search for: 200ms

I''m trying to use tc and netem to delay packets from several different machines as they exit via eth0. Assume two source IPs, 10.0.0.122 and 10.0.0.133. I''d like to delay packets from the first one by 200ms, and packets from the second one by 300 ms. Any other traffic should be sent out normally. Here''s what I tried: # make three classes, 1:1, 1:2, and 1:3: tc qdisc add dev eth0 root handle 1: prio # attach netem with 200ms delay to priority 2 hook: tc qdisc add dev eth0 parent 1:2 handle 2...

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

...ss wants to run. Our A*k is 1.2.18, and his is 1.4.x I'm not sure if either of these messages will actually cause issues, how I can debug them further? Any ideas? Adrian Jun 19 09:24:45 NOTICE[6632]: chan_sip.c:10029 handle_response_peerpoke: Peer '227' is now TOO LAGGED! (204ms / 200ms) Jun 19 09:24:55 NOTICE[6632]: chan_sip.c:10023 handle_response_peerpoke: Peer '227' is now REACHABLE! (148ms / 200ms) Jun 19 09:24:59 NOTICE[6632]: chan_sip.c:11777 sip_poke_noanswer: Peer '464' is now UNREACHABLE! Last qualify: 101 Jun 19 09:24:59 NOTICE[6632]: chan_sip.c:10029 h...

Anyone have a practical experience/knowledge relative to why a 5ESS central office switch would require a "w" in the Dial statement to handle analog pstn-fxo calls? I fully understand what "w" is doing, just trying to better understand why a 5ESS doesn't accept dtmf a little quicker then it does. Does that switch make use of dtmf receiver cards (or something) that

I'm using 1.8.3, and have 2 sip providers. Both are set with qualify=yes. Each of them sometimes have qualify times 10+ times the other. For instance, one will be at 10-15ms, the other at 200ms. Is there a way I can route an outgoing call to the provider with the lower qualify time? sean

...SIP.conf configured voicemail.conf configured. extensions.conf configured. sip.conf [user] callerid="User Name" type=friend nat=no username=user secret=password host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away defaultip=10.1.1.53 dtmfmode=rfc2833 mailbox=1115 voicemail.conf 1115 => 1234,User Name,user@somecomapany.com extension.conf exten => 1115,Macro(stdexten,1115,${user}) exten => 1115,1,Dial,SIP/user/s|15 exten => 1115,2,Voicemail,u1115 exten => 1115,102,Voicemail,b1115 exten...

...works fine for first few seconds. Bur after a certain period [usually 20 sec or more] it does not work or works very bad [produces distorted or noisy echo]. -- If no one talks for some time [around 4/6 mins] EC works again for some period and after that period it fails again. -- I used echo tail 200ms. Using long echo tail [2000ms], EC works very fine but in that case CPU usage becomes very high. What should be the solution in this situation? I mean what can be the reason that EC fails after some time. 2: ==================== -- I read the played packets from DirectSound Buffer and use it...

...quot;pool" parameter in database.yml I''m using rails 3, ruby 2, postgres, elasticsearch, unicorn, nginx My server : 12 coeurs 24 threads ( I dont know if it is necessary for define pool value) unicorn : worker_processes 24 I have around 1500 pages requested per minutes, and around 200ms per request Is it also useful/necessary to use this ? allow_concurrency: true thanks -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To unsubscribe from this group and stop receiving emails from it, send an email to rubyonr...

...articipant with the same codec). For the two-(or more) speaker case, each speaker gets the other speaker's frame (transcoded if needed), and we mix and recode the summation of each speaker for all others. In the application we're using, there can be a _lot_ of jitter (not just the 200ms worth that your jitterbuffer seems to account for, but 1 second or more), and if we don't dejitter first, we can easily end up with cases where: a) We send out subsequent frames for different speakers with overlapping timestamps. b) Different speakers have different clock skews, and over ti...

I''ve been thinking a lot about running mcollective on my production servers recently, especially about its nice integration with puppet (using facts and so on). However I''ve got some concern about its scalability - how publish/subscribe middleware scales in terms of speed and flooding the network. I run about 2000 servers. Has anyone got any experience running mcollective on

...(kgdb) vpath vp 0xffffff02ea3d7ce8: 18381 0xffffff02eab1dce8: 16384 0xffffff02eaaf0588: base 0xffffff01616d8b10: data 0xffffff01616d8ce8: pgsql 0xffffff002af9f588: <mount point> 0xffffff002af853b0: d3 0xffffff002abd6b10: / (kgdb) During such an fsync, DTrace shows me that syncer sleeps of 50-200ms are happening up to 8 or 10 times a second. When this happens, a bunch of postgres threads become blocked in vn_write() waiting for the vnode lock to become free. It looks like the write-clustering code is limited to using (nswbuf / 2) pbufs, and FreeBSD prevents one from setting nswbuf to anything...

...work. See the manual. > > Sorry I don't get you. I can make sense of that *if* you're talking about testecho sample application, since in testecho the tail is set to 1024 samples. But, as I explained, I used my test program for producing the result, and there I set the tail length to 200ms (1600 samples). *And*, in addition to that, I delay the frames that's fed to speex_echo_cancellation() by 100ms, to reflect the playback/recording delay when the mic.pcm file was recorded. This means, if I play the frame now, I will give it to speec_echo_cancellation() 100ms later (similar to y...

...Config, and explain that it is insecure. ~ Documentation update. ~ Repair mismatch between driver and client regarding parameter names. ~ Implement "corner coasting". Matt Domsch (4): ~ Add synaptics-0.14.6-poll-delay.patch from Fedora ~ Add synaptics-0.14.6-poll-200ms.patch from Fedora ~ Add synaptics-0.14.6-poll-delay.patch from Fedora ~ Add synaptics-0.14.6-poll-200ms.patch from Fedora Matthew Garrett (1): ~ On Linux 2.6, the evdev interface provides support for requesting the Mattia Dongili (1): ~ Add 03-escape-dashes-in-manpages.patch f...

Hi All, I have a question regarding the nominal jitter buffer configuration: The call was setup as G.729A (annexb=no), ptime=20ms and nominal jitter buffer size = 50ms, and round trip delay is 200ms, the TDM side will experience intermittent one way voice during the call, but IP side can always heard the voice from TDM side. My question is, should this possible caused by the nominal jitter buffer? If there didn't have large network jitter, should the nominal jitter buffer set to 10 or larg...

...til the phone reregisters. Here is what the entry looks like for a Cisco7960: [cisco] type=friend username=cisco secret=supersecret host=dynamic defaultip=192.168.0.55 canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away context=local callerid="Cisco Phone" <2010> mailbox=2010 Of course, if I set the host= parameter to the phone's IP and set the phone not to register, everything works also, but the icons on the phone indicate that it is not registered. Is there any way around this? John

hi i wonder why my outbound calls via asterisk->sipgate->german telecom have such high delay rates (about 500 or mor ms) while inbound signals are quite ok (max ca 200ms). any idea? joerg

Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130220/70e10d34/attachment.htm>

...2345 at sip provider as 1234 here. ;allow=g729 ; ;[cisco] type=friend username=9873 secret=pwd ;nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away defaultip=192.168.0.5 mailbox=9873 ;[cisco2] type=friend username=9874 secret=pwd nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms awa...

So I was doing some experimenting with prototype and ajax and discovered something interesting. When I use the first ajax call the response from the server take 53ms. When I use the second on using the prototype functions the request from the server is over 200ms. My question is why, and can it be fixed? <p><a href="#" onclick="serverSideAlert();">Call server-side function</a></p> <p id="responseTxt"></p> <script type="text/javascript"> function serverSideAlert() { va...

...o an FFT on it. More precisely, I'd like to do an NFFT (non-equispaced / non-uniform time FFT) on the data. The data is timestamped samples from a cheap self-logging accelerometer. The data is weakly regular, with the following characteristics: - short gaps every ~20ms - large gaps every ~200ms - jitter/noise in the timestamp The gaps cover ~10% of the acquisition time. And they occur often enough that the uninterrupted portions of the data are too short to yield useful individual FFT results, even without timestamp noise. My searches have revealed no NFFT support in R, but I'm h...