search for: 80ms

...te, &page); firstPacketInPage = 1; packetsToFetch = 1; while(packetsToFetch) { ogg_packet packet; packetOutRet = ogg_stream_packetout(&m_streamState, &packet); // This is the test case. If you try and seek, using op_pcm_seek, // to a position, that when adjusted by 80ms for the discarded data // lies within the first half of a packet that has been split over // an ogg page boundary, op_pcm_seek will return OP_EBADLINK. This // seems to be because the library doesn't take into account that // split packets are not counted towards the granule posit...

...other attempts, and there have never been official > plans. It's difficult to partition input for opus at anything other than > the track level, because of the way the decoder derives its adaptive > state from recently-seen audio. I guess cutting together streams with at > least an 80ms overlap wouldn't glitch too much? According to the Opus standard, after 80 ms the encoding would converge. That is, only the previous 80 ms of audio would be needed to get a perfectly merged stream. You could play safe and do, lets say, 200 ms overlapping. For example, read https://wiki.xiph....

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

...it determined by observation. In the first test, I limit the SUBCLASS_OUTRATE to 200Kbit. Both pings are around 20ms before I start the VoIP services. However, once I start the services, the pings jump up to 1800ms. In the second test, I limit the SUBCLASS_OUTRATE to 180Kbit. The pings jump up to 80ms, which is perfectly acceptable. After a few tests, I noticed that 180Kbit is a magic number, anything exceed that will generate 1800ms pings, and below it is 80ms. In my senario, the weird point is that the determining factor is the ceiling, but not the rate. That''s the "rate"...

Yes, that's a start. Ultimately, though, I'm hoping to reduce the 80ms requirement, and am trying to get a handle on what state in the decoder must converge and what complications I might be up against. I'm also only considering CELT-based encodings if that simplifies things. I know that Opus can do inter-frame energy envelope prediction, but that dependency can...

...the OS. Since libguestfs doesn't use a boot disk, and the appliance disk is limited to 4GB, and we use virtio-scsi, this is all useless. EDD is also broken currently on RHEL 7.3, see: https://bugzilla.redhat.com/show_bug.cgi?id=1404287 Also the EDD probing takes significant extra time (about 80ms on my laptop), and using edd=off reduces this time although doesn't entirely eliminate it. --- src/launch.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/launch.c b/src/launch.c index 78bf46d..46d7ab9 100644 --- a/src/launch.c +++ b/src/launch.c @@ -363,6 +363,7 @@ guestfs_int_applia...

I use Speex with dialup modem users. Even if the dialup modem is "56K", you should assume that upstream bandwidth available is only 20-30kbps at the most. I use 16kHz wideband mode and VBR quality 2. Also, I send 80ms (4 frames) per packet, because there is an overhead of approximately 33 bytes per packet due to UDP (8 bytes), IP (20 bytes), and PPP (~5 bytes) headers. If you sent 20ms (1 frame) per packet, this overhead would be 13.2kbps, which is totally unacceptable for a dialup modem. 80ms packets res...

...Along with that I'm unable to enable all my addons due to the fact that my frame rate will drop to at extremely low levels. But when I run as root (I know this is a bad idea..that's why I'm trying to fix it) I'm able to enable all 215 addons and will get latency of any were between 80ms to 180ms and a frame rate of about 40-50fps with almost everything turned to its max with the exception of shadows. [Rolling Eyes] I start wow from the shell Code: wine ~/Documents/World\ of\ Warcraft\ 3.2.0/Wow.exe Please let me know if you need any other information. System Specs GeForc...

Before I get too far in my attempts... Has anyone already implemented support in scp for larger buffers/windows that would take advantage of wscaled TCP windows? Paul Hyder NOAA Forecast Systems Lab Boulder, CO FYI: Linux 2.4.20, 30-80ms RTT, data rates 100-1000Mbps, and a need to fill TCP windows of 2-8MBytes. (Existing limits appear to be about 256KB.)

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 65536 4194304 net.core.rmem_max = 8388608 net.core.wmem_max = 8388608 net.ipv4.tcp_bic = 0 (tcp westwood and vegas are also turned off for now) Now if I run 50 netpe...

Hi, I am checking out the quality at a few vendors, and althought I know it doesn`t totally reflect call quality I am using ping as a cheap subsitute to having a real VoIP testing system The question I have is this one: given that one service gives me a 80ms ping (pretty consistantly) and another one gives me 30ms (again very consistently), is this 50ms difference enough to impact perceived call quality? Or will the quality be impossible to differenciate, and I should choose based on some other criteria? (customer service, price, etc) The thing is I...

I had CC readers going over the internet (with pings over 80ms) connected to Linksys PAP2. It was only successful once every 3 attempts. I had 100% reliability when it was connected on LAN. Timing is an issue, if you doing everything on LAN it should not be a problem. Just make sure you use G.711 protocol. > -----Original Message----- > From: asterisk...

...(alphalabs), lan ip: 192.168.2.103, tinc-ip: 10.243.1.10 internet vserver (slowpoke), no lan ip, tinc-ip: 10.243.232.121 Everything works fine until both nodes are in the same LAN. The first 2-3 minutes everything is fine. Pings between the machines go through other servers so they are between 50-80ms, 0% packet loss. But as soon as localDiscovery starts to kick in the pings drop to 20ms (which is good) but packet loss goes up to 90% (tested with ping -f). For every lost packet tincd (started with -D -d4) shows the Error: "Received UDP packet from unknown source 192.168.2.103 port 655"...

Hello, I would like to know how Asterisk deal in this case: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk

Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don't really know where to start on measuring jitter

...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 from what's in iax.conf or their experiences or maybe some adjustments I should try that'd be really really helpful. Thanks!! Chris

I am interested in being able to encode a single Opus stream using several CPU cores. I get a raw audio input and "opusenc" can transcode it at 1200% speed (Raspberry PI 3B+). It saturates a single CPU core, but the other three are idle. Is out there any project to add multithreading options to "opusenc", or something in that line? Looking around, I have found this:

...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 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 "ave...

...ent for -l13 very high compression setting. For x265 (HEVC), I use BPG codec, and this time I use -m 0 setting which uses the fastest encoding setting of x265.I run the .exe 10x times and I pick the best scores. So here are the timings at -l13 very high compression: Encode time: x265 (BPG -m 0): 80ms NHW: 20ms Decode time: x265 (BPG): 90ms NHW: 10ms So with an equivalent level of optimization (x4-5, multithreading, SIMD,...), the NHW Project is 80/20x5=20x faster to encode and 90/10x5=45x faster to decode than x265 (optimized HEVC)! Yes, 45x faster to decode than x265 at very high compressi...

Hi guys, I have a bunch grandstream phones using ulaw and my users are complaining they are jittery when I use "canreinvite=yes". The data connection is an ADSL link dedicated for phone traffic. At any given time, I have at most 2 calls in parallel. I'm not a huge fan of asterisk being in media path doing buffering because the delay (jbmaxsize=80,jbimpl=fixed) is pretty long and