similar to: Jitter buffer

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 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] On Behalf Of Jean-Marc Valin Sent: Tuesday,

kapul@ai-bot.net wrote: >Hello, > >I've only one question: how does a jitter buffer work? > > > Have you tried typing "jitter buffer" into google? Were the results there not satisfactory? -SteveK

[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

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 to use with other codecs and reduce the

Are these things possible? 1) Set the local Asterisk jitterbuffer size, but only for a particular connection. I'd like to force Asterisk to use a particularly large buffer in certain cases. Should I expect this to work? [general] jitterbuffer=no register => username:password@parcelfarce.domain.net ;parcelfarce register => username:password@iaxtel.com ;iaxtel [parcelfarce]

I'm assuming asterisk does not have a SIP jitter buffer in place? Any ideas on how to help with this going over a data T1 where VoIP is shared with regular traffic? Problem is when people are downloading the voice is jittery, even lossy. Matt

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:

We have a customer who is connected to our PSTN gateway using IAX and noticing that even when the traffic from their site is modest their outbound audio has short dropouts. Inbound audio is fine. (They have ADSL so it is expected that outbound audio would be the first to experience problems.) We have several questions to pose to the collective wisdom of this list. Q1: Are there any statistics

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

Jean-Marc Valin wrote: >>OK, I'm actually about ready to start working on this now. >> >>If people in the speex community are interested in working with me on >>this, I can probably start with the speex buffer, but I imagine >>there's going to be a lot more work needed to get this where I'd like >>it to go. >> >> > >And where

Danny Chan wrote: >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. > > 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

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

Hi everyone, Judging from the lack of complaints about the split, I think I can go safely forward with that. The only thing to decide is the name of the new library that will contain the preprocessor, the AEC, jitter buffer, and resampler. So far, I've got a few potential names: - libspeaks - libspeexproc - libspeexvoip - libspeech - libspoke Let me know (in private so as to not flood the

Hi Folks, I know the PAP2T-NA has a jitterbuffer.... however, it seems to be adaptive, which is fine for most situations... however, is there some way I can either: A) Specify how long it waits before it starts to shrink? B) Specifiy a fixed sized jitterbuffer? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Jean-Marc Valin wrote: >>In particular, (I'm not really sure, because I don't thorougly >>understand it yet) I don't think your jitterbuffer handles: >> >>DTX: discontinuous transmission. >> >> > >That is dealt with by the codec, at least for Speex. When it stops >receiving packets, it already knows whether it's in DTX/CNG mode.

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 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

> Yes. Jean-Marc has made the API more similar. > > Jean-Marc: Have you looked at the API we have for the > asterisk/iaxclient jitterbuffer? Just did. > It's pretty close to what you have now -- the major difference is that > your jb still assumes it can "own" the data passed in -- it copies it, > and it destroys it at will. With the API I put together,

On 5/3/06, Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca> wrote: > > I must say I really like the generalized jitter buffer though :) It's a > > cleaner and more flexible implementation and can more easily be adjusted > > to contain additional information with each packet. This looks interesting to tie into asterisk's jb and plc code as well.