search for: jitter

Dear all, i have noticed some strange behaviour in the ?jitter? function in R. On the help page for jitter it is stated that "The result, say r, is r <- x + runif(n, -a, a) where n <- length(x) and a is the amount argument (if specified).? and "If amount is NULL (default), we set a <- factor * d/5 where d is the smallest difference betwee...

Dear all, i have noticed some strange behaviour in the ?jitter? function in R. On the help page for jitter it is stated that "The result, say r, is r <- x + runif(n, -a, a) where n <- length(x) and a is the amount argument (if specified).? and "If amount is NULL (default), we set a <- factor * d/5 where d is the smallest difference betwee...

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 fo...

Hi, I thought I'd repost this to the -users list for some background on the jitter buffer and its workings and remaining issue.s I'll also pu a little executive summary here at the top: Where a channel is native bridged to another iax2 channel: 1) Lag is not measured and will usually show 0ms. Any other number is an old measurement from the start of the call 2) Th...

http://www.asterisk.org/node/48317 does a nice job of explaining the 1.4 jitter buffer, however it raised a question in my mind. In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the UDP RTP packets renumbered on transmit, or is the original sequence number preserved in the UDP header? A comment is made on the referenced blog that jitter buffering is bes...

How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070803/c6d473...

Good question. I'm coming to the conclusion that using plain UDP and "home-grown" packet construction for transmitting the speex data (with timestamp/sequence counter) and implementing jitter control on the receiver end is an adequate implementation for a VoIP application. Assuming of course that I don't care about any interoperability issues with other applications etc. I was under the impression from reading some RTP documents that such stack would be necessary to achieve accepta...

Hi Jean-Marc, Thanks for your response. Given a worst case scenario, what is the "worst case" latency (in terms of Speex frames) that the jitter buffer algorithm will incur? We're trying to determine the worst case hard number. Sorry for unclear question below; what I was trying to ask is that given a worst case latency (which I'm asking in the first question) inherent in using the jitter buffer feature, is that any way in the cod...

...u 20.04, sessionInfo at end. This came up in r-help, I'm answering to the OP and also posting to r-devel since I believe it is more appropriate there. I can confirm this. The original instructions are the first and the last, but even with smaller numbers the error shows up. set.seed(2020) jitter(c(1,2,10^4)) # desired behaviour #[1] 1.058761 1.957690 10000.047401 jitter(c(0,1,10^4)) # bad behaviour #[1] -92.43546 -1454.61126 8269.53754 jitter(c(-1,0,10^4)) # bad behaviour #[1] -1484.3895 -427.5283 8010.3308 jitter(c(1,2,10^5)) # bad behaviour #[1] 4809.238 10578.561...

Hi, I know that there was a jitter buffer patch (for sip) for the 1.0.9 branch some time agin. At this time, we can not upgrade to 1.4.x. Is there a useable, fairly stable INCOMING sip jitter buffer patch? That is.. I want Asterisk to jitter buffer incoming SIP packets. -------------- next part -------------- An HTML attachme...

Hi, I'm one of the people working on the Rockbox project (http://www.rockbox.org) which is an open source alternative firmware for a range Digital Audio Players. Recently we integrated support for the Speex codec using libspeex and seems to work well. If you could add Rockbox to your list of software that supports Speex, that'd be great. So that's the introduction done. Now for

Thank you for a very good explanation which shed light on some of the questions that I had after reading the source code. Reading your text however, I wonder if I'm perhaps missing an important point on the proper use of the jitter buffer: ... > Now, clearly, if early_ratio is high and late_ratio is very > low, the buffer is buffering more than it needs to; it will > skip a frame to reduce latency. ... Question: Do I understand it that I should not put every incoming packet through the jitter buffer? The way my...

Hi... In Stata, there is the ability to display scatter plots with data points at the same (x,y) location, using the 'jitter' command of the twoway scatter stata command. Anyone know of a way that I can do the equivalent thing in R? For non-Stata readers, if jitter is enabled in stata, and n-points would be at the same (x,y) location, the points are offset a bit (according to the jitter parameter) and you see...

Hi Jean and Steve, Can you tell me whether the jitter filter / buffer is adaptive type, I saw the description of speex_jitter.h say it is "adaptive", anyone of the group has implemented it and confirm it. Thank you all. Regards, Danny Chan -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org]...

Hi, I'd like know about anyone using the current jitter buffer in Speex. I'm planning on changing it to make it more general and I'd like some feedback about how to make it better. Also, let me know if you're doing anything serious with it and want to make sure I don't break your stuff. Basically, I want to make the jitter buffer easier...

Hmm, I tried... I completly understand an idea of jitter buffer and I know there is a lot of kinds of this solution (eg. AJB - Adaptive Jitter Buffer). I simply want to know what type is used in speex codec and how could I use that. What is the reason for using jitter buffer implemented in speex against to my own (implemented at lower layer - transmissio...

I'd like to use stripplot for some plots because I want to use the jitter parameter. On the other hand, I'd like to use dotplot because I'd like to have the horizontal lines that it includes. dotplot doesn't have a jitter option and I'm not having any success with getting panel.grid(h=-1) with stripplot. Can anyone show me how to make dotplot-like lines...

Is is possible to give a short hint about how the jitter buffer would "catch up" when network condition have been bad and then get better? I'm using the jitter buffer with success now, but sometimes I have a long delay that's caused by bad network conditions and then later when the conditions get better, I would think we would want th...

I use the speex version of your jitter, and in speex_jitter_get, you always call the jitter_buffer_update_delay. -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: dimanche 18 mars 2007 13:06 To: Ouss Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Problem with the svn jitter buffer >...

Hi, I have a SIP client which is registered to asterisk. Asterisk is registered to a SIP trunk and also handles the media. Now since my client has some issues in its RTP Tx, which seems to have some amount of jitter (mean jitter as per ethereal trace is about 17ms, max jitter is 20 ms and max delta is 85 ms), to over come that I have enabled jitter buffer in the SIP channel by setting sip.conf parameters jenable=yes, jbforce=yes, jbmaxsize=200 and jbimpl=fixed. However on setting these parameters I am unable t...