jitter and playout delay computations!!!

Discussion in 'VOIP' started by John, Dec 19, 2005.

  1. John

    John Guest

    Hi,
    I had posted a query lastweek regarding jitter computation and
    playout delay.

    since, no one responded to it i am posting it again:

    We have the interarrival jitter computation:

    int transit = arrival - r->ts;
    int d = transit - s->transit;
    s->transit = transit;
    if (d < 0) d = -d;
    s->jitter += (1./16.) * ((double)d -
    s->jitter);

    how do we compute the jitter buffer required (num of packets to be
    buffered before we begin to playout)

    how playout delay and num of packets to buffer (size of jitter buffer)
    is computed from s-> jitter(inter arrival jitter).



    Regards,
    John
     
    John, Dec 19, 2005
    #1
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  2. "John" <> writes:
    > I had posted a query lastweek regarding jitter computation and
    > playout delay. since, no one responded to it i am posting it again:
    > We have the interarrival jitter computation:
    >
    > int transit = arrival - r->ts;
    > int d = transit - s->transit;
    > s->transit = transit;
    > if (d < 0) d = -d;
    > s->jitter += (1./16.) * ((double)d -
    > s->jitter);
    >
    > how do we compute the jitter buffer required (num of packets to be
    > buffered before we begin to playout) how playout delay and num of
    > packets to buffer (size of jitter buffer) is computed from s->
    > jitter(inter arrival jitter).


    What class is this for and what are you offering if we do your
    homework for you? ;-)

    -wolfgang
    --
    Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
    Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
     
    Wolfgang S. Rupprecht, Dec 19, 2005
    #2
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  3. John

    John Guest

    Well, what can i offer you expect, that i will be greatful to u and i
    promise that if anybody needs
    to know something which i already knew i will share my knowledge i have
    gained from all of you.

    i am not a student. i am looking for work and am trying to understand
    rtp so that it could enhance
    my prospects.


    Regards,
    James
    Wolfgang S. Rupprecht wrote:
    > "John" <> writes:
    > > I had posted a query lastweek regarding jitter computation and
    > > playout delay. since, no one responded to it i am posting it again:
    > > We have the interarrival jitter computation:
    > >
    > > int transit = arrival - r->ts;
    > > int d = transit - s->transit;
    > > s->transit = transit;
    > > if (d < 0) d = -d;
    > > s->jitter += (1./16.) * ((double)d -
    > > s->jitter);
    > >
    > > how do we compute the jitter buffer required (num of packets to be
    > > buffered before we begin to playout) how playout delay and num of
    > > packets to buffer (size of jitter buffer) is computed from s->
    > > jitter(inter arrival jitter).

    >
    > What class is this for and what are you offering if we do your
    > homework for you? ;-)
    >
    > -wolfgang
    > --
    > Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
    > Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
     
    John, Dec 20, 2005
    #3
  4. "John" <> writes:
    > Well, what can i offer you expect, that i will be greatful to u and i
    > promise that if anybody needs
    > to know something which i already knew i will share my knowledge i have
    > gained from all of you.
    >
    > i am not a student. i am looking for work and am trying to understand
    > rtp so that it could enhance
    > my prospects.


    Sorry for doubting. The lack of verbiage and detail made it sound a
    bit like a homework problem.

    When you posted it the first time, I did wonder what google had on it
    so I poked around a bit. Now I may have gotten things a bit wrong,
    but it sure looked to me like the units were dimensionless (not time
    at all) but simply the audio sample count. Typical 20ms rtp packets
    had a value of 160 samples and the "timestamp" would increment by 160.
    (I take it at some point in the past that timestamp was truly a time
    value, but as people used different sample rates it was easier to just
    count samples directly.)

    (* 8000 ; samples per second for alaw/ulaw
    20e-3) ; 20ms rtp packet
    160.0 ; samples/packet

    It might be interesting to grab a copy of some open source voip
    software (kdephone, linphone) and observe it in action (say by adding
    printf's for logging the RTP headers). Both of those programs are
    still fairly rough around the edges and an interested party tweaking
    things and fixing bugs could still make a significant contribution.

    -wolfgang
    --
    Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
    Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
     
    Wolfgang S. Rupprecht, Dec 20, 2005
    #4
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