search for: 40kbps

...add dev eth0 root handle 1: htb default 20 #add main rate limit class tc class add dev eth0 parent 1: classid 1:1 htb rate 100mbit #add leaf classes, 1:2 is lan, 1:3 is outbound max tc class add dev eth0 parent 1:1 classid 1:2 htb rate 100mbit tc class add dev eth0 parent 1:1 classid 1:3 htb rate 40kbps # 1:31 is ftp with lower prio, 1:32 is ACk AND email higher prio tc class add dev eth0 parent 1:3 classid 1:31 htb rate 20kbps ceil 40kbps prio 2 tc class add dev eth0 parent 1:3 classid 1:32 htb rate 20kbps ceil 40kbps prio 1 #filter traffic into classes tc filter add dev eth0 parent 1:0 prio...

...in wget on the stream and trying to play it with mpg123 results in silence. The problem mp3s were encoded with lame --alt-preset cbr 40 and lame --alt-preset cbr 80. You can access some example mp3s here: http://www.svaudio.org/test1.mp3 (works) http://www.svaudio.org/test2.mp3 (doesn't work-40kbps) http://www.svaudio.org/test3.mp3 (doesn't work-80kbps) My DNS is flaky sometimes, so use 68.53.92.187 if you have problems. All of the problem mp3s work fine when played independently, and mp3_check finds no problems. I realize that I could just reencode these, but I want to know what setti...

Dear all, I know this is not imq mailing list. But many of the users over here have done exactly what i want. Requirement:- I want to tight bound eth1 for 100 kbps but after i want to create many classes of 64 kbps or 50 kbps and others. But the total sum of classes is more than 100 kbps so my eth1 is not restrciting total bandwidth at 100kbps. According to search on google imq is the solution.

...ow modules like 72 Module Description Use Count codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g729.so Raw G729 data 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_g723.so G.723.1 Simple Timestamp File Format 0 The codec_g729a.so doesn't appear.......... Any idea how to solve the problem..... Thanks Best Regards, Marc LEURENT -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using Gnu...

.... Generate a bunch of terminal output(500k) 2. Sleep 15 seconds 3. Go back to step 1 After repeating steps 1-3 for some random amount of time(sometimes 3 minutes, sometimes 50+), the SSH server will go from streaming the output back to the client @ 4-4.5 mbps(normal-behavior.png), down to 30-40kbps(bad-behavior.png). Most of the time, SSH stays in this 30-40kbps state for as long as their is data in the TCP queue. ie. during peaks, netstat will show the queue having 90-100k of data waiting to be transmitted. We think that Nagle may be taking effect randomly for some reason. When I '...

hi, there is a low bitratestream available for testing. it's around 40kbps and it sounds really bad, because it's 11khz stereo :). http://ogg.radiostudio.org:8007/ostream.ogg -- Sent through GMX FreeMail - http://www.gmx.net --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this lis...

hi, there is a low bitratestream available for testing. it's around 40kbps and it sounds really bad, because it's 11khz stereo :). http://ogg.radiostudio.org:8007/ostream.ogg -- Sent through GMX FreeMail - http://www.gmx.net --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this lis...

...run in icy mode. > > icy mode doesn't have mountpoints, so your mountpoint would be ignored. > > jack. > Thank for prompt answer. It help me. But I have another problem. I tried multistreaming with two streams where is one mp3 stream reencoded at lower bitrate (to 24 kbps from 40kbps). When I listen to original bitrate there is no problem, but when I listen to reencoded stream, there are every ~ two second drop outs. When I feed icecast with only one reencoded stream, there is no problem. This is on localnet (10MB) and cpu usage is on 30% so this shouldn't be a problem. I&...

...r is > it two fxs ports with the same extension? Hi, I have some time using Sipuras 2000 as base sistem and have to say they have very good performance on Asterisk, from Inside the network and also as a remote client, even thou its true they lack ilbc and gsm you can use G.711, G.726-16/24/32/40kbps G.729A (just one line) and G.723.1 6.3Kbps or 5.3Kbps, But I dont know if ther is any plugin G.723.1 plugin for Asterisk...

...iber connection --------- (eht2) RT1 (eth1) --------- wireless connection ------------ (eth4) RT2 (eth0) -------- users RT1 - Slackware router RT2 - Slackware router Delay between Test PC and ISP is approx 450 ms. I have the following shapers RT1 ETH2 - dedicated bandwidth for Test PC 40kbps RT2 ETH4 - dedicated bandwidth for Test PC 30kbps RT2 ETH4 - dedicated bandwidth for users to upload is 250kbps The thing is if I use Test PC to download from RT1 it makes straight line of 40kbps. If I use TestPC to download from RT2 the traffic varies from 3kbps to 30kbps. It is very rand...

...bitrate up to 56, which resulted in a noticeable audio increase > in quality but the buffering was unacceptable. If anyone has settings that > work i would be interested in hearing about them. You're not going to be able to send 56kbps over a modem, as has been stated already. IMHO, 40kbps would probably be your absolute top for a 56k modem. If you want to be accessible to 33.6/28.8k modems, don't go any higher than 24kbps. At this rate, you hit the stereo vs mono argument. If you want clearer audio, go mono. But if you want stereo, you'll have poorer audio. In my expe...

...scribing to the icecast forum (http://icecast.imux.net/) I also tried it a few months ago, this mail address is also the mail address for the forum subscription. --------------------------------------- I also have an problem with android on icecast, With low buffering streams for example with 40kbps, i have to wait up to 10 seconds before the player plays music. I haired that the problem might be that the buffering time of the new android 4.x versions has been multiplied a view times causing this problem. Do you know and alternative (webplayer) to use for android devices where i can set m...

hi how does one control traffic of ftp using tc.I tried few things ,dont know how to do it.Could you please guide me in it. Thanks in advance Nampreet _________________________________________________________________ All that you wanted to know about Ms Beautiful Lips http://server1.msn.co.in/Profile/katrina.asp

...as of this morning 3/30/04 of asterisk, zap and libpri. The SIP device I am using is a Sipura SPA-2000 with G726-32 "Forced". When I 'make clean" and recompiled zaptel, libpri, asterisk and start asterisk I can see: [format_g726.so] [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data) == Registered file format g726-40, extension(s) g726-40 == Registered file format g726-32, extension(s) g726-32 == Registered file format g726-24, extension(s) g726-24 == Registered file format g726-16, extension(s) g726-16 I 'Ass'ume this indicates that g726 is installed.....

...ional calls are made at once - I'm on a 1024kbps download and 256kbps upload DSL line (only the asterisk server uses this DSL line). Today I switched the codec from ulaw to ilbc in an attempt to lower the bandwidth requirements. I have read many times that IAX2 has a signaling overhead (30 or 40kbps?) and then every channel uses the codec bandwidth. Does Asterisk automatically know to use one overhead bandwidth management for IAX2 remote peer gateways? I'm afraid of the possibility that I have not configured the VOIPJET IAX2 link correcty and that I'm making lots of single connection...

...he fly on my P2-266 at 44100 fast enough to be able to stream it. Ostream looked the more promising of the two, with breaks in the audio but otherwise coping OK ... until it crashed, presumably because it filled some buffer somewhere. It can maintain a 44.1khz mono stream at lowest bitrate (aprox. 40kbps) but only just. I only tried oddcast once, at 64kbps stereo. It sounded very weird, as if it was only recording half the data in a given time period. It sounded as if it was playing at about double speed but wasn't pitch shifted. What I want to know is this. Am I expecting too much of my P2...

Hi! It''s me again bothering you guys, what I want to do is to give full bandwidth to VPN traffic and limit the rest to 30KB/s (kilobytespersecond), ok? Here''s what I have: tcclasses ################################## eth0 1 1kbps 70kbps 1 eth0 2 1kbps 30kbps 2 default eth1 3 15kbps 10000kbps 1 eth1 4

...Shang <gshang@uq.net.au> wrote: > Hi: > > My guess is that ices can't handle 16000 or 32000hz > sampling rates. The > 32kbps file is 22050hz and the 80kbps file is > 32000hz. I couldn't get > mpg123 to play test2 remotely, but lame defaults to > 32000hz with 40kbps > mono and 16000hz with 40kbps stereo, so I'm guessing > this is the problem. > You can either use --resample to force a different > sampling rate or pick a > different bitrate. > > Geoff. > > > -- > Geoff Shang <gshang@uq.net.au> > ICQ number 43634...

...add dev eth0 root handle 1: htb default 20 #add main rate limit class tc class add dev eth0 parent 1: classid 1:1 htb rate 100mbit #add leaf classes, 1:2 is lan, 1:3 is outbound max tc class add dev eth0 parent 1:1 classid 1:2 htb rate 100mbit tc class add dev eth0 parent 1:1 classid 1:3 htb rate 40kbps # 1:31 is ftp with lower prio, 1:32 is ACk AND email higher prio tc class add dev eth0 parent 1:3 classid 1:31 htb rate 1kbps ceil 40kbps prio 2 tc class add dev eth0 parent 1:3 classid 1:32 htb rate 20kbps ceil 40kbps prio 1 #filter traffic into classes tc filter add dev eth0 parent 1:0 prio 0...

Hello, I've got two streams, one for broadband, one for dialup. Well, having had occation to use a dialup connection recently i checked the dialup stream. Although it was streaming what the broadband stream was, the audio quality was audibly worse. It didn't buffer, but it didn't sound as clear as the broadband stream. I used lame to encode the tracks to mp3 and used it's