Simple static Jitter buffer dimensioning

Discussion in 'VOIP' started by Breezer, Feb 23, 2006.

  1. Breezer

    Breezer Guest

    Hi all, how can I choose the size (in frames or packets, provided they are
    always of the same size) of a simple static jitter buffer implemented as a
    ring buffer that can overflow or underflow (so, no menagement of timestamps)
    if I know the variance of the interarrival times?
    The interarrival are very correlated because there are a lot of bursts ( a
    late packet and the subsequents all back to back).
    If the distribution was normal and the interarrivals independents, I could
    have said that taking three times the standard deviation I catch the 99% of
    them...

    Thanks in advance

    Breezer
     
    Breezer, Feb 23, 2006
    #1
    1. Advertising

  2. "Breezer" <> writes:
    > Hi all, how can I choose the size (in frames or packets, provided they are
    > always of the same size) of a simple static jitter buffer implemented as a
    > ring buffer that can overflow or underflow (so, no menagement of timestamps)
    > if I know the variance of the interarrival times?
    > The interarrival are very correlated because there are a lot of bursts ( a
    > late packet and the subsequents all back to back).
    > If the distribution was normal and the interarrivals independents, I could
    > have said that taking three times the standard deviation I catch the 99% of
    > them...


    The fact that the delay on each packet isn't independent of the delay
    on previous and subsequent packets kills most of the assumptions you
    need in order to extrapolate to a Gaussian curve. At least on the
    small time-scale. I bet if you only did random sampling (say looked
    once every 10 seconds or once every 100 seconds then things would look
    more Gausian.

    There are many systems where sampling at too high a rate give you too
    much high frequency crap (which can seriously change your observed
    curve shape). For a similar problem that ntp (the network time
    protocol) has to deal with, Google for "allan variance".

    From a practical voip standpoint, I'm not sure what is gained by
    shooting for an arbitrarily chosen "benchmark" number like 99%. What
    you really need to optimize for user annoyance. You need to find the
    point in the delay / packet loss curve where the users total annoyance
    from highly delayed speech is balanced with the raspy sound from lost
    packets. Hopefully this won't be a common occupance but when it is,
    I'd hope the equipment has some reasonable *sounding* trade-off's that
    are set via real user tests.

    -wolfgang
    --
    Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
    Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
     
    Wolfgang S. Rupprecht, Feb 23, 2006
    #2
    1. Advertising

Want to reply to this thread or ask your own question?

It takes just 2 minutes to sign up (and it's free!). Just click the sign up button to choose a username and then you can ask your own questions on the forum.
Similar Threads
  1. Dmitry Melekhov

    smokeping, SAA, jitter

    Dmitry Melekhov, Nov 25, 2003, in forum: Cisco
    Replies:
    2
    Views:
    1,963
    Dmitry Melekhov
    Dec 5, 2003
  2. Dmitry Melekhov

    SAA, jitter, strange values

    Dmitry Melekhov, Dec 12, 2003, in forum: Cisco
    Replies:
    0
    Views:
    498
    Dmitry Melekhov
    Dec 12, 2003
  3. Colin chaplin
    Replies:
    0
    Views:
    680
    Colin chaplin
    Jun 3, 2004
  4. Jungleboy

    Jitter Buffer Calculation

    Jungleboy, Nov 25, 2004, in forum: VOIP
    Replies:
    1
    Views:
    6,395
    Daniel Rakel
    Nov 25, 2004
  5. Stephen Regan

    Convert RTCP Jitter to milliseconds

    Stephen Regan, Apr 7, 2005, in forum: VOIP
    Replies:
    0
    Views:
    1,446
    Stephen Regan
    Apr 7, 2005
Loading...

Share This Page