similar to: SIP Jitter Buffer Patch for 1.2.x branch?

Oh, I forgot to mention one thing. I currently append a few bits of custom information to each speex packet which I fetch out with things like speex_bits_unpack_unsigned(&sjJitter.current_packet, 1); It would be very usefull if the jitter buffer didn't actually decode the packet, but instead returned a pointer to it (or NULL if you should play silence and -1 if you should repeat the

Hello. I'm having Asterisk 1.6.0.x and trying to solve the issue concerning with a bad quality of voice for incoming SIP calls into the app_meetme. As I know, in my case of calls, jitter buffer is NOT executed on anyone channel. So, after reading Russell Bryant's post ( http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/) I added following scheme

Greetings list, Some time ago (probably about a year ago now) we disabled IAX jitter buffering on all our boxes because it was causing issues in a mixed 1.0 and 1.2 environment. One thing I've noticed over the last few months as more and more clients have moved from the 512k/1mb/2mb ADSL connections they were using onto "up to 8mb" connections is that whilst overall throughput is a

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.

> We just return a frame with the return value JB_DROP, which tells the > caller to drop this frame, and call jb_get again. > > When the caller is done with the jitterbuffer, it calls jb_getall() > repeatedly, until it's empty, and then it can discard all the frames. Hmm, looks a bit error-prone to me. Especially considering I still have to explain that "no, you

Hi, We've been having mixed success (mostly good) with the jitter-buffer, and I'm trying to understand what my options are for tuning JB parameters to our particular use-case. In what may become a series of emails, my first question is how to properly set the margin parameter. I have set up a test harness that allows me to synthesize arbitrary sequences of puts/gets, so that I can test

Hi, I would like to test chan_sip with a bigger jitter buffer. Does anybody know where in the code this is defined? I looked through it but could not find where. If anybody else can find it please let me know. Regards, Andres

> Perhaps, but then you need to assume that the jitterbuffer can just > throw away the data, and that limits how you can use it. In object- > oriented terms, you might want to pass objects to the JB, and then > call a destructor on them. In C terms, you may want to allocate > frames via malloc(), and then call free() on them later. You might > want to pass in

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.

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

> 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,

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

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

>> After changing my code to construct a JitterBufferPacket and passing that >> to speex_jitter_put, my program works with the new jitter buffer using the >> wrapper functions (speex_jitter_* instead of the new jitter_buffer_*). > > Oops, I forgot to make sure I keep the API stable for the old buffer. > Any thoughts on the change (revert or continue as is)? Well, I

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:

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 this function? Say I have channel A which is not in need of buffering, while channel B do need it. If A

> > 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). > Depending on the version of Asterisk, it will also place the

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

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. > >

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