similar to: How to set the jitter buffer

Displaying 20 results from an estimated 5000 matches similar to: "How to set the jitter buffer"

2005 Sep 08
10
voice over atlantic
Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup?
2003 Dec 09
2
Need help with jitter buffer/quality settings
I'm using Asterisk to do audio as well as H.263 video over SIP. Actually the video works pretty well but I have trouble with the audio. I'm wondering if someone can suggest codec/jitter settings or other tweaks. The system looks like this: Softphone <---ulaw----> Asterisk #1 <------IAX (usually GSM)---------> Asterisk #2 <-------- IAX (usually GSM) -------->
2004 Sep 10
3
call quality monitoring
I need to debug a call quality issue with remote users on the other end of a satellite link. The symptoms are: we here on the Internet side can hear them just fine. On their end, things work sorta OK most times, but they often suffer from severe dropouts and digital warbling, both of which I attribute to them missing packets. Often times they can't make out a word we are saying while we can
2005 May 07
1
Setting the jitter buffer in AIX
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]
2005 Feb 08
5
jitterbuffers - suggested settings
Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A & B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know
2007 Apr 20
6
How can I improve call quality?
Hi All, I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used for PSTN calls via IAX2. Our 'net link is a dedicated 2Mb fibre connection (of which we have ever used 50% max bandwidth). We've no E1/T1 links, everything is IP based. My boss complains that many of the calls he holds with others has a bad quality. He also says that its not just him. My iax.conf file
2006 May 22
1
SIP to IAX - forcing codec pass thru
hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP
2003 Jul 08
5
Using multiple iconnecthere accounts
Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that
2005 May 13
0
Problem with IAX trunking
Hi all, I'm trying to get IAX2 trunking between two * boxes and am having extreme difficulty :) What happens is when the sending * server (the one initiating the call) receives the ACCEPT back from the receiving server it immediately replies with INVAL. I've checked the code and it seems to be not matching the accept packet with the relevant item in the iaxs array due to the following
2004 Aug 27
5
iaxtel and jitterbuffer
I am trying to work out IAX <--> IAX communications with my * box. I have a registration with iaxtel and I thought I would start there for my learning. I am able to call the number for Digium's support line (700-428-6000), but the sound is horribly chopping. Some reading revealed the jitterbuffer settings, so I enabled them in iax.conf. I have the following now: ; Inter-Asterisk
2004 Aug 06
2
RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)
I've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and
2004 May 23
1
IAX2 REACHABLE/UNREACHABLE
All, I have an issue with IAX that I can't comprehend. Approximately every eight minutes my servers go unreachable. They stay unreachable for exactly 10ms. I have two servers running IAX and it happens on both servers simultaneously. I have searched the archives and see similar issues, but not the exact same one. I am on the current CVS stable version of *. Also, during IAX calls,
2005 Jan 17
1
here's my IAX callthrough app and some questions about problems I have.
Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,
2005 Sep 18
2
How does the jitter buffer "catch up"?
> FYI: The below is just my interpretation of the code, I might be wrong. Most of it is right. Actually, would you mind if I use part of your email for documenting the jitter buffer in the manual? > Each time a new packet arrives, the jitter buffer calculates how far ahead > or behind the "current" timestamp it is; this is called arrival_margin. > The "current"
2004 Aug 06
1
Speex settings and jitter
[Just curious, and seizing the opportunity to communicate with other folks who are doing the same kind of thing I am...] How are you measuring the latency? I tried measuring it with my program (also Win32-based, also using DirectSound[Capture]) and came up with around 130ms. To measure it, I placed the mic near a speaker to get feedback going, had my program connect to itself (local
2020 Sep 23
3
jitter-bug? problematic behaviour of the jitter function
Dear all, i have noticed some strange behaviour in the ?jitter? function in R. On the help page for jitter it is stated that "The result, say r, is r <- x + runif(n, -a, a) where n <- length(x) and a is the amount argument (if specified).? and "If amount is NULL (default), we set a <- factor * d/5 where d is the smallest difference between adjacent unique (apart from fuzz) x
2020 Sep 23
3
jitter-bug? problematic behaviour of the jitter function
Dear all, i have noticed some strange behaviour in the ?jitter? function in R. On the help page for jitter it is stated that "The result, say r, is r <- x + runif(n, -a, a) where n <- length(x) and a is the amount argument (if specified).? and "If amount is NULL (default), we set a <- factor * d/5 where d is the smallest difference between adjacent unique (apart from fuzz) x
2006 Oct 24
2
IAX2 goes "one way audio" when lag gets bad
Hi, I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. The customer is connected via IAX2 to our softswitch. On the customer's end I have the following config in iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all
2005 Feb 12
2
Intermediary jitter buffering
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
2007 Apr 11
3
SIP Jitter Buffer Patch for 1.2.x branch?
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: