Video Over Internet

8/6/2019 Video Over Internet

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Abstract Media streaming is an interesting concept that

requires certain components to be successful. This project is a

critical review of the paper Streaming Video over the Internet:

Approaches and Directions. There are certain factors that must

be considered when tackling streaming media including

compression, quality of service, storage, synchronization,

protocol, and distribution services. Streaming media has seen

significant changes over the years. Streaming and viewing media

has become popularized over the last decade because of easy to

use services and programs to accomplish streaming media.

Various applications for streaming media are reviewed and

tested to compare features and abilities. The future of streamingmedia will continue to grow as it has in the past years, so many

areas of media streaming must continue to improve to support

faster, more efficient, and higher quality streaming.

Index Terms Streaming video, Internet, Video compression,

Streaming server, Transport protocols

I. INTRODUCTIONhe internet has always encouraged the idea of putting

real-time multimedia up where anyone can view it.

There are different ways of getting media from one userto another across the Internet including full file

download and media streaming. Media streaming is very

important as it offers a method for immediate viewing

and does not have the significant transfer time that can

be experienced by a full download. The chosen paper

focuses on the ways in which streaming media can be

accomplished.

There are many programs available to view streaming media.

The different programs have unique features for streaming

media playback. Using one of these programs to view and

stream media is a simple task that can allow streaming using

one of the given protocols.

The six areas that must be touched on when explaining

video streaming are shown in Figure 1. This paper

reviews the six concepts of video compression, quality

of service control, continuous media distribution,

streaming servers, media synchronization, and protocols

for streaming media.

II. REVIEW OF PAPER[1]The following is a summary of the critiqued paper

touching on the important topics covered. The basic

design for media streaming is depicted Figure 1. Raw

media is compressed and saved into a storage device as

audio and video components. The storage device is

located on a server. For the media to pass from the

storage device to the Internet, the media must pass

through certain quality of service controls as well as

transport protocols. While travelling through the

Internet, continuous media distribution services are used.

When the media arrives at the receiver or clients end, it

again goes through transport protocols and quality of

service controls. At this point, the separate streams for

audio and video are passed through audio and video

decoders respectively. After decoding, media

synchronization is used to align the streams to assure

proper playback.

Figure 1 Video Streaming Setup (adapted) [1]

A. Video Compression

One of the main purposes of video compression is to

reduce the bandwidth necessary to transmit video. There

are many approaches and requirements for encoding and

Critical Review of Streaming Video over the

Internet: Approaches and Directions

Patrick Liby

T


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decoding media in streaming applications. For the most

part, video compression schemes can be split up into

scalable and non-scalable video coding.

Non-scalable video coding is the more basic process. For

non-scalable coding, raw video is put through some kind

of transform like the discrete cosine transform. The data

is then quantized from continuous values to discretevalues. Next, variable length coding is applied to create a

compressed bit-stream. At this point, all of the encoding

is completed

Figure 2 Non-scalable video encoding process

(adapted) [1]

The decoding process is very similar. The compressed

bit-stream goes through variable length decoding. Then,

an inverse quantization process converts the discrete

system back to a continuous system. Next, the data goesthrough a reverse of the transform implemented in the

encoder.

Figure 3 -Non-scalable video decoding process

(adapted) [1]

Scalable video coding is different from non-scalable

because raw video is compressed into numerous sub-

streams of data. Part of the scalable video code bit-

stream is a base sub-stream that can be decoded alone

for a low quality video. Additional sub-streams are

encoded to enhance the base sub-stream. These sub-

streams must be decoded along with the base stream andimprove video quality.

Scalable coding can create base sub-streams with

reduced quality, image size, or frame rate. For each

scalability factor, raw video is compressed to a base

layer bit stream and enhancement layer bit stream.

Figure 4 SNR-scalable encoding process (adapted) [1]

When only the base layer bit stream is decoded, the

decoded video has the reduced quality, image size, or

frame rate. When the enhancement layer is included, the

top quality video is achieved.


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Figure 5 - SNR-scalable decoding process (adapted) [1]

A new scalability factor is proposed called fine

granularity scalability. It is most similar to the SNR

encoder. However, FGS encoders use bit planes in the

coding process, which separates transform data into

planes of bits from most significant to least significant.

The enhancement stream is made up of these bit planes.

This approach allows the FGS encoder to have

continuous rate control over the enhancement stream.

The enhancement stream can include as many or has few

of the bit planes to attain the desired bit rate.

Many factors for video streaming impact the video

compression aspect. The need for a certain quality level

of video that is acceptable for viewing demands a

requirement for a minimum bandwidth. Over the

Internet, this presents an issue, because there is not

support for minimum bandwidths. The Internet also

causes the need for a video streaming application to deal

with congestion, as sometimes there is heavy traffic on a

network. In video compression, this is handled by rate

control, which is more easily done with scalable coding

rather than non-scalable coding.

Another concern for streaming video is delay. Having a

video pause momentarily during playback due to a

packet taking too long to arrive must be overcome. The

most common solution for this is a buffer at the playback

end so that there is a certain amount of video ready to go

in case of a delay in receiving a packet. Packet loss is

another video streaming issue, which can best be

handled by making the video stream tough with a

compression that helps with packet loss. Decoding can

be impacted by the end device for streaming video if

there is a specific requirement, like low power for a

handheld device. To consume less power, a simple

decoding process is necessary.

B. Quality of Service Control

Application-layer QoS control techniques are used to

handle changes in quality of transmission and network

conditions. The main areas of focus that are used are

error control and congestion control. The purpose of

error control is to improve quality when there is lost

data. Mechanisms in place include error concealment,

retransmission, forward error correction, and error

resilient encoding. Congestion control is helpful in

minimizing delays and avoiding packet losses.Application-layer QoS control techniques are used by

the receiving systems and do not need help from the

network to implement the techniques.

The error control mechanisms in use include the

following four types.

Forward error correction is an error control that adds

redundant data called an error-correction code to allow

the receiver to detect and correct errors to some extent.

FEC is very beneficial because retransmission of datacan normally be avoided and a back channel is

unnecessary. The type of FEC implemented can be

categorized into one of three schemes; channel coding,

source coding-based FEC, and joint source/channel

coding. Channel coding involves breaking the video

stream into pieces and creating packets to transmit.

Additional packets are added to each piece in a way that

when transmitted, only the original number of packets

must be successfully obtained in order to perfectly

decode the piece of video. Source coding FEC also

includes redundant data into the transmission. In sourcecoding FEC, a packet contains the data for its current

packet and the redundant compressed version of the data

from the previous packet. The combined source and

channel coding attempts to achieve the best balance of

channel and source coding.

Retransmission is an approach to error control that can

be overlooked since it is possible for the retransmission


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time to be inside the maximum delay. In this way,

retransmission can make a reasonable means for error

control, because it is possible for the lost data to be

retransmitted before its intended display time.

Error-resilient encoding improves the strength of

compressed video over packet loss. Basic schemes

include data partitioning, data recovery, andresynchronization marking. These are not quite as

applicable to video streaming on the Internet, because

the medium is not as susceptible to error. In the Internet

video-streaming realm, a less standardized error-resilient

tool like multiple description coding is more promising.

MDC breaks a raw video into many streams of data

called descriptions, which provide acceptable quality on

their own and better quality when combined. The benefit

of MDC is that it is strong against loss because even if

descriptions get lost, it only needs one to have a

minimum quality video. The other benefit is that it hasenhanced quality for each additional description that is

received. The down side of MDC is that each description

must contain a certain extent of information about the

original video signal. This means that the amount of

attainable compression is less. Each description must

also have data to combine any of the other descriptions,

which also impacts the amount of compression. A trade

off of compression efficiency and reconstruction quality

must be determined when using multiple description

coding.

The receiving end uses error concealment when there

has been a loss. Its purpose is to cover up packet losses

and make the remaining data more appealing to the

viewer. Error concealment can use either spatial or

temporal interpolation. Spatial interpolation recreates

missing pixel data by using the data of the surrounding

pixels to determine the most suitable values. Temporal

interpolation reconstructs missing pixel data by using the

data for that pixel in the previous frames. Spatial

interpolation is more common for intra-coded frames,

while temporal interpolation is more popular for inter-coded frames. Some proposed error concealment

techniques have been put include maximally smooth

recovery, projection onto convex sets, and motion

compensated temporal prediction.

Congestion control is necessary when bursting loss and

excessive delay occurs in a video stream due to network

congestion. Congestion control can help to minimize

packet loss and delay. The most common congestion

control for video streaming is rate control, which tries to

reduce congestion by matching the video stream rate to

the currently usable network bandwidth.

There are three types of rate control schemes: source-

based, receiver-based, and hybrid rate control. Source-

based rate control makes the transmitting endresponsible for changing the video transmission rate.

The sender uses feedback about the network to regulate

the video stream rate. Source-based rate control can be

used for unicast and multicast. Unicast video uses probe-

based and model-based approaches to rate control.

Probe-based rate control probes the network to

determine the available bandwidth by varying the

sending rate. Model-based rate control uses a model of

the TCP connection including the throughput, packet

size, round trip time, and packet loss ratio of the

connection. The model determines the sending rate forthe stream; therefore, the connection can avoid

congestion just like the TCP. Multicast video has the

transmitter use one channel to send the video, which is

called single channel multicast. Only probe-based rate

control may be used with single channel multicast. It is

less flexible in supplying services of receivers with

varying needs, but is more efficient due to all receivers

being on the same channel. Multicast video on separate

unicast video streams would be less efficient but would

offer more flexibility in services available.

Receive-based rate control allows the receiving end to

determine the receiving rate of the video stream by

varying the number of channels without send

involvement in rate control. Receiver-based rate control

uses the same control mechanisms of probe-based and

model-based control. Probe-based rate control probes the

bandwidth by joining additional channels and checking

the impact on congestion. If congestion is detected, a

layer is dropped. The model-based control approach uses

similar characteristics as the source-based model

approach to explicitly estimate the available bandwidth.Hybrid rate control has the receivers adjust the receiving

rate by adjusting the number of channels and the senders

change the transmission rate on each channel using

information from the receivers. A layered multicast

scheme is a form of hybrid rate control.

Rate shaping is an attempt to equalize the raw video bit-

stream and the target rate constraint. It is necessary in


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source-based rate control, since stored video can be pre-

compressed at a different rate from the usable network

bandwidth. Common filters to accomplish the task

include codec, frame-dropping, layer-dropping,

frequency, and re-quantization filters. Codec filters

affect the compression of a stream by using transcoding

between compression schemes. Frame-dropping filters

reduce the data rate by lower the number of frames sentby removing less important frames until a desired rate is

achieved. Frame dropping can be done at the source or

anywhere in the network. Layer-dropping filters discard

layers based on significance. The last layer to be

dropped is the base layer. Frequency filters include low

pass, color reduction, and monochroming.

Monochroming changes an image from color to

grayscale. Frame rate remains the same in frequency

filters; instead it lowers quality to reduce the bandwidth.

Re-quantization filter reduces the rate by de-quantizing

and re-quantizing the data with a larger quantizationstep.

C. Continuous Media Distribution Services

Streaming data is continuous media because it is a

sequence of data that is time sensitive. Continuous

media distribution services are used to improve QoS and

efficiency over the network and Internet from streaming

media. Important areas of continuous media distribution

services include network filtering, application-level

multicast, and content replication.

Network filtering is intended to improve quality during

congestion in the network. While a video server can

filter and adjust stream rates based on the level of

congestion, it may be more beneficial to have filters in

the network that are not going to get as busy as the video

server. Filters placed in a network can go anywhere

along the path from server to client, including placing

more than one filter on a pathway. A filter in the

network takes in client information and varies the data

from the server to meet the needs of the clients.

Figure 6 Filters within a network(adapted) [1]

Filter operation includes, at a minimum, a server, client,

filter, and two channels each from the filter to the server

and client. The control channel is bidirectional, but thedata channel flows only from the server in the direction

of the client. There are the same two channels all of the

way from the server to the client, even if there are

multiple filters along the path. In this system, the client

need only interact with a single host, which is the last

filter on the path to the client. The host (filter) either

passes information along to previous pieces in the path,

or it will accomplish the request from the client. A filter

can acquire data from the previous piece in the path

chain and it can pass data along to the next piece of the

chain or the client.

Figure 7 Network filtering(adapted) [1]

Frame-dropping filters are commonly used as networkfilters, where the receiver can request a different

bandwidth and the filter adjusts the frame-dropping rate.

The determining factor for frame dropping is the packet

loss ratio, which is determined by the receiver and sent

to the filter. If the packet loss ratio is outside of the

threshold boundaries, the receiver requests a new frame-

dropping rate. The benefits of the frame-dropping filter

in the network include better quality higher bandwidth

efficiency.


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The filter provides better quality because instead of

packets being corrupted when the network gets

congested, the filter prevents the congestion by dropping

packets in a way that delicately affects quality.

Bandwidth efficiency is improved because late frames

are deleted as soon as they are late, and they do not

continue along the path to the receiver and bog downnetwork resources.

Application-level multicast is an attempt to help with the

Internets inability to accommodate large data transfers

like streaming-media multicast. IP multicast is one

attempt to address this issue, which is able to offer

efficient multipoint packet delivery. This technology is

efficient by transporting only one copy of the original IP

packet on any path in the IP multicast tree. IP multicast

has many issues including scalability, network

management, deployment, and support for higher layerfunctionality. An application-level multicast mechanism

has been suggested to overcome these issues. The

purpose of the application-layer multicast is to establish

a multicast service over the Internet, which would allow

CSPs and ISPs to create Internet multicast networks and

route them into media multicast networks. Basically, in

the same way that the Internet is a huge web of

interconnected networks from peering among ISPs,

media multicast networks can be established through a

web of content-distribution networks from application-

level peering. Media multicast networks are comprisedof nodes that route within the application layer. These

nodes are wired to neighboring nodes, and altogether

perform optimal routing for sending data across the

network. The media multicast network is also

responsible for rerouting data when the network is

congested along a certain path. Bandwidth is conserved

in this method because only one copy of the multicast

data is sent over the pathways. Application-level

multicast is able to overcome the issues of scalability,

network management, and congestion control.

Content replication is essential to increase the scalability

of the media delivery system. It is achieved through

caching and mirroring to put data nearer on the pathway

to clients. Caching and mirroring are advantageous

through lowering bandwidth on the network, lowering

latency, and improving availability. Mirroring is done

when copies of multimedia are placed on other locations

around the Internet. Basically, the original file is on the

server, and backup copies are on duplicate servers. This

improves performance as clients can acquire the file

from the nearest server, original or duplicate. The

downside to mirroring is that it is a complex and

potentially expensive process. Also, scripts and server

setup vary from server to server making the process

difficult. Caching creates local copies of files for clients

to obtain, commonly from a single server, which iscalled the clients cache. The cache works as a gopher

and acquires media from the server and stores it locally

to pass on to the client. Caches are also designed to get

media from other caches when local to reduce the load

on the original server, thereby improving bottleneck

issues. Caching techniques have tended to be generic,

however, in a case like caching even just a few frames

can significantly improve performance. By storing part

of a video stream in cache, strength of the video stream

can be improved over network congestion. The proposal

of cache hints has helped to reduce latency and improvecache hit rate by scheduling data retrieval. There are

content hints and application hints. Content hints deal

with data and delivery method, while application hints

describe data needs of the receiving application. When

multicast is used, hints are able to help decide if and

when multicast channels should be combined and for

how long.

D. Streaming Servers

Streaming servers are an absolute requirement for videostreaming with good quality. Servers are needed to

process data fast enough to prevent video and audio

skips and glitches when the client plays the media across

the Internet. Streaming servers must also be able to

provide stop, pause, rewind, and fast forward playback

options. A streaming server must also be able to

synchronize its components for video, audio, and

anything else like lecture slides.

Streaming servers are commonly made up of three

components: the communicator, the operating system,and the storage system. The communicator works in the

application layer and deals with the transport protocol

used by the server. A client uses the communicator to

talk to the server and obtain streaming videos and media.

The operating system for a streaming server must focus

on the servers ability to provide time sensitive service

and quality performance to its users. The storage system


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for the server must be able to continuously store and

fetch media.

A main purpose of the operating system on a computer is

to separate the hardware from the rest of the software

running on the computer. The operating system connects

the CPU, RAM, storage, and I/O devices to software and

each other.

Process management is used to put individual single

processes onto the CPU based on scheduling rules to

allow all processes to be completed. Since continuous

media has certain timing specifications, the operating

system needs to use real-time scheduling. The majority

of ways to rectify real-time scheduling issues come from

two algorithms for media systems. Earliest Deadline

First assigns a deadline for every task and they are

processed based on priority of deadline. Rate-monotonic

assigns each tasks priority based on the tasks requestrate. The higher a tasks rate, and therefore the lower the

period, the higher the priority. Tasks are processed from

highest priority to lowest.

Figure 8 EDF and rate-monotonic scheduler[1]

Earliest Deadline First and Rate-monotonic are both

preemptive with scheduling. An interrupted task resumes

at a later time when the interruption is done. Earliest

Deadline First scheduling uses a single-priority queue

where the processor runs the task with the first deadline.

Rate-monotonic uses a multi-priority queue where the

processor will always work on the highest priority first,

including interrupting a task if one with a higher prioritycomes about. Rate-monotonic scheduling is more likely

to switch tasks than Earliest Deadline First. The Rate-

monotonic schedule boasts that all deadlines can be met

as long as the processor is only 69% utilized. Earliest

Deadline First can achieve 100% utilization of the

processor, but sometimes some tasks are not processed

when the queue is overloaded.

Resource Management must allocate resources such as

the CPU, memory, and storage on the server. Resources

are finite, so the server can only provide a given quality

of service to a certain number of clients at once. The

server must also use admission control to determine if an

additional client will overload the resources and break

performance expectations. If the client is allowed to

connect, the resource manager then assigns properresources to the connection. Admission control

algorithms come in two types: statistical admission

control and deterministic admission control. Statistical

admission control algorithms give the client a statistical

guarantee for requirements like continuity. Deterministic

admission control algorithms create firm guarantees.

Deterministic admission control is a simplified but strict

means of obtaining quality assurance, which makes it

advantageous compared to statistical admission control.

On the other hand, statistical admission control better

uses server resources by taking advantage of humantolerances and server average performance versus

minimum performance. Resource allocation can also be

classified as deterministic or statistical. Deterministic

resource allocation sets aside maximum necessary

resources for a task, regardless of whether those

resources are used all throughout the task. Statistical

resource allocations are able to get higher utilization by

allowing short-term overflows as well as a minimal

quality of service violations.

For file management, the file system is used to accessand control files to store and retrieve. For continuous

media, two schemes can be used for the file systems.

The first scheme organized files on the hard disk the

same way that it is done for discrete data where a file is

in one location on a hard disk. To access in real time,

disk-scheduling algorithms are performed and buffer

capacity is increased enough to prevent skipping in the

media during playback. The other scheme distributes the

storage of media across multiple places. This allows

throughput to be improved by pulling pieces of data

from multiple places at once so that the data does not bottleneck from one storage location to the processor.

Disk scheduling is important for either scheme, but

standard disk-scheduling algorithms have no real time

guarantees. Possible disk-scheduling algorithms for real-

time include SCAN-EDF, grouped sweeping scheduling,

and dynamic circular SCAN. SCAN-EDF uses seek

optimization combined with real-time abilities provided

by Earliest Deadline First. Grouped sweeping scheduling


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forms groups out of the video streams, where each

stream in a group has a comparable deadline, and SCAN

is used to determine scheduling for the streams within a

group. DC-SCAN tries to minimize disk seek overhead

by using a circular SCAN, which allows higher

throughput. Start-up delays can be minimized using this

method. Real-time conditions for continuous media can

be achieved from these algorithms. File managementmust also support pause, resume, and fast forward

functions. The pause and resume functions make

efficient design difficult because they dont allow

multiple viewers to share a single stream. Fast-forward

can be achieved by increasing the playback speed or

playing at normal speed and skipping data. Skipping

data is more commonly used as this does not run into the

issue of an increased data rate seen when playback speed

is increased. This results in video playback where the

resolution does not change, but there are minor jumps

between frames, which are difficult for the viewer tonotice.

Storage systems for multimedia in a streaming server are

difficult to design. These challenges include high

throughput, large capacity, and fault-tolerance.

Throughput is a serious issue because if a media file is

stored entirely on one disk, it can only be accessed by a

limited number of users at the same time based on the

throughput of the given disk. Data striping is the

technique used to solve this issue, because it splits a file

into pieces and spreads them out over different disks sothat pieces can be accessed simultaneously by a higher

number of users. This also creates the challenge of

balancing heavy-trafficked pieces across different disks.

Data striping does not create additional copies of a file

across multiple disks, making it more effective than just

replicating the file to multiple locations. Streaming video

can be performed much better with a hierarchical storage

architecture or tertiary storage. Hierarchical storage

systems move data to and from high-cost and low-cost

storage devices. Data is typically stored on cheaper but

slow areas like optical discs, and it is moved to fasterhard disk arrays when it is needed. When the data is

stored on the slower, low cost devices like a series of

optical discs like DVD-ROMs, this storage system is

called tertiary storage. For large amounts of streaming

video, a Storage Area Network is used to connect

various storage devices (high and low cost and speed) to

host servers. An SAN typically uses a fiber channel to

connect all the components for faster access. Network

Attached Storage is another way to allow large amount

of data storage. This approach attaches a computer on

the network to offer file storage services to the rest of

the network. Both approaches improve the standard

storage by offering simplification with a central storage

space and scalability. It is also possible to use both SAN

and NAS, which can offer both file-level and block-level

protocols. Fault tolerance is a must for a storage system, because inevitably a disk will fail. In this situation, a

server needs to recreate the data that is lost, most

commonly by an error correction code or multiple copies

of data at separate locations. The multiple copies

approach has been found to be cheaper per bit than the

error correction method.

E. Media Synchronization

Synchronization for media systems deals with temporal

relations between media items such as audio, video, andany accompanying materials. Since the significant media

for streaming is continuous, it is made up of time-

dependent media items, including audio that must sync

up with video as well as must be played in chronological

order. [mm-sync]

Media synchronization focuses on the integration of

media streams in a synchronized fashion. An example of

this would be making sure the proper piece of an audio

stream is playing when either a video clip or slide show

presentation is playing. If not, the a confusion issuearises when, for example, the current audio is referring

the previous or next slide or the current audio is the

delayed voice of a person in a video, where the audio

does not line up with their motions and movements.

For media to be properly synchronized, the intent is to

have the receiver show the media as it was intended. To

achieve this goal, timing requirements called temporal

relationships must be maintained not only among media

streams, but also inside a media stream.

There are a few levels of synchronization that include

intra-stream synchronization, inter-stream, and inter-

object synchronization. Intra-stream synchronization

deals with factors within a media stream like syncing an

audio file to play normally. Intra-stream focuses on

synchronization at the media layer, the lowest level.

Inter-stream synchronization deals with the next level

above the media layer, the stream layer. On this level,


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the focus is on the entire stream interacting with other

streams. An example for inter-stream synchronization is

having an audio stream and video stream playing

together. Inter-stream synchronization keeps the two

streams in line so that one is not playing ahead or faster

than the other. Inter-object synchronization is the top

level of synchronization called the object layer. This

layer incorporates media streams with non-continuousmedia. Such time-independent data needs to be

displayed or incorporated for a desired duration, which

is the job of inter-object synchronization. Subtitles are

an example of this, because subtitles are only useful

when displayed at the proper time for a necessary

duration. Inter-object synchronization makes sure the

subtitle does this and prevents delayed or early subtitles.

Figure 9 depicts a continuous audio stream with several

subtitles being displayed at the appropriate times.

Figure 9 Example of synchronization between audio

and subtitles

One of the main concerns with media synchronization is

that media streams get out of sync during transit. Every

component that the media travels through causes certain

delays, which do not necessarily affect all streams the

same way. The purpose of media synchronization

mechanisms is to realign the streams in an attempt to fix

the transition issues. Realigning a stream requirestemporal relations to be specified either manually or

automatically. In a situation where subtitles or slides

must be synced up with audio, manual specifications are

used. Specifications are automatic in audio and video

media. When audio and video are recorded together,

these specifications are embedded and upon playback,

they are automatically in place.

There are numerous methods for temporal relation

specification. Methods can be interval-based, axes-

based, control flow-based, or event-based. Axes-based is

the most common method as it is very simple and

successful. It works by placing timestamps on each

media stream before transmission. When the data is

streamed over to the receiver, the time stamps can be

used to sync up the streams. Timestamps allow eachstream to individually sync up itself as well as sync

multiple streams together.

Temporal relations are also beneficial along the

transmission route by each element the data goes

through. Having data available so that it is quick to

obtain minimizes latency. Allowing the necessary

bandwidth is effective enough to counteract the jitter that

would otherwise impact synchronization.

One type of mechanism used to improve synchronizationis a preventative mechanism. This is implemented on the

transport end of the transmission. A few examples of

preventative mechanisms include network transport

protocol, disk-reading scheduling algorithms, and

synchronization schedulers. The aim of preventative

mechanisms is to make latency and jitter irrelevant from

the media streaming.

The opposite type of mechanism is corrective.

Corrective mechanisms are implemented on the receiver

end of the transmission. Stream synchronization protocolis the main corrective mechanism example. The goal of a

corrective mechanism is to reacquire synchronization.

Media synchronization is a big concern for media

streaming. It is one of the most common hiccups and it is

also one of the most noticeable issues from the end

users perspective. For many of the other concerns in a

media streaming system, an issue with that concern will

cause an issue with media synchronization.

F. Protocols for Streaming Media

When streaming media, there are many protocols

available to use between sender and receiver. The

protocols used for streaming media can be distinguished

into three categories: network-layer protocol, transport

protocol, and session control protocol.


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Network-layer protocol is used for beginning to end

packet transfer. The network-layer protocol allows the

transfer of data of various lengths. Network support such

as network addressing is given by network-layer

protocol.

Transport protocol is how the network is able to offer

end-to-end services. Transport protocols are the mostfamiliar and commonly used or chosen by providers.

These protocols include UDP, TCP, RTP, and RTCP.

UDP and TCP are the more commonly known of the

protocols. Real-time transport protocol (RTP) and real-

time control protocol (RTCP) are both higher-level

protocols that can be implemented along side either UDP

or TCP, both lower level protocols.

Certain functions are achieved through using TCP or

UDP. Streams can be multiplexed using TCP or UDP

from a single device and IP address. Error correction isused by both protocols. Corrupted data is removed

before it reaches the layers above TCP and UDP, which

are RTP and RTCP. In the event of data being lost, TCP

resends the data while UDP does not. TCP also attempts

to counteract overflow via flow control, but UDP has no

such feature. UDP is commonly used for media streams

since the latency felt when TCP must resend data is too

noticeable to the end user.

Real-time transport protocol is a type of data protocol

focusing on transmitting data from the sender end to thereceiver end. RTP uses time stamping, sequencing, and

identification to support media streaming. The time

stamping done by RTP is what the media

synchronization techniques use to realign data. Sequence

numbering verifies that the data at the receiver end is

reassembled into the correct order. One kind of

identification employed by RTP is payload type

identification, which determines the kind of payload in

the packet. It helps the recipient with the proper

interpretation of the data packet. The other identification

is source identification. This tells the recipient where thedata in question came from, as in who sent it.

Unlike RTP, real-time control protocol is, as it is named,

a control protocol. RTCP is used to give information

back to the sender, such as quality of service,

synchronizations, or identifications. Quality of service

feedback is the main intended purpose of RTCP. This

helps both parties in the channel know how well the

transmissions are going as far as success rates, delay

durations, and current productivity. Synchronizations are

used to send the current time stamp so that each end

knows what should be assembled and played back at the

given moment. Participation packets transfer information

about the involved parties in a form that can be

understood by users.

Session control protocol allows multiple exchanges

between a sender and receiver over one TCP connection.

The protocol is based off of TCP protocol. It is also

intended to be simple to put in place. Session control

protocols help with message transfer and confirmation

while there is a firm connection.

One session control protocol is real-time streaming

protocol. It is a control protocol for streaming media

servers used to create and manage media data transfers

between sender and receiver. RTSP uses RTP for theactual data transfer, but it can be used to acquire

description information or to introduce more media to a

session.

Another session control protocol is session initiation

protocol. This protocol is used to control sessions of

voice transmission or calls over an IP. SIP can help to

establish, organize, customize, or end communication

between users. SIP is a much more transient protocol as

it can handle a users location changing and continue to

keep a channel going.


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Figure 10 Media streaming protocol(adapted) [1]

Figure 10 shows how all of the protocols are used in the

scheme for media streaming.

III. UPDATE TO PAPERVideo compression has gone through significant

improvements. Currently, the newest widely usedcompression standard is H.264/MPEG-4, which was

introduced in 2003 with ongoing updates. It compresses

video more efficiently than previous methods such as

H.262/MPEG-2. The H.264 format is used for

applications ranging from low quality Internet streaming

to high quality movie theater digital films. One of the

benefits of the H.264/MPEG-4 standard is its use of

entropy coding designs that provide nearly lossless

compression. It allows fairly high quality to be

transferrable without huge overhead or bandwidth being

necessary. The standard also has a set of profiles, whichare chosen from depending on what the purpose or

application of the given video is. Concerns such as cost

and quality are used to help with what profile is most

useful. H.264/MPEG-4 also uses error resilience tools to

help reduce error alongside other error control methods.

One error resilience tool, flexible macroblock ordering,

creates the ability to split an image into group of slices.

Each slice may be decoded individually. If a slice is lost,

neighboring slices are used to rebuild it. Another feature

used is redundant slices, which works hand in hand with

flexible macroblock ordering, as it provides lower

quality backup slices in case the main slice is lost. The

standard also implements UEP, which applies stronger

protection to more important pieces of data. [3]

In the wireless realm, studies have shown that scalable

video streaming is much more beneficial than its non-scalable counterpart. When it comes to wireless mobile

communication, bandwidth varies with time so having a

scalable approach allows the receiver to continue to

stream video. Even though it is at a reduced quality, it is

hugely beneficial since the user notices quality change

less than delay or jitter. [8]

A few factors that have changed when it comes to

streaming media include network bandwidth, access to

networks on the Internet, the use of standard protocols,

and commercialization of the Internet.

The increase in network bandwidth over the years has

helped allow the average Internet user to use streaming

media. The increase in bandwidth has made it practical

to watch streaming videos in real time. The high

bandwidth allows high quality video to be streamed.

Where eight years ago, nearly all of the streaming video

was in standard quality 320 by 240, today, high

definition 1280 by 720 and even better is being pushed

for as the norm. [2]

Since the completion of the critiqued paper, the single

most important addition to the Internet in the realm of

streaming media was the creation of the website

YouTube and similar sites. YouTube allows users to

upload unlimited videos and watch those posted on the

website through the sites search engine. YouTube uses

the Adobe Flash Player engine to play videos. The site

also allows users to embed videos from the site into

other web pages. Adobe Flash Player is the engine used

for nearly 75% of streaming video online. [6] The site

helped popularize streaming media to the general public.[4]

Internet television has also been a huge jump for

streaming media. Internet television provides

mainstream television shows to be viewed online. The

show is streamed to the user in real time with either high

definition or standard definition depending on the


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connection speed of the user and the stream quality

being offered by the provider. [9]

Mobile devices have had one of the most recent impacts

on streaming media.

IV. AVAILABLE APPLICATIONS FORSTREAMINGHere is an overview of some of the best and easy to use

applications for media streaming. The applications

discussed in this section are the top in the media

streaming area.

VLC Media Player is a cross-platform open-source

multimedia player, framework, and server that handles

the majority of formats for audio and video, DVDs and

VCDs, and streaming protocols. VLC can be used as a

media converter or server to stream in unicast or

multicast on networks. VLC can also be used as a plug-

in with other programs like web browsers or video

encoders. VLC has a modular design that allows it to

easily add new codecs, streaming methods, and file

formats. VLC is capable of playing content that is

corrupt, incomplete, or damaged. Files that are mid-

download can be played and continue to download, due

to the program being a packet-based player. VLC can

handle the streaming protocols UDP, HTTP, RTP,

RTSP, RTMP, and MMS. The application can play

formats with inputs UDP/RTP unicast or multicast,

HTTP, FTP, TCP and SFTP. While the application has

no native formats, the default and most common formats

for VLC include AVI and MPEG-4 files.

QuickTime is a multimedia framework from Apple that

can handle a large variety of digital media formats. The

QuickTime framework allows encoding and transcoding

between formats compatible with QuickTime. The

application is capable of a plug-in architecture to support

additional codecs. QuickTimes Streaming Server is a

RTSP server that delivers video and audio on request

over a network or the Internet. The software uses HTTP

Live Streaming, which works by breaking a stream into

small HTTP-based file downloads to transfer data.

QuickTime fully supports standards including H.264,

MPEG-4, and 3GPP. QuickTime is also part of the

foundation used by Apple to create its music playback

software iTunes. QuickTime is also the native player for

QuickTime Movie files, QTVR movies, and Apple

Lossless audio files.

Windows Media Player is a digital media player and

media library used for playing video, audio, and other

media. WMP can transcode media to desired formats. It

supports local playback and streaming playback.

Windows Media Player also has a Video Smoothingfeature that improves frame-rate through interpolation,

which can give a smoother playback on low-frame-rate

videos. WMP can use video overlays as well as video

interlacing, scaling, and improved color accuracy. The

application is also capable of media sharing, which

allows content to be streamed to and from devices

including personal computers and gaming consoles.

WMP supports streaming formats including HTTP,

RTSP, and in early versions MMS. With Windows

Media Services streaming media server, streaming media

can be generated. Multicast and unicast streams aresupported. Windows Media Player is the native

application for Windows Media Video, Windows Media

Audio, and Advanced Systems Format.

RealPlayer is a cross platform closed source media

player produced by RealNetworks, which plays a variety

of formats including MPEG-4. RealPlayer was one of

the first media players with the capability of streaming

media across the Internet. RealPlayer was popular in the

early years of the Internet, but has been surpassed by

Windows Media Player and Apple iTunes based onQuickTime formats. Streaming formats supported by

RealPlayer include HTTP, RTSP, and MMS. RealPlayer

has a feature for streaming media playback that allows a

user to pause the video clip without stopping the

buffering. RealPlayer is the native software for file

formats RealAudio, RealVideo, and RealMedia Shortcut

among others.

To showcase how some of these applications work with

streaming, basic streaming will be performed and

explained for the VLC program. For QuickTime andWindows Media Player, the receiver side of streaming

was tested by watching a video placed on a webpage.

Streaming media over a wireless network using VLC

Player was tested. The first thing to note was the benefit

of the same application providing the features to both

stream the video and watch it from a secondary

computer. VLC has a very easy to use streaming wizard


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that goes step by step through setting up the stream. The

following steps will show how to achieve the same

stream manually.

The process begins with opening VLC and selecting a

video file to open. The Streaming/Saving box is

checked. All of these directions mentioned so far can be

seen in Figure 11.

Figure 11 VLC Opening a file dialog

The next step is to click the settings tab in the above

figure. When the new dialog opens, there is an option to

display the stream locally, which was selected to make

playback features easier and more manageable. The

button to Stream the file is selected and the selected

Type is UDP. Other selectable types include RTP andHTTP. The network address of the secondary computer

to which the stream should be sent is entered in the

address field. A specific port may be chosen if desired,

VLC defaults to port 1234 for simplicity. There are also

options for transcoding the video and audio streams

including the method of encoding and the bitrate. All of

this is shown in the figure below. There is also the

ability to add a web interface by selecting it from the

VLC menu bar. This web interface allows the user on

the secondary computer to open a web browser and enter

the network address of the primary computer followed by the port being used. This web interface allows the

secondary computer to control the media stream for fast-

forwarding, rewinding, and jumping position.

Figure 12 VLC settings for streaming settings

With the chosen setup, features available to the

secondary computer include pausing and playing. If the

stream is paused, the stream stores up while it is not

playing. Once it starts playing, it continues from the

position it was paused at, it does not jump to the current

location that the stream is at on the primary computer.

For QuickTime, the streaming portion of the application

is separated and available as QuickTime StreamingServer. This program is available only with the Mac OS

X Server operating system. Since no Mac OS Server is

available, only the receiver side of using QuickTime for

streaming will be tested and documented.

Using QuickTime Player to view streaming media is

simple. After opening QuickTime Player, go to the File

menu and select Open URL. Type in the address for the

streaming movie and press Open. The streaming file will

open in a self-contained window.

Figure 13 QuickTime Open URL window

An alternative method to streaming a QuickTime movie

file is to embed or hyperlink it in a webpage, which just


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requires the file being loaded on the website. QuickTime

is fully integrated into the web browser so that if a .mov

file is loaded on a web page, it will play it back through

the browser with pause, play, and jump position

controls.

Windows Media Player works fairly similar with

streaming video as QuickTime. The software to streammedia self-contained is part of the Windows Server

setup. In a similar manner to QuickTime, a video can be

uploaded to a website, and the receiver end of the

streaming process can be seen. To accomplish this, a

.wmv file was loaded onto the users plaza account

through plaza.ufl.edu. Upon opening Windows Media

Player, the option for Open URL is selected from the

File menu. The address can then be entered into the

Open URL dialog and select OK as shown in Figure 14.

Figure 14 Windows Open URL window

The file buffers and then begins playing. Available

features from the receiver end include pause/play and

jump to position.

Another means of streaming video is using a service like

YouTube to upload media content to YouTubes web

server and play it back through a web interface. The first

step is to log into YouTube or create an account. Then

click the Upload link at the top of the webpage. Once a

file is selected, YouTube beings uploading the file and

the webpage will show the progress as seen in Figure

14.. Once it is loaded, YouTube provides a user with a

URL for the video and a code snippet to embed the

video into another webpage.

Figure 15 Uploading a video to YouTube

To view the video, a new web browser is opened. Then

the URL provided by YouTube is entered. The YouTube

page loads the webpage with the uploaded video.

Features available include play/pause and jump to

position.

V. CONCLUSIONThe improvements in the Internet as well as the quality

of video have allowed many changes to streaming

media. The demand for high quality streaming media

forces servers to push the limits for bandwidth and

throughput.

The impacts on the six features are significant in

themselves to consider. Video compression must find

more innovative ways to reduce size without affecting

media quality. For quality of service, congestion control

and minimizing delay will be paramount for the

continued improvements to streaming video quality to be

successful. Continuous media distribution services is

highly involved in the future of streaming media as the

Internet continues to be inefficient in terms of

application layer support for heavy data transferring.

Streaming servers are continually growing as companies

constantly need more streaming space for any video or


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media functions. As the quality of streaming video

continues to improve upon itself, media synchronization

will be more and more important for media playback as

there is the possibility for multiple video channels,

numerous audio channels, and slideshows to be

transmitted synchronously but a media synchronization

necessary to retain timing for data file. Protocol layers

will have one of the biggest impacts with the leastknowledge that it is being performed. More advanced

protocols will likely be made to work on top of existing

protocols like UDP and TCP.

It is clear that streaming media has had significant hikes

in importance over the past decade, especially in recent

years. The significance of factors, specifically bandwidth

and quality of service, was somewhat underestimated.

The high demand of streaming media such as internet

television has seen jumps in popularity that in turn

require a jump in available bandwidth as well as videoquality. This also brings into question the need for good

video compression to minimize data size, while retaining

quality.

One of the most unexpected impacts in recent years is

the impact that mobile media streaming has had. Mobile

Network Infrastructures will grow and expand rapidly

from the need for more and more coverage for wireless

networks or broadcast areas for multimedia services.

Mobile networks are one of the hot areas for future

expansion related to media streaming. With companiestying to unveil their 4G networks with improved features

like bandwidth, it is clear what is to come.

Streaming media over a server can be done efficiently

and cheaply given the proper software. VLC provided

the most adaptable software with the most available

features. VLC was the only application researched that

offered streaming media from the same console as the

video player. The applications simplicity and tutorial

help for its features more advanced than basic media

playback make it a great choice for streaming media.

The advent of websites like YouTube has opened up the

realm of media streaming to what is arguably the

simplest means possible. Since the web server takes care

of all the concerns of overhead, bandwidth, servers,

compression, and playback, it clearly creates an

environment with the least involved process for

implementation.

Applications with the capability of playback tend to have

similar features. Any of those used are of nearly equal

standing for quality and appeal. QuickTime and

Windows Media Player both had simple setups with the

easy option of streaming from a website. This kind of

feature is commonly overlooked. However, having the

media players console features to go with a videostreaming over http, is arguably a much more convenient

setup.

The need and use for media streaming has and will

continue to be on the rise. From personal enjoyment of

television shows streamed online to secure business web

meetings, the application for streaming media is still

unfolding.

REFERENCES[1] D. Wu, T. Hou, W. Zhu, Y.-Q. Zhang, J. Peha, "Streaming Video over

the Internet: Approaches and Directions," IEEE Transactions on Circuits

and Systems for Video Technology, Special Issue on Streaming Video,vol. 11, no. 3, pp. 282-300, March 2001.

[2] Grant and Meadows. (2009). Communication Technology Update andFundamentals 11

thEdition

[3] T. Wiegand and G. J. Sullivan, The H.264/AVC video codingstandard,IEEE Signal Processing Mag., vol. 24, pp. 148-153, 2007.

[4] Josh Lowensohn. (2008). YouTube to Offer Live Streaming This Year.http://news.cnet.com/8301-17939_109-9883062-2.html?tag=mncol

[5] Krasic, C. and Li, K. and Walpole, J., The case for streaming multimediawith TCP, Lecture Notes in Computer Science, Springer, 2001

[6] "Flash moves on to smart phones". Jonathan Fildes. BBC News.Retrieved from http://news.bbc.co.uk/2/hi/8287239.stm

[7] Fischer. (2010). Digital Video and Audio Broadcasting Technology.Chapter 29: Television over the Internet. 3rd Edition

[8] H.-H. Juan, H.-C. Huang, C. Huang, and T. Chiang, Scalable videostreaming over mobile WiMAX, in Proceedings of IEEE Int.

Symposium on Circuits and Systems (ISCAS), May 2007, pp.

34633466.

[9] Barkhuus, L. (2009) Television on the Internet: new practice, newviewers. San Diego:University of California

[10] Apple Inc. (2010). Apple QuickTime. Retrieved fromhttp://www.apple.com/quicktime/

[11] VideoLAN(July 2009). VLC media player - Open SourceMultimedia Framework and Player. Retrieved from

http://www.videolan.org/vlc/

[12] RealNetworks Inc. (2009). Media Player for mp3, Flash, Audio,Video | RealPlayer on Real.com. Retrieved from

http://www.real.com/

[13] Microsoft Windows (2010). Microsoft Windows Media YourDigital Entertainment Resource. Retrieved from

http://www.microsoft.com/windows/windowsmedia/default.mspx

Patrick Liby is the author. He is currently attending the University of Floridaand working on his Bachelors and Masters Degrees in Electrical Engineering.

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