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B.Bharat Shetty
4th semester CS&E
SJCE
DEFNITION: Multimedia is often described as a
holy grail by some people.Literally the term
multimedia is just two or more media.To be precise
“Multimedia” generally means the combination of
two or more continuous media i.e. media that has to
be played during some well defined interval usually
with some user defined interaction.
Before Proceeding concept of a protocol must be clear to you
Multimedia Networking Applications
 Streaming stored audio and video
RTSP
 Protocols for Real-Time Interactive Applications
RTP
RTCP
SIP.
 RSVP
 H.323 standard
 Summary
MM Networking Applications
Classes of MM applications: Fundamental
characteristics:
1) streaming stored audio
• Typically delay
and video
sensitive.
2) streaming live audio and
• End-to-end delay.
• Delay jitter.
video
• But loss tolerant:
3) real-time interactive
infrequent losses cause
audio and video
minor glitches.
• Antithesis of data,
which are loss
Jitter -the variability
intolerant but delay
of packet delays within
tolerant.
the same packet stream
Multimedia, Quality of Service: What is it?
Multimedia applications:
network audio and video
(“continuous media”)
QoS
level of performance
needed for application to
function.
Streaming Stored Multimedia
• Streaming:
• media stored at source
• transmitted to client
• streaming: client playout
begins before all data has
arrived
• timing constraint for still-tobe transmitted data: in time
for playout
SSM: What Is It?
1. video
recorded
2. video
sent
network
delay
3. video received,
played out at client
streaming: at this time, client
playing out early part of video,
while server still sending later
part of video
time
Streaming Stored Multimedia:
Interactivity
functionality similar to a VCR:
client can pause, rewind, FF, push
slider bar
• 10 sec initial delay causes no
problem
• 1-2 sec until command effect
OK
• RTSP often used
• timing constraint for still-to-be
transmitted data: in time for playout
Streaming Live Multimedia
Examples:
• Talk shows over the web.
• Events brought live to home.
Streaming.
• Playback buffer.
• Playback can lag tens of seconds after
transmission.
• Still have timing constraint.
Interactivity.
• Fast forward impossible.
• Rewind, pause possible!
Video
servers
switch
Customer’s
House
ATM or SONET
Backbone network
Local spooling server
LDN-local distribution network
LDN*
User Control of Streaming Media:
RTSP
RTSP:
• Client-server application layer protocol.
• User can control display: rewind, fast forward, pause,
resume, repositioning, etc…
Disadvantages:
•
•
•
Does not define how audio/video is encapsulated for streaming
over network.
No restrictions on how sm is transported. It can be
transported over UDP or TCP.
No specifications on how the media player buffers
audio/video.
Real-time Protocol (RTP)
RTP specifies a packet structure for packets
carrying audio and video data.
RFC 1889.
RTP packet provides.
Payload type identification.
Packet sequence numbering.
Timestamping.
RTP runs in the end systems.
RTP packets are encapsulated in UDP segments.
Interoperability: if two internet phone
applications run RTP, then they may be able to
work together.
RTP Runs on Top of UDP
RTP libraries provide a transport-layer
interface that extend UDP:
• port numbers, IP addresses
• payload type identification
• packet sequence numbering
• time-stamping
RTP Header
Payload Type (7 bits): Indicates type of encoding currently being
used. If sender changes encoding in middle of conference, sender
informs the receiver through this payload type field.
•Payload type 0: PCM mu-law, 64 kbps
•Payload type 3, GSM, 13 kbps
•Payload type 7, LPC, 2.4 kbps
•Payload type 26, Motion JPEG
•Payload type 31. H.261
•Payload type 33, MPEG2 video
Sequence Number (16 bits): Increments by one for each RTP packet
sent, and may be used to detect packet loss and to restore packet
sequence.
Real-time Control Protocol (RTCP)
Works in conjunction with RTP.
Each participant in RTP session periodically
transmits RTCP control packets to all other
participants.
Each RTCP packet contains sender and/or
receiver reports.
Key things:
-Number of packets sent.
-Number of packets lost.
-Inter arrival jitter.
RTCP - Continued
- RTP session typically has a single multicast address; all RTP
and RTCP packets belonging to the session use the multicast address.
- RTP and RTCP packets are distinguished from each other through the use of
distinct port numbers.
- To limit traffic, each participant reduces his RTCP traffic as the number
of conference participants increases.
RTCP Packets
Receiver report packets:
Fraction of packets lost, last sequence number,
average interarrival jitter.
Sender report packets:
SSRC of the RTP stream,the current
time,the number of packets sent and the
number of bytes sent.
Source description packets:
E-mail address of sender,sender's name,
SSRC of associated RTP stream.
Enable mapping between the SSRC and
the user/host name.
SIP
• Session initiation protocol.
• Comes from IETF.
• Usually a single module suitable for
internetworking.
• People are identified by names or e-mail
addresses, rather than by phone numbers.
• You can reach the callee, no matter where the
callee roams, no matter what IP device the
callee is currently using.
Calling a Known IP Address
Bob
Alice
167.180.112.24
INVITE bob
@193.64.2
10.89
c=IN IP4 16
7.180.112.2
4
m=audio 38
060 RTP/A
VP 0
193.64.210.89
port 5060
port 5060
Bob's
terminal rings
200 OK
.210.89
c=IN IP4 193.64
RTP/AVP 3
3
75
m=audio 48
ACK
port 5060
port 38060
time
• Bob’s 200 OK message
indicates his port number,
IP address & preferred
encoding (GSM)
• SIP messages can be
sent over TCP or UDP;
here sent over RTP/UDP.
m Law audio
GSM
• Alice’s SIP invite
message indicates her
port number & IP address.
Indicates encoding that
Alice prefers to receive
port 48753
time
•Default SIP port number
is 5060.
Example of SIP message
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 167.180.112.24
From: sip:[email protected]
To: sip:[email protected]
Call-ID: [email protected]
Content-Type: application/sdp
Content-Length: 885
c=IN IP4 167.180.112.24
m=audio 38060 RTP/AVP 0
Notes:
r HTTP message syntax
r sdp = session description protocol
r Call-ID is unique for every call.
More details
• Here we don’t know
Bob’s IP address.
Intermediate SIP
servers will be
necessary.
• Alice sends and
receives SIP messages
using the SIP default
port number 506.
• Alice specifies in Via:
header that SIP client
sends and receives
SIP messages over UDP
Definition of H.323 standard
H.323 is a standard that specifies the components,
protocols and procedures that provide multimedia
communication services—real-time audio, video, and
data communications—over packet networks,
including Internet protocol (IP)–based networks.
H.323 is part of a family of ITU—T recommendations
called H.32x that provides multimedia communication
services over a variety of networks
-Emergence of voice-over–IP (VoIP) applications
and IP telephony
-The absence of a standard for voice over IP
meant products that were incompatible.
- Such requirements forced the need for a
standard for IP telephony.
E.g.:Version 2 of H.323—packet-based multimedia
communications systems
H.323 -
lications
H.323 is applied in a variety of
mechanisms
 Audio
 Audio
 Audio
 Video
only
and
and
and
(IP telephony)
video (video telephony)
data and audio
data.
H.323 can also be applied to
multipoint-multimedia communications.
H.323 provides myriad services and,
therefore, can be applied in a wide
variety of areas—
consumer, business, and entertainment
application The features being added
currently include fax-over-packet
networks, gatekeeper-gatekeeper.
Control
Speech
RTP
RTCP
H.225
(RAS)
UDP
Q.931
H.245(call
control)
TCP
IP
Data Link Protocol
Physical Layer Protocol
H.323 Packet Network
The H.323 architectural Model for Internet Telephony
Terminal
GATEWAY
TELEPHONE
NETWORK
INTERNET
Gatekeeper
ZONE
Similarities b/w H.323 and SIP
Allow two Party and multiparty calls
Support parameter negotiation
Support Encryption
Support Media transport on RTP/RTCP protocols
Feature sets are almost similar
ITEM
H.323
SIP
Designed By
ITU
ITEF
Compatibility with PSTN
Yes
Largely
Compatibility with net
No
Yes
Architecture
Monolithic
Modular
Completeness
Full Protocol Stack
Handles only setup
Message Format
Binary
ASCII
Multimedia conferences
Yes
No
Addressing
Host or Telephone no
URL
Instant Messaging
No
Yes
Implementation
Large and complex
Moderate
Standards size
1400 pages
250 pages
RSVP designed at
-MIT
-PARC
-California University.
RSVP Features:
RSVP is a novel signaling protocol in at least 4 ways:
1. It accommodates multicast, not just point-tomultipoint (one-to-many) reservations.
2. QoS routing can be deployed separately (in more
operations, and so is
3. relatively low cost
4.
Scalability.
RESERVATION IMPLEMENTATIONS
Reservations are implemented through two types of RSVP
messages:
PATH and RESV.
 The PATH messages are sent periodically from the sender to the
multicast address. A PATH message contains flow spec to describe
sender template (data format, source address, source port) and traffic
characteristics.
RESV messages are generated by the receivers and contains
reservation parameters including flow spec and filter spec.
S1
R1
R4
H5
H4
S2
R2
R3
H3
S1
R1
R4
H5
H4
S2
R2
R3
H3
Multimedia Networking: Summary
Video conferencing,distance learning.
Distributed networking and sharing of
data and info resources.
Future holds so much promise.
Virtual reality,digital animation,net
telephony.
Interactivity enabled in all sectors.
Next generation internet: Intserv,
RSVP, Diffserv.
Graphical Analysis Ahead
Multimedia market A graphical analysis
Worldwide Statistics of multimedia market compiled by NASDAQ
Movies
Arcades
Video
CDs/Tapes
Info servers
Home video
Cable networks
Values in Billions
TV/Radio
Catalog
Publishing
Computers
Telecom
0
20
40
60
80
100
Market Value in Billions
120
140
160
Growth of Multimedia Networks
N
E
T
W
O
R
K
s
2003
Years
Source: NASDAQ and NASSCOM survey
The real-time challenges
However, multimedia networking is not a trivial
task. We can expect at least three difficulties.
1.
When compared with traditional textual
applications, multimedia applications usually
require much higher bandwidth.
2.
Most multimedia applications require the real-time
traffic
3.
In addition to the delay, network congestion also has
more serious effects on real-time traffic
4.
Multimedia data stream is usually bursty
Books:
1.Multimedia Handbook- Jessica Keyes-TMH
2.Computer networks – Andrew Tanenbaum
3.Multimedia systems- a perspective (IEEE)
Internet:
1.www.cis.ohio-state.edu/~jain/
2.www.google.com
3.University of colombia website.
4.IEEE papers on net
B.Bharat Shetty 4th sem CS&E SJCE