[SIPForum-discussion] Jitter measurement.
HM Kias
hmkias at gmail.com
Sat Mar 16 09:04:11 UTC 2013
Hi,
It's a 32-bit field indicates in relative terms the time when the payload
was sampled. This field allows the receiver to remove jitter and to play
back the packets at the right interval assuming sufficient buffering.
Regards,
On Tue, Aug 16, 2011 at 4:00 PM, Badri Ranganathan <badri at arcatech.com>wrote:
> Hi,
>
> Can anyone tell me how "jitter" is "measured" ?
>
> >From what I could gather from the internet -
>
> <<=========================
>
> Jitter is defined as a statistical variance of the RTP data packet
> inter-arrival time. In the Real Time Protocol, jitter is measured in
> timestamp units. For example, if you transmit audio sampled at the usual
> 8000 Hertz, the unit is 1/8000 of a second.
>
>
> In RFC1889, I can see this definition under RTCP sender reports -
>
> interarrival jitter: 32 bits
>
> An estimate of the statistical variance of the RTP data packet
> interarrival time, measured in timestamp units and expressed as an unsigned
> integer. The interarrival jitter J is defined to be the mean deviation
> (smoothed absolute value) of the difference D in packet spacing at the
> receiver compared to the sender for a pair of packets. As shown in the
> equation below, this is equivalent to the difference in the "relative
> transit time" for the two packets; the relative transit time is the
> difference between a packet's RTP timestamp and the receiver's clock at the
> time of arrival, measured in the same units.
>
> ===========================>>
>
>
> Here it says its measured in timestamp units. Cant understand this
> explanation much. Why cant it be expressed in milliseconds ?
>
> Thanks,
> Badri.
>
>
> -----Original Message-----
> From: discussion-bounces at sipforum.org [mailto:
> discussion-bounces at sipforum.org] On Behalf Of Steve Underwood
> Sent: 13 August 2011 08:58
> To: Rohan Almeida
> Cc: discussion at sipforum.org
> Subject: Re: [SIPForum-discussion] SIP Faxing
>
> A key question is 15% of what kind of calls fail? If you are talking
> about a public FAX server, open to all, you might well see 15% to 20% of
> failed calls from wrong numbers, voice calls to FAX numbers, etc. If you
> are making test calls into a well controlled server and get 15%
> failures, that pretty nasty. You should be getting less than 1%
> failures, even if the calls are sending tens of pages each.
>
> Steve
>
>
> On 08/13/2011 01:14 PM, Rohan Almeida wrote:
> > 15% failure is too high. 2-10 % is on an average is acceptable.
> > but again the fault might not be only in your system. the interconnect
> > devices might also be the cause.
> >
> > On Tue, Aug 9, 2011 at 9:52 AM, Greg Settle <gsettle at opentext.com
> > <mailto:gsettle at opentext.com>> wrote:
> >
> > What is the topology?
> >
> > SIP trunk - SBC/gateway - fax server? If yes, who is the SIP trunk
> > provider? Do they support T.38 FoIP?
> > T1/PRI - gateway - fax server? If yes, which gateway(s) is involved?
> > Support for T.38?
> >
> > Assuming a fax server is involved? If yes, which fax server?
> >
> > Generally, if T.38 FoIP is supported among all endpoints, and proper
> > testing was done to verify configuration among the components, SIP
> > faxing failure rates should remain low. 15% failure rate is high,
> and
> > points to weak links in the topology where FoIP standards (T.38)
> > are not
> > being met. Attached is a document which may assist.
> > Greg
> >
> > -----Original Message-----
> > From: discussion-bounces at sipforum.org
> > <mailto:discussion-bounces at sipforum.org>
> > [mailto:discussion-bounces at sipforum.org
> > <mailto:discussion-bounces at sipforum.org>] On Behalf Of Steve
> Underwood
> > Sent: Monday, August 08, 2011 9 <tel:2011%209>:47 AM
> > To: discussion at sipforum.org <mailto:discussion at sipforum.org>
> > Subject: Re: [SIPForum-discussion] SIP Faxing
> >
> > On 08/04/2011 09:40 PM, Melissa Parsons wrote:
> > >
> > > Hello,
> > >
> > > Can anyone tell me what an acceptable failure rate would be when
> > > faxing over SIP? Currently we are averaging around 15%. This
> > number is
> >
> > > high I would expect it to be somewhat higher than traditional
> faxing
> > > but not this high.
> > >
> > > Thank you,
> > >
> > > *Melissa Parsons*|Enterprise Systems Engineer
> > >
> > > *MarineMax, Inc.* ( 727.531.1700 <tel:%28%20727.531.1700> office
> > | (727.524-3954 <tel:%28727.524-3954> fax
> > >
> > > 18167 US Hwy. 19 N. Clearwater, Florida 33764 <tel:33764>
> > >
> > >
> > That's vague. Are you using A-law or u-law? Are you using G.726?
> > Are you
> >
> > using T.38? Pretty much anything else you might use will give 0
> > success
> > rate, so I assume its one of those. The failure rate you get will
> > depend
> >
> > on many factors. A high packet loss rate is a disaster for FAX. High
> > jitter levels for the packet delivery time can be too. T.38
> > implementations are quite variable in their behaviour, and not always
> > that compatible. Many quirks in the SIP signalling arrangements
> > exist in
> >
> > various implementations, too. At the end of the day you'll get
> > somewhere
> >
> > between 0 and 100% success, depending on all these factors. Between
> > servers in large data centres, connected straight on to the
> internet's
> > backbone you might be able to send FAXes all week without an error.
> In
> > really bad tributaries of the internet you might get a near 100%
> > failure
> >
> > rate.
> >
> > Regards,
> > Steve
> > _______________________________________________
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> >
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> >
> >
>
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--
HM Kias
91-9443467600
Fonicom
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