[asterisk-users] Bad Echo between SIP calls

[Darryl Dunkin]

The echo cancellation card is for SIP->Zap calls only, no echo
cancellation is done in Asterisk for SIP only calls. SIP to SIP, media
is just passed through the server  untouched (using media flow through,
which is the option in sip.conf of canreinvite=no) if you are not
handling any translation, even when handling translation between SIP
calls there shouldn't be any echo cancellation done in Asterisk for SIP
only calls.
 
The place to look at would be the remote SIP devices which is typically
what is adding the echo, this is usually a gain issue of some sort
depending on which handsets you are using.

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Deepak
Naidu
Sent: Monday, June 11, 2007 16:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bad Echo between SIP calls


Sounds crazy right? even was I, more over support guy logged in unloaded
the zap modules to test them, still an echo.

Ya, I was clear saying that we have SIP--- SIP issue ie internal
extension echo problem.  It seems the echo with SIP--SIP has many
factors.  I am just curios to eliminate any possibility of Asterisk
failing to cancel the echo.

OK, one question here howz the call flow when a SIP---SIP call is
established ie.  is the connection between 2 phones when an Internal
call is made or does the SIP call goes via Asterisk once the SIP--SIP
call is establised.

--
Deepak

Matthew Fredrickson <creslin at digium.com> wrote: 


	On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote:
	
	> Hi,
	>           We have a PRI connection & when its was on test
networks we 
	> had echo problems withoutside line. 
	>
	> So I bought a TE212P card resolve the echo problem.  Which did
to an 
	> extent. Its using asterisk 1.2.18 & RHEL4-Update 4.
	>
	>
	> But now when we are live, there is a terrible echo between 2
SIP 
	> calls. If I call the same extension from outside the voice is
clear.
	>
	> I am not sure whats the problem.  Also there's slight echo
when 
	> calling Digium support.
	>
	> Totally lost Digium says we need to remove the echo module to
resolve 
	> SIP echo problems. Then ? the heck we pay for..
	
	Are you sure that they understood that you were having this
problem 
	between 2 SIP endpoints? That advice only makes sense to test if
one 
	side is Zap and the other side is SIP.
	
	
	---
	Matthew Fredrickson
	Software Engineer
	Digium, Inc.
	
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