search for: 90ms

...Factors = FALSE, header=TRUE, nrows=2.5e6) # read the file; 'nrows' is a slight # overestimate dat <- dat[,1:3] # select just the first 3 columns head(dat, 10) # print the first 10 rows # Select using which() as the final step ~ 90ms total time on my macbook air system.time( head( dat[which(dat$gender2=="other"),],), gcFirst=TRUE) # Select skipping which() ~130ms total time system.time( head( dat[dat$gender2=="other", ]), gcFirst=TRUE) Now I would think that the second one without which() w...

...g. For x265 (HEVC), I use BPG codec, and this time I use -m 0 setting which uses the fastest encoding setting of x265.I run the .exe 10x times and I pick the best scores. So here are the timings at -l13 very high compression: Encode time: x265 (BPG -m 0): 80ms NHW: 20ms Decode time: x265 (BPG): 90ms NHW: 10ms So with an equivalent level of optimization (x4-5, multithreading, SIMD,...), the NHW Project is 80/20x5=20x faster to encode and 90/10x5=45x faster to decode than x265 (optimized HEVC)! Yes, 45x faster to decode than x265 at very high compression!, this would really save on battery li...

hello I have a Cisco ATA working with asterisk, but I have east error when it is tried to validate in asterisk. Feb 2 10:04:42 NOTICE[1125342512]: File chan_sip.c, Line 5210 (handle_request): Registration from '<sip:501@216.xxx.xxx.72;user=phone>' failed for '200.xxx.xxx.24' Feb 2 10:04:42 NOTICE[1125342512]: File chan_sip.c, Line 5210 (handle_request): Registration from

...half Of Jonathan Tew Sent: Monday, February 02, 2004 1:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoicePulse IAX2 lag Is anyone else noticing high lag on their voicepulse IAX2 connections? We're seeing 500-600ms lag time when our pings to gw5.voicepulse.com are only 90ms. Our other IAX2 connections to this box are not experiencing anything else like this. Our server has hundreds of megs of free ram and is very idle CPU wise so I don't our Asterisk is bogging down. Just curious if anyone can shed some light on this. Thanks, Jonathan _____________________...

Hi, I have two analog phones connected on the digium tdm22b; I can't put calls on hold by pressing the R button on the phone. I can do it only by hook flash. How can I configure asterisk to use the R button? Thank you in advance.

I want to speed-up dialing on X101P clone (Ambient modem). I probably must change wcfxo.c, but what line to change? (On usual modems, I can type ATS11=50 to get tone dialing much faster (50ms instead of default 90ms). After that, I can write configuration to nvram (AT&W) to be permanent)

You cant go by pings. ICMP traffic is given lowest priority on internet routers, where voip rtp or iax might be given much higher priority. Plus I have 2 providers, the provider with the 90ms ICMP ping time is way better than the provider with the 15ms ping time. It depends on so many factors, including their equipment. I have a continuing problem with the voice dropping out for 1 second or less during a call and both providers have this problem but I haven't been able to figure o...

...4.22 released). It uses SIP to connect to the PSTN via a provider who is on the same LAN as the box (it is co-located at the provider). They also have about 20 SIP phones as extensions that connect to the box over the internet. "sip show peers" indicates that most phones have a latency of 90ms-100ms. The provider is at 1ms. All links use the digium G.729 codec. They have reported that while call quality is normally very good, if a call is transferred from one extension to another, the transferred call starts to experience considerable audio latency. Transferring the call again also incr...

I was wondering what could be pros and cons of ztdummy vs proper timer device (i.e. X100P). I am going to set up an asterisk server in europe (to do trunking, to save bandwith) and I was wondering if it'll be OK to get it going with ztdummy. Furthermore, I have only a 1024/256kbps PPPOE DSL link and I need to squeeze 12 channels through, I was wondering which codec would be suitable

Hi, I am checking out the quality at a few vendors, and althought I know it doesn`t totally reflect call quality I am using ping as a cheap subsitute to having a real VoIP testing system The question I have is this one: given that one service gives me a 80ms ping (pretty consistantly) and another one gives me 30ms (again very consistently), is this 50ms difference enough to impact perceived call

Hi. I wrote in a reply to a mail on here recently that you can''t set mpu (minimum packet unit) on HTB as you can on CBQ. I''ve just noticed that there is a patch on devik''s site which does mpu and overhead. http://luxik.cdi.cz/~devik/qos/htb/ For dsl users mpu is, for practical purposes going to be 106 - overhead is still variable though, depending on packet size.

...ransfer_20090207_124642.log 2>&1 Side note - anyone have numbers for running rsync over ssh over a WAN? Even with several hundred megabits of bandwidth available on each side it seems most often each file copy caps out at about 700kB/s with hpnssh(lower with normal ssh), latency is about 80-90ms between the sites. There are about 45 servers so we still get good performance as an aggregate but it'd be nice to get better performance on a per-server level as well if possible. I think hpnssh is the right approach with it's auto tuning and stuff but my expectations were for higher throu...

Hi, This might not be a very popular question, but I was just wondering if anyone have ever tried to run Asterisk on a Windows computer using Microsoft Virtual Server (http://www.microsoft.com/windowsserversystem/virtualserver/default.mspx). I am told that you can run Linux on a virtual server using this software, so in theory it should be possible. I run a windows based domain, but am also

Interesting. Might be worth giving a try again to the idea of creating the file in anonymous memory and using a write to output it. Cheers, Rafael On 23 November 2016 at 02:41, Sean Silva via llvm-dev <llvm-dev at lists.llvm.org> wrote: > > > On Wed, Nov 16, 2016 at 12:44 PM, Rui Ueyama via llvm-dev > <llvm-dev at lists.llvm.org> wrote: >> >> LLD supports

I suppose this is rather an informative e-mail than a question. However if people had similar experiences or could comment what the differences are in other countries or with business analog lines, it would be interesting. It took me a week until a BT engineer was sent to my home home, since BT tech support was unable to provide information about the problem. Problem: Calling party controls how

Hi I share my bandwith (adsl 512/128) between 12 users. I set up simple qos script for incoming (IMQ) and outgoing traffic using htb in root, 4 classess and esfq qdisc at leafs. Interactive traffic goeas to class1 , http,mail etc. to class2, p2p, ftp to class3, and rest to class4. Classess divide link in proportion 20% (prio 1), 40% (prio 2), 20% (prio 3) ,20% (prio 4) This works

Hi, with the attached config my gaming ping is still +20ms, even if the line isn''t saturated..can anybody give me a hint how to get a better response time? my line: 1024/128kbit outbound: one htb qdisc for gaming (7kbps) prio 0 ceil 14kbps<- should get more traffic when needed. Htb again for irc and default. inbound: css, irc, p2p, default <- same shema as before, give css

...rows=2.5e6) # read the file; 'nrows' is a slight > > # overestimate > > dat <- dat[,1:3] # select just the first 3 columns > > head(dat, 10) # print the first 10 rows > > > > # Select using which() as the final step ~ 90ms total time on my macbook air > > system.time( > > head( > > dat[which(dat$gender2=="other"),],), > > gcFirst=TRUE) > > > > # Select skipping which() ~130ms total time > > system.time( > > head( > > dat[dat$gender2=="...

Hello, 1. I currently have a 128kbps cable link to the internet..and I''m sharing this connection with others. I''ve made the following script(for alocating bandwidth depending on the services used: browsing, squid, games like counter-strike, icmp, ssh), which unfortunately isn''t working very well as the response times I''m getting upon pinging the server

Hi there! I am working a lot with VoIP in my company, so I thought to use linux bridge functionality together with tc to emulate delay, jitter, packet loss, duplication, reordering etc. for testing purposes in our lab against our VoIP products. I just recognized, that a basic bridge just with it''s minumum configuration of 2 network interfaces creates latency of approx. 5ms on very low