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voice problem in vicidial.php

PostPosted: Tue Sep 18, 2007 4:07 pm
by vicidial
hello

i m using
asterisk-1.2.17
zaptel-1.2.16
VD_app_conference_0.6
astguiclient_1.1.12-3


when i m doing manual dialing from softphone the voice is clear and good
but on the same time if i m trying to dial through the vicidial.php page i m getting lot of voice problem like voice cracking
echo problem .

i m using ulaw in client side and server side open source g729

this is my top command result:


Tasks: 132 total, 3 running, 129 sleeping, 0 stopped, 0 zombie
Cpu(s): 26.5% us, 26.8% sy, 0.2% ni, 46.2% id, 0.0% wa, 0.0% hi, 0.3% si
Mem: 2071512k total, 2000296k used, 71216k free, 56580k buffers
Swap: 2031608k total, 0k used, 2031608k free, 1678400k cached

PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND
19086 root 25 0 25216 10m 5648 R 99.9 0.5 42:17.56 asterisk
19258 root 16 0 6844 5204 1880 S 1.0 0.3 0:11.17 AST_update.pl
2782 mysql 16 0 126m 22m 3904 S 0.3 1.1 1:59.38 mysqld
1 root 16 0 1744 568 492 S 0.0 0.0 0:01.89 init
2 root RT 0 0 0 0 S 0.0 0.0 0:00.03 migration/0
3 root 34 19 0 0 0 S 0.0 0.0 0:00.00 ksoftirqd/0
4 root RT 0 0 0 0 S 0.0 0.0 0:00.00 watchdog/0
5 root RT 0 0 0 0 S 0.0 0.0 0:00.04 migration/1
6 root 34 19 0 0 0 S 0.0 0.0 0:00.00 ksoftirqd/1
7 root RT 0 0 0 0 S 0.0 0.0 0:00.00 watchdog/1
8 root 10 -5 0 0 0 S 0.0 0.0 0:00.00 events/0
9 root 10 -5 0 0 0 S 0.0 0.0 0:00.00 events/1
10 root 10 -5 0 0 0 S 0.0 0.0 0:00.00 khelper
11 root 10 -5 0 0 0 S 0.0 0.0 0:00.00 kthread
14 root 20 -5 0 0 0 S 0.0 0.0 0:00.00 kacpid
108 root 10 -5 0 0 0 S 0.0 0.0 0:00.00 kblockd/0
109 root 10 -5 0 0 0 S 0.0 0.0 0:00.00 kblockd/1


this is my cpu information:



[root@localhost asterisk]# cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cpu family : 6
model : 15
model name : Intel(R) Xeon(R) CPU 3040 @ 1.86GHz
stepping : 6
cpu MHz : 1867.177
cache size : 2048 KB
physical id : 0
siblings : 2
core id : 0
cpu cores : 2
fdiv_bug : no
hlt_bug : no
f00f_bug : no
coma_bug : no
fpu : yes
fpu_exception : yes
cpuid level : 10
wp : yes
flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm pni monitor ds_cpl est tm2 cx16 xtpr lahf_lm
bogomips : 3194.88

processor : 1
vendor_id : GenuineIntel
cpu family : 6
model : 15
model name : Intel(R) Xeon(R) CPU 3040 @ 1.86GHz
stepping : 6
cpu MHz : 1867.177
cache size : 2048 KB
physical id : 0
siblings : 2
core id : 1
cpu cores : 2
fdiv_bug : no
hlt_bug : no
f00f_bug : no
coma_bug : no
fpu : yes
fpu_exception : yes
cpuid level : 10
wp : yes
flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm pni monitor ds_cpl est tm2 cx16 xtpr lahf_lm
bogomips : 3727.36

i m having 2mbps bandwidth in wireless connection and there is no packet loss in my bandwith
and i m having only 10 agents

can any one help me out

PostPosted: Tue Sep 18, 2007 10:32 pm
by ramindia
app_conference_0.6


app_conference is not stabled and no one tested on live envirionment
other than some people on the group posted some results
but it was not 100% success i see.

So i recomend to move to app_meetme

ram

PostPosted: Thu Sep 20, 2007 7:40 pm
by vicidial
i have been move to app_meetme but the problem is same

PostPosted: Thu Sep 20, 2007 11:08 pm
by ramindia
Hi

after moving to meetme

post relevent config files, and errors related to that perticular call

Voice break, choppy voice, other related problem related most of the time

bandwidth, internet latency, consistency.

ram

PostPosted: Sun Sep 23, 2007 12:17 pm
by gardo
the biggest issue w/ meetme is the need for a timing device. get a cheap x100p card instead of using ztdummy and this will help a lot.

PostPosted: Sat Sep 29, 2007 11:47 am
by vicidial
i m having the fxo card and i have install but the problem is same

PostPosted: Sat Sep 29, 2007 2:48 pm
by gardo
check if you have loaded the module for x100p (wcfxo) and not ztdummy.

PostPosted: Sat Sep 29, 2007 3:22 pm
by Op3r
echo is not possible using voip. Check your equipments.

PostPosted: Sun Sep 30, 2007 5:10 am
by gardo
do you have proof for this op3r? any link that can backup your statement? i get echo sometimes but not a lot calling someone in the US using different voip providers.

PostPosted: Sun Sep 30, 2007 11:11 am
by Op3r
I used to have that kind of problem. Then i checked its my headset's fault.

I tried many different voip providers ranging from airborn to voxee some of them gave me echo using that headset others don't.

Its all about the equipment.

a simple query gave me this yield.

Echo in telephony systems is caused by two main phenomena: the first is electrical echo due to imperfect impedance matching, and the second is acoustic echo due to microphone pickup of audio output. Both these sources produce similar effects and have to be treated similarly. The major difference is electrical echo is a property of the line connection and remains mostly constant throughout the call, while acoustic echo varies in strength and delay depending on the changing acoustic environment of the echo source. For instance, on a hands-free cell-phone call, the echo characteristics change as the speaker moves around.

and oh here's another one.

http://forums.digium.com/viewtopic.php? ... =echo+voip



Please learn more about it than starts asking me anything if I have proof. or any link that may back up my statement. Thank you and have a nice day.

PostPosted: Sun Sep 30, 2007 11:37 am
by gardo
Dude, what you posted doesn't say anything about voip not having echoes. It just explain what an echo is. The link you also posted doesn't contain any relevant info regarding voip echoes.

Doing a little research, I found this link: http://www.commsdesign.com/main/1999/10/9910feat4.htm. It's about echo cancellation for voip.

This disproves your statement that echo is not possible in voip lines.

PostPosted: Sun Sep 30, 2007 12:45 pm
by Op3r
We're sorry. The page that was requested does not exist.

Please hit the back button on your browser to return to the last page you visited. Or explore any of our sections via the links to the left.

Thank you for visiting CommsDesign.

Yep.

edit***
found the link you gave.

Basically it is stating the obvious... and which is way out of asterisk control :) and which way out of the user's control .



check the digium's forum link.

PostPosted: Sun Sep 30, 2007 12:56 pm
by gardo
Hehehe! Omit the period at the end of the link.

What I'm trying to point here is that voice echoes also happen in voip. That it's not true that it's not possible as you stated earlier. That's just it. Nothing more nothing less.