similar to: TC not rate limiting for me

hi all I just tried to test whether the HTB works well or not .. (policy is very simple ^^;;) I tried HTB simulator ''Ethloop'' with lo (loopback ) and I found HTB works well....(very well) So I changed lo to eth0 and tested .. ./tc qdisc add dev eth0 root handle 1: htb default 10 ./tc class add dev eth0 parent 1: classid 1:1 htb rate 500kbps ceil 500kbps ./tc class add

hi all I just tried to test whether the HTB works well or not .. (policy is very simple ^^;;) I tried HTB simulator ''Ethloop'' with lo (loopback ) and I found HTB works well....(very well) So I changed lo to eth0 and tested .. ./tc qdisc add dev eth0 root handle 1: htb default 10 ./tc class add dev eth0 parent 1: classid 1:1 htb rate 500kbps ceil 500kbps ./tc class add

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

I''m trying to shape traffic on a Devil-Linux box. This note was originally sent to their maillist, because the LARTC list appears to have been down for the past few days. My mailbox was just flooded with a half dozen or so confirmation requests in response to my repeated attempts to subscribe to this list. ---------- Forwarded message ---------- From: drew einhorn

I use the Opus multistream encoder for both mono and stereo encodings and after updating from 1.1.3 to 1.3 I noticed the size of the produced Opus files had doubled for 1-channel encodings whereas switching to the standand Opus encoder gave me roughly the same sizes as before. In these tests, I'm encoding an 8kHz mono stream containing only speech with the following options set: frame length

Hello, some days ago, I was asking for help here about not able to do anything when I had bittorrent running, I will post the problem here: I''m using ipp2p to mark p2p packets, and then send them with -j CLASSIFY to the correct HTB class, I see traffic in the class when I start azurerus, and traffic does get shaped, but then I''m still not able to surf the web nor chat nor

Reading along the Net it seems that MAC marking is not working with egress HTB (because ipables marks packages based on --mac-source ). So my only choice is using ingress or u32. So this is how I did it: I called bellow script add_shaping DEV="eth0" tc qdisc add dev $DEV root handle 1: htb default 20 tc class add dev $DEV parent 1: classid 1:1 htb rate 200kbps ceil 200kbps tc class

With the below script, whenever I ping 10.0.16.10 (which matches the only filter I have), traffic still get''s sent to the default 1:2 class instead of 1:1 and I don''t know why... Any hints? (kernel 2.6.12, iproute2-2.6.15) tc qdisc del dev eth0 root > /dev/null 2>&1 tc qdisc add dev eth0 handle 1: root htb default 2 tc class add dev eth0 classid 1:1 parent 1: htb rate

On 31.05.2013 07:05, vikrant verma wrote: > Hello Michal, > > Throttling is working now after setting inbound.average = 10 (i have > set other values to 0 ) > But i am observing an inconsistent behavior in the observed throttling > bandwidth > > when i set inbound.average = 10 , i observed bandwidth of 24 kbps > when i set inbound.average = 100 , i observed

I''m getting a slow network speed (download/upload) domU and dom0 limits in 10kbps/20kbps. Booting a kernel without Xen, this doesnt happens /etc/xen/xend-config.sxp (network-script network-bridge) (vif-script vif-bridge) (dom0-min-mem 196) (dom0-cpus 0) cat /etc/xen/XXX.cfg kernel = ''/boot/vmlinuz-2.6.26-1-xen-amd64'' ramdisk =

what is the script for deleting filter that already exist??? the htb script is: tc qdisc add dev eth0 root handle 1:0 htb default 4 tc class add dev eth0 parent 1:0 classid 1:1 htb rate 500kbps ceil 500kbps tc class add dev eth0 parent 1:1 classid 1:2 htb rate 300kbps ceil 300kbps tc qdisc add dev eth0 parent 1:2 handle 12:0 pfifo tc class add dev eth0 parent 1:1 classid 1:3 htb rate 150kbps

Hi, I''m using tc and HTB to shape my outgoing ADSL traffic. I was trying to make some graphs on the classes by meassuring the "sent bytes" of each class using rrdtool to store the data (as kbps after conversion). I expected that meassuring the root class I would get values similar that the ones I get measuring the interface counters but they differ by a large amount. Is

I've been using tc/htb for network control previously to control bandwidth available to different services running on their own IPs on a unvirtualized host. Now, I have put them into their own guest VM. I would like to be able to do something similar to ensure the more crucial service gets more bandwidth as well as ensuring ssh always get reserved bandwidth. However, when I try the good old

Running "tc -s class show dev eth1" against an HTB qdisc results in the output of class summaries similar to this: class htb 1:10 parent 1:1 leaf 10: prio 0 rate 150000bit ceil 1466Kbit burst 2Kb cburst 2Kb Sent 158641651 bytes 771351 pkt (dropped 0, overlimits 0 requeues 0) rate 8064bit 7pps backlog 0b 0p requeues 0 lended: 680985 borrowed: 90366 giants: 0 tokens: 173443 ctokens:

Hello, I have been trying to playback a video file via Playback() in Asterisk 1.8.7.1 but the file format seems to fail. [2011-12-02 18:46:24] WARNING[7665]: file.c:653 ast_openstream_full: File /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 does not exist in any format [2011-12-02 18:46:24] WARNING[7665]: file.c:959 ast_streamfile: Unable to open /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 (format 0x4

Hi guys, I'm using Opus for speech in wide-band mode (sampling rate 16000) and 20ms frames with signal type set to SIGNAL_VOICE. I have a few questions here: 1. When I choose VBR mode, the codec seems to choose the bitrate on its own. However, that seems to be an issue on mobile devices. In some cases, when I configure the bitrate to say 20kbps, I see that the outgoing codec bitrate at

The Opus RFC seems to recommend a frame size of 20ms for most applications. For wideband speech, the sweet spot range is recommended to be 16-20kbps. 20ms frames => 50 frames per second. For a VoIP application, the header overhead per frame (IP+UDP+RTP+SRTP) is 44bytes => 17.6kbps at 50 frames per second. So a 20ms frame size seems to cause a roughly 100% overhead of header data. Therefore,

Hi, I am currently using the following script to shape traffic on my linux box. I am getting ready to move the script to my linux firewall box so it shapes traffic for my home lan. I am looking to refine my rules a bit so to make bandwidth sharing a bit more efficient. Specifically I am looking to give out bound email more priority so that when a large email is sent, I want it to borrow all

In this simple htb setup: # tc -s -d class ls dev eth0 class htb 1:1 root rate 300000bit ceil 300000bit burst 1749b/8 mpu 0b overhead 0b cburst 1749b/8 mpu 0b overhead 0b level 7 Sent 13171835 bytes 13169 pkt (dropped 0, overlimits 0 requeues 0) rate 45848bit 10pps backlog 0b 0p requeues 0 lended: 5272 borrowed: 0 giants: 0 tokens: -84429 ctokens: -84429 class htb 1:2 parent 1:1 prio 0

On 01/21/2015 07:51 PM, Daniel K wrote: > Hi guys, > > I'm using Opus for speech in wide-band mode (sampling rate 16000) and > 20ms frames with signal type set to SIGNAL_VOICE. > > I have a few questions here: > > 1. > When I choose VBR mode, the codec seems to choose the bitrate on its own. > However, that seems to be an issue on mobile devices. In some cases,