similar to: To jitter buffer or not to jitter buffer?

Greetings list, Quite a few of our users seem to be experiencing poor voice quality when they're using internet connections over which we have little or no control (i.e. they're using their own router with no QoS, etc.). Some of these connections are giving a qualify time within asterisk of 130ms+. Are there any recommendations as to phones with particularly good buffering that might

This are "suppose" to be SIP based phones. I just got one to play with, no docs, no nothing. before I start spending time trying to sort it out I thought I'd ask to see if anyone had made one work with AST, and if they have any pearls of wisdom. mucho thanks

Hi, I'm just wondering how other people cost/charge for * boxes they have installed in small businesses? Asterisk is such a complex beast and is capable of so much it seems equally complex to figure out a charging model!? I'm contemplating installing a pair of servers at a small company with two separate locations, one main aim is to provide a IAX link between the two but I'm sure

On 9/29/2017 3:37 PM, Rolf Turner wrote: > On 30/09/17 07:45, JLucke at ria.buffalo.edu wrote: > > <SNIP> > >> >> The conceptual paradigm for R is only marginally commensurate with >> that of >> standard statistical software. >> You must immerse yourself in R to become proficient. > > Fortune nomination. For newer list members wondering what

Hi all, Is it possible to use 2 incoming fxo lines (one is for my company the other for the family) with Asterisk@home? Best regards, Francois Random Thought: --------------- Errors like straws upon the surface flow: Who would search for pearls must dive below. - John Dryden, 1631 - 1700

Hi, We're being hit by occasional very high load (up to 23 on a SUN Fire v440) on our LDAP server coming from a query of (objectClass=posixAccount). This currently returns over 54,000 entries ... This is coming from the getpwent() call in util_getent.c:getpwent_list() This is coming from srv_samr_nt.c:get_memberuids() But we're having trouble working out what triggers this call: it

Hi all, Does anyone happen to know if there are *any* 32-bit architectures (on which Linux runs) for which the ABI for a "long long" is different from passing two "longs" in the appropriate order, i.e. (hi,lo) for bigendian or (lo,hi) for littleendian? I'd like to switch klibc to use the 64-bit file ABI thoughout, but it's a considerable porting effort, and

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 - transmission layer - eg. rtp). Kapul On Tue, Sep

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 attachment was scrubbed... URL:

Hi All, I have a question regarding the nominal jitter buffer configuration: The call was setup as G.729A (annexb=no), ptime=20ms and nominal jitter buffer size = 50ms, and round trip delay is 200ms, the TDM side will experience intermittent one way voice during the call, but IP side can always heard the voice from TDM side. My question is, should this possible caused by the nominal jitter

Hi, Our project is using the jitter buffer feature built in Speex. We noticed there are some latency when using the jitter buffer. Does anyone know what is the "worst case" latency inherent in the jitter buffer algorithm? I believe someone already mentioned that it's adaptive but is there a worst case hard number (in terms of 20ms Speex frames)? I'm not familiar with the

Hi, sorry for the repost again, but does anyone have a code snippet example of how to use the jitter buffer? Regards, Andy ----- Original Message ---- From: Andy Ngo <ndno72-speex@yahoo.com> To: speex-dev@xiph.org Sent: Wednesday, January 10, 2007 8:09:10 PM Subject: [Speex-dev] Sample snippet of how to use jitter buffer Hi, I searched around in the Speex manual and API but

Hi, I'm trying to use the jitter buffer feature that comes with Speex but I'm getting unexpected latency. I wrote a client application that does VOIP-like functions and without using jitter buffer, the end-to-end latency is around 250 ms (I'm using lowband 5.97 kpbs). However, when I tried to incorporate the jitter buffer feature, the latency would grow as time elapsed (up to a few

Hello, I've only one question: how does a jitter buffer work? Regards Kapul

('binary' encoding is not supported, stored as-is) I'm working on a voip project with Speex and the only major design obstacle left (that I'm aware of) is the jitter buffer. Before I attempt to write something of my own I'd like to try out the jitter buffer functions in Speex's unstable release. Can anyone give me a basic run down of how the experimental jitter buffer is

Hi all I have seen that, for speex 1.1.6, there is a jitter buffer implementation. I´d like to know, please, what kind of jitter algorithm it´s been used to implement it. Is there anything like an algorithm overview or something else that someone could send me? Thanks in advance and best regards Guild __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new

> 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 to use with

Hello, I think that the new Jitter Buffer have a problem. It works perfectly when I call the speex_jitter_put every 20ms (on my lan) but in other case (with big latency on Internet connexion), it doesn't works. The old version is OK in all cases. Hope it will helps. Thanks Ouss -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 > I think that the new Jitter Buffer have a problem. > >

We have a customer on a wireless connection that has very bad jitter. They can hear people fine, but people have a very hard time hearing them. They are connected via a SPA-2102. It is a SIP client going to a SIP trunk. Something like this in sip.conf [general] would be in effect for all SIP clients: jbenable = yes jbmaxsize = 150 jbresyncthreshold = 1000 jbimpl = fixed jblog = yes I only want