This document provides an overview of multimedia networking. It discusses various multimedia applications including streaming stored video, voice over IP, and protocols for real-time conversational applications. It describes key characteristics of video and audio, different multimedia network applications, challenges of streaming stored video including continuous playback and client-side buffering. The document also discusses protocols for multimedia networking including UDP streaming, HTTP streaming, and adaptive HTTP streaming. It covers topics such as content distribution networks, voice over IP, limitations of VoIP including packet loss and end-to-end delay, and techniques for removing jitter like fixed and adaptive playout delay.
This slide-share contains details about the Multimedia networking and why it is important for us. Also this contains details regarding performance issues, applications, technical challenges and features of a multimedia system.
Pgp-Pretty Good Privacy is the open source freely available tool to encrypt your emails then you can very securely send mails to others over internet without fear of eavesdropping by cryptanalyst.
Real Life Applications of Distributed Systems:
1. Distributed Rendering in Computer Graphics
2. Peer-To-Peer Networks
3. Massively Multiplayer Online Gaming
This slide-share contains details about the Multimedia networking and why it is important for us. Also this contains details regarding performance issues, applications, technical challenges and features of a multimedia system.
Pgp-Pretty Good Privacy is the open source freely available tool to encrypt your emails then you can very securely send mails to others over internet without fear of eavesdropping by cryptanalyst.
Real Life Applications of Distributed Systems:
1. Distributed Rendering in Computer Graphics
2. Peer-To-Peer Networks
3. Massively Multiplayer Online Gaming
File Replication : High availability is a desirable feature of a good distributed file system and file replication is the primary mechanism for improving file availability. Replication is a key strategy for improving reliability, fault tolerance and availability. Therefore duplicating files on multiple machines improves availability and performance.
Replicated file : A replicated file is a file that has multiple copies, with each copy located on a separate file server. Each copy of the set of copies that comprises a replicated file is referred to as replica of the replicated file.
Replication is often confused with caching, probably because they both deal with multiple copies of data. The two concepts has the following basic differences:
A replica is associated with server, whereas a cached copy is associated with a client.
The existence of cached copy is primarily dependent on the locality in file access patterns, whereas the existence of a replica normally depends on availability and performance requirements.
Satynarayanana [1992] distinguishes a replicated copy from a cached copy by calling the first-class replicas and second-class replicas respectively
TCP/IP have 5 layers, whereas OSI model have 7 layers in its Model. TCP/IP is known for the secured connection and comunication. I have explained all functions and definitions of layers in TCP/IP Model
Parallel computing and its applicationsBurhan Ahmed
Parallel computing is a type of computing architecture in which several processors execute or process an application or computation simultaneously. Parallel computing helps in performing large computations by dividing the workload between more than one processor, all of which work through the computation at the same time. Most supercomputers employ parallel computing principles to operate. Parallel computing is also known as parallel processing.
↓↓↓↓ Read More:
Watch my videos on snack here: --> --> http://sck.io/x-B1f0Iy
@ Kindly Follow my Instagram Page to discuss about your mental health problems-
-----> https://instagram.com/mentality_streak?utm_medium=copy_link
@ Appreciate my work:
-----> behance.net/burhanahmed1
Thank-you !
Defines a framework for authentication service using the X.500 directory.It is the Repository of public-key certificates,Based on use of public-key cryptography and digital signatures.
A NOVEL ADAPTIVE CACHING MECHANISM FOR VIDEO ON DEMAND SYSTEM OVER WIRELESS M...IJCNCJournal
Video on Demand (VOD) system over the wireless mobile network is a system that provides video services to mobile clients. The main problem with these systems is the high service delay where the mobile clients have to wait to view their favorite movie. The importance of this paper is based on finding a solution on how to reduce the delay time in the VOD system. This paper introduces a novel caching mechanism named
Proxy Server Cache mechanism to tackle the issue of service delay. This delay happens when the broadcasting phase that is related to the first segment is missed by a client from the current broadcasting channels. In this mechanism, the video’s first segment is stored on a server of a stationary proxy type. The
delayed clients will directly acquire the first segment from the proxy server instead of waiting for the following broadcasting channel pertaining to the first segment. The proposed scheme ensuresobtaining the first segment from mobile clients when they arrive. Additionally, the performance of the proposed scheme is validated by applying the VOD system, which can involve the balancing mechanism to retain particular requests through to the local proxy server to provide a fair dissemination for these requests. The obtained result confirms that the proposed scheme reduces the time delay of the system in comparison with the best existing schemes. The results of the average time delay in the Proxy-Cache scheme is 179.2505
milliseconds when 10 clients arrive each minute (Client/minute), the average time delay is 140 milliseconds when the video lengths are 30, 60 and 90. Meanwhile, the failure probability for obtaining the first segment of the video remains zero when the number of arrived requests is set to2, 4, 6, 8 and 10.
File Replication : High availability is a desirable feature of a good distributed file system and file replication is the primary mechanism for improving file availability. Replication is a key strategy for improving reliability, fault tolerance and availability. Therefore duplicating files on multiple machines improves availability and performance.
Replicated file : A replicated file is a file that has multiple copies, with each copy located on a separate file server. Each copy of the set of copies that comprises a replicated file is referred to as replica of the replicated file.
Replication is often confused with caching, probably because they both deal with multiple copies of data. The two concepts has the following basic differences:
A replica is associated with server, whereas a cached copy is associated with a client.
The existence of cached copy is primarily dependent on the locality in file access patterns, whereas the existence of a replica normally depends on availability and performance requirements.
Satynarayanana [1992] distinguishes a replicated copy from a cached copy by calling the first-class replicas and second-class replicas respectively
TCP/IP have 5 layers, whereas OSI model have 7 layers in its Model. TCP/IP is known for the secured connection and comunication. I have explained all functions and definitions of layers in TCP/IP Model
Parallel computing and its applicationsBurhan Ahmed
Parallel computing is a type of computing architecture in which several processors execute or process an application or computation simultaneously. Parallel computing helps in performing large computations by dividing the workload between more than one processor, all of which work through the computation at the same time. Most supercomputers employ parallel computing principles to operate. Parallel computing is also known as parallel processing.
↓↓↓↓ Read More:
Watch my videos on snack here: --> --> http://sck.io/x-B1f0Iy
@ Kindly Follow my Instagram Page to discuss about your mental health problems-
-----> https://instagram.com/mentality_streak?utm_medium=copy_link
@ Appreciate my work:
-----> behance.net/burhanahmed1
Thank-you !
Defines a framework for authentication service using the X.500 directory.It is the Repository of public-key certificates,Based on use of public-key cryptography and digital signatures.
A NOVEL ADAPTIVE CACHING MECHANISM FOR VIDEO ON DEMAND SYSTEM OVER WIRELESS M...IJCNCJournal
Video on Demand (VOD) system over the wireless mobile network is a system that provides video services to mobile clients. The main problem with these systems is the high service delay where the mobile clients have to wait to view their favorite movie. The importance of this paper is based on finding a solution on how to reduce the delay time in the VOD system. This paper introduces a novel caching mechanism named
Proxy Server Cache mechanism to tackle the issue of service delay. This delay happens when the broadcasting phase that is related to the first segment is missed by a client from the current broadcasting channels. In this mechanism, the video’s first segment is stored on a server of a stationary proxy type. The
delayed clients will directly acquire the first segment from the proxy server instead of waiting for the following broadcasting channel pertaining to the first segment. The proposed scheme ensuresobtaining the first segment from mobile clients when they arrive. Additionally, the performance of the proposed scheme is validated by applying the VOD system, which can involve the balancing mechanism to retain particular requests through to the local proxy server to provide a fair dissemination for these requests. The obtained result confirms that the proposed scheme reduces the time delay of the system in comparison with the best existing schemes. The results of the average time delay in the Proxy-Cache scheme is 179.2505
milliseconds when 10 clients arrive each minute (Client/minute), the average time delay is 140 milliseconds when the video lengths are 30, 60 and 90. Meanwhile, the failure probability for obtaining the first segment of the video remains zero when the number of arrived requests is set to2, 4, 6, 8 and 10.
Learn about why Mesh may not be the answer for #WebRTC and why the better approach is to centralize and mix all the media in an MCU. As well as an in-depth explanation on the benefits of using an MCU. As presented by Chad Hart at WebRTC Expo V.
IBM Connect session
See also Call Speed Bandwidth Calculator spreadsheet
https://greenhouse.lotus.com/files/app#/file/a5817fad-b7b4-4fe5-94ce-00c11cf20ba4
Chaining Algorithm and Protocol for Peer-to-Peer Streaming Video on Demand Sy...ijwmn
As the various architectures and protocol have been implemented a true VoD system has great demand in the global users. The traditional VoD system does not provide the needs and demands of the global users. The major problem in the traditional VoD system is serving of video stream among clients is duplicated and streamed to the different clients, which consumes more server bandwidth and the client uplink bandwidth is not utilized and the performance of the system degrades. Our objective in this paper is to send one server stream sufficient to serve the many clients without duplicating the server stream. Hence we have proposed a protocol and algorithm that chains the proxy servers and subscribed clients utilize client’s uplink bandwidth such that the load on the server is reduced. We have also proved that less rejection ratio of the clients and better utilization of the buffer and bandwidth for the entire VoD system.
The International Journal of Engineering & Science is aimed at providing a platform for researchers, engineers, scientists, or educators to publish their original research results, to exchange new ideas, to disseminate information in innovative designs, engineering experiences and technological skills. It is also the Journal's objective to promote engineering and technology education. All papers submitted to the Journal will be blind peer-reviewed. Only original articles will be published.
The papers for publication in The International Journal of Engineering& Science are selected through rigorous peer reviews to ensure originality, timeliness, relevance, and readability.
Increasingly video content is becoming part of the enterprise web environment. The promise of HTML5's video element was supposed to solve a lot of the issues around serving videos to the web. But has it succeeded? And what of Accessibility?
This seminar will cover the state of video delivery on the web today, the issues, the promises, and, importantly, how to ensure that it all meets accessibility requirements.
Optimal Streaming Protocol for VoD Using Clients' Residual BandwidthIDES Editor
A true VoD system has tremendous demand in the
market. The existing VoD system does not cater the needs
and demands of the market. The major problem in the VoD
system is serving of clients with expected QoS is difficult. In
this paper, we proposed a protocol and algorithm that
chains the proxy servers and subscribed clients. Our
objective is to send one server stream and this stream should
be served to N asynchronous clients. The server bandwidth
is scarcity and on the client uplink bandwidth is
underutilized. In this protocol, we are using client’s residual
bandwidth such that the load on the server bandwidth is
reduced. We have proved that optimal utilization of the
buffer and bandwidth for the entire VoD system and also
less rejection ratio of the clients.
Building Cloud-ready Video Transcoding System for Content Delivery Networks (...Zhenyun Zhuang
GLOBECOM 2012
Video streaming traffic of both VoD (Video on
Demand) and Live is exploding. Various types of businesses
and many people are relying on video streaming to attract
customers/users and for other purposes. Given the vast number
of video stream formats (e.g., MP4, FLV) and transmission
protocols (e.g., HTTP, RTMP, RTSP) for supporting varying
types of playback terminals (particularly mobile devices such as
iphone/ipad and Android phones), video content providers often
need to transcode videos to multiple formats in order to stream
to different types of users.
Being time-sensitive and requiring high bandwidth, video
streaming exerts high pressure on underlying delivery networks.
Content Delivery Network (CDN) providers can help their
customers quickly and reliably distribute stream contents to end
users. In addition to distributing video streams, CDN providers
typically allow their customers to perform video transcoding on
CDN platforms. With the high volume of video streams and the
bursty transcoding workload, CDN providers are eager to deploy
elastic and optimized cloud-based transcoding platforms.
Multimedia Video transmission is over Wireless Local Area Networks is expected to be an important component of many
emerging multimedia applications. However, Wireless networks will always be bandwidth limited compared to fixed networks due to
background noise, limited frequency spectrum, and varying degrees of network coverage and signal strength One of the critical issues
for multimedia applications is to ensure that the Quality of Service (QoS) requirement to be maintained at an acceptable level. Modern
mobile devices are equipped with multiple network interfaces, including 3G/LTE WiFi. Bandwidth aggregation over LTE and WiFi
links offers an attractive opportunity of supporting bandwidth-intensive services, such as high-quality video streaming, on mobile
devices. Achieving effective bandwidth aggregation in wireless environments raises several challenges related to deployment, link
heterogeneity, Network congestion, network fluctuation, and energy consumption. In this work, an overview of schemes for video
transmission over wireless networks is presented where an acceptable quality of service (QoS) for video applications required realtime
video transmission is achieved
Live multimedia streaming and video on demand issues and challengeseSAT Journals
Abstract
Live Streaming and Video on Demand are the trending technologies nowadays over the internet. It provides the mechanism to
deliver multimedia content such as audio or video to the large number of audience. However internet based services face the
problem of QOS (Quality of Service) due to the instability faced in networks. Performance gets degraded when serving content to
large number of consumers. Despite following the modern architectural design, precise estimate of resources such as bandwidth
and server load is a challenging task.. In this paper we delve into the architectural and performance issues of running these kinds
of services. Our study demonstrates that the streaming architecture and Security issues are the challenges faced by these
technologies. Moreover resources such as bandwidth and design of networks degrade the quality of multimedia data delivered to
users. Thus in order to have best experience of streaming and Video on demand services, these issues must be addressed.
Keywords— Live Streaming, VOD (Video On Demand), P2P Streaming, Client-Server Model.
Kubernetes & AI - Beauty and the Beast !?! @KCD Istanbul 2024Tobias Schneck
As AI technology is pushing into IT I was wondering myself, as an “infrastructure container kubernetes guy”, how get this fancy AI technology get managed from an infrastructure operational view? Is it possible to apply our lovely cloud native principals as well? What benefit’s both technologies could bring to each other?
Let me take this questions and provide you a short journey through existing deployment models and use cases for AI software. On practical examples, we discuss what cloud/on-premise strategy we may need for applying it to our own infrastructure to get it to work from an enterprise perspective. I want to give an overview about infrastructure requirements and technologies, what could be beneficial or limiting your AI use cases in an enterprise environment. An interactive Demo will give you some insides, what approaches I got already working for real.
UiPath Test Automation using UiPath Test Suite series, part 3DianaGray10
Welcome to UiPath Test Automation using UiPath Test Suite series part 3. In this session, we will cover desktop automation along with UI automation.
Topics covered:
UI automation Introduction,
UI automation Sample
Desktop automation flow
Pradeep Chinnala, Senior Consultant Automation Developer @WonderBotz and UiPath MVP
Deepak Rai, Automation Practice Lead, Boundaryless Group and UiPath MVP
DevOps and Testing slides at DASA ConnectKari Kakkonen
My and Rik Marselis slides at 30.5.2024 DASA Connect conference. We discuss about what is testing, then what is agile testing and finally what is Testing in DevOps. Finally we had lovely workshop with the participants trying to find out different ways to think about quality and testing in different parts of the DevOps infinity loop.
Dev Dives: Train smarter, not harder – active learning and UiPath LLMs for do...UiPathCommunity
💥 Speed, accuracy, and scaling – discover the superpowers of GenAI in action with UiPath Document Understanding and Communications Mining™:
See how to accelerate model training and optimize model performance with active learning
Learn about the latest enhancements to out-of-the-box document processing – with little to no training required
Get an exclusive demo of the new family of UiPath LLMs – GenAI models specialized for processing different types of documents and messages
This is a hands-on session specifically designed for automation developers and AI enthusiasts seeking to enhance their knowledge in leveraging the latest intelligent document processing capabilities offered by UiPath.
Speakers:
👨🏫 Andras Palfi, Senior Product Manager, UiPath
👩🏫 Lenka Dulovicova, Product Program Manager, UiPath
Builder.ai Founder Sachin Dev Duggal's Strategic Approach to Create an Innova...Ramesh Iyer
In today's fast-changing business world, Companies that adapt and embrace new ideas often need help to keep up with the competition. However, fostering a culture of innovation takes much work. It takes vision, leadership and willingness to take risks in the right proportion. Sachin Dev Duggal, co-founder of Builder.ai, has perfected the art of this balance, creating a company culture where creativity and growth are nurtured at each stage.
Connector Corner: Automate dynamic content and events by pushing a buttonDianaGray10
Here is something new! In our next Connector Corner webinar, we will demonstrate how you can use a single workflow to:
Create a campaign using Mailchimp with merge tags/fields
Send an interactive Slack channel message (using buttons)
Have the message received by managers and peers along with a test email for review
But there’s more:
In a second workflow supporting the same use case, you’ll see:
Your campaign sent to target colleagues for approval
If the “Approve” button is clicked, a Jira/Zendesk ticket is created for the marketing design team
But—if the “Reject” button is pushed, colleagues will be alerted via Slack message
Join us to learn more about this new, human-in-the-loop capability, brought to you by Integration Service connectors.
And...
Speakers:
Akshay Agnihotri, Product Manager
Charlie Greenberg, Host
GDG Cloud Southlake #33: Boule & Rebala: Effective AppSec in SDLC using Deplo...James Anderson
Effective Application Security in Software Delivery lifecycle using Deployment Firewall and DBOM
The modern software delivery process (or the CI/CD process) includes many tools, distributed teams, open-source code, and cloud platforms. Constant focus on speed to release software to market, along with the traditional slow and manual security checks has caused gaps in continuous security as an important piece in the software supply chain. Today organizations feel more susceptible to external and internal cyber threats due to the vast attack surface in their applications supply chain and the lack of end-to-end governance and risk management.
The software team must secure its software delivery process to avoid vulnerability and security breaches. This needs to be achieved with existing tool chains and without extensive rework of the delivery processes. This talk will present strategies and techniques for providing visibility into the true risk of the existing vulnerabilities, preventing the introduction of security issues in the software, resolving vulnerabilities in production environments quickly, and capturing the deployment bill of materials (DBOM).
Speakers:
Bob Boule
Robert Boule is a technology enthusiast with PASSION for technology and making things work along with a knack for helping others understand how things work. He comes with around 20 years of solution engineering experience in application security, software continuous delivery, and SaaS platforms. He is known for his dynamic presentations in CI/CD and application security integrated in software delivery lifecycle.
Gopinath Rebala
Gopinath Rebala is the CTO of OpsMx, where he has overall responsibility for the machine learning and data processing architectures for Secure Software Delivery. Gopi also has a strong connection with our customers, leading design and architecture for strategic implementations. Gopi is a frequent speaker and well-known leader in continuous delivery and integrating security into software delivery.
The Art of the Pitch: WordPress Relationships and SalesLaura Byrne
Clients don’t know what they don’t know. What web solutions are right for them? How does WordPress come into the picture? How do you make sure you understand scope and timeline? What do you do if sometime changes?
All these questions and more will be explored as we talk about matching clients’ needs with what your agency offers without pulling teeth or pulling your hair out. Practical tips, and strategies for successful relationship building that leads to closing the deal.
Transcript: Selling digital books in 2024: Insights from industry leaders - T...BookNet Canada
The publishing industry has been selling digital audiobooks and ebooks for over a decade and has found its groove. What’s changed? What has stayed the same? Where do we go from here? Join a group of leading sales peers from across the industry for a conversation about the lessons learned since the popularization of digital books, best practices, digital book supply chain management, and more.
Link to video recording: https://bnctechforum.ca/sessions/selling-digital-books-in-2024-insights-from-industry-leaders/
Presented by BookNet Canada on May 28, 2024, with support from the Department of Canadian Heritage.
JMeter webinar - integration with InfluxDB and GrafanaRTTS
Watch this recorded webinar about real-time monitoring of application performance. See how to integrate Apache JMeter, the open-source leader in performance testing, with InfluxDB, the open-source time-series database, and Grafana, the open-source analytics and visualization application.
In this webinar, we will review the benefits of leveraging InfluxDB and Grafana when executing load tests and demonstrate how these tools are used to visualize performance metrics.
Length: 30 minutes
Session Overview
-------------------------------------------
During this webinar, we will cover the following topics while demonstrating the integrations of JMeter, InfluxDB and Grafana:
- What out-of-the-box solutions are available for real-time monitoring JMeter tests?
- What are the benefits of integrating InfluxDB and Grafana into the load testing stack?
- Which features are provided by Grafana?
- Demonstration of InfluxDB and Grafana using a practice web application
To view the webinar recording, go to:
https://www.rttsweb.com/jmeter-integration-webinar
3. INTRODUCTION
Multimedia-Technology that enables humans to use computers
capable of processing textual data, audio and video, still pictures, and
animation.
Today, people not only use the internet to watch movies but also to
upload videos ( YouTube), make internet calls (Skype and Google
talk).
By the end of the decade and with emerging technologies like 4G
and Wi-Fi access, Internet will not only provide phone service for
less money, but will also provide numerous value-added services,
such as video conferencing, online directory services, and voice
messaging
MULTIMEDIA NETWORKING
3
4. Multimedia Networking Applications
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.
MULTIMEDIA NETWORKING
4
5. Properties of Audio
Basic encoding technique of converting analog audio into digital audio is called pulse
code modulation (PCM)
Examples of Sampling Rates
Speech encoding (uses PCM) ;8000 samples per second and 8 bits per sample
Audio compact disk (also uses PCM) ; 44,100 samples per second and 16 bits per
sample
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
Note that digital audio has lower bandwidth compared to video although users are more
sensitive to audio malfunctions than video.
MULTIMEDIA NETWORKING
5
6. TYPES OF MULTIMEDIA NETWORK
APPLICATIONS
Multimedia applications are classified into three broad categories:
Streaming Stored Video/Audio
◦Underlying medium is prerecorded video, such as a movie or a prerecorded
user generated video (on YouTube).
◦These are stored on servers, and users send requests to the servers to view the
videos
Features of stored streaming video
Streaming
Interactivity
Continuous Playout
MULTIMEDIA NETWORKING
6
7. 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, QQ, and Google Talk
Two Important Application service requirement for CVVOIP
Timing Considerations
Audio and video conversational applications are highly delay-sensitive
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
MULTIMEDIA NETWORKING
7
8. Streaming Live Audio &
Video
Similar to traditional broadcast radio and television, except that transmission
takes place over the Internet such as a live sporting event or news
Because the event is live, delay can also be an issue, although the timing
constraints are much less stringent than those for conversational voice
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
8
9. Streaming Stored Video
For streaming video applications, prerecorded videos are placed on servers, and
users send requests to these servers to view the videos on demand.
User may watch the video from beginning to end without interruption, may stop
watching the video well before it ends, or interact with the video by pausing or
repositioning to a future or past scene
Streaming systems can be categorized as:
UDP Streaming
HTTP Streaming
Adaptive HTTP Streaming
MULTIMEDIA NETWORKING
9
10. Cumulative data
Streaming Stored Video
1. video
recorded (e.g., 30
frames/sec)
2. video
sent
network delay
(fixed in this
example)
3. video received,
played out at client
(30 frames/sec)
time
streaming: at this time, client
playing out early part of video,
while server still sending later
part of video
MULTIMEDIA NETWORKING
10
11. Streaming Stored Video: Challenges
Continuous Playout constraint: once client Playout begins, playback must
match original timing
… but network delays are variable (jitter), so will need client-side buffer
to match Playout requirements
Other challenges:
Client interactivity: pause, fast-forward, rewind, jump through video
Video packets may be lost, retransmitted
MULTIMEDIA NETWORKING
11
12. Client-side Buffering, Playout
Buffer fill level, Q(t)
Playout rate,
e.g., CBR r
Variable fill
rate, x(t)
Client application
buffer, size B
Video Server
Client
1. Initial fill of buffer until Playout begins
2. Playout begins a
3. Buffer fill level varies over time as fill rate x(t) varies and playout rate r
is constant
MULTIMEDIA NETWORKING
12
13. Client-side Buffering, Playout
buffer fill level, Q(t)
playout rate,
e.g., CBR r
variable fill
rate, x(t)
client application
buffer, size B
video server
Playout buffering: average fill rate (x), playout rate (r):
x < r: buffer eventually empties (causing freezing of video playout until buffer
again fills)
x > r: buffer will not empty, provided initial playout delay is large enough to absorb
variability in x(t)
Initial playout delay tradeoff: buffer starvation less likely with larger delay, but
larger delay until user begins watching
MULTIMEDIA NETWORKING
13
14. Advantages of client-Side Buffering
Client side buffering can absorb variations in server-to-client delays;
particular piece of video data is delayed, as long as it arrives before the
reserve of received-but-not yet played video is exhausted, this long delay
will not be noticed.
If the server-to-client bandwidth briefly drops below the video
consumption rate, a user can continue to enjoy continuous playback, as
long as the client application buffer does not become completely drained.
MULTIMEDIA NETWORKING
14
15. UDP Streaming
UDP streaming, ..Server transmits video at a rate that matches the client’s video
consumption rate by clocking out the video chunks over UDP at a steady rate.
Drawbacks of UDP Streaming
Its unpredictable and varying amount of available bandwidth between server and
client, constant-rate UDP streaming can fail to provide continuous playout
It requires a media control server, such as an RTSP server, to process client-toserver interactivity requests and to track client state
Third drawback is that many firewalls are configured to block UDP traffic,
preventing the users behind these firewalls from receiving UDP video
MULTIMEDIA NETWORKING
15
16. Streaming Multimedia: HTTP
Multimedia file retrieved via HTTP GET
Send at maximum possible rate under TCP
variable rate,
x(t)
Video
file
TCP send buffer
TCP receive
buffer
Server
Application playout
buffer
Client
Fill rate fluctuates due to TCP congestion control, retransmissions (in-order
delivery)
Larger playout delay: smooth TCP delivery rate
HTTP/TCP passes more easily through firewalls and NATS
MULTIMEDIA NETWORKING
16
17. Streaming Multimedia: DASH
DASH: Dynamic, Adaptive Streaming over HTTP
Server:
Divides video file into multiple chunks
Each chunk stored, encoded at different rates
Manifest file: provides URLs for different chunks
Client:
Periodically measures server-to-client bandwidth
Consulting manifest, requests one chunk at a time
◦Chooses maximum coding rate sustainable given current bandwidth
◦Can choose different coding rates at different points in time (depending on
available bandwidth at time)
MULTIMEDIA NETWORKING
17
18. DASH Cont.……………
DASH: Dynamic, Adaptive Streaming over HTTP
“intelligence” at client: client determines
◦When to request chunk (so that buffer starvation, or overflow does
not occur)
◦What encoding rate to request (higher quality when more bandwidth
available)
◦Where to request chunk (can request from URL server that is “close”
to client or has high available bandwidth)
MULTIMEDIA NETWORKING
18
19. Content Distribution Networks (CDN)
Challenge: How to stream content (selected from millions of videos) to hundreds
of thousands of simultaneous and distributed users?
Option 1: Build single, large “Mega-Server”
Drawbacks
◦Single point of failure
◦Point of network congestion
◦Long path to distant clients
◦Multiple copies of video sent over outgoing link
This solution: Use of Content Distribution Networks (CDNs)
MULTIMEDIA NETWORKING
18
20. Content Distribution Networks
Challenge: How to stream content (selected from millions of videos) to hundreds of
thousands of simultaneous users?
Option 2: Store/Serve multiple copies of videos at multiple geographically
distributed sites (CDN)
Enter deep: Push CDN servers deep into many access networks
◦Close to users
◦Used by Akamai, 1700 locations
Bring home: smaller number (10’s) of larger clusters in POPs near (but not within)
access networks
◦Used by Limelight
MULTIMEDIA NETWORKING
20
21. CDN Cont.……………
CDN manages servers in multiple geographically distributed
locations.
Stores copies of the videos (and other types of Web content,
documents, images, and audio).
Direct each user request to a CDN location that will provide
the best user experience.
MULTIMEDIA NETWORKING
21
22. CDN: Simple Operation
Bob (client) requests video http://netcinema.com/6Y7B23V
Video stored in CDN at http://KingCDN.com/NetC6y&B23V
1. Bob gets URL for for video
http://netcinema.com/6Y7B23V
from netcinema.com
web page
2
1
6. request video from
KINGCDN server,
streamed via HTTP
netcinema.com
netcinema’s
authorative DNS
2. resolve http://netcinema.com/6Y7B23V
via Bob’s local DNS
5
3. netcinema’s DNS returns URL
http://KingCDN.com/NetC6y&B23V
3
4
4&5. Resolve
http://KingCDN.com/NetC6y&B23
via KingCDN’s authoritative DNS,
which returns IP address of KIingCDN
server with video
KingCDN
authoritative DNS
KingCDN.com
MULTIMEDIA NETWORKING
23. CDN Cluster Selection Strategy
Challenge: how does CDN DNS select “good” CDN node to stream to client
◦Pick CDN node geographically closest to client
◦Pick CDN node with shortest delay (or min # hops) to client (CDN nodes
periodically ping access ISPs, reporting results to CDN DNS)
◦IP anycast
Alternative: Let client decide - give client a list of several CDN servers
◦Client pings servers, picks “best”
◦Netflix approach
MULTIMEDIA NETWORKING
23
24. Case study: Netflix
30% downstream US traffic in 2011
owns very little infrastructure, uses 3rd party services:
own registration, payment servers
Amazon (3rd party) cloud services:
◦Netflix uploads studio master to Amazon cloud
◦create multiple version of movie (different endodings) in cloud
◦Upload versions from cloud to CDNs
◦Cloud hosts Netflix web pages for user browsing
Three 3rd party CDNs host/stream Netflix content: Akamai, Limelight, Level-3
MULTIMEDIA NETWORKING
24
25. Case Study: Netflix
upload copies of multiple
versions of video to CDNs
Amazon cloud
Netflix registration,
accounting servers
2. Bob browses
Netflix video
2
3. Manifest file
returned for
requested video
Akamai CDN
Limelight CDN
3
1
1. Bob manages
account
Netflix
4. DASH streaming
MULTIMEDIA NETWORKING
Level-3 CDN
25
26. Voice-Over-IP
Involves taking analog audio signals and converting them into digital data
which can be transmitted over the Internet.
Voice telephony over the internet
Example: the sender generates bytes at a rate of 8,000 bytes per second; every
20 msecs the sender gathers these bytes into a chunk. A chunk and a special
header are encapsulated in a UDP segment.
UDP segment is sent every 20 msecs.
When each packet manages to get to the receiver with a continuous end-to-end
delay, then packets arrive at the receiver occasionally every 20 msecs.
Receiver plays back each chunk as it is received
Problem: some packets get lost and don’t have the same end-to-end delay
MULTIMEDIA NETWORKING
26
27. LIMITATIONS OF VoIP: PACKET LOSS
The UDP segment is encapsulated in an IP datagram.
While datagram moves through network, it goes through router buffers as it
waits for transmission on outbound links.
Problem: One or more of the buffers in the path from sender to receiver may
be full therefore arriving IP datagram may be discarded (not arrive to receiver)
This loss can be removed by using TCP rather than UDP
Problem: This retransmission increases end-to-end delay
Upside: Loss of packets between 1 and 20 can be tolerated depending on how
loss is obscured at receiver and voice encoding
MULTIMEDIA NETWORKING
27
28. END-TO-END DELAY
Accumulation of transmission, processing, and queuing delays in routers;
propagation delays in links; and end-system processing delays.
For real-time conversational applications, end-to-end delays;
Under 150 msecs are not known to a human listener (good)
Between 150 and 400 msecs can be acceptable though not good (bad)
Above 400 msecs can be problematic in voice conversations. (really bad)
Packets delayed for a very long time may be effectively lost by receiver
MULTIMEDIA NETWORKING
28
29. PACKET JITTER
During end-to-end delays there are varying queuing delays that a packet faces in the network’s
routers. As a result of these delays, the time between sending and receiving a packet can fluctuate
from packet to packet which is referred to as packet jitter.
Difference between the two consecutive packets may be more or less than 20msec
Once jitters are ignored by receivers and chunks are played out as they come, then the audio
quality can become meaningless to the receiver.
Upside: jitters can be removed by using sequence numbers, timestamps, and a playout delay
MULTIMEDIA NETWORKING
29
30. REMOVING JITTERS: FIXED PLAYOUT
DELAY
Receiver attempts to play out each chunk exactly q msecs after the chunk is
generated.
Chunk is timestamped at the sender at time t
Receiver plays out the chunk at time t + q, if chunk has arrived by stipulated
time.
Packets that arrive late are lost.
Variation of q
Large q: less packet loss
Small q: Better conversational experience
q much smaller than 400msec: high packet loss
MULTIMEDIA NETWORKING
30
31. ADAPTIVE PLAYOUT DELAY
With large initial playout delays, most packets will make their deadlines hence there will be
slight loss;
Problem: in conversational services such as VoIP, long delays may become annoying and
intolerable so playout delay should be minimized so as to decrease the loss percentage.
So as to address the above problem, there should be an estimate of the network delay and
the variance of the network delay.
Sender’s silent periods should be condensed and elongated
So as to estimate the network delay; the algorithm below is used by the receiver
di = (1 – u) di–1 + u (ri – ti)
Average
network delay
after ith packet
fixed
constant, e.g.
0.1
time received time sent
(timestamp)
end-to-end delay of ith packet
MULTIMEDIA NETWORKING
31
32. RECOVERING FROM PACKET LOSS
Retransmitting lost packets may not be practical in a real-time conversational application such as VoIP and
may not easily be accomplished on time for the conversation to remain understandable.
VoIP applications use forward error correction (FEC) and interleaving to anticipate loss.
FEC
Add redundant information to original packet stream which is used to reconstruct exact or approximate
version of lost packet.
Simple FEC
when a group on n chunks are sent, add redundant encoded chunk by exclusive OR-ing the n original
chunks
If a packet from a group of n + 1 packets is lost, the receiver is able reconstruct the lost packet.
Problem: If two or more packets in a group are lost, the receiver cannot reconstruct the lost packets.
Transmission rate will be increased by a factor of 1/n if n + 1 group size is small.
Play out delay increased because receiver has to wait for the whole group of packets to begin play out
MULTIMEDIA NETWORKING
32
33. RECOVERING FROM PACKET LOSS
Another mechanism under FEC
send a lower-resolution audio stream as the redundant information.
For example
Stream PCM encoding at 64 kbps (nominal stream) with low resolution audio, a GSM
encoding at 13kbps)
sender creates the nth packet when it takes nth chunk from the nominal stream and appends it to
the (n – 1)st redundant chunk (low bit rate)
Receiver conceals loss by playing out low bit rate chunk if nonconsecutive packet loss occurs.
Low bit rate chunks give less quality than nominal chunks.
MULTIMEDIA NETWORKING
33
34. RECOVERING FROM PACKET LOSS
Interleaving
Before transmission, sender breaks down the audio parts and separates packets by a certain
distance in stream.
This lowers the effects of packet loss because loss of packet results in smaller gaps in the
reconstructed stream compared to if the gaps is large
Error concealment
Produce replacement for lost packet that is similar to the original
Works because audio signals are usually almost the same in short term.
Works for small loss rates (less than 15 percent) and for small packets (4–40 msecs).
Packet repetition can also be used for recovery from packet loss where lost packets are replaced
with copies that arrive immediately before loss.
MULTIMEDIA NETWORKING
34
35. VoIP APPLICATION: SKYPE
It’s a proprietary application
control and media packets are encrypted
Sends audio and video packets over UDP but control packets are sent over TCP as well as
media packets once UDP streams are blocked by firewalls.
Uses FEC for packet loss recovery (voice and video streams) sent over UDP.
Uses P2P where peers communicate with each other in real time.
Important for also determining user location
Peers are organized into a hierarchical overlay network (super peer or an ordinary peer).
Keeps an index that maps Skype usernames to current IP addresses (and port numbers)
Index is distributed over the super peers.
MULTIMEDIA NETWORKING
35
36. MULTIMEDIA NETWORKING
PROTOCOLS
There are some protocols that are used to support real time traffic over the
internet. The following named protocols will be discussed:
RTP (Real Time Protocol)
RTCP (Real Time Control Protocol)
SIP (Session Initiation protocol)
RTSP (Real Time Streaming Protocol)
MULTIMEDIA NETWORKING
36
37. RTP
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.
The Audio-Video Transport Working Group of the
Internet Engineering Task Force (IETF) developed RTP and was first published
in 1996 as RFC 1889 but this was later superseded by RFC 3550 in 2003.
MULTIMEDIA NETWORKING
37
38. RTP PROTOCOL COMPONENTS
The following information is needed to send streaming data:
Timestamps (for synchronization),
Sequence numbers (for packet loss and reordering detection)
The payload format which indicates the encoded format of the data.
Frame Indication (this marks the beginning and end of each frame).
Source identification (identifies the originator of the frame)
MULTIMEDIA NETWORKING
38
39. RTP runs on UDP Cont’d
MULTIMEDIA NETWORKING
39
41. RTP – Time Stamp, Sequence Number and
Source Identifier
Sequence number field.
The sequence number field is 16 bits long. The sequence number increments
by one for each RTP packet sent, and may be used by the
Timestamp field.
The timestamp field is 32 bits long. It reflects the sampling instant of the first
byte in the RTP data packet.
Timestamp clock rate for video is 90,000 Hz and the timestamp clock rate for
audio 8000 Hz.
Synchronization source identifier (SSRC).
The SSRC field is 32 bits long. It identifies the source of the RTP stream.
Typically, each stream in an RTP session has a distinct SSRC.
MULTIMEDIA NETWORKING
41
42. Why RTP can not use TCP
TCP forces the receiver application to wait for retransmission(s) in case of
packet loss, which causes large delays.
TCP cannot support multicast.
TCP headers are larger than a UDP header (40 bytes for TCP compared to 8
bytes for UDP).
TCP doesn’t contain the necessary timestamp and encoding information needed
by the receiving application.
MULTIMEDIA NETWORKING
42
43. 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:
Feedback on performance of the application and the network
Correlation and synchronization of different media streams generated by the same sender (e.g.
combined audio and video)
The way to convey the identity of sender for display on a user interface
MULTIMEDIA NETWORKING
43
44. 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; I
Intermedia synchronization, which enables the synchronization of separately
transmitted audio and video streams.
MULTIMEDIA NETWORKING
44
45. 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 …)
MULTIMEDIA NETWORKING
45
47. Setting up a call to a known IP Address
MULTIMEDIA NETWORKING
47
48. SIP Cont’d
User Agent
User agent client (end-device, calling party)
User agent server (end-device, called party)
SIP Proxy Server
An intermediary entity that acts as both a server and a client for the purpose of
making requests on behalf of other clients. A proxy server primarily plays the
role of routing, which means its job is to ensure that a request is sent to another
entity "closer" to the targeted user.
SIP Registrar
Maintains user’s whereabouts in location database (location service). It is a
server that accepts REGISTER requests and places the information it receives
in those requests into the location service for the domain it handles.
MULTIMEDIA NETWORKING
48
49. 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.
MULTIMEDIA NETWORKING
49
51. Network Support for Multimedia
Observed how application-level mechanisms are used by multimedia
applications to improve multimedia performance.
Also learned how content distribution networks and P2P overlay networks
provide a system-level approach for delivering multimedia content.
Such techniques and approaches are all designed to be used in today’s besteffort Internet.
Internet provides only a single, best-effort class of service.
MULTIMEDIA NETWORKING
51
52. Network Support for Multimedia
There are three approaches
Making the best of best-effort service.
Differentiated service.
Per-connection Quality-of-Service (QoS) Guarantees.
MULTIMEDIA NETWORKING
52
53. Dimensioning Best-Effort Networks
Approaches to improving the quality of networked multimedia
Provide enough link capacity throughout the network
Such that network congestion, and its consequent packet delay and loss never occur
No queuing delay or loss.
Drawbacks:
Bandwidth Provisioning.
o How much capacity to provide at network links
Network Dimensioning
o How to design a network topology
MULTIMEDIA NETWORKING
53
54. Providing Enough Capacity
Issues to be addressed in order to predict application-level performance between two
network end points;
Models of traffic demand between network end points.
Well-defined performance requirements.
Models to predict end-to-end performance for a given workload model, and
techniques to find a minimal cost bandwidth allocation that will result in all user
requirements being met.
MULTIMEDIA NETWORKING
54
55. Providing Multiple Classes of Service
Simplest development to the one-size-fits-all best-effort service model is;
To divide traffic into classes,
Provide different levels of service to these different classes of traffic.
motivating scenarios.
H1
H2
H3
R1
R1 output
interface
queue
R2
1.5 Mbps link
MULTIMEDIA NETWORKING
H4
55
56. Scenario 1: mixed HTTP and VoIP
Example: 1Mbps VoIP, HTTP share 1.5 Mbps link.
HTTP bursts can congest router, cause audio loss
In the best-effort Internet-transmitted in a first-in-first-out (FIFO) order
How should we solve this potential problem?
want to give priority to audio over HTTP
R1
R2
Insight 1: Packet marking allows a router to distinguish among packets belonging to different classes of traffic .
MULTIMEDIA NETWORKING
56
57. Scenario 1: Mixed HTTP and VoIP
Suppose: Router is configured to give priority to packets marked as belonging to the 1 Mbps audio
application.
What happens if the audio application starts sending packets at a rate of 1.5 Mbps or higher?
Similarly; If multiple applications (for example, multiple audio calls), were sharing the link’s bandwidth;
they too could collectively starve the FTP session.
Ideally; Need a degree of isolation among classes of traffic so that one class of traffic can be protected from
the other.
Insight 2: It is desirable to provide a degree of traffic isolation among classes so that one class is not
adversely affected by another class of traffic that misbehaves.
MULTIMEDIA NETWORKING
57
58. Traffic isolation Mechanisms
Traffic policing
Policing mechanism ensures that these criteria are indeed observed.
Policed application misbehaves-policing mechanism takes some action.
Traffic entering the network conforms to the criteria.
Marking & Policing - network edge
1 Mbps
phone
R1
R2
1.5 Mbps link
packet marking and policing
MULTIMEDIA NETWORKING
58
59. Traffic isolation Mechanisms; Cont.
Link-level packet-scheduling
Explicitly allocate a fixed amount of link bandwidth to each class
A class only uses the amount of bandwidth that has been allocated
Wastes bandwidth
1 Mbps
phone
1 Mbps logical link
R1
R2
1.5 Mbps link
0.5 Mbps logical link
Insight 3: While providing isolation among classes or flows, it is desirable to use resources (for example, link
bandwidth and buffers) as efficiently as possible.
MULTIMEDIA NETWORKING
59
60. Scheduling Mechanisms
Link-scheduling discipline is the manner in which queued packets are selected for transmission on the link.
First-In-First-Out (FIFO)
o Packets are send in order of arrival to queue.
Packet-discarding policy:
If there is not space for the arriving packets, determines whether;
o the packet will be dropped (lost) - tail drop
o other packets will be removed from the queue to make space for the arriving packet
◦ priority: drop/remove on priority basis
◦ random: drop/remove randomly
packet
arrivals
queue
(waiting area)
link
(server)
MULTIMEDIA NETWORKING
packet
departures
60
61. Scheduling Mechanisms: cont.
Priority Queuing
Packets arriving at the output link are classified into priority classes at the output queue.
Packet’s priority depends:
o Explicit marking ,destination IP address or destination port number.
Each priority class typically has its own queue.
Choosing a packet to transmit, from the highest priority class
The choice among packets in the same priority class is typically done in a FIFO manner
2
high priority queue
(waiting area)
1
5
4
3
arrivals
arrivals
departures
packet in
service
classify
low priority queue
(waiting area)
link
(server)
1
4
2
3
5
departures
1
MULTIMEDIA NETWORKING
3
2
4
5
61
62. Scheduling Mechanisms: Cont.
Round Robin
Packets are sorted into classes as with priority queuing.
Round Robin scheduler alternates service among the classes.
A work-conserving queuing discipline will never allow the link to remain idle whenever there are packets (of
any class) queued for transmission.
2
1
5
4
3
arrivals
packet in
service
1
2
3
4
5
departures
1
3
3
4
MULTIMEDIA NETWORKING
5
62
65. The Leaky Bucket - Implementation
Before a packet is transmitted into the network, it must first remove a token from the token
bucket.
If the token bucket is empty, the packet must wait for a token.
The maximum burst size for a leaky-bucket policed flow is b packets.
The token generation rate is r
The maximum number of packets that can enter the network of any interval of time of length t
is rt + b.
MULTIMEDIA NETWORKING
65
66. Diffserv
Provides service differentiation
The ability to handle different classes of traffic in different ways within the Internet in a
scalable manner.
Need for scalability arises from:
The fact that millions of simultaneous source-destination traffic flows may be present at a
backbone router.
The Diffserv architecture consists of two sets of functional elements:
Edge functions: packet classification and traffic conditioning.
At the incoming edge of the network arriving packets are marked.
Marks packets as in-profile and out-profile
Core function: forwarding.
DS-marked packet arrives at a Diffserv MULTIMEDIA NETWORKING
capable router,
66
67. Diffserv Architecture
The packet is forwarded onto its next hop according to the per-hop behavior (PHB) associated
with that packet’s class.
Buffering and scheduling based on marking at edge.
Preference given to in-profile packets over out-of-profile packets
MULTIMEDIA NETWORKING
67
68. PHB - Per-Hop Behavior
Key component of the Diffserv architecture
PHB is a description of the externally observable forwarding behavior of a Diffserv node applied to a particular Diffserv
behavior aggregate”
Important considerations embedded within:
A PHB can result in different classes of traffic receiving different performance/behaviors
It does not mandate any particular mechanism for achieving these behaviors.
Differences in performance must be observable and hence measurable.
Two PHBs have been defined:
Expedited forwarding
• PHB specifies that the departure rate of a class of traffic from a router must equal or exceed a configured rate.
Assured forwarding
• Divides traffic into four classes, where each AF class is guaranteed to be provided with some minimum amount of
bandwidth and buffering.
MULTIMEDIA NETWORKING
68
70. Per-Connection (QoS) Guarantees: Cont.
If a call is to be guaranteed a given quality of service once it begins, The Network
mechanisms and protocols needed are:
Resource reservation. Process to guarantee that a call will have the resources
(link bandwidth, buffers) needed to meet its desired QoS is to explicitly allocate
those resources to the call.
Call admission. If resources are to be reserved, then the network must have a
mechanism for calls to request and reserve resources.
Call setup signaling. The call admission process described requires that a call be
able to reserve sufficient resources at each and every network router on its sourceto-destination path to ensure that its end-to-end QoS requirement is met.
MULTIMEDIA NETWORKING
70
Insight 3: While providing isolation among classes or flows, it is desirable to use resources (for example, link bandwidth and buffers) as efficiently as possible.