search for: 32ms

I apologize about doubling these up, I forgot the subject! I have a cisco VPN from router to router over a Data T-1. The ping times are consistently 32ms with random ping responses of 295ms -408ms about every 30 secs to a minute, I have jitter buffer enabled. The connection goes like this... Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B and then T-1 to a PBX, the calls are internal so they are terminating on Toshiba digital phones. Loud c...

...------------- [Apr 22 17:50:48] NOTICE[2550] cdr.c: CDR simple logging enabled. [Apr 22 17:50:48] NOTICE[2550] loader.c: 33 modules will be loaded. [Apr 22 17:50:48] WARNING[2550] translate.c: plc_samples 160 format f [Apr 22 17:50:48] NOTICE[2561] chan_sip.c: Peer '104' is now Reachable. (32ms / 2000ms) [Apr 22 17:50:48] NOTICE[2561] chan_sip.c: Peer '105' is now Reachable. (34ms / 2000ms) [Apr 22 17:50:48] NOTICE[2561] chan_sip.c: Peer '107' is now Reachable. (30ms / 2000ms) [Apr 22 17:50:48] NOTICE[2561] chan_sip.c: Peer '101' is now Reachable. (31ms / 2000ms) [...

Hi Jean-Marc, Regarding you points: 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent to the client and decoded when it is recevied so the AEC is always performed on raw PCM16 8KHZ ? 2) The audio is moved in 32ms (512 byte) chunks and the reading and writing to the AEC code will be done by separate threads at regular 32 ms intervals. 3) Occasionaly audio is dropped if it has become delayed but a jitter buffer of 120ms is in use. People at different distances from the server will have a slightly dif...

...SIP/101,30,Ttm) [outgoing] exten =>_94XXXXXXX,1,Dial(ZAP/g1/${EXTEN},45,tTwW) exten =>_94XXXXXXX,2,Hangup() exten =>_94XXXXXXX,102,Hangup() zapata.conf: [channels] signalling=fxs_ks usecallerid=yes callwaiting=no threewaycalling=no transfer=yes cancallforward=yes ; valores validos 256(32ms),512(64ms),1024(128ms) echocancel=yes echotraining=yes echocancelwhenbridged=no rxgain=0 txgain=0 group=1 callgroup=1 pickupgroup=1 immediate=no faxdetect=incoming ;busydetect=yes ;busycount=10 answeronpolarityswitch=yes hanguponpolarityswitch=yes polarityonanswerdelay=600 ;callprogress=no progzone...

...Number of Interfaces: 1 Attributes: 80 MaxPower Needed: 100mA Interface Number: 0 Name: hid Alternate Number: 0 Class: 03(HID ) Sub Class: 0 Protocol: 0 Number of Endpoints: 1 Endpoint Adderss: 81 Direction: in Attribute: 3 Type: Int. Max Packet Size: 8 Interval: 32ms If someone could help me, it'll be really nice, as I've tried newhidups - fentoups - energized drivers without success. I think my problem come from the USB to serial chipset from Cypress which is not included into newhidups but it's out of what I can manage. Best regards, Guillaume...

...Number of Interfaces: 1 Attributes: 80 MaxPower Needed: 100mA Interface Number: 0 Name: hid Alternate Number: 0 Class: 03(HID ) Sub Class: 0 Protocol: 0 Number of Endpoints: 1 Endpoint Adderss: 81 Direction: in Attribute: 3 Type: Int. Max Packet Size: 8 Interval: 32ms If someone could help me, it'll be really nice, as I've tried newhidups - fentoups - energized drivers without success. I think my problem come from the USB to serial chipset from Cypress which is not included into newhidups but it's out of what I can manage. Best regards, Guillaume...

...oded (using Nelly Moser ASAO) and sent > to the client and decoded when it is recevied so the AEC is always > performed on raw PCM16 8KHZ ? No. The entire path from AEC to loudspeaker and from mic back to AEC must be free of any non-linearity, codec, drift, ... > 2) The audio is moved in 32ms (512 byte) chunks and the reading and > writing to the AEC code will be done by separate threads at regular > 32 ms intervals. You're free to do the processing/blocking as you like. > 3) Occasionaly audio is dropped if it has become delayed but a jitter > buffer of 120ms is in use...

I have a 12 channel PRI with SNOM 190's and asterisk CVS from January. Most calls are fine, all incoming calls are fine, but I am getting echo on a significant number of outgoing calls. The person on the other side hears a perfect call, but the SIPphone side gets to hear themselves. It happens 100% of the time to some numbers (outgoing only), and only sporadically to others. Has anyone

...oded (using Nelly Moser ASAO) and sent > to the client and decoded when it is recevied so the AEC is always > performed on raw PCM16 8KHZ ? No. The entire path from AEC to loudspeaker and from mic back to AEC must be free of any non-linearity, codec, drift, ... > 2) The audio is moved in 32ms (512 byte) chunks and the reading and > writing to the AEC code will be done by separate threads at regular > 32 ms intervals. You're free to do the processing/blocking as you like. > 3) Occasionaly audio is dropped if it has become delayed but a jitter > buffer of 120ms is in use...

I have a cisco VPN from router to router over a Data T-1. The ping times are consistently 32ms with random ping responses of 295ms -408ms about every 30 secs to a minute, I have jitter buffer enabled. The connection goes like this... Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B and then T-1 to a PBX, the calls are internal so they are terminating on Toshiba digital phones. Loud c...

...nut-server.service; enabled; vendor preset: enabled) ???? Active: active (running) since Wed 2024-03-27 12:04:45 CET; 3min 48s ago ??? Process: 17682 ExecStart=/sbin/upsd (code=exited, status=0/SUCCESS) ?? Main PID: 17683 (upsd) ????? Tasks: 1 (limit: 19016) ???? Memory: 620.0K ??????? CPU: 32ms ???? CGroup: /system.slice/nut-server.service ???????????? ??17683 /lib/nut/upsd mars 27 12:04:45 nicolas-MS-7B19 systemd[1]: Starting Network UPS Tools - power devices information server... mars 27 12:04:45 nicolas-MS-7B19 upsd[17682]: fopen /run/nut/upsd.pid: No such file or directory mars 2...

Hi, I am looking for AEC software which can be run on the server side. This means there will be a fairly constant 600ms or so gap between sending out an audio frame and getting it back with echo. Could Speex AEC be configured to handle these conditions? If so, how good can I expect it to be? Thanks --------------------------------- Yahoo! Mail is the world's

We have a zpool made of 4 512g iscsi luns located on a network appliance. We are seeing poor read performance from the zfs pool. The release of solaris we are using is: Solaris 10 10/09 s10s_u8wos_08a SPARC The server itself is a T2000 I was wondering how we can tell if the zfs_vdev_max_pending setting is impeding read performance of the zfs pool? (The pool consists of lots of small files).

....17 make clean make make install make samples make config Mi configuration files: zaptel.com loadzone=es defaultzone=es fxsks=1 zapata.conf [channels] signalling=fxs_ks usecallerid=yes callwaiting=no threewaycalling=no transfer=yes cancallforward=yes ; valores validos 256(32ms),512(64ms),1024(128ms) echocancel=yes echotraining=yes echocancelwhenbridged=no rxgain=0 txgain=0 group=1 callgroup=1 pickupgroup=1 immediate=no faxdetect=incoming ;busydetect=yes ;busycount=10 answeronpolarityswitch=yes hanguponpolarityswitch=yes polarityonanswerdelay=600 ;callprogress=no progzone...

Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too expensive. Can anyone recommend a decent channelbank that won't break the bank? TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

Hi, What kind of latency is expected using 8,16,32 khz? I am trying to do a realtime stream server, and I am having latency above >500ms ( I capture the sound using the mic, encode it and send it to the client). I am using ALSA system, a SB128 PCI and a 800Mhz P3 . What can I do to lower the latency? I tried a test : arecord -t raw | speexenc - - | speexdec - and I found that I also have a 500

....yahoo.com> Content-Type: text/plain; charset="iso-8859-1" Hi Jean-Marc, Regarding you points: 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent to the client and decoded when it is recevied so the AEC is always performed on raw PCM16 8KHZ ? 2) The audio is moved in 32ms (512 byte) chunks and the reading and writing to the AEC code will be done by separate threads at regular 32 ms intervals. 3) Occasionaly audio is dropped if it has become delayed but a jitter buffer of 120ms is in use. People at different distances from the server will have a slightly different ro...

...Prot=00 MxPS=64 #Cfgs= 1 P: Vendor=051d ProdID=0003 Rev= 1.06 S: Manufacturer=American Power Conversion S: Product= FW:COM 02.3 / S: SerialNumber= C:* #Ifs= 1 Cfg#= 1 Atr=60 MxPwr= 2mA I: If#= 0 Alt= 0 #EPs= 2 Cls=03(HID ) Sub=00 Prot=00 Driver=usbhid E: Ad=81(I) Atr=03(Int.) MxPS= 64 Ivl=32ms E: Ad=01(O) Atr=03(Int.) MxPS= 64 Ivl=32ms I have it configured in ups.conf as: [apc] driver = usbhid-ups vendorid=051d productid=0003 port=auto upshid-ups is unable to communicate with the UPS, giving 'Protocol error'. I've also tried the snapshot and 2...

...MSN, DDI, SUB) o Call offering services (TP, CFU, CFB, CFNR) o Call completion services (CW, HOLD, ECT) o Charging services (AoC) o Three-party conference o Large Conference * VoIP Gateway support o Echo cancellation (G.168) 128ms (32ms for VoIP) <<<<< it seems to support echo cancellation o Real Time Protocol (RTP framing) o Dynamic Anti-Jitter Buffer o Comfort Noise Generation o Voice Activity Detection o Voice compression (GSM, G.726) ===== capi.conf ==...

...ls=00(>ifc ) Sub=00 Prot=00 MxPS= 8 #Cfgs= 1 P: Vendor=0665 ProdID=5161 Rev= 0.02 S: Manufacturer=Cypress Semiconductor S: Product=USB to Serial C:* #Ifs= 1 Cfg#= 1 Atr=80 MxPwr=100mA I: If#= 0 Alt= 0 #EPs= 1 Cls=03(HID ) Sub=00 Prot=00 Driver=usbhid E: Ad=81(I) Atr=03(Int.) MxPS= 8 Ivl=32ms usbhid-ups Checking device (0665/5161) (002/006) - VendorID: 0665 - ProductID: 5161 - Manufacturer: Cypress Semiconductor - Product: USB to Serial - Serial Number: unknown - Bus: 002 Trying to match device Device matches HID descriptor, method 1: (9 bytes) => 09 21 00 01 00 01 22 1b 00 i=0, ext...