VoIP echo problem

Discussion in 'VOIP' started by Andy&Alice, Jul 4, 2003.

  1. Andy&Alice

    Andy&Alice Guest

    Hi,

    Can anyone tell me what happen of VoIP conversation over broadband, the
    sound have great echo and feed back from PSTN to FXO to internet to FXO to
    PSTN.
     
    Andy&Alice, Jul 4, 2003
    #1
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  2. Andy&Alice

    RealWiild Guest

    Hi Andy&Alice

    > Can anyone tell me what happen of VoIP conversation over broadband, the
    > sound have great echo and feed back from PSTN to FXO to internet to FXO to
    > PSTN.


    Basically You need to look into RX /TX gains all places where you hvae PSTN
    <-> FXO (ip) Conversions. You need to make sure you have a Echo Canceller in
    the VoiceGateway if not you can not do anything about the Echo.


    I posted this a while back :

    Echo cannot origin from an VoIP network. But delay times due to codecs and
    buffering quickly makes even the slightest echo recieved very annoying. Echo
    is generated by digital (4 wire) <-> anolog (2 Wire) conversions either in
    the PSTN or "at the other end" (overlaps of ear & mouth)".

    If you hear echo then the source of the echo is either the far end or
    somewhere in the PSTN network.

    If the far end hears echo then you generate the Echo.

    There are a couple of mechanism to prevent echo that is ERL (Echo Return
    Loss) og ERLE (Echo Return Loss Enhance). ERLE is often named Echo
    Canceller.

    ERL use adjustment of powerlevels for recieve and transmit audio stream, and
    should be adjusted so that the echo is as low volume as possible without
    loosing the ability to communicate with the far end. You have 2
    possibilities :
    1) Lower powerlevels of what you send out - Risk : The far end cannot
    hear you
    2) Lower powerlevels of what you recieve - Risk : You cannot hear the
    far end

    If the Echo is Doubletalk - then ERLE cannot distinguish Real talk from
    Echo, and the ERLE stops working.
    If this happens you should adjust ERL values.

    So ERLE stops working with doubletalk. But there is an other more annoying
    way to stop the ERLE from working - The is delay times outside all gateways
    from the VoIP network. Being the gateway toward PSTN og The Analog Pots.
    Every POTS should have an ERLE on at least 8 ms, this would normally be
    sufficient to removed ECHO generated bye overlap between Ear & Mouth pieces
    or in the short cable connecting the phone to the POTS. Of course IP phones
    should also have at least 8 ms ERLE.

    The gateway towards the PSTN network should have sufficient ERLE to
    compensate for the delay in the PSTN (which is about ~1 ms pr. 1000 km
    fiber, 2-4 ms pr. interconnection - ADMs and so on) ... The Standard for
    ERLE i G.164 which specify at least 128 ms - But 128 would make the price of
    the gateway astronimical for us non-Telco operators due to the complexity in
    ERLE. Therefore typical values 16-64 ms typically 16-32 ms ERLE. ERLE
    buffers constantly 16 ms of outgoing talk, converts it to a reverse pattern
    (a ^ function - e.g. a HAT - function) and compare it to the incoming voice
    stream. If it finds a matching pattern on the incoming voice stream it will
    be applied if not it will be discarded. If delay times is more that the
    buffer in ERLE then ERLE will never work. And echo cannot be eliminated. You
    can only minimize the annoyence using ERL operations.

    There are allways ERLE deployed in the Public PSTN network where it
    interconnects with international destinations, and towards Mobile networks
    (PLMNs). But not allways between PSTN operators. Thats because all other
    telephony network and connection can tolerate fairly large amount of delay
    in the public network - You can have echo but it is not hearable by the
    human ear.

    With VoIP networks which have to convert voice stream to data streams and
    back again so that the round tripdelay times can often be more that 160 ms -
    Which is well in to the hearable area (I belive all types of Echo with a gap
    or roundtrip delay above 32 ms is hearable first talking in a large room but
    the larger roundtrip delay is the more distinct the echo becomes.
    This also means that lag times in the IP network is not a probable
    contributor to the Echo - The operation around the DSPs width roundtrip
    delays over 160 ms is the probable contributor. Reducing buffering in the
    DSP function is a possibility, but at the cost of bandwidth and does
    normally not help much.

    The only way to eliminate echo is adjusting ERL and have to sufficient ERLE
    in the gateways. Where ERL reduces the powerlevel of the Echo, so that ERLE
    can kickin and remove the echo completely (Actually reduces the powerlevel
    of the Echo to very low values (unhearable) . The combined loss Echo by the
    combined ERL and ERLE is ACOM - I cant remember the threshold values of the
    power levels for hearable sounds but ACOM should be below this value if ERLE
    works. You can only completely remove echo by removing the source of the
    Echo.

    I Hope you can use this information - Even if it is written in a bit poor
    english.


    Cheers, Real
     
    RealWiild, Jul 4, 2003
    #2
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  3. Andy&Alice

    shido Guest

    What are you using? Please be very specific.

    -Greg

    "Andy&Alice" <> wrote in message
    news:be47gr$...
    > Hi,
    >
    > Can anyone tell me what happen of VoIP conversation over broadband, the
    > sound have great echo and feed back from PSTN to FXO to internet to FXO to
    > PSTN.
    >
    >
     
    shido, Jul 5, 2003
    #3
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