I am using Asterisk 1.6 with Konference, in Centos 5.4.
with Mizuphone softphone latest version(trial/free) in Windows XP & 1 custom made phone which use for call.
For Echo cancellation at softphone’s headphone speaker i use SolicallPro latest version.
For Call management i use custom made java application(i write/edit that) which use Asterisk-java API.
No problem in call when 2 people talks both can hear each others voice clearly & without echo.
- when i listen audio file which generated by Konference as wav format i can hear voice with echo. It observe as Person talk from Mizuphone his voice has only echo in wav file.
- If call time is 10 min then wav generate only 8.19 or 8.50 min but it contain all speech, if i read book in call then all lines cover in that call which i read it.
- If 2 person talks same time some time both cant listen anything in speaker even in recorded wav file also i can observe that both peoples all sentences are covered but for understand it very difficult. I have to listen that wav file 4-5 times. that’s not issue but it not come in speaker of phones but in wav it comes?
1.In asterisk i see Mizusoftphone use ulaw codec where my device use G722,
In Mizuphone’s log i can see codec used as PCMU.
By java application i hard-coded send Asterisk G722 for both channels(Dialer & Receiver).
- When 2 person talks each other both not observe Echo or call sound quality issues but when i listen wav file echo is there.
What i done:-
I try lots of call by following checking
- I check sound drivers settings in Windows XP.
- Sound setting for Recording , Master volume etc which ever available for slide bar or checkboxes etc.
- In Softphone Mizuphone audio related settings.
I not understand how softphone & custom made phone communicate with each other by different codecs.(PCMU,ulaw, G722).
My main problem is i can change any version of Asterisk/Konference/Softphone/Centos/windows/API
But have to solve the problem.
Any clue or help? If needed i have lots of logs which i not understand clearly completely.