search for: 150ms

Does anyone know the default E&M Wink timings for Nortel DID ports? The default settings on Asterisk are: ; prewink: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ; rxflash: Receiver flashtime (default 1250ms) ; debounce: Debounce timing (default 600ms)

Does anyone have X-Lite build 1059 working fully with Asterisk? The GSM Codec works very well now but we have problems when using G711 in that when I setup a ping between the two sites and then watch the latency, it steadily increases and starts at about 150ms and goes up to 2500ms within about 20 seconds. I have not investigated fully but I guess that its sending ever increasing size packets. When we use the GSM codec its fine. When we have used the G711 codec in previous releases of X-Lite, it's been fine. Also we get one way speech in the followi...

...quality voice. The problem is that voice over calls routed through SimpleTelecom and nNufone occassionally breaks. We also have a digium card and the calls over the digium card using the Zaptel Interface have a very good quality. We have enough bandwidth, the latency to the servers is 100-150ms and the packet loss is around 1%. We tried using the G711 and G729 codec and also have the jitterbuffer enabled. How can I solve this problem of voice quality? Can a better implementation of jitterbuffer with packet loss concealment help? If so how do I get the newer implementation. I would rea...

...andrake 9.2 (kernel: 2.4.22-32mdk) with a dual 800mhz PIII with 256M Ram 4port FXO digium card, no IRQ sharing I can find (cat /proc/pci & cat /proc/interrupts), vmstat reports a minimum of 80+% CPU idle when problem occurs. connect to a Grandstream 101 (GS) via vpn (no nat). Link has 100ms - 150ms ROUND TRIP latency (constant 'ping' during test). Codec is alaw or ulaw. TDM card is plugged to an NEC PBX (old NEAX 2000 IVS) via the analog station card in the PBX. PROBLEM: Establish a call from the GS to an NEC phone (Dterm III) connected to the PBX. The voice quality on the GS sound...

...n speech for far-end input and same speech with attenuation, a bit of reverb and 50-150 ms delay, all this done in sound editor, for mic input, i get 5-8 db attenuation. But when i use the same speech played and recorded for mic input, i see about 5 db of attenuation only if i remove the delay (150ms) at the beginning of mic input file to make far-end and mic input files to go almost ?in sync?, and even in this case there?s almost no attenuation during about a second after pauses in speech. Looks like the filter re-adapts after each pause. When the delay is at least 50 ms, the output of echo ca...

...r digging, delete logs and restart icecast we found this: [2020-11-15 22:09:38] INFO auth/queue_auth_client auth on (default mount) has 101 pending [2020-11-15 22:09:38] WARN auth/auth_add_listener too many clients awaiting authentication Before those errors our auth api was logging ~60 rps in ~150ms (during the error the api was not getting requests). Has anyone else has already faced this problem? Any tips? Best, Thiago -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/icecast/attachments/20201115/0311f293/attachment.html>

...full*9/10 2 default,flow=nfct-src After upgrading I was getting this message in my log: Nov 14 13:42:42 Setting up Traffic Control... HFSC: Illegal "umax" HFSC: Illegal "sc" ERROR: Command "tc class add dev eth1 parent 1:1 classid 1:110 hfsc sc umax b dmax 150ms rate 1575kbit ul rate 3150kbit" Failed After some investigation of /var/lib/shorewall/firewall I found the above line to be: run_tc class add dev eth1 parent 1:1 classid 1:110 hfsc sc umax ${eth1_mtu}b dmax 150ms rate 1550kbit ul rate 3100kbit But eth1_mtu was not defined anywhere! Hence it...

...TC -s qdisc show dev ${IFACE}, I see, that the packets go to correct qdiscs. But when I start FTP data transfer, then the ping time is same with and without this shaping. I should also mention, that I am testing it on WiFi with hostap drivers, where the ping times are about 2-3ms when idle and 100-150ms durring high traffic. Is the first/fastest prio class really 1:1, and the last/slowest is 1:4? Or did I miss something else? A part of the code follows: $TC qdisc add dev ${IFACE} root handle 1:0 prio bands 4 priomap 2 2 2 2 2 2 0 0 1 2 2 2 2 2 2 2 2>/dev/null $TC qdisc add dev ${IFACE}...

...kilobytes??) for the internet bandwith. i cannot use CRTP because i cant be sure if the ISP's routers are CRTP aware. Installing ADSL from ISP1 on the asterisk place will give a clear advantage Please correct any of my prior statements if wrong. should I maintain packet latency below 300ms or 150ms? How can I measure this latency all the way to the asterisk? Should I ping from SITE_A to the asterisk box with 8k packets? If I can't install ADSL for the moment, will the above setup work? thanks in advance for all your help. Erick.

I''m building a new index from scratch based on a number of documents stored in a database loaded using my Rails env (using Ruby Ferret 0.9x (installed today with Gem) on Windows). At first everything goes nice but after a number of documents it starts to go slower and slower until it grinds to a halt (at least feels like it). Am I doing something wrong? Is there some way to work

...oup B Rcv Idle Tones :::::::: B-2 & B-3 Group B Rcv Busy Tones :::::::: B-1 & B-2 ANI Request ::::::::::::::::::: Do not request ANI ANI Length ANI Category Request Tone ANI Tone Request ANI Category :::::::::::::::::: I-1 ANI Calling Party Category :::: II-1 Seizure Ack Timeout ::::::::::: 150ms Release Guard Timeout ::::::::: 600ms * Always double-check the DNIS length with your carrier *************************************************** example of inopportune disconnection: Jan 17 10:04:17 WARNING[-1120736336]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 Rx bits 0x9 [1/ 20/...

...EventMachine to proxy HTTP to other processes. On OS X, it works great, adding only about 20% extra latency into the connection when proxying versus connecting to the original backend process directly, so a 5ms connection might now take 6ms. On a Linux deployment, however, that 5ms was going up to 150ms! I noticed that when I removed EventMachine.epoll, everything corrected itself and it worked as quickly as in OS X. So, for now I''ve removed EventMachine.epoll, and all is well. The first question, therefore, is does anyone have any experience or knowledge of why using epoll could make an...

Hello, I have an interesting question. I can''t find a solution anywhere to this problem. I have an adsl connection which is able to send a full data packet in about 150ms. I want to cut this time down to decrease latency while serving files, so I thought I''d try and reduce the packet size for the offending upload streams (my webserver). The trick is that I also want all other tcp connections to have the normal full packet size. Enter MSS-clamping. My pppoe...

Why ov_open_callbacks(fd_, &vf, NULL, 0, OV_CALLBACKS_NOCLOSE) takes so long to open OGG file? The OGG file has just 210 MB. It takes a lot of time open the file. I am working on OGG audio recorder and player. Thanks in advance! With kind regards, Pavel

...n speech for far-end input and same speech with attenuation, a bit of reverb and 50-150 ms delay, all this done in sound editor, for mic input, i get 5-8 db attenuation. But when i use the same speech played and recorded for mic input, i see about 5 db of attenuation only if i remove the delay (150ms) at the beginning of mic input file to make far-end and mic input files to go almost ?in sync?, and even in this case there?s almost no attenuation during about a second after pauses in speech. Looks like the filter re-adapts after each pause. When the delay is at least 50 ms, the output of echo ca...

...same speech with attenuation, a bit of reverb and 50-150 ms delay, all this > done in sound editor, for mic input, i get 5-8 db attenuation. > > > > But when i use the same speech played and recorded for mic input, i see about 5 > db of attenuation only if i remove the delay (150ms) at the beginning of mic > input file to make far-end and mic input files to go almost ?in sync?, and even > in this case there?s almost no attenuation during about a second after pauses in > speech. Looks like the filter re-adapts after each pause. When the delay is at > least 50 ms, t...

...An observation is that icecast buffers seem to keep growing -- information from liquidsoap is more and more in advance compared to what one hears. So I decided to ask delay() if liquidsoap's centralized scheduling is correct for shout outputs. It appears that delay() is nearly a constant, about 150ms. I'm wondering about the precise meaning of this. First interpretation: icecast is warning us that the buffer is growing too fast, something is wrong, liquidsoap shouldn't be so fast. Second interpretation: everything's ok, the advance is a constant so it has no impact on the buffer. I...

...describe what happended, but I'll try. After changing from 3.5p1 to 3.6.1p2 without any changes in compilation, or configuration, sshd started to "choke". Data displayed on the terminal is displayed partially for example half of the screen in 200ms and the rest after a pause of 100-150ms. It's realy annoying. I have 100mbit/s full-duplex cards. Load average is about 0.02. Everything was fine, till i've changed the version. 'm running 2.4.20+grsec1.9.9.+cc0.23pre5 with libc-2.2.4 and openssl-0.9.7b. Data on tty's is displayed fluently. I tried to change sshd's...

Hi, When I use the analog phone connected to Zap/1 how do I transfer & hold the caller ? When I hit the flash key, all that happens is the caller hears a beep (sounds like DTMF). But no stutter dial tone on the Zap/1 Port, just continuing conversation with the caller. What could be wrong here? Cheers Jay