Does anyone offer VOIP (SIP or IAX) dial-in numbers WITHOUT echocancellation

Discussion in 'UK VOIP' started by jrwalliker, Jul 24, 2008.

  1. jrwalliker

    jrwalliker Guest

    All I want is a UK (and maybe also a USA) dial-in number which
    forwards A-law (or ulaw in USA) audio packets from the PSTN to my
    asterisk server without changing the audio in any way. This means no
    echo cancellation.

    Can anyone offer such a service?

    I could do it myself with a group of ISDN2 lines or preferably an
    ISDN30 connection, but this is rather expensive.

    In any case, why do VOIP service providers insist on echo cancelling
    their connections when this job is best done at the end points?

    John
     
    jrwalliker, Jul 24, 2008
    #1
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  2. jrwalliker

    Dave Higton Guest

    I don't know; but here are some points to think about.

    VoIP shouldn't inherently have any echo. The acoustic path from
    earpiece to microphone is at a /very/ low level, surprisingly
    low, unless the incoming audio is on speakers - and in this case
    it would definitely be best done at the endpoint.

    Most echo is caused by imperfect termination of an analogue line.
    Any echo cancellation is easiest performed when the path delay
    is shortest. Look at the segments of a call, see which are
    analogue and which are VoIP, and then work out where the echo is
    best performed.

    Another point: acoustic paths are unpredictable and subject to
    variation as the speaker, microphone and other people and objects
    move around in a room, whereas imperfect analogue termination is
    constant over a very long period.

    Dave
     
    Dave Higton, Jul 24, 2008
    #2
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  3. we do this
     
    Peter Gradwell, Jul 24, 2008
    #3
  4. jrwalliker

    jrwalliker Guest

    Peter,

    I'm very pleased to hear it as I am already one of your customers! I
    emailed your support staff the other day and received the following:
    If there really is a way in which you could set some of my inbound
    numbers up (or create new ones specially for this purpose) without e/c
    that would be really useful.

    For the benefit of those who wonder "why?" - I'm doing automated
    testing of mobile handsfree echo cancellers using test software on an
    asterisk server. Having extra cancellers in the way just messes
    things up.

    Regards,

    John
     
    jrwalliker, Jul 25, 2008
    #4
  5. jrwalliker

    jrwalliker Guest

    With a standard BT badged phone on an "analogue" line I typicall see
    an echo return loss of about -25dB.

    John
     
    jrwalliker, Jul 25, 2008
    #5
  6. jrwalliker

    Dave Higton Guest

    Sure, but my point is that that's caused by imperfect termination
    of the analogue line, and not by acoustic feedthrough from earpiece
    to microphone.

    Dave
     
    Dave Higton, Jul 25, 2008
    #6
  7. jrwalliker

    jrwalliker Guest

    With a standard BT badged phone on an "analogue" line I typicall see
    That will depend a lot on the particular phone and line, but my
    experience has been that the echo is often a mixture of mismatch
    reflection and acoustic coupling. It is certainly interesting to
    experiment with the source impedance settings on voip - analogue
    adapters. An incorrect setting here can easily dominate any acoustic
    echo.

    I still wonder why VOIP suppliers seem so keen on echo cancellation,
    when the end points are normally VOIP phones or computers which do the
    echo cancellation themselves. Why spend money on a process which is
    probably not necessary and which if overdone can degrade the signal?

    John
     
    jrwalliker, Jul 27, 2008
    #7
  8. jrwalliker

    alexd Guest

    Perhaps the cost of implementing it across the board is less than the
    cost of fielding support queries from people who do have echo problems.
    Perhaps all the ITSPs you've tested use voice platforms that don't allow
    echo cancellation settings to be set per-extension.
     
    alexd, Jul 27, 2008
    #8
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