Thread View: comp.multimedia
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Started by Wilko Reinhardt
Wed, 30 Sep 1992 14:46
Comminication requirements of MM applications
Author: Wilko Reinhardt
Date: Wed, 30 Sep 1992 14:46
Date: Wed, 30 Sep 1992 14:46
274 lines
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12102 bytes
Hello everybody:
We are working on the project EuroBridge founded by European
commission. The aim of EuroBridge is to find flexible
communication structures for distributed Multimedia
applications. The problem we have is, that at the moment we
haven't enough experience on the communication requirements of
Multimedia application. That means we are looking for exact
values of various performance parameters.
If you are involved in developing distributed multimedia
applications or you have knowledge on this item, please take
five minutes time fill the questionnaire below and mail it to
Wilko Reinhardt, Lehrstuhl f. Informatik, RWTH Aachen, Ahornstr.
55
FAX: +49-241-80-21429, phone: -37,
email: wilko@informatik.rwth-aachen.de
At the end of the questionnaire you will find an explanation of
all items. Many thanks in advance for your co-operation!
Wilko Reinhardt (RWTH Aachen, Germany)
--------------------------------------------
Questionnaire
Multimedia Application:
...........................................................
(Please give us a short description of your multimedia system)
Performance QoS
Video
| range:(max. / av. / min) | not relevant
| |
jitter | |
| |
delay - one way | |
| |
throughput (steady) | |
| |
throughput (bursty) | |
| |
connection set up time| |
| |
response time | |
| |
max. acceptable error rate| |
for the application | |
(without error contr. | |
mechanism) | |
| |
| |
others? | |
| |
| |
| |
Performance QoS
Audio | range:(max. / av. / min) | not
relevant
| |
jitter | |
| |
delay - one way | |
| |
throughput (steady) | |
| |
throughput (bursty) | |
| |
connection set up time| |
| |
response time | |
| |
max. acceptable error rate| |
for the application | |
(without error contr. | |
mechanism) | |
| |
| |
others? | |
| |
| |
| |
Performance QoS
Data
| range:(max. / av. / min) | not relevant
| |
jitter | |
| |
delay - one way | |
| |
throughput (steady) | |
| |
throughput (bursty) | |
| |
connection set up time| |
| |
response time | |
| |
max. acceptable error rate| |
for the application | |
(without error contr. | |
mechanism) | |
| |
| |
others? | |
| |
| |
| |
Functional QoS
| relevant | not relevant
| |
QoS re-negotiation | |
| |
security mechanisms | |
| |
- encryption | |
| |
- password/id | |
| |
resource reservation | |
| |
multipoint connections | |
| |
data flow synchronisation | |
| |
Error correction scheme | |
(ARQ / FEC / none) | |
Channel Prioritisation | |
| |
others? | |
| |
| |
| |
Appendix: Explanation of terms
Jitter
However due to statistical queuing delays and network errors
(lost or corrupted packets) there are delay variations that we
call jitter. An additional source of jitter could be the system
clocks at the source and sink running at different rates, the
variation in quartz oscillators is small but at the high speeds
required for videoconferencing this may be a factor.
Jitter may also be caused by unnecessary multiplexing in a
layered protocol stack.
Delay
The transmission delay should be measured as an end to end
metric, i.e. measured from the point where the sample is
generated to the point at which it is presented to the user.
The total delay consists of the packetisation delay, the network
transmission delay and the presentation delay. The packetisation
delay is the time taken to generate a sample and deliver it to
the network, the network transmission delay is the time taken to
transmit the sample over the network, and the presentation delay
is the time spent buffering and processing the sample be-fore
presenting it to the user.
Throughput
The throughput of a channel is the amount of data that can be
transmitted through that chan-nel in a unit of time (bits per
second usually). Research has identified a need to extend the
definition of throughput to contain some measure to account for
burstiness.
Connection time or set-up time
The time taken to set up a channel or virtual circuit should be
as short as possible and not re-quire a large amount of network
resources since this will be a common event on networks
supporting videoconferencing applications. During channel set-up
the service requirements for that association will need to be
met or negotiated.
Response Time
The response describes the time between submission of a message
and receipt of the remote system's reaction. This time indicates
how long a user has to wait for answers. The value may vary from
some ms (e.g. interactive work with the remote system) to a few
minutes (e.g. non urgent file retrieve).
Error Rate
The reliability of a channel depends on what type of information
it contains. Both video and audio can tolerate a certain amount
of loss of information. Audio can tolerate a 1% loss as long as
each error burst is less than 4ms. The acceptable error loss
for video is dependant on the encoding and compression schemes
used. Data can not tolerate any loss. Research has indicated
that the residual error rate should be revised to distinguish
between bit and packet errors as video and audio are affected
differently depending on the type of error.
QoS Parameter Renegotiation
As the network resources may change during the lifetime of an
association a higher level re-source manager may wish to change
the associations characteristics in mid connection. The
application may also request that the characteristics change,
e.g. in a video surveillance appli-cation the guard may have
numerous slow scan windows until he sees something that he
wishes to investigate. He then may wish to increase the
resolution and frame rate for a better view.
Security Mechanisms
Your application may asked for a secure transfer of confidential
data (e.g. medical diagnos-tics, financial information, love
letters, etc.). The communication system can provide several
security mechanisms like encryption or user identification by
passwords or user id's.
Resource Reservation
This communication feature provides the reservation of
communication resources (e.g. bandwidth) in advance. The user is
able to reserve a channel which meets his QoS demands. At the
scheduled time the communication system will guarantee the
required QoS.
Data Flow Synchronisation
Three types of synchronicity have been identified :
Asynchronous: has unspecified delay, which implies unspecified
jitter
Synchronous: has an upper bound on delay, which implies a coarse
bound on jitter.
Isochronous: has a specified delay and delay jitter.
For videoconferencing applications we require isochronous
synchronisation, where this is the synchronisation of streams in
real-time.
Error Correction
For the case of video and audio in a real time video conference
retransmitting a lost packet is not generally considered
applicable. The desired quality of service is usually reached
by ne-gotiating a guarantied reliability. For the case of data
there must be some mechanism to re-trieve lost or corrupted
data.
There are two possible methods that would be applicable,
Automatic Repeat Request (ARQ) and Forward Error Correction
(FEC).
In order for ARQ to work efficiently, the receiver must have
enough buffer space to buffer all packets from the lost/damaged
one until the retransmitted packet has reached the receiver.
ARQ requires a duplex connection.
FEC includes redundant information in the transmission to enable
retrieval of corrupted or lost data at the sink, if necessary.
FEC is applied to all packets transmitted and the overhead is
known beforehand. FEC calculations are complex and may need to
be implemented in hardware to meet delay and jitter limits.
Channel Priorities
Research has identified a need to be able to assign priorities
to channels so the transport sys-tem can be aware of the
relevant
importance of the various channels.
Re: Comminication requirements of MM applications
Author: Magnus Lindgren
Date: Thu, 01 Oct 1992 09:53
Date: Thu, 01 Oct 1992 09:53
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Could you please post a summary of the result of your survey when it's done? I think a lot of people will be interested in that. (I will at least) Thanks. Magnus Lindgren Phone: +46 920 75425 Telia Research Telefax: +46 920 75490 Aurorum 6 E-mail: jml@lulea.trab.se S-951 75 Lulea, Sweden
Re: Comminication requirements of MM applications
Author: Julian Lau
Date: Fri, 02 Oct 1992 02:09
Date: Fri, 02 Oct 1992 02:09
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Greetings all, I'm doing some personal research into multi-media and was wondering if theres any courses out there that can suggest some good de-facto reference text. Please email personally to me: laucy@newton.ccs.tuns.ca Thanks much and best regards. Julian.
Re: Comminication requirements of MM applications
Author: Konrad Froitzheim
Date: Wed, 07 Oct 1992 11:28
Date: Wed, 07 Oct 1992 11:28
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In article <1992Sep30.144650.7973@Urmel.Informatik.RWTH-Aachen.DE>, Wilko Reinhardt <wilko@informatik.rwth-aachen.de> wrote: > > > Hello everybody: > > ... > > If you are involved in developing distributed multimedia > applications or you have knowledge on this item, please take > five minutes time fill the questionnaire below and mail it to > > Wilko Reinhardt, Lehrstuhl f. Informatik, RWTH Aachen, Ahornstr. > 55 > FAX: +49-241-80-21429, phone: -37, > > email: wilko@informatik.rwth-aachen.de > > At the end of the questionnaire you will find an explanation of > all items. Many thanks in advance for your co-operation! > > Wilko Reinhardt (RWTH Aachen, Germany) Did you check out our document: 'Requirements of Isochronous Audio/Data/Video Service on the Communication Platform'. This is deliverable 10b by the RACE II project CIO (R2060). You can get a copy from the RACE central office: CEC DG XIII Telecommunications Directorate F TR61 2/2 Rue de la Loi 200 B-1049 Brussels Belgium ********************************************************************* * Dr. Konrad Froitzheim * * University of Ulm, (Germany) Dept. of Distributed Systems * * Internet: frz@informatik.uni-ulm.de * * Fax: ++ 49 731 502 4142 * *********************************************************************
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