search for: bursti

ALL: <<diffserv-gred_10-06>> I have attached the current script that I am using. $TC qdisc add dev $EDEV parent 2:20 gred setup DPs 3 default 2 grio $TC qdisc change dev $EDEV parent 2:20 gred DP 1 limit $lim min $minTh max $maxTh avpkt $avgPL burst $bursty bandwidth $netBand probability 0.02 prio 2 $TC qdisc change dev $EDEV parent 2:20 gred DP 2 limit $lim min $minTh max

Hello again, Is it possible to split a bandwidth equally among clients regardless of its current link speed? I have a link that can get bursty at times. At any given time the N active sessions (the ones with non-empty queues) need to be serviced simultaneously, each at a rate of 1/N''th of the link speed. My case is an internet connection that oscillates between 96 kbps and 256 kbps. I

Hi all, Newbie to the group. Just started using Opus as part of a WebRTC project and amazed by the versatility of the codec. Great stuff!!!! I have been trying to understand the performance of Opus inband FEC in the presence of bursty loss. Although I do not have exact characterization of the loss profile, we are seeing issues over WiFi. RTCP reports about 33% loss, but I am guessing a lot of it

Hi all, I have an average 2mbit multicast stream that once in a while bursts high (up to 20mbit/s) in short periods (about 200ms). Could anyone please help me with directions using tc for configuing leaky bucket shaping to this stream? I have a 5mbit/s ceiling. My system is running gentoo linux 2.6.14, and I have compiled in all QoS modules. Thanks. Oivind

I have a 24 x 1TB system being used as an NFS file server. Seagate SAS disks connected via an LSI 9211-8i SAS controller, disk layout 2 x 11 disk RAIDZ2 + 2 spares. I am using 2 x DDR Drive X1s as the ZIL. When we write anything to it, the writes are always very bursty like this: ool 488K 20.0T 0 0 0 0 xpool 488K 20.0T 0 0 0 0 xpool

Hi, I have a large journal of 400MB (the max for 4k blocks), and I'd like to be able to check how much of it is being used. If it's not being used that much, I'd like to know if the writes *from* the journal to the rest of the disk could be delayed until a certain percentage of the journal has been used. It would be nice to be able to get more contiguous writes to the journal for

Hi everyone, I''m doing some networking experiments over high BDP topologies. Right now the configuration is quite simple -- two Xen boxes connected via a dummynet router. The dummynet router is set to limit bandwidth to 500Mbps and simulate an RTT of 80ms. I''m using the following sysctl values: net.ipv4.tcp_rmem = 4096 87380 4194304 net.ipv4.tcp_wmem = 4096

Hey list I have almost gotten my shaping setup up and running as planned. The last barrier seems to be tcp overshooting availible bandwidth when its starting a transfer, and thereby bursting the line, so ping rises for a moment. At least this is my best guess at the problem :) There is a possibility that its just plain old traffic being bursty for some reason.. I am using bittorrent to test this, as it seems to be what stresses the line the most. Would it be possible to lower the def...

Just showing someone how great DTrace is and then we spot something I do not understand. Of course it can be a major misunderstanding of myself. Would appreciate another (expert) look upon this. The goal we try to achieve is trying to get insight if there is a bursty nature in the time slot when system calls are done. Following DTrace snippet is tried: dtrace -q -p 3173 -n

...nd I've decided to share it. Testing and contributions are welcome! VoIPmonitor is open source live network packet sniffer which analyze SIP and RTP protocol. It can run as daemon or analyzes already captured pcap files. For each detected VoIP call voipmonitor calculates statistics about loss, burstiness, latency and predicts MOS (Meaning Opinion Score) according to ITU-T G.107 E-model. These statistics are saved to MySQL database and each call is saved as pcap dump. Web PHP application (it is not part of open source sniffer) filters data from database and graphs latency and loss distribution....

2.0.6pre3 compiles and installs under FreeBSD 3.2-RELEASE; however, throughput from an NT 4 SP6 workstation with domain security and only TCP_NODELAY as a socket option is clocking around 200K/sec for large files (65MB). Hardware is PPro200, single CPU kernel (never have gotten 2.0.5 or later to work SMP), Adaptec 2944UW/3 Quantum 9GB Atlas II wide diff drives (clock 9MB/sec using bonnie), Intel

Hello, My question is : is there a way to have Icecast buffer the input / increase the input buffering ? Even though the latency would increase ? Why : On some radio streams, the listeners to a Icecast stream experience very frequent silences / disconnections. I strongly suspect that the source has a slow/bursty connection to Icecast so that the data sent to the Icecast server arrives in

Hello, Besides a very good functionality, I have a small, yet stressing, problem. Let''s say I have a bandwidth of N bits. I have X clients, everyone with his fixed bandwidth. I create htb qdsics for each client. Each leaf has sfq queueing discipline. A sample config would be like this : tc class add dev eth0 parent 1:2 classid 1:346 htb rate 32Kbit ceil 32Kbit burst 4k

Using Samba v2.0.10 on a RedHat v7.1 box, I see that the Win98 clients on our network are making sporatic requests for user "nobody" (as shown below). Anyone know what's going on here? My network has both Linux and Win98 clients, but only the Win98 clients are showing this behavior. Both the Linux and Win98 clients access the same shared drives, but only the Win98 clients use

I''m streaming audio over a WAN that is 1.5Mbit/s. The audio has to go both ways and can be over tcp or udp. The manufacturers recommend udp as it has less overhead but has a higher chance of dropping out. The audio requires 130-200kbit/s in each direction for a combined 260-400kbit/s. I have two transparent bridges on either side of the WAN to do traffic control. I split it so

Hi all, I''m having a hell of a time getting my IFB to work. I know I''ve done this before, so I''m missing something stupid. Can anybody tell me what it might be? Configs as follows: -------- #!/bin/sh modprobe ifb numifbs=1000 modprobe act_mirred modprobe 8021q brctl addbr br0 brctl setfd br0 0 brctl stp br0 off brctl addif br0 eth1 brctl addif br0 eth2 ifconfig eth1

That "program" will be detected by your ISP within a day or so, determined to be a virus, and your service will get disconnected...which n turn will not help your latency or jitter at all. VoIP can tolerate a fair amount of latency; latency over about 100ms is heard as a perceptible delay resulting in a connection that appears to be half duplex. Jitter, on the other had, is the real

(By the way, upgrading to .11 seemed to take care of the skipping. Thanks!) The stream seems to be working just fine, although Winamp is showing bandwidth at 130 or higher, occasionally up to 145 or so. Most of my MP3s are 128 kbps, and ices should be re-encoding everything as 128 anyway. This is really a minor issue... but I'm wondering what could be taking up the extra bandwidth? Or

> behavior you are seeing....(that also is a difference between iceast1 and > icecast2 - no server-side buffering in icecast2 and there was in > icecast1)... What do you mean stream buffering? We never buffered in either version that I remember. If you fall behind, you get kicked. It's always been that way. jack. --- >8 ---- List archives: http://www.xiph.org/archives/

Has anyone here implemented Jon Kleinberg's burst detection algorithm ("Bursty and Hierarchical Structure in Streams" http://www.cs.cornell.edu/home/kleinber/bhs.pdf)? I'd rather not reimplement if there's already running code available.... Thanks, -s [[alternative HTML version deleted]]