Velocity Reviews > VOIP > jitter and playout delay computations!!!

# jitter and playout delay computations!!!

John
Guest
Posts: n/a

 12-19-2005
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

Wolfgang S. Rupprecht
Guest
Posts: n/a

 12-19-2005

"John" <(E-Mail Removed)> 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
Guest
Posts: n/a

 12-20-2005
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" <(E-Mail Removed)> 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
Guest
Posts: n/a

 12-20-2005

"John" <(E-Mail Removed)> 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