search for: 30ms

Here's the story... Nortel system set to use g711 @ 30ms payload ... Asterisk box would need to communicate to that box @ 30 ms and another end point at 20 ms. I've seen discussions of setting this to a different size, but seems to be limited to the entire codec and not on a per peer basis. Anyone have luck with this? The Asterisk can be 1.4 o...

...can change time slice of the scheduler at runtime via xl command line, a very nice feature. However, it is not *correctly* implemented. Problem description -------------------- say you set the ''cap'' of one VM to 50 (a half core), -when setting the time slice to be *greater* than 30ms, the VM gets much *less* CPU cycles than its allocation -when setting the time slice to be *smaller* than 30ms, the VM gets much *more* CPU cycles than its allocation Problem happens in sched_credit.c/csched_sys_cntl(): -------------------- after changing prv->tslice_ms, other parameters *shou...

Hi all, I need to be able to adjust packet sizes and found the patch at http://bugs.digium.com/view.php?id=5162 Thus, I checked out and compiled http://svn.digium.com/view/asterisk-old/team/group/5162_rtp_packetization I added the line "packetization = 30" for one peer in my sip.conf and started asterisk with the "-I" switch for async RTP. That's all it takes

When I start my script: * - Creating classes on br1 for upload control ... * - tc class add dev br1 parent 2:0 classid 2:46 hfsc ls m1 576.0Kbit d 2000ms m2 192.0Kbit ul m2 384Kbit ... [ ok ] * - tc class add dev br1 parent 2:46 classid 2:47 hfsc sc umax 1500b dmax 30ms rate 80Kbit . [ ok ] * - tc class add dev br1 parent 2:46 classid 2:48 hfsc ls m2 152.0Kbit ul m2 152.0Kbit ... [ ok ] * - tc class add dev br1 parent 2:46 classid 2:49 hfsc rt m2 76.00Kbit ls m2 152.0Kbit ul m2 304Kbit ... [ ok ] * - Creating classes on wivl4 for download control...

...outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.116 5060 Unmonitored 101/101 68.156.63.111 D N 1038 OK (133 ms) This seems pretty high when my ping time from a host on the same network is ~30ms: Pinging 209.251.157.93 with 32 bytes of data: Reply from 209.251.157.93: bytes=32 time=30ms TTL=51 Reply from 209.251.157.93: bytes=32 time=27ms TTL=51 Reply from 209.251.157.93: bytes=32 time=36ms TTL=51 Reply from 209.251.157.93: bytes=32 time=28ms TTL=51 Any suggestions or is this normal? Sh...

Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on "sip show peer" shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify? Thanks. -- James

...re will be different categories of traffic (VoIP, P2P, General). The VoIP subclass (Class 11) should offer a real-time bandwidth of 64Kbps minimum with a maximum of 128Kbps, sharing anything over 80Kbps with other services. The VoIP subclass should also be able to dequeue the packets within a 30ms timeframe, but more importantly, it should not vary the time in which it dequeues the packets by more than 2-3ms between each packet. The P2P subclass (Class 12) should offer a mimimum of 0Kbps and a maximum of 384Kbps, sharing all of its allotted bandwidth with anything that needs it. The G...

Hello, I am very slowly working on the NHW Project.I recently made a speed comparison with x265 and wanted to share it with you. On my processor Intel Core i5-6400, in average the NHW encoder (totally unoptimized) is x10 times faster to encode than x265 (png decoding time removed), in average 30ms vs 300ms for a 512x512 24bit color image, and in average the NHW decoder (totally unoptimized) is x3 times faster to decode than x265 (output to .ppm file), in average 10ms vs 30ms. As a reminder, x265 is ultra optimized, so with good C optimization, SIMD optimization and multithreading, the NHW c...

Hi, We're using R in an application where asking for a probability of an event takes about 130ms. What could we do to take that down to 30ms-40ms? The query code uses randomforest, knn. -- M.

...2ms. When I increase the policing rate and shaping rates to 48.4687. The combined increase is 31.2 kbit which is very small. there are some packets queued for a short period and some dropped which clears the queue. The maximum number of packets dropped was 20 per second. But the delay goes up to 30ms. check out the graphs at http://frink.nuigalway.ie/~jlynch/queue/ I cant seem to explain this. Even if the queue was full all the time and each packet was of maximum size, the delay imposed by queueing should be a maximum of 10 * 1500 * 8 /100,000,000 which equals 1ms. How can so much delay...

...i there, (context: I was optimizing Roundcube mailbox list server response, and in that 300-400ms response time, around 170ms is spent on single fgets() call which is waiting IMAP repsonse to "SELECT MyMailbox" command) I straced dovecot and of the whole request/response process, around 30ms is spent for everything else, and overwhelming majority of time (150-170ms) is spent for reading dovecot-uidlist file for given mailbox. I skimmed over src/lib-storage/index/maildir/maildir-uidlist.c and src/src/lib/istream.c, but I am out of ideas on how to optimize mentioned file reading elegant...

Hello. I have the following situation. AEC is used in network chat software over DirectSound API. Echo and reference signals are almost aligned (delay is no more than 30ms). When echo is emulated in notebook (built-in speakers + mic) everything goes fine and echo is cancelled. But when configuration includes stand-alone speakers and mic no echo is removed. Audio is in 22050 hz at 16 bit mono format, number of samples to process were 441 and tail was used 4096...

Hi there! I am working a lot with VoIP in my company, so I thought to use linux bridge functionality together with tc to emulate delay, jitter, packet loss, duplication, reordering etc. for testing purposes in our lab against our VoIP products. I just recognized, that a basic bridge just with it''s minumum configuration of 2 network interfaces creates latency of approx. 5ms on very low

...I have already done what is described on http://wiki.xenproject.org/wiki/Tuning_Xen_for_Performance It is better now but still significantly worse than with centos6 dom0 my settings: xen parameter: dom0_mem=1024M cpufreq=xen dom0_max_vcpus=2 dom0_vcpus_pin xl sched-credit Cpupool Pool-0: tslice=30ms ratelimit=1000us Name ID Weight Cap Domain-0 0 1024 0 samael 1 256 0 satan 2 512 0 amon 3 256 0 leviathan...

...10/3/15 Anton A. Shpakovsky <saa at tomsksoft.com> > > > Hello. > > > > I have the following situation. AEC is used in network chat software > > > > over DirectSound API. Echo and reference signals are almost aligned > > > > (delay is no more than 30ms). When echo is emulated in notebook > > > > (built-in speakers + mic) everything goes fine and echo is cancelled. > > > > But when configuration includes stand-alone speakers and mic no echo is > > > > removed. Audio is in 22050 hz at 16 bit mono format, number o...

Hello list, I have nat=no and qualify=no in my sip peer definition and still my CLI is flooded with : [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (24ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (25ms / 2000ms) [Mar 12 10:17:26] N...

...8 kbps mode to maximise compatibility. > Page 8: > > Optional parameters: > > ptime: see RFC 4566. SHOULD be a multiple of 20 msec. > > maxptime: see RFC 4566. SHOULD be a multiple of 20 msec. > > In real world, many SIP application use either 20 or 30ms. This > ptime parameter is really not reliable for negotiation... On possible > way to handle non multiple would be to take the right above value: > if 30ms is specify, then recommand to use 40ms for speex. Actually, it needs to be "MUST" be a multiple of 20 ms because no matter...

...The average RTT is not unusual; since this is only occasional packets it only affects the worst case and standard deviation. What seems likely is that these servers are overloaded for CPU. I have tried the various tweaks of the credit scheduler but the fact remains that the credit scheduler has a 30ms time slice, so I believe that when the server is so loaded that domUs are competing for CPU time, I could expect that a CPU hog gets the CPu for 30ms before handing over to a non-hog who gets a 30ms penalty on an RTT measurement. Clearly the answer is to not overload the servers, and it''s...

...d ones. >> Page 8: >> >> Optional parameters: >> >> ptime: see RFC 4566. SHOULD be a multiple of 20 msec. >> >> maxptime: see RFC 4566. SHOULD be a multiple of 20 msec. >> >> In real world, many SIP application use either 20 or 30ms. This >> ptime parameter is really not reliable for negotiation... On possible >> way to handle non multiple would be to take the right above value: >> if 30ms is specify, then recommand to use 40ms for speex. > > Actually, it needs to be "MUST" be a multiple of 20...

...act that iLBC is > designed > to encode frames independently (more on this below). Even from their > site, you see that even at 13.3 or 15 kbps (they don't specify which), > they have the same quality as G.729A at 8 kbps. The two different bitrates are just the difference between 30ms and 20ms frame sizes; you use a bit more bps with the 20ms frames. > > Now, their point is that since each frame is independent, the codec is > more robust to packet losses which is good for VoIP. However, in my > opinion (and that of others) is that the price in terms of bit-rate is...