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MULTIMEDIA NETWORKING

Jay Gholap (110803061)



What is Multimedia ?
Multimedia is an integration of text, graphics,
still and moving images, animation, sounds,
and any other medium where every type of
information can be represented, stored,
transmitted and processed digitally
Technologies of Multimedia Networking
Media compression – better to reduce the data volume before
transmission
Image compression
Video compression
Audio compression

Multimedia transmission technology


Protocols for real-time transmission
Rate / congestion control
Error control
Classes of MM applications:
1) Streaming stored audio and video
2) Streaming live audio and video
3) Real-time interactive audio and video
Streaming Stored Multimedia
Stored streaming:

media stored at source

Transmitted to client

Streaming: client playout begins before all data has arrived.

Gives VCR functionalities: fast fwd,rewind ,pause etc.
Downloading a file
1st approach:using a web server with a metafile
Downloading a file
2nd approach:using a media server
Streaming Live Multimedia
Examples:

Internet radio talk show

live sporting event e.g. Live streaming
match.
playback buffer

playback can lag tens of seconds after
transmission.

still have timing constraint
Interactivity

fast forward impossible

rewind, pause possible!
Real time interactive multimedia

Applications:

IP telephony

Video conference
allows people to communicate
visually and orally
End-End delay requirements :

audio: < 150 msec good, < 400 msec OK

includes application-level (packetization) and network delays

Higher delays are noticeable.
Protocols

HTTP GET
Web Web
browser presentation desc. server

RTSP
media
player RTCP media
server

RTP

client server

RTSP: Real-Time Signaling Protocol


RTCP: Real-Time Control Protocol
RTP: Real-Time Transport Protocol


They provide:
-Decompression
- Jitter Removal
-Error Correction
RTSP operation

Setup:A client can issue a SETUP request , so


server allocates resources to stream and starts a
RTSP session.
Play: The PLAY method tells the server to start
sending data via the mechanism specified in
SETUP.
Pause: The PAUSE request causes the stream
delivery to be interrupted (halted) temporarily
Teardown: Frees resources associated with
stream. The RTSP session ends.
RTSP client FSM

state message_sent next_state_ after_response


Init SETUP Ready
TEARDOWN Init
Ready PLAY Playing
RECORD Recording
TEARDOWN Init
SETUP Ready
Playing PAUSE Ready
TEARDOWN Init
PLAY Playing
SETUP Playing (changed transport)
Recording PAUSE Ready
TEARDOWN Init
RECORD Recording
SETUP Recording (changed transport)
RTSP server FSM

state message received next state


Init SETUP Ready
TEARDOWN Init
Ready PLAY Playing
SETUP Ready
TEARDOWN Init
RECORD Recording
Playing PLAY Playing
PAUSE Ready
TEARDOWN Init
SETUP Playing
Recording RECORD Recording
PAUSE Ready
TEARDOWN Init
SETUP Recording
Real Time Transport protocol (RTP)


RTP is protocol designed to handle real time traffic on the
internet.

It doesnt have a delivery mechanism , so it must be used with
UDP.

RTP specifies packet structure for packets carrying audio, video
data .
RTP libraries provide transport-layer interface
•that extends UDP:
• payload type identification
• packet sequence numbering
• Time-stamping
Position of RTP
RTCP
• RTCP is a control protocol that works in
conjunction with RTP
• RTCP uses odd numbered port UDP port that
follows port number selected for RTP.
• Provides useful statistics: packets sent, lost,
jitter, round-trip time
• Sources can use this to adjust their data rate
• Other information includes email address, phone
number, name – allow users to know the
identities of other users in the session
Problems for multimedia


Packet Jitter
- variation in packet delays.

Dropped Packets
-if there is congestion in network .i.e.if no of packets sent to
network is greater than capacity of network.
-Some of the packets are lost , so reception is not good .
-Sometimes errors are introduced causing corrupted packets.
Streaming Multimedia : Delay jitter

constant bit
rate video client video constant bit
Cumulative data

transmission reception rate video


playout at client
variable
network

buffered
video
delay
jitter

client playout time


delay
Recovery from Packet loss

FEC technique :
• “piggyback lower
quality stream”

• Example: send lower


resolution audio stream as
the redundant
information

 whenever there is non-consecutive loss,


receiver can conceal the loss.
Interleaving:Recovery from Packet loss

Interleaving

Intentionally alter the sequence of packets before transmission

Now, packet contains small units from different chunks

If packet is lost,still have most of every chunk.

Better against “burst” losses of packets

Results in increased playout delay from inter-leaving
Techniques to improve quality of service

 FIFO

 Priority Queue
Techniques to improve quality of service


FiFO queue holds the
packets, if traffic consists of
fixed size packets, the
process removes fixed no of
packets from queue at
constant rate.

A leaky bucket algorithm
shapes bursty traffic into
fixed-rate traffic by averaging
data rate .

It may drop packets if queue
is full.
Thank you!

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