I really don't want the jitterbuffer to handle decoding at all,
> because in some cases, I want to dejitter the stream, but not decode
> it.
In that case, your callback can just send the encoded stream somewhere
else, it doesn't have to actually decode anything.
> For example, I will be running this in front of a conferencing
> application. This conferencing application handles particip...
2004 Nov 16
2
Jitter buffer
...o
>modify the current implementation to make that easier (though it's
>already not very hard).
>
>
I can see how you'd do that, but I don't think that would work for me.
I really don't want the jitterbuffer to handle decoding at all, because
in some cases, I want to dejitter the stream, but not decode it.
For example, I will be running this in front of a conferencing
application. This conferencing application handles participants, each
of which can use a different codec. Often, we "optimize" the path
through the conferencing application by passing th...
2004 Aug 27
5
iaxtel and jitterbuffer
I am trying to work out IAX <--> IAX communications with my * box. I have a
registration with iaxtel and I thought I would start there for my learning.
I am able to call the number for Digium's support line (700-428-6000), but the
sound is horribly chopping. Some reading revealed the jitterbuffer settings,
so I enabled them in iax.conf. I have the following now:
; Inter-Asterisk
2005 May 13
0
Problem with IAX trunking
...;
; You can adjust several parameters relating to the jitter buffer.
; The jitter buffer's function is to compensate for varying
; network delay.
;
; All the jitter buffer settings except dropcount are in milliseconds.
; The jitter buffer works for INCOMING audio - the outbound audio
; will be dejittered by the jitter buffer at the other end.
;
; jitterbuffer=yes|no: global default as to whether you want
; the jitter buffer at all.
;
; dropcount: the jitter buffer is sized such that no more than "dropcount"
; frames would have been "too late" over the last 2 seconds.
; Set t...
2004 Nov 17
3
Jitter buffer
Jean-Marc Valin wrote:
>>Heh. I guess after playing with different jitter buffers long enough,
>>I've realized that there's always situations that you haven't properly
>>accounted for when designing one.
>>
>>
>
>For example? :-)
>
>
I have a bunch of examples listed on the wiki page where I had written
initial specifications:
2004 Nov 15
2
Jitter buffer
Jean-Marc Valin wrote:
>>I believe it is adaptive, but no, I haven't used it, because it's
>>coupled only to the speex codec. We're working on a generic
>>application and codec-independent jitter buffer algorithm, for use in
>>asterisk and iaxclient (at least). Some information is available at
2006 Apr 28
1
mISDN: No DID/extension information returns busy to caller
...o
; Set this to no to disable echotraining. You can enter a number > 10
; the value is a multiple of 0.125 ms.
;
; default value: no
; yes = 2000
; no = 0
;
;echotraining=no
echotraining=2000
;
; chan_misdns jitterbuffer, default 4000
;
jitterbuffer=4000
;
; change this threshold to enable dejitter functionality
;
jitterbuffer_upper_threshold=0
;[intern]
; define your ports, e.g. 1,2 (depends on mISDN-driver loading order)
;ports=1,2
; context where to go to when incoming Call on one of the above ports
;context=Intern
;[internPP]
;
; adding the postfix 'ptp' to a port number is...