Unit 7 Multimedia Networking - genuinenotes.com · Multimedia Networking Compiled by: Krishna...

Unit 7

Multimedia Networking

Compiled by: Krishna Bhandari

www.genuinenotes.com

Multimedia applications

•Multimedia is content that uses a combination of different content forms such as

text, audio, images, animations, video and interactive content.

•In recent years, there has been an explosive growth of new applications on the

Internet like streaming video, IP telephony, teleconferencing, interactive games,

distance learning, and so on.

• Those multimedia networking applications are referred as continuous-media

applications and require services different from those for traditional applications

like e-mail, Web, remote login, etc.

• They are also different from download and then play applications. Especially, the

new applications require high quality on the communication latency and the latency

variation(delay-sensitive) but may not require high quality on the error rate (loss-

tolerant).

• One key issue for supporting new multimedia networking applications is how to get

the high quality for the communication latency on the best effort Internet which

provides no latency guarantee.

•Another key issue is how to improve the Internet architecture to provide support for

the service required by multimedia applications.www.genuinenotes.com

• Audio, video processing

• Education and training

• Multimedia analysis and Internet

• Artificial Intelligence

• Virtual reality and 3-D imaging

• Wireless, Mobile Computing

• Animation and Graphics

• Visual Communication


Classes of multimedia application

• Streaming stored audio\video

• Streaming live audio\video

• Real-time interactive audio\video

Streaming stored audio\video

Applications have the following key features:

➢ Stored media: the contents has been prerecorded and is stored at the server. So, a

user may pause, rewind, or fast-forward the multimedia contents. The response

time to the above actions should be in the order of 1-10 seconds.

➢ Streaming: a user starts playout a few seconds after it begins receiving

the file from the server. So, a user plays out the audio/video from one location

in the file while it is receiving later parts of the file from the server. This

technique is called streaming and avoids having download the entire file before

starting playout.

➢ Continuous playout: once playout begins, it should proceed based on the

original timing of the recording. This requires high quality on the end-to-end

delay.www.genuinenotes.com

Streaming live audio\video

➢Applications are similar to traditional radio and television, except that

audio/video contents are transmitted on the Internet. In these applications, many

clients may receive the same program.

➢A key issue here is how to deliver the program efficiently to multiple clients on

the Internet. IP multicasting technologies play a key role for this.

➢ Similar to streaming stored audio and video applications, applications here

require continuous playout and high quality on the end-to-end delay.

Real-time interactive audio\video

➢Applications allow users using audio/video to communicate with each other in

real time.

➢ Real-time interactive audio on the Internet applications are IP telephony, video conference, distributed interactive worlds.

➢Applications in this category require very high quality on the end-to-end delay,

usually a fraction of one second.


Server for stored streaming audio\video

• In these applications, clients request audio/video data stored at servers. Upon

client's request, servers send the data into a socket connection for transmission.

• Both TCP and UDP socket connections have been used in practice. The data are

segmented and the segments are encapsulated with special headers appropriate

for audio/video traffic.

• The real time protocol(RTP) is a public-domain standard for encapsulating such

segments.

•Audio/video streaming applications usually provide user interactivity which

requires a protocol for client/server interaction. The real time streaming protocol

(RTSP) is a public-domain protocol for this purpose.

• Clients often request data through a Web browser. A separate helper application

(called media player) is required for playing out the audio/video.


Multimedia for web server

➢ The stored audio/video files can be delivered by a Web server or by audio/video

streaming server. When an audio file is delivered by a Web server, the file is

treated as an ordinary object in the server's file system, like HTML and JEPG

files.

➢ To get the file, a client establishes a TCP connection with the server and sends an

HTTP request for the object. On receiving the request, the Web server

encapsulates the audio file in an HTTP response message and sends the message

back to the TCP connection.

➢ It is more complicated for the video case because usually the sounds (audio) and

images are stored in two different files. In this case, a client sends two HTTP

requests over two separate TCP connections and the server sends two responses,

one for sounds and the other for images, to the client in parallel. It is up to the

client to synchronize the two streams.


Fig: Multimedia web server


Multimedia for streaming server

➢Audio/video files can be delivered by a streaming server to a media player.

➢ Streaming servers include those marketed by RealNetworks and Microsoft,

and those of public-domain servers.

➢With a streaming server, audio/video files can be transmitted over UDP which

has much smaller end to end delay than TCP.

➢ RTSP is a protocol which allows a media player to control the transmission of

a media stream.

➢ The control actions include pause/resume, repositioning of playback, fast-

forward, and rewind.

➢ RTSP messages use a different port number from that used in the media stream

and can be transmitted on UDP or TCP.


Fig: Multimedia streaming server


Real-time protocol (RTP)

• Real-Time Transport Protocol (RTP) is an Internet Protocol standard that

specifies the way programs manage the real-time transmission of multimedia

data over unicast or multicast network services.

• In comparison to TCP (Transmission Control Protocol) which favors data

integrity rather than delivery speed, RTP favors rapid delivery and has

mechanisms to compensate for any minor loss of data integrity.

• RTP defines the standardized packet format for delivering audio and video over

IP networks and used in conjunction with Real-Time Transport Control

Protocol (RTCP) to ensure that multiple streams of media can be synchronized

and Quality of Service (QoS) can be maintained.

• The most important recent application of RTP is the introduction of VoIP (Voice

over Internet Protocol) systems which are becoming very popular as

alternatives to regular telephony circuits.


RTP components include:

• a sequence number, which is used to detect lost packets;

• payload identification, which describes the specific media encoding so that it

can be changed if it has to adapt to a variation in bandwidth;

• frame indication, which marks the beginning and end of each frame;

• source identification, which identifies the originator of the frame; and

• Intra-media synchronization, which uses timestamps to detect different delay

jitter within a single stream and compensate for it.


RTP Packet Format

• Usually, the media data is encapsulated in RTP packets which are encapsulated in

UDP segments.

• Version: (2 bits) Indicates the version of the protocol. Current version is 2.

• P (Padding): (1 bit) Used to indicate if there are extra padding bytes at the end of

the RTP packet. A padding might be used to fill up a block of certain size, for

example as required by an encryption algorithm. The last byte of the padding

contains the number of padding bytes that were added (including itself).

• X (Extension): (1 bit) Indicates presence of an Extension header between standard

header and payload data. This is application or profile specificwww.genuinenotes.com

• CC (CSRC count): (4 bits) Contains the number of CSRC identifiers that follow

the fixed header.

• M (Marker): (1 bit) Used at the application level and defined by a profile. If it is

set, it means that the current data has some special relevance for the application.

• PT (Payload type): (7 bits) Indicates the format of the payload and determines its

interpretation by the application. This is specified by an RTP profile.

• Sequence number: (16 bits) The sequence number is incremented by one for each

RTP data packet sent and is to be used by the receiver to detect packet loss and to

restore packet sequence.

• Timestamp: (32 bits) Used to enable the receiver to play back the received

samples at appropriate intervals. When several media streams are present, the

timestamps are independent in each stream, and may not be relied upon for media

synchronization.

• SSRC: (32 bits) Synchronization source identifier uniquely identifies the source of

a stream.

• CSRC: (32 bits each) Contributing source IDs enumerate contributing sources to a

stream which has been generated from multiple sources.www.genuinenotes.com

RTSP(Real-Time Streaming Protocol )

• The Real-Time Streaming Protocol (RTSP) is an application-level protocol for

control over the delivery of data with real-time properties.

• RTSP provides an extensible framework to enable controlled, on-demand

delivery of real-time data, such as multimedia data.

• The stream controlled by RTSP may use Real-Time Transport Protocol (RTP),

but RTSP and the underlying protocol used to carry the media stream will not

have any dependence over each other.

• Three of the required RTSP methods are SETUP, PLAY and TEARDOWN.

• These three methods are necessary and sufficient for a client from initiation of

multimedia data streaming to termination of the streaming session.

•A client issues a SETUP request to initiate a server to start a new streaming

session, and then issues a PLAY request to initiate streaming of multimedia

data, and finally issues a TEARDOWN request to close the streaming session.


RTSP client server interaction

• The sequence of requests and responses, and the corresponding RTSP methods

involved in a typical RTSP session, where an RTSP proxy is in between the path of

the RTSP server and the RTSP client,


Real Time Traffic

• Real Time Network traffic also known as data traffic refers to the amount of data

moving across a network at a given point of time.

• Network data is mostly encapsulated in network packets, which provide the load

in the network.

• Network traffic is the main component for network traffic measurement, network

traffic control and simulation.

• The proper organization of network traffic helps in ensuring the quality of service

in a given network.


Loss Recovery Schemes in Real Time Traffic (RTP Protocol):

A packet is lost either if it never arrives at the receiver or if it arrives after itsscheduled playout time. Retransmitting lost packets may not be feasible in a real-time conversational application. Indeed, retransmitting a packet that has missedits playout deadline serves absolutely no purpose. And retransmitting a packetthat overflowed a router queue cannot normally be accomplished quicklyenough. Because of these considerations, real time multimedia applications oftenuse some type of loss anticipation scheme. Two types of loss anticipationschemes are:

Forward error correction (FEC)

Interleaving

Forward Error Correction (FEC):

The basic idea of FEC is to add redundant information to the original packetstream. For the cost of marginally increasing the transmission rate, the redundantinformation can be used to reconstruct approximations or exact versions of someof the lost packets. There are two simple FEC mechanisms.

The first mechanism sends a redundant encoded chunk after every n chunks. Theredundant chunk is obtained by exclusive OR-ing the n original chunks. In thismanner, if any one packet of the group of n + 1 packets is lost, the receiver canfully reconstruct the lost packet. But if two or more packets in a group are lost,the receiver cannot reconstruct the lost packets.


The second FEC mechanism is to send a lower-resolution stream as the redundant

information. For example, the sender might create a nominal stream and a

corresponding low-resolution, low-bit rate stream. The low-bit rate stream is

referred to as the redundant stream.

As shown in Figure below, the sender constructs the nth packet by taking the nth

chunk from the nominal stream and appending to it the (n – 1)st chunk from the

redundant stream.

In this manner, whenever there is nonconsecutive packet loss, the receiver can

conceal the loss by playing out the low-bit rate encoded chunk that arrives with

the subsequent packet. Of course, low-bit rate chunks give lower quality than the

nominal chunks. However, a stream of mostly high-quality chunks, occasional

low-quality chunks, and no missing chunks gives good overall quality.


Fig: Forward Error correction using redundant stream


Interleaving:

As an alternative to redundant transmission, a VoIP application can send interleaved audio. As shown in Figure below, the sender re-sequences units of data before transmission, so that originally adjacent units are separated by a certain distance in the transmitted stream. Interleaving can reduce the effect of packet losses.

If, for example, units are 5 ms in length and chunks are 20 ms (that is, four units per chunk), then the first chunk could contain units 1, 5, 9, and 13; the second chunk could contain units 2, 6, 10, and 14; and so on. Figure below shows that the loss of a single packet from an interleaved stream results in multiple small gaps in the reconstructed stream, as opposed to the single large gap that would occur in a non-interleaved stream. Interleaving can significantly improve the perceived quality of data stream. It also has low overhead. The obvious disadvantage of interleaving is that it increases latency. This limits its use for conversational applications such as VoIP, although it can perform well for streaming stored audio. A major advantage of interleaving is that it does not increase the bandwidth requirements of a stream.


Fig: Interleaving the stream to reduce the effect of packet loss


Jitter

• Jitter is defined as a variation in the delay of received packets.

• The sending side transmits packets in a continuous stream and spaces them

evenly apart. Because of network congestion, improper queuing, or configuration

errors, the delay between packets can vary instead of remaining constant, as

shown in the figure.

• This variation causes problems for audio playback at the receiving end. Playback

may experience gaps while waiting for the arrival of variable delayed packets.


Delay

• The delay of a network specifies how long it takes for a bit of data to travel

across the network from one node or endpoint to another. It is typically measured

in multiples or fractions of seconds.

• Delay may differ slightly, depending on the location of the specific pair of

communicating nodes.

• Types of Delay

➢ Processing delay – time routers take to process the packet header

➢ Queuing delay – time the packet spends in routing queues

➢ Transmission delay – time it takes to push the packet's bits onto the link

➢ Propagation delay – time for a signal to reach its destination


Playback Buffer

• It has been observed that available bandwidth fluctuates with time, if buffering is

not used.

• Playback disruption would occur, if the instantaneous available bandwidth is

lower than the media rate, in order to reduce the number of playback disruptions,

clients would need a buffer .

• Generally all the clients of streaming media typically employ a buffer of 5 to 15

seconds of buffering capacity, with such buffering, a client can pre fetch the data

that is not immediately needed when the available bandwidth is above the media

rate and then the client drains the buffer when the available bandwidth is below

the media rate.

•A video packet that arrives beyond its payout time is useless and can be regarded

as lost. Hence in order to provide continuous playback, a buffer at receiver/client

has been introduced before decoding, it is called playback buffer, and the major

aim of introducing the playback buffer is to smoother the bit rate variations


Content Distribution Network

• A content distribution network—also known as a content delivery network—is

a system of distributed servers (network) that deliver pages and other Web

content to a user, based on the geographic locations of the user, the origin of the

webpage and the content delivery server.

• This service is effective in speeding the delivery of content of websites with

high traffic and websites that have global reach. The closer the CDN server is

to the user geographically, the faster the content will be delivered to the user.

• The primary technique that a content distribution network (CDN) uses to speed

the delivery of web content to end users is edge caching, which entails storing

replicas of static text, image, audio, and video content in multiple servers

around

the "edges" of the internet, so that user requests can be served by a nearby edge

server rather than by a far-off origin server.

• To also accelerate the delivery of dynamically generated web content that's

difficult or impossible to cache, an advanced content distribution network uses

a range of techniques such as: route optimization, TCP connection

optimization, and pre-fetching.www.genuinenotes.com

Working of CDN: Finding base server

• Servers nearest to the website visitor respond to the request. The content

delivery network copies the pages of a website to a network of servers that are

dispersed at geographically different locations, caching the contents of the page.

• When a user requests a webpage that is part of a content delivery network, the

CDN will redirect the request from the originating site's server to a server in the

CDN that is closest to the user and deliver the cached content.

• CDNs will also communicate with the originating server to deliver any content

that has not been previously cached.

Advantages:

• Improving website load times

• Reducing bandwidth costs

• Increasing content availability and reachability

• Improving website security


CDNs typically adopt one of two different server placement philosophies:

Enter Deep: One philosophy, is to enter deep into the access networks of Internet

Service Providers, by deploying server clusters in access ISPs all over the world.

The goal is to get close to end users, thereby improving user-perceived delay and

throughput by decreasing the number of links and routers between the end user and

the CDN cluster from which it receives content. Because of this highly distributed

design, the task of maintaining and managing the clusters becomes challenging.

Bring Home: A second design philosophy is to bring the ISPs home by building

large clusters at a smaller number of key locations and connecting these clusters

using a private high-speed network. Instead of getting inside the access ISPs, these

CDNs typically place each cluster at a location that is simultaneously near the

PoPs (Point of Presence) ISPs. Compared with the enter-deep design philosophy,

the bring-home design typically results in lower maintenance and management

overhead, possibly at the expense of higher delay and lower throughput to end

users.



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Unit 7 Multimedia Networking - genuinenotes.com · Multimedia Networking Compiled by: Krishna...

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