[asterisk-users] Echo problem in VoIP-calls

[Jonas Kellens]

Hello,

I stated in my first post that both ends hear an echo when one speaks to 
the other...

The only place where echo cancellation is being applied is in the 
Asterisk server. I have the following in sip.conf :


;------------------------------ JITTER BUFFER CONFIGURATION 
--------------------------
jbenable = yes              ; Enables the use of a jitterbuffer on the 
receiving side of a
                               ; SIP channel. Defaults to "no". An 
enabled jitterbuffer will
                               ; be used only if the sending side can 
create and the receiving
                               ; side can not accept jitter. The SIP 
channel can accept jitter,
                               ; thus a jitterbuffer on the receive SIP 
side will be used only
                               ; if it is forced and enabled.

jbforce = no                ; Forces the use of a jitterbuffer on the 
receive side of a SIP
                               ; channel. Defaults to "no".
;-----------------------------------------------------------------------------------


Thank you for your replies.

Kind regards.
Jonas.


On 06/30/2010 11:36 AM, Gareth Blades wrote:
> Routers wont cause echo. In order for them to do so they would have to
> store the outbound voice traffic, delay it and then mix it into the
> inbound voice.
>
> Telephones inherently cause echo. For domestic calls the audio path is
> normally so short that any echo arrives back so quick the human ear does
> not detect it. For international calls the telco uses expensive echo
> cancelation technology.
> When you switch to voip you are often suddenly introducing a much larger
> delay so any excho which was present before but not noticed suddenly
> becomes noticable.
>
> You need to analyse the audio path your calls are taking, where the
> delays are being introduced and where echo cancelation is being applied.
>
> You also havent stated which end of the conversation is hearing the echo.
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