similar to: Jitter buffer latency

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

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, Is there a white paper describing how the jitter buffer in Speex works and what "side effects" (such as introducing delay) can happen? In the past, I have implemented a similar "dual buffer" algorithm to try to smooth out lost or delayed packets with a fixed know delay (no larger than 2 times the Speex frame). Thanks. Regards, Andy -------------- next part

Hi, I looked at the jitter buffer code and it seems like the maximum number of frames that Speex can hold up to is 200 (SPEEX_JITTER_MAX_BUFFER_SIZE). This is equivalent to 4 seconds (20msec/frame * 200 frames). Can I just reduce this constant to limit the size? I know I'll reduce the smoothness of delayed frames but I want to reduce the delay in case my audio application isn't

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

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. > >

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

Since r12660, the speex_jitter_get with high latency doesn?t works, I have no sound. Before this release, the speex_jitter_get works in all conditions. speex_jitter_get return void, then I cannot know the reason of this problem. Regards Ouss -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: dimanche 18 mars 2007 23:07 To: Ouss Cc:

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

('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

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, Hi :) First off, could you try to set your email client to break long lines before transmitting? In my (somewhat outdated) pine, the lines appear as VERY long lines when I try to reply, making it hard to read :) Minor detail though, I should probably fix pine. Some day. > The way you describe how the jitter buffer should be implemented makes me > wonder: How does the

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:

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:

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

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 the audio to gradually catch up with real-time

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

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

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

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: