similar to: Asterisk, newbie, fwd and is this jitter?

Hi I have been playing around with asterisk for a few days now. I have asterisk running with a single x100p card and a couple of x-lite "extensions". Here's where I am at: I can make calls between the extensions. Voice mail seems to work OK. I can use the x100p card to dial out to the PSTN over the analogue interface and receive calls fine. I have my FWD account hooked up as well

How do you setup the timing in Meetme conference? I have a x100p and tdm4x card. When I dialing to my conference I get a request to schedule in the past error message. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 10, 2004 10:48 AM To:

Hello, R 4.0.2 on Ubuntu 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

Hi folks! I'm using an X100P (connected to my phone line) and an S100U, and when I calls out from the phone connected to the S100U it is a very bad sound quality, it "pops" and "jitters" a lot. But internal calls from for example a SIP client to the phone on the S100U sounds good. Calls from an SIP client to the outside world using the X100P also works good! My second

I've been trying to resolve some quality issues and I was hoping someone might be able to provide some insight. To give you an idea the calls are coming in via a SIP DID and sent out via an IAX2 connection. Latency to both the SIP equipment and IAX equipment are around 80ms with 0 packet loss accoridng to ping tests. The server is located in a data centre so bandwidth is not an issue. Most

> 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

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) The jitter

> 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

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

Hello, The way you describe how the jitter buffer should be implemented makes me wonder: How does the jitter buffer works when there is no transmission? Let's say my "output" thread gets a speex frame from the jitter buffer every 20ms. What happen when there is no frame that arrived on the socket? No frames at all for a pretty long time (ie many seconds). This is my case because I

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

> > 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

Right - and I deal with that on the receiver end based on an approximation of sender's and receiver's responsiveness - the minimum latency I've been able to get into the system is about 150 ms. Of that, jitter buffering is about 40-100ms. I'd love to figure out how to get that down without killing myself on thread switching or Win32 kernel calls, but ms has to actually implement

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

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

The audio frame speex generates sounds pretty terrible most of the time, and I don't use it for jitter correction instead I just use it for dropped packets - so I usually drop the late packet. It sounds acceptable as long as I drop less than 5% of traffic (dropping 2 in a row makes a bad robot noise, so I reset the stream in that case). The good news is that on an unsaturated DSL line jitter

> 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 > http://www.voip-info.org/tiki-index.php?page=Asterisk%20new% > 20jitterbuffer Yes, this

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

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