[asterisk-users] SIP / Echo Cancellation

[Vineet Bhojnagarwala]

Very informative post Vinícius !

2010/3/5 Vinícius Fontes <vinicius at canall.com.br>

> ----- "Chandrakant Solanki" <solanki.chandrakant at gmail.com> escreveu:
>
> > Hello
> >
> > I have successfully compiled OSLEC for echo cancellation for DAHDI
> > channel.
> >
> > Is there any way to do echo cancellation for SIP Channel.
> >
> > Is any, please suggest me.??
> >
> > Thanks in advance..
> >
> > --
> > Regards,
> >
> > Chandrakant Solanki
>
> Short answer: Maybe. Depends on the SIP device you're using.
>
> Long answer:
> *takes a deep breath*
>
> First you gotta understand why echo occurs. Every single call you've ever
> made on your life has echo. You can hear yourself when you're speaking. If
> that was not the case, it would feel like talking on a push-to-talk system.
> So echo is a natural and even desirable phenomenom. What makes echo
> unconfortable is when the echo is *delayed* too much.
>
> There's a number of causes for this to happen. First and foremost,
> sometimes a part of the signal you're transmitting is reflected back to you.
> That usually happens on the analog part of the system (analog phones as a
> whole, the handset of an IP phone, the headset connected to your computer's
> sound card, etc). When we're talking about VoIP, the latencies involved are
> much higher than a completely TDM system. There's the encoding latency,
> easily understood as the time the device takes to convert the analog signal
> (your voice) in RTP packets, then there's the transmission latency, inherent
> to any network, and so on. All those latencies add up to each other, making
> the total latency go skyhigh and making you hear your own voice delayed by
> some milisseconds - the infamous echo.
>
> Asterisk cannot cancel echo when the call is entirely IP, from an IP phone
> to another, for example. There's simply no need for that. That's because
> it's the device's job to cancel the echo caused by its own TX reflections or
> analog/digital conversions. On the other hand, Asterisk can and will cancel
> echo if you have a hardware echo canceller or a software based one, like
> OSLEC -- which is by far the best software echo canceller I've ever seen.
>
> Finally, in order to solve your problem, you'll need to check a few things.
> If the call is entirely VoIP, from one end to other, then the IP phones,
> ATAs, gateways, softphones, whatever, are the sole responsibles on
> cancelling the echo. You'll need to turn on echo cancelling on this devices
> or tweak its parameters. Also, don't forget that latency makes echo much
> worse. If you control the entire network between the two phones, you MUST
> set up a QoS policy in order to minimize the latency as much as possible.
> I've solved many echo problems by just implementing end-to-end QoS on the
> network.
>
> Lastly (I swear I'm finishing this essay right here :), if that's not your
> case and you're having echo issues calling from a SIP phone to an external
> number, double check if OSLEC is indeed set as the echo canceller on
> /etc/dahdi/system.conf and enabled with echocancel=yes on your
> chan_dahdi.conf. You can always check if the echo canceller is active on a
> certain DAHDI channel by issuing the command "dahdi show channel XX" on
> Asterisk CLI, where XX of course is the said DAHDI channel.
>
> --
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