search for: 70ms

...Having issues getting a replace command working correctly. The error reported is "RTNETLINK answers: Invalid argument" which isn''t descriptive or helpful. The command i''m running is: tc qdisc replace dev ppp0 parent 8001:D handle D: tbf rate 5Kbit burst 5kb latency 70ms The idea being to replace an sfq with handle D and hopefully limit a certain user in my WRR class. If I delete the qdisc first and add with the below line it works. tc qdisc del dev ppp0 parent 8001:D handle D: tc qdisc add dev ppp0 parent 8001:D handle D: tbf rate 5Kbit burst 1540 latency 70m...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I understand it, Asterisk currently uses the timestamps in incoming RTP packets to build outgoing voice frames. Is this true? Would it be possible for me to use i.e. zaprtc as a timing source for the outgoing stream? I.e. in a setup like below I'd like to use zaprtc timing on Ast1 because I don't trust the timestamps coming from

...locations): # create three bands, and place all default traffic in the first one: tc qdisc add dev eth0 root handle 1: \ prio bands 3 priomap 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 # the second and third bands of the prio above will delay packets # by 50ms for one simulated remote location, and by 70ms for the other: tc qdisc add dev eth0 parent 1:2 handle 20: netem delay 50ms tc qdisc add dev eth0 parent 1:3 handle 30: netem delay 70ms # use filters to place ''remote'' traffic into the appropriate band: tc filter add dev eth0 protocol ip parent 1:0 \ prio 2 <...match fo...

Hey Guys, I am considering leasing a new server in Germany to run my Asterisk infrastructure and I was wondering how response time would affect the performance of the system. Right now I have a response time of around 60-70ms with my server in California. The server in Germany would have a response time of around 140ms (both ways). My DID/Termination providers are in Canada and the USA, and all my voip boxes are also in the USA. Any suggestions or recommendations? Thanks in advance!

...air 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 enemy. Jitter is the variation in packet timing, for example, packet A arrives in 80ms, packet B in 120ms, and packet C in 70ms. The jitter for this scenario would be 120ms-70ms = 50ms. Of course the jitter time is only half of the story, the number of packets that are "outliers" in the RTP stream will also have an impact. Typical jitter measurements are stated as "average jitter" which helps masks the p...

Hi have set the following tbf tc qdisc add dev eth0 root tbf rate 0.5mbit \ burst 5kb latency 70ms peakrate 1mbit \ minburst 1540 I want to add a filter so the ip traffic pass from it.. plz help me __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

...y 10ms? 2) How much tokens are used for one packet (1500)? 3) And token unrelated question. Does packet droping alway ocurrs or it happens because of bad settings? I have applied this settings on 10mbit ethernet card: tc qdisc add dev eth1 root tbf rate 0.5mbit burst 5kb latency 70ms / peakrate 1mbit minburst 1540 Result: qdisc tbf 8007: dev eth1 rate 512Kbit burst 5Kb/8 mpu 0b peakrate 1Mbit mtu 1539b/8 mpu 0b lat 68.4ms Sent 1284481 bytes 852 pkts (dropped 48, overlimits 1743) Thanks! _______________________________________________ LARTC mailing list /...

...o do with being trying rails on Windows maybe? Could it be RubyMine? (I guess I should run WEBrick outside of Rubymine to see if it affects anything later). WEBbrick seems to add another 10ms overhead on top of all of this in most of my pages as well. It does seem weird for thinks to be taking 60-70ms per page. That does seem a little slow for pages so simple. Comments? -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFF+G/Ez6ZCGd0@public.gmane.org To uns...

...y choppy (random). A call from the same GS to an internal GS here (same latency, same IP path) is much better (some chop). SOLUTIONS tried to date: Tried a local network test, sound excellent. Tried a 'low latency vpn' test bed to reduce latency (is latency the issue?). Latency down to ~70ms round trip, better but still noticeably choppy. Tried different codecs (ulaw, alaw, iLBC), no impact. Tried various echo cancel/train on/off values, cancel and train on seem best. Tried modifing Samples/TX packet changes, 2 and 64 tried as 'ends of spectrum' values. little noticable i...

...it help against echos heard on the lines or just the 'clicks' - are they jitter? Usually my cross-IAX latency is under about 35ms (commonly only 5-6ms) and SIP latency (I know this doesn't really matter as there's no buffer for SIP) as recorded by 'sip show peers' as about 70ms (but in reality ping I think is about half this). My current setup is : SIP and IAX clients --> my * --> providers via IAX (one 5ms away, one 80ms away) jitterbuffer=yes dropcount=5 maxjitterbuffer=100 maxexcessbuffer=45 If anyone can post any amplification on these settings apart fro...

...resulting in a > > connection that appears to > > be half duplex. > > > > Jitter, on the other had, is the real enemy. Jitter > > is the variation in > > packet timing, for example, packet A arrives in > > 80ms, packet B in 120ms, > > and packet C in 70ms. The jitter for this scenario > > would be 120ms-70ms = > > 50ms. Of course the jitter time is only half of the > > story, the number of > > packets that are "outliers" in the RTP stream will > > also have an impact. > > Typical jitter measurements are...

Hi, is it just me, or are the VoIP providers for Germany more expensive than going via call-by-call? sipgate.de lists a price of 1.76ct a minute, a couple of call-by-calls are listed at 1.3ct-1.5ct a minute. Did anyone else thought this to be strange? What about other countries? Same thing?! It seems to me, VoIP providers only make sense for headquarter to subsidiary, but not for regular

...:17.097839 145.99.132.3.1030 > 145.99.70.68.139: P 1:73(72) ack 1 win 64000 (DF) [tos 0xd] ... etc. ... and there is the browselist. Does anyone got a clue? Is this due to the ping times? (This behaviour first showed up after a software upgrade in our ADSL modems, which introduced latencies of 70ms and more. Some weeks ago these were cut down to 35ms and browsing seems to work slightly better). Valentijn Valentijn Sessink - valentyn@dds.nl - Email (o.; g.mv.) glasachtige, ondoorzichtige of transparante massa, in een dunne laag aangebracht ter bedekking of versiering (brandverf) van metalen,...

Hi, I tried hunting for the algorithmic delay of vorbis, but searching Google for "vorbis algorithmic.delay" just gives a lot of hits about the delay in speex. :-\ I'm wonder what the delay would be, using vorbis for high quality conferencing VoIP. Regards, Steve

...ons, :fields].zip [ Thread.start {Section.all.to_a}, Thread.start {Field.all.to_a}, ].map &:value] end But in that case, I don''t even know how this should be reported ideally. For instance, suppose Section.all.to_a takes 50ms from DB while Field.all.to_a takes 70ms. How would you explain an output like this? Action completed in 80ms (view: 5ms, db: 120ms) So, except for such cases, using thread locals would yield to more reliable results in the production logs for a quick inspect. Is the overhead of using Thread.current so high when compared to "se...

...ess any other key on the phone besides #. But in the Record test, it all works fine. I used dahdi_monitor to try to debug this, and found that during dialing the extension, each DTMF tone lasts about 100ms. But whatever DTMF is sent during the Echo test (# or anything else) gets cut off at about 70ms, and that tone is then transmitted continuously on the channel until I hang up, even though I don't hear it on the phone. Here's what I see in the log when this happens: Set channel DAHDI/1-1 to write format ulaw Launching 'Echo' Begin DTMF digit: 0x23 '#' on D...

...assid 1:20 htb rate 0.5mbit #tc qdisc add dev eth0 parent 1:10 handle 10: sfq perturb 10 #tc qdisc add dev eth0 parent 1:20 handle 20: sfq perturb 10 #tc filter add dev eth0 parent 1: protocol ip u32 match ip dst a.b.c.d/24 flowid 1:10 #tc qdisc add dev eth0 root tbf rate 0.5mbit burst 5kb latency 70ms peakrate 1mbit minburst 1540 (yes that does 0.5mbit limiting, but it would be a start...) #tc qdisc add dev eth0 root handle 1: htb default 20 #tc class add dev eth0 parent 1: classid 1:1 htb rate 1920kbit #tc class add dev eth0 parent 1:1 classid 1:10 htb rate 128kbit prio 1 #tc class add dev eth...

[Just curious, and seizing the opportunity to communicate with other folks who are doing the same kind of thing I am...] How are you measuring the latency? I tried measuring it with my program (also Win32-based, also using DirectSound[Capture]) and came up with around 130ms. To measure it, I placed the mic near a speaker to get feedback going, had my program connect to itself (local

Hi. I wrote in a reply to a mail on here recently that you can''t set mpu (minimum packet unit) on HTB as you can on CBQ. I''ve just noticed that there is a patch on devik''s site which does mpu and overhead. http://luxik.cdi.cz/~devik/qos/htb/ For dsl users mpu is, for practical purposes going to be 106 - overhead is still variable though, depending on packet size.

In order to get more information on IO performance problems I created the script below: #!/usr/sbin/dtrace -s #pragma D option flowindent syscall::*write*:entry /pid == $1 && guard++ == 0/ { self -> ts = timestamp; self->traceme = 1; printf("fd: %d", arg0); } fbt::: /self->traceme/ { /* elapsd =timestamp - self -> ts; printf("