Another question: some theory.

Discussion in 'VOIP' started by Jack L., Feb 23, 2004.

  1. Jack L.

    Jack L. Guest

    VoIP-developers know the problems VoIP applications suffer during a
    conversation, eg. delay, echo in the far-end and loss of data packets.
    Numerous mechanisms have been developed to remove some of them, but in the
    end, the Internet cannot guarantee data packets to arrive within a specific
    time which means that a VoIP conversation (currently) won't be as good as
    with PSTN networks.

    Would it make sense to research on whether opening X channels to the other
    side and thereby sending the SAME packet X times at the same time will
    ensure that data arrives to the destination? We assume that both sides have
    broadband connection so we have enough bandwidth to "waste". The idea is
    that we hope the packets will be routed differently so even if one packet
    disappears on one of the channels, the same packet will still arrive to the
    destination as the routers (hopefully) let it take another path.

    The idea probably sounds crazy but just come up with your thoughts and
    ideas; I am looking for a subject to my final thesis to write about. :)
    Actually, any inspirations to a subject is highly welcome. :) Thanks!

    --
    Mvh / Best regards,
    Jack, Copenhagen

    The email address is for real. :)
     
    Jack L., Feb 23, 2004
    #1
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  2. Jack L.

    Hank Karl Guest

    There are a lot of issues that can cause VoIP quality problems.
    Packet loss is one; Jitter and delay are others.

    Broadband connections don't always have bandwidth to waste--ADSL may
    be 1.5M down, but only 128K up. If you are using G.711, you may need
    over 90K of the bandwidth for voice (considering the TCP/IP overhead).
    If you choose the worst voice packet size, you will have a total
    message size of 49 bytes. This can be a problem if your link uses ATM
    (like ADSL normally does). ATM requires two 53 byte cells to send a
    49 byte message (ATM uses a 5 byte header and 48 byte payload per
    cell), in which case the ADSL bandwidth utilization just about
    doubles..

    So if you are losing packets because any link is congested, sending
    the same packet four times will be counter-productive.

    Assuming that you can get the four different channels to be routed
    differently, sending the same packet four times over four different
    links will give you delay problems, you would be setting your delay to
    that of the slowest link.

    For more information on VoIP voice quality, see

    http://www.voiptroubleshooter.com

    and http://www.telchemy.com/techref.html

    Regards,
    Hank

    On Tue, 24 Feb 2004 00:51:53 +0100, "Jack L." <>
    wrote:

    >VoIP-developers know the problems VoIP applications suffer during a
    >conversation, eg. delay, echo in the far-end and loss of data packets.
    >Numerous mechanisms have been developed to remove some of them, but in the
    >end, the Internet cannot guarantee data packets to arrive within a specific
    >time which means that a VoIP conversation (currently) won't be as good as
    >with PSTN networks.
    >
    >Would it make sense to research on whether opening X channels to the other
    >side and thereby sending the SAME packet X times at the same time will
    >ensure that data arrives to the destination? We assume that both sides have
    >broadband connection so we have enough bandwidth to "waste". The idea is
    >that we hope the packets will be routed differently so even if one packet
    >disappears on one of the channels, the same packet will still arrive to the
    >destination as the routers (hopefully) let it take another path.
    >
    >The idea probably sounds crazy but just come up with your thoughts and
    >ideas; I am looking for a subject to my final thesis to write about. :)
    >Actually, any inspirations to a subject is highly welcome. :) Thanks!
     
    Hank Karl, Feb 24, 2004
    #2
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  3. Hank Karl <> wrote in
    news::

    > (considering the TCP/IP overhead).


    I think you mean UDP/IP overhead ;-)

    --
    Andreas
     
    Andreas Sikkema, Feb 24, 2004
    #3
  4. Jack L.

    shope Guest

    "Jack L." <> wrote in message
    news:Omw_b.96441$...
    > VoIP-developers know the problems VoIP applications suffer during a
    > conversation, eg. delay, echo in the far-end and loss of data packets.
    > Numerous mechanisms have been developed to remove some of them, but in the
    > end, the Internet cannot guarantee data packets to arrive within a

    specific
    > time which means that a VoIP conversation (currently) won't be as good as
    > with PSTN networks.
    >
    > Would it make sense to research on whether opening X channels to the other
    > side and thereby sending the SAME packet X times at the same time will
    > ensure that data arrives to the destination? We assume that both sides

    have
    > broadband connection so we have enough bandwidth to "waste". The idea is
    > that we hope the packets will be routed differently so even if one packet
    > disappears on one of the channels, the same packet will still arrive to

    the
    > destination as the routers (hopefully) let it take another path.


    the problem is that packets normally arent diverse routed at the places
    where losses are common - over the 1st and last hops and at peering points.

    Even if this "helps" with your individual stream - imagine the effect on a
    Voip based carrier where most of the traffic is voip - you are making 4
    copies of the data.

    You need to think about what you mean by "better" - if more packets have a
    copy arrive, then you have lower loss rate, but what about the other
    characteristics? if your paths are not stable and you are pushing load on
    some links higher then i suspect that jitter is going to get worse not
    better - or maybe better on average, but worse in the worst case. No idea
    what effect that has on voip but i doubt it is good.

    finally - the end point has to sort out the mess. Just using the 1st copy
    packet is probably going to work.
    >
    > The idea probably sounds crazy but just come up with your thoughts and
    > ideas; I am looking for a subject to my final thesis to write about. :)
    > Actually, any inspirations to a subject is highly welcome. :) Thanks!


    you may not know but voip already allows for error correction in the packets
    (cisco phones for example can mask a 30 mSec error) -
    >
    > --
    > Mvh / Best regards,
    > Jack, Copenhagen
    >
    > The email address is for real. :)

    --
    Regards

    Stephen Hope - remove xx from email to reply
     
    shope, Feb 24, 2004
    #4
  5. Good thought. Look into rtp redundancy.

    ---

    ~ VoIP-developers know the problems VoIP applications suffer during a
    ~ conversation, eg. delay, echo in the far-end and loss of data packets.
    ~ Numerous mechanisms have been developed to remove some of them, but in the
    ~ end, the Internet cannot guarantee data packets to arrive within a specific
    ~ time which means that a VoIP conversation (currently) won't be as good as
    ~ with PSTN networks.
    ~
    ~ Would it make sense to research on whether opening X channels to the other
    ~ side and thereby sending the SAME packet X times at the same time will
    ~ ensure that data arrives to the destination? We assume that both sides have
    ~ broadband connection so we have enough bandwidth to "waste". The idea is
    ~ that we hope the packets will be routed differently so even if one packet
    ~ disappears on one of the channels, the same packet will still arrive to the
    ~ destination as the routers (hopefully) let it take another path.
    ~
    ~ The idea probably sounds crazy but just come up with your thoughts and
    ~ ideas; I am looking for a subject to my final thesis to write about. :)
    ~ Actually, any inspirations to a subject is highly welcome. :) Thanks!
     
    Aaron Leonard, Feb 24, 2004
    #5
  6. Jack L.

    Jack L. Guest

    Jack L. wrote:

    > The idea probably sounds crazy but just come up with your thoughts and
    > ideas; I am looking for a subject to my final thesis to write about.
    > :) Actually, any inspirations to a subject is highly welcome. :)
    > Thanks!


    Thank you so much for sharing your thoughts on this idea - the arguments
    sound convincing enough not to put further effort on it.

    --
    Mvh / Best regards,
    Jack, Copenhagen

    The email address is for real. :)
     
    Jack L., Feb 24, 2004
    #6
  7. Jack L.

    Hank Karl Guest

    On 24 Feb 2004 15:23:08 GMT, Andreas Sikkema <> wrote:

    >Hank Karl <> wrote in
    >news::
    >
    >> (considering the TCP/IP overhead).

    >
    >I think you mean UDP/IP overhead ;-)


    Ok, I slipped. But there is also RTP, RTCP, and possibly PPPoEoA to
    consider.
     
    Hank Karl, Feb 26, 2004
    #7
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