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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
Database (Course Information, Content, Assignments,
Problems and Solutions, Audio-video Information,
Examinations, Announcements, Student Records)
Web server
Internet
Browser
(student 1)
Browser
(student N)
Browser
(teacher 1)
Browser
(teacher M)
Real Time Multimedia has the following advantages for Distance Learning :
* Liveliness
* Efficiency
* Interactivity
Most of the real-time applications are based on the end-to-end real time
transport protocol (RTP) and real time control protocol (RTCP).
RTCP sender report
Camera &
Receiver 1
Data
microphone
Receiver 2
Check
result
Internet
RTCP Receiver Report
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
User Datagram Protocol
Network Layer
Link Layer
Physical Layer
RTCP
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
Receiver classifies loss error
Loss rate(%)
Network
Bit-rate
state
100%
Packet loss
--- Decrease
Loaded
--- Hold
Unloaded
--- Increase
c
probability
n
Congested
u
0
Problems lie in dynamics of bit-rate adaptation
Priority
weighting
“congested”
high priority weighting
or low priority weighting
“congested”
Packet loss
probability
Speech packet
unusable
probability
“loaded”
Average
jitter
Management
“unloaded”
“unloaded”
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  (400Pd) ms
Pd is the processing delay.
2 Management
• The priority of a given packet
• 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.
Average Jitter
Speech transmission rate
Decreasing
Hs + 
Constant
Hs
Increasing
Hs  
* Hsmin  Hs    Hsmax
Table: Determination of the requested sending rate of speech packet
Transmission
rate (kb/s)
Fig.5 Speech packet transm ission rate
100
50
40
30
20
40
50
80
60
70
Tim e (sec)
60
40
Previ ous method
New method
20
0
0
100
200
300
Tim e (second)
Fig.6 Speech transm ission rate
Packet loss probability
Transmission
rate (kb/s)
Loss probability (%)
Fig.4 Loss probability
60
60
55
50
45
245
250
255 260
265 270
275
Tim e (sec)
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.