search for: 120kbps

...to trouble again..... After go through www.lartc.org I have implemented the HTB instead of CBQ for the same scenario. Now following files are under /etc/sysconfig/htb directory. eth0 DEFAULT=30 R2Q=10 eth0-2.root RATE=256kbps BURST=25k eth0-2:10.comp1 RATE=120kbps BURST=12k PRIO=0 LEAF=sfq RULE=192.168.200.0/24 eth0-2:20.comp2 RATE=80kbps BURST=8k PRIO=1 LEAF=sfq RULE=192.168.100.0/24 eth0-2:30.server RATE=56kbps BURST=6k PRIO=3 LEAF=sfq RULE=203.145.134.120/29 -------------------- eth1-2:30.root RATE=56kbps BURST=6k eth1-2:30:3...

...ll analyse WAV files and determine the "Nominal Deviation Factor" or something like that to figure out what quality setting I need to use to fit them in a specific amount of space. Is this even consistant? By that I mean if a certain song encoded at q4 (128k nom.) gets an average of 120kbps does that mean the same song encoded at q8 (256k nom.) will get an average of 250kbps? Or if it's not linear is there a way to predict the q setting to use. So for files that take up 775 Megs @ q10 to fit in 700 Megs I need to use what? q9? q9.5? q9.7? q9.8? There has...

I have a problem with IAX call transfer. The call goes successful but consumes lot of BW in the middle tier. The actual connection is like this (NAT) DIAX(IAX2) -----> *1 ------> *2 *1 & *2 were public IP with asterisk. It consumes around 120kbps in total to forward a single GSM call. I have the following configuration in my iax.conf [general] ... disallow=all allow=gsm [provider] type=peer username=userid secret=password host=myprovider.com i can successfully make a call to my provider through the following settings in extensions.conf...

Hi, I have two issues with the --bwlimit parameter to rsync which are not mentioned in the FAQ (please correct me if I am wrong): 1. When both an rsync client and an rsync daemon specify the --bwlimit parameter, the client's value overrides the server's value. This could be used in a DOS attack if the client uses a larger value than the server intended. The documentation also is

Hi! So here is what I got with different encoder settings; still not sure what the "best" setting is. 6kbps seems to add distortions, so I tried 12kbps. MP3-Original: LAME 3.99r, 120kbps, 44100Hz, Stereo, VBR V5 (22:23, 19.8MB) Opus (--raw-rate 44100 --bitrate 56 --vbr --comp 5): (44:45?, 23.4MB): Broken Opus (--bitrate 56 --vbr --comp 5 --ignorelength - %d): (22:23, 12.1 MB, 74kbps) Opus (--bitrate 6 --vbr --comp 10 --ignorelength - %d): (1MB, 6kbps) Opus (--speech --set-ctl-int 4...

Hi! I have some MP3 audio material which is basically speech with some background noises, essentially > 120Hz and < 5kHz. I had the idea to reduce the file size by recoding the material to Opus at 56kbps. Unfortunately the result is a file sampled at 48kHz much larger than the original. I hope you agree that it does not make sense to create a file larger than the original (MP3). Of course

...l tc qdisc del dev $DEV ingress 2> /dev/null > /dev/null tc qdisc del dev $DEV root 2> /dev/null > /dev/null iptables -F #también las relgas iptables #creacion del arbol de bandas tc qdisc add dev $DEV root handle 2: htb default 60 tc class add dev $DEV parent 2: classid 2:1 htb rate 120kbps ceil ${RATEUP}kbps tc class add dev $DEV parent 2:5 classid 2:50 htb rate $[70*$RATEUP/100]kbps ceil ${RATEUP}kbps tc class add dev $DEV parent 2:6 classid 2:60 htb rate $[20*$RATEUP/100]kbps ceil ${RATEUP}kbps prio 1 tc class add dev $DEV parent 2:7 classid 2:70 htb rate $[10*$RATEUP/100]kbps ceil...

Hi! I have read all informations i could find, but some things are still not clear. My setup is: ---INTERNET1(eth0)-\ /- Local net1 (eth2) GW ---INTERNET2(eth1)-/ \- Local net2 (eth3) I have NAT and a working setup using HTB,SFQ, classifying with the iptables -j CLASSIFY way. I shape only the traffic coming from the internet heading to the intranet. I would like