B22 - Multimedia Networking

Computer Networks 2
Multimedia Networking
Team Details
1) Manjunath Mankani 01FE19BCS421
2) Gagandeep Ramgiri 01FE19BCS423
3) Shahid A Hasnabadi 01FE19BCS431
4) Akhilesh Harti 01fe17bcs023
5) Suraj Meharwade 01FE19BCS420
INTRODUCTION
Multimedia network application is any network application that employs audio or video.
• To Understand Internet Multimedia applications, first we look at the characteristics
(Properties) of video & audio.
Properties of Video
• Video has high bit rate (100Kbps for low-quality video conferencing to 3Mps HD Movies)
• Video can be compressed, Compression can be used to create different versions of a
video with each having different video quality so users can choose whichever version
they can watch according to their available bandwidth.
Properties of Audio
• Digital Audio data has much less bandwidth requirements than that of Video.
• Basic encoding technique of converting analog audio into digital audio is called pulse
code modulation (PCM)
• PCM encoded speech not commonly used on the internet so compression techniques
are used to lower bit rates of audio streams.
• MP3 is a compression technique and the encoders can compress rates most commonly
to 128 kbps
Types Of Multimedia Network Application

• Multimedia application are divided into 3 categories
1. Streaming stored audio/video.
2. Conversational voice/video over IP.
3. Streaming live Audio/Video.

1. Streaming stored audio/video.

• In this case of application, the videos are pre-recorded such as movie, TV show, etc.
• These videos are placed on servers and users, and users send requests to the server to
view videos on demand.
• Streaming stored audio/video has three key features
1. Streaming - Means client can play video from 1 location while receiving later parts at
same time.
2. Interactivity - Pause, reposition forward, reposition backward, fast-forward.
3. Continuous playout - Once playout begins, it should proceed in accordance of original
timing.
• Important performance measure for Streaming stored audio/video is Average

Throughput.
• To provide continuous playout one has to maintain Average throughput.
• Streaming stored audio/video uses CDN rather than single data center to give better
performance
2. Conversational Voice/Video Over IP
• Real-time conversational voice over the Internet commonly known as internet telephony
(VoIP)
• Video conversational systems allow users to create conferences with three or more
participants. Conversational voice and video are widely used in the Internet today,
Skype and Google Talk.
• Two Important Application service requirement for CVVOIP
1) Timing Considerations Audio and video conversational applications are highly delay-
sensitive
2) Tolerance of Data loss Conversational multimedia applications are loss-tolerant
occasional loss only causes occasional glitches in audio/video playback, and these
losses can often be partially or fully concealed
3. Streaming Live Audio and Video
• Live streaming comes from a content source such as video cameras and microphones. It
is made available at the same time as the event being filmed occurs.
• Similar to traditional broadcast radio and TV, except that transmission is on Internet.
• Delays of up to ten seconds or so from when the user chooses to view a live transmission
to when playout begins can be tolerated
Multimedia Networking Protocols

There are some protocols that are used to support real time traffic
over the internet.
1) RTP (Real Time Protocol)
2) RTCP (Real Time Control Protocol)
3) SIP (Session Initiation protocol)
4) RTSP (Real Time Streaming Protocol)
1. RTP (Real Time Protocol)
This protocol is used for real time data transport, in this case video and audio.
It specifies a standard packet format for delivering audio and video over IP networks.
RTP Protocol Components

• The following information is needed to send streaming data:
1) Timestamps - for synchronization
a. The timestamp field is 32 bits long. It reflects the sampling instant of the first byte in
the RTP data packet.
b. Timestamp clock rate for video is 90,000 Hz and the timestamp clock rate for audio
8000 Hz.
2) Sequence numbers - for packet loss and reordering detection.
• The sequence number field is 16 bits long. The sequence number increments by one
for each RTP packet sent.
3) Frame Indication - this marks the beginning and end of each frame.
4) Source identification - identifies the originator of the frame.
• The SSRC(Synchronization source identifier) field is 32 bits long. It identifies the source
of the RTP stream. Typically, each stream in an RTP session has a distinct SSRC.
2. RTCP – Real Time Control Protocol
• RTP is a protocol that provides basic transport layer for real time applications but does
not provide any mechanism for error and flow control, congestion control, quality
feedback and synchronization. For that purpose the RTCP is added as a companion to
RTP to provide end-to-end monitoring and data delivery, and QoS.
• RTCP is responsible for three main functions:
1) Feedback on performance of the application and the network
2) Correlation and synchronization of different media streams generated by the same
sender (e.g. combined audio and video)
3) The way to convey the identity of sender for display on a user interface
RTCP Components
• Quality of service (QoS)
• Feedback: Includes the numbers of lost packets, round-trip time, and jitter, so that the
sources can adjust their data rates accordingly;
• Session control: uses the RTCP BYE packet to allow participants to indicate that they are
leaving a session;
• Identification: which includes a participant's name, e-mail address, and telephone
number for the information of other participants;
• Intermedia synchronization: which enables the synchronization of separately transmitted
audio and video streams.
3. SIP – Session Initiation Protocol

• SIP is a text-based transaction protocol similar to HTTP.
It uses commands (called methods) and responses.
• This lightweight protocol provides mechanisms that
make it possible to make calls over an IP network.
Peer-to-peer application protocol transported
via UDP or TCP
• Designed to establish, modify and terminate stateful
multimedia communication sessions/conferences/instant
messaging.
4. RTSP – Real Time Streaming Protocol

• This is protocol is used to enhance control in streaming media servers in the
entertainment and communications industry.
• It provides “network remote control”.
• RTSP messages are sent out of band over port 544
• Works with unicast and multicast
• RTSP doesn’t mandate encapsulation. Can be proprietary over UDP or TCP, or RTP
(preferred)
• Clients of media servers issue VCR-like commands, such as play and pause, to facilitate
real-time control of playback of media files from the server.
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Computer Networks 2

Types Of Multimedia Network Application

2. Conversational voice/video over IP.

3. Streaming live Audio/Video.

1. Streaming stored audio/video.

• Important performance measure for Streaming stored audio/video is Average

Multimedia Networking Protocols

RTP Protocol Components

3. SIP – Session Initiation Protocol

4. RTSP – Real Time Streaming Protocol

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