L.R.He, B.M.G. Cheetham

Mobile Systems Architecture Group, Department of Computer Science, University of Manchester, Oxford Rd, M13 9PL, U.K.

•Application of real time multimedia communications to distance learning over IP networks. .

• Current RTP systems are based on comparing frame loss rate, as reported by RTCP, with thresholds. • We presents a flow/congestion control mechanism for adapting to congestion using measurements of time delay, jitter & speech packet loss rate.

•We propose a dynamic assignment of priority to the speech, image & data as appropriate to distance learning activity.

• To improve the flow/congestion control mechanism based on RTP and RTCP

•To allow dynamic priority assignment determining how the available capacity is divided between speech / image / data.

Distance learning is applied in these fields:

* Providing open learning environments

* Offering more information for traditional teaching

* Providing continuing education after graduation

* Developing academic co-operation

Three types of distance education

-Paper based learning

-TV and radio based learning

-Internet based learning

Real Time Multimedia has the following advantages for Distance Learning :

* Liveliness

* Efficiency

* Interactivity

Internet

Browser (student 1)

Browser (student N)

Browser(teacher 1)

Browser(teacher M)

Web server

Database (Course Information, Content, Assignments, Problems and Solutions, Audio-video Information, Examinations, Announcements, Student Records)

Camera &

microphone

Data

RTCP sender report

Internet

Receiver 1

Receiver 2

RTCP Receiver Report

Check result

Most of the real-time applications are based on the end-to-end real time transport protocol (RTP) and real time control protocol (RTCP).

Real time transport protocol is used for end-to end data transfer. RTP transports real-time media data along with synchronization information over a datagram protocol.

G.711,G.722,G.723.1,G.728,G.729

H.261, H.263

RTP

RTCP

User Datagram Protocol

Network Layer

Link Layer

Physical Layer

Real-Time control Protocol has a feedback function. Feedback from the receivers is necessary for diagnosing distribution faults.

RTCP produces sender and receiver reports.

• Stream statistics

• Packet counts

• Sender identification

• Quality of service

• Lost packets

Packet loss

probability

Loss rate(%)

100%

c

0

Congested --- Decrease

Unloaded --- Increase

Loaded --- Hold

Receiver classifies loss error

Bit-rateNetwork

state

u

Problems lie in dynamics of bit-rate adaptation

n

Packet loss probability

Speech packet unusable

probabilityAveragejitter

“loaded”Management

Priority weighting

“unloaded”

“unloaded”

“congested”

“congested”

high priority weighting or low priority weighting

1 Determination of the network states

As before, the network will be defined to have three different states, “congested”, “loaded”, and “unloaded”.

If 1 λn λc or (λn λu and λsn >λsc), the network state is “congested”.

λsc is a threshold.

If λn λu, then the network is “unloaded”.

If (λn λu and λsn λsc), the network is “loaded”.

The probability of a received speech packet being unusable λsn as follows:

nth control period

T (400Pd) ms

Pd is the processing delay.

• If the network state is “congested”, H = H – H.

• If the network state is “unloaded”, H = H + H.

• If the network state is “loaded”, the sender hold the image and other packets transmission rate, checks the average jitter, Jn, to decide how to adjust the speech

packet transmission rate.

2 Management

• The priority of a given packet

Average Jitter Speech transmission rate

Decreasing Hs +

Constant Hs

Increasing Hs * Hsmin Hs Hsmax

Table: Determination of the requested sending rate of speech packet

Fig.4 Loss probability

0

20

40

60

80

100

0 100 200 300

Time (second)

Lo

ss p

rob

abili

ty (

%)

Packet loss probability

Fig.5 Speech packet transmission rate

20

30

40

50

60

40 50 60 70

Time (sec)

Tra

nsm

issio

n

rate

(kb

/s)

Previous method New method

Fig.6 Speech transmission rate

45

50

55

60

245 250 255 260 265 270 275

Time (sec)

Tra

nsm

issio

n

rate

(kb

/s)

Previous method New method

• A flow/congestion control mechanism based on RTP and RTCP has been investigated.

• A more appropriate variation in transmission rate is achieved by the new method.

• It can be appropriately applied to distance learning over internet.