The document summarizes different techniques for streaming media, including HTTP Live Streaming (HLS) and Real Time Streaming Protocol (RTSP). It then describes the architecture and process of HLS in detail over 4 steps:
1) The media is encoded and segmented by the encoder.
2) The segmenter splits the encoded media into short chunks and encrypts the segments if needed.
3) The distribution system stores the segments and playlists on an HTTP server.
4) The client downloads playlists and segments through HTTP and plays the media.
The document also compares RTSP and HLS, noting advantages of HLS like reliability, firewall traversal, and use on YouTube. It surveys different video-
Streaming stored video allows video playback to begin before the entire file has been downloaded. It works by storing/buffering portions of the video at the client. There are three main types of streaming: UDP streaming, HTTP streaming, and adaptive HTTP streaming. HTTP streaming is most common today and works by transmitting the video file over HTTP as quickly as the network allows. Adaptive streaming addresses limitations of standard HTTP streaming by allowing clients to switch between multiple encodings of the video to adapt to changing network conditions.
Streaming stored video involves pre-recording videos on servers and allowing users to view them on demand. There are three main types of streaming: UDP, HTTP, and adaptive HTTP streaming. HTTP streaming is most common, storing video as HTTP files that are transmitted over TCP. This allows video to pass through firewalls more easily. Adaptive streaming divides video into multiple encoded quality levels to adapt to changing bandwidth conditions. The client measures bandwidth and selects the highest quality available. All streaming requires client-side buffering to smooth out network delays and jitter.
Different Types of Live Video Streaming Protocols and the Way They WorkChetu
The document discusses several popular live streaming protocols: HTTP Live Streaming (HLS), Real-Time Messaging Protocol (RTMP), and Web Real-Time Communication (WebRTC). HLS was developed by Apple and has wide support across devices and platforms, making it the most widely used protocol. RTMP was originally created by Macromedia and is commonly used for ingesting live streams. WebRTC allows for real-time audio and video transfer between browsers without plugins. Secure Reliable Transport (SRT) is an open-source protocol developed by Haivision that provides security, reliability, and low latency streaming. Choosing the right protocol depends on factors like compatibility, security, quality, latency, codec requirements, and platform
Inlet Technologies - Powering Smooth StreamingSematron UK Ltd
The white paper discusses Microsoft's Smooth Streaming technology which allows for adaptive bitrate streaming of video over HTTP. It works by encoding video into small chunks at multiple bitrates, allowing clients to switch between chunks to maintain high quality playback based on bandwidth. Inlet provides products that help content creators generate and deliver Smooth Streaming video assets for live and on-demand streaming.
A review over multimedia and how to share multimedia data between clients and servers.
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This document discusses key areas in streaming video over the internet including video compression techniques, application-layer quality of service control, continuous media distribution services, streaming servers, media synchronization mechanisms, and protocols for streaming video. It covers nonscalable and scalable video encoding, bandwidth and delay requirements, congestion and rate control, error control, streaming server components, and protocols like RTP, RTSP, and SIP. The document also outlines future directions in these areas like improved video compression schemes, evaluating rate control effectiveness, building scalable distribution infrastructure, and enhancing streaming server and protocol functionality.
1. The document proposes a new protocol called Connection Initiation Protocol (CIP) to improve performance for real-time streaming data transmission over networks.
2. CIP would encapsulate basic connection management tasks and be used alongside RTP for streaming data and RTCP for control signaling.
3. Unlike SIP-based systems, using CIP would reduce network traffic and improve performance for real-time data delivery by establishing a virtual communication channel between parties before transmitting streaming data and not involving the CIP server for RTP/RTCP transmission.
EE4414 Multimedia Communication Systems IIFranZEast
This experiment involves video streaming over the internet using different protocols and applications. Students will stream video from a server using RealServer and RealPlayer, observing video quality and network statistics under varying conditions. They will also use a custom RTSP client and server to observe protocol requests and responses. The goal is to understand commercial and underlying protocol operations for video streaming.
Streaming stored video allows video playback to begin before the entire file has been downloaded. It works by storing/buffering portions of the video at the client. There are three main types of streaming: UDP streaming, HTTP streaming, and adaptive HTTP streaming. HTTP streaming is most common today and works by transmitting the video file over HTTP as quickly as the network allows. Adaptive streaming addresses limitations of standard HTTP streaming by allowing clients to switch between multiple encodings of the video to adapt to changing network conditions.
Streaming stored video involves pre-recording videos on servers and allowing users to view them on demand. There are three main types of streaming: UDP, HTTP, and adaptive HTTP streaming. HTTP streaming is most common, storing video as HTTP files that are transmitted over TCP. This allows video to pass through firewalls more easily. Adaptive streaming divides video into multiple encoded quality levels to adapt to changing bandwidth conditions. The client measures bandwidth and selects the highest quality available. All streaming requires client-side buffering to smooth out network delays and jitter.
Different Types of Live Video Streaming Protocols and the Way They WorkChetu
The document discusses several popular live streaming protocols: HTTP Live Streaming (HLS), Real-Time Messaging Protocol (RTMP), and Web Real-Time Communication (WebRTC). HLS was developed by Apple and has wide support across devices and platforms, making it the most widely used protocol. RTMP was originally created by Macromedia and is commonly used for ingesting live streams. WebRTC allows for real-time audio and video transfer between browsers without plugins. Secure Reliable Transport (SRT) is an open-source protocol developed by Haivision that provides security, reliability, and low latency streaming. Choosing the right protocol depends on factors like compatibility, security, quality, latency, codec requirements, and platform
Inlet Technologies - Powering Smooth StreamingSematron UK Ltd
The white paper discusses Microsoft's Smooth Streaming technology which allows for adaptive bitrate streaming of video over HTTP. It works by encoding video into small chunks at multiple bitrates, allowing clients to switch between chunks to maintain high quality playback based on bandwidth. Inlet provides products that help content creators generate and deliver Smooth Streaming video assets for live and on-demand streaming.
A review over multimedia and how to share multimedia data between clients and servers.
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This document discusses key areas in streaming video over the internet including video compression techniques, application-layer quality of service control, continuous media distribution services, streaming servers, media synchronization mechanisms, and protocols for streaming video. It covers nonscalable and scalable video encoding, bandwidth and delay requirements, congestion and rate control, error control, streaming server components, and protocols like RTP, RTSP, and SIP. The document also outlines future directions in these areas like improved video compression schemes, evaluating rate control effectiveness, building scalable distribution infrastructure, and enhancing streaming server and protocol functionality.
1. The document proposes a new protocol called Connection Initiation Protocol (CIP) to improve performance for real-time streaming data transmission over networks.
2. CIP would encapsulate basic connection management tasks and be used alongside RTP for streaming data and RTCP for control signaling.
3. Unlike SIP-based systems, using CIP would reduce network traffic and improve performance for real-time data delivery by establishing a virtual communication channel between parties before transmitting streaming data and not involving the CIP server for RTP/RTCP transmission.
EE4414 Multimedia Communication Systems IIFranZEast
This experiment involves video streaming over the internet using different protocols and applications. Students will stream video from a server using RealServer and RealPlayer, observing video quality and network statistics under varying conditions. They will also use a custom RTSP client and server to observe protocol requests and responses. The goal is to understand commercial and underlying protocol operations for video streaming.
The document discusses several standard and proprietary streaming media protocols. It introduces Real-Time Transport Protocol (RTP) and Real-Time Control Protocol (RTCP) which transport streaming media and provide quality of service reports. It also describes Real Time Streaming Protocol (RTSP) which provides playback controls. Synchronized Multimedia Integration Language (SMIL) is mentioned as an XML language for multimedia content. Major companies like Real, Microsoft, and Apple are noted to use similar but proprietary protocols instead of the standards.
This document discusses methods for distributing computationally difficult video encoding work across multiple computers to reduce the total encoding time. It presents an application developed that splits video files into chunks, distributes the chunks to client computers to be encoded in parallel, then recombines the encoded chunks. Testing showed encoding time decreased significantly as more computers were added, with up to a 90% reduction in time when using 12 computers compared to one. However, diminishing returns were observed from additional computers. Room for improvement includes using a peer-to-peer model and supporting additional formats/operating systems.
Speaking of experiences web, the one of video in web is one of most popular at the moment. In this session they will see the possibilities of support of those experiences of video with Flash Media Server 3.5.
This document provides release notes for Cambium Network Updater Release 4.0. Key features of Release 4.0 include improved support for secure file serving, an MD5 checksum utility, and support for Windows 7 and additional Cambium device firmware versions. The release also requires Java Runtime Environment 1.5 or later and supports installation on Windows 2000/2003/XP/7 and Red Hat Enterprise Linux 4 and 5.
Scalable Service Oriented Architecture for Audio/Video ...Videoguy
The document summarizes research on developing a scalable service-oriented architecture for video conferencing using publish/subscribe event brokers to distribute real-time audio and video streams. It outlines the GlobalMMCS architecture, which separates media processing from delivery for scalability. Performance tests showed a single broker could support 1500 audio or 400 video participants, and distributed brokers improved scalability further.
The document discusses streaming media technology which enables real-time distribution of audio, video, and multimedia over the Internet. Streamed data is transmitted by a server and received/displayed by client applications. There are advantages like steady service and on-demand access, but also disadvantages like difficulty maintaining steady service with low bandwidth and high maintenance costs for streaming servers. The document then describes different types of streaming including stored, live, and interactive audio/video. It also outlines four approaches to streaming using web servers and media servers with protocols like HTTP, RTSP, and RTP.
This document provides an addendum to the user's guide for WhatsUp software version 2.5. It describes new features such as IPX monitoring and enhanced auto scan options. It also lists changes to system requirements for IPX support and notes documentation updates and corrections to the original user's guide.
The document provides system requirements for InMotion Web Conferencing software. The host and presenter software requires Internet Explorer 5.0 or later and there are no other client applications needed. The server requires Windows 2000 or later, 1.2GHz processor, 256MB RAM, and port 80 available. All users need IP access to the server and the server needs a static IP address. Bandwidth needs vary depending on the features used, with video conferencing requiring the most bandwidth at 256kbps.
The document provides an introduction and overview of transcoding including:
- Transcoding converts media formats to facilitate distribution across different platforms and ecosystems.
- Codecs, profiles, containers, and platforms are key terminology. H.264 is a widely used and patented codec.
- Formats combine containers and codecs with parameters for playback.
- Transcoding allows content to be optimized and customized for different destinations and viewer requirements.
SIP is a protocol for establishing multimedia sessions over IP networks. It originated from work in the 1990s on protocols like SCIP and SIP drafts. SIP eventually became standardized as RFC 3261 and is now widely used for voice and video calling. Cisco supports SIP in products like Cisco Unified Communications Manager, Cisco Unified Border Element, and Cisco Unified Presence to enable VoIP calling and integration between SIP and other protocols. The future of SIP includes more peer-to-peer implementations and using presence as a foundation for new services.
Barry Hesk: Cisco Unified Communications Manager training deck 1Barry Hesk
Cisco Unified Communications Manager (CUCM) training will cover CUCM basics and advanced configurations over three days. The instructor will use a demo environment including two CUCM servers and a Cisco Unity Connection voicemail server. Topics will include CUCM architecture, installation, upgrades, backups/restores, protocols, phones, gateways, and more. The goal is to explain how CUCM works from the perspective of a customer.
This presentation given by Zhaolin Wang, VIA Embedded Software Manager, provides an overview of the Android multimedia framework and then moves on to tackle critical areas such as enabling full support for RTSP and HTTP protocols, and optimizing the Android multimedia framework to match streaming server parameters, as well as shortening processing time (from start to initial playback) to ensure smooth playback, how to solve problems of clock drift (out of sync audio and video), and improving fault tolerance to avoid crashes and freezes due to corrupt data.
Session Initiation Protocol (SIP) is an application layer protocol for setting up and managing multimedia communication sessions over IP networks. It allows users to initiate, modify and terminate multimedia sessions that include voice, video and messaging applications. SIP supports mobility through proxy servers that can forward calls to a user's current location. Common security threats to SIP include registration hijacking, message modification and denial of service attacks. Recommended security mechanisms include TLS for hop-by-hop security, S/MIME for end-to-end encryption, and digest authentication.
Review of video over IP testing tools including: video syntax analyzer, pixel based measurement indexes like PSNR and SSIM and the tools to measure them, IP based video quality testing.
Matthew Kaufman Future Of Communication With Rtmfp Final Revisedstoem
Matthew Kaufman introduced RTMFP, a new protocol developed by Adobe that allows for direct peer-to-peer communication between Flash Players. RTMFP uses UDP instead of TCP to avoid limitations of client-server communication over TCP. It includes security features like encryption and authentication. RTMFP can be used with Flash Media Server for one-to-many streaming, or directly between Flash Players without a server for lower latency one-to-one communication by traversing firewalls and NAT.
This presentation is devoted to the architecture of streaming services, special features of adaptive streaming, benefits and disadvantages of various streaming technologies and specific issues of media streaming apps development.
This presentation by Nazariy Mamrokha, GlobalLogic expert, was delivered at GlobalLogic Lviv C++ TechTalk on September 15, 2016. Learn more here: https://www.globallogic.com/ua/gl_news/globallogic-lviv-c-techtalk-summary/
The document discusses multimedia requirements and techniques for streaming audio and video over the internet. It covers three classes of multimedia applications: streaming, unidirectional real-time, and interactive real-time. Streaming applications can tolerate some delay but not packet loss. Real-time applications have strict delay requirements to avoid jitter. The document discusses protocols like RTP and RTSP that are used for multimedia streaming and techniques for recovering from jitter and packet loss like buffering, FEC, and interleaving.
Sara and jacqui's steaming video projectsaralorenz17
The document discusses streaming video and audio. It defines streaming media as content sent compressed over the internet and displayed in real time, as opposed to downloading where content is saved. It then explains how streaming works, involving a server sending compressed video/audio files to clients through specialized streaming applications. There are two options for streaming - operating your own server or using a hosted streaming plan. Advantages of streaming include handling traffic loads and live events, while disadvantages include potential issues with playback quality and short videos. The document then details various streaming technologies, formats, and services.
The document discusses video streaming, including definitions and concepts. It covers topics such as the difference between streaming and downloading, common streaming categories like live and on-demand, protocols used for streaming like RTSP and RTP, and the development process for creating streaming video including content planning, capturing, editing, encoding, and integrating with servers.
The document discusses several standard and proprietary streaming media protocols. It introduces Real-Time Transport Protocol (RTP) and Real-Time Control Protocol (RTCP) which transport streaming media and provide quality of service reports. It also describes Real Time Streaming Protocol (RTSP) which provides playback controls. Synchronized Multimedia Integration Language (SMIL) is mentioned as an XML language for multimedia content. Major companies like Real, Microsoft, and Apple are noted to use similar but proprietary protocols instead of the standards.
This document discusses methods for distributing computationally difficult video encoding work across multiple computers to reduce the total encoding time. It presents an application developed that splits video files into chunks, distributes the chunks to client computers to be encoded in parallel, then recombines the encoded chunks. Testing showed encoding time decreased significantly as more computers were added, with up to a 90% reduction in time when using 12 computers compared to one. However, diminishing returns were observed from additional computers. Room for improvement includes using a peer-to-peer model and supporting additional formats/operating systems.
Speaking of experiences web, the one of video in web is one of most popular at the moment. In this session they will see the possibilities of support of those experiences of video with Flash Media Server 3.5.
This document provides release notes for Cambium Network Updater Release 4.0. Key features of Release 4.0 include improved support for secure file serving, an MD5 checksum utility, and support for Windows 7 and additional Cambium device firmware versions. The release also requires Java Runtime Environment 1.5 or later and supports installation on Windows 2000/2003/XP/7 and Red Hat Enterprise Linux 4 and 5.
Scalable Service Oriented Architecture for Audio/Video ...Videoguy
The document summarizes research on developing a scalable service-oriented architecture for video conferencing using publish/subscribe event brokers to distribute real-time audio and video streams. It outlines the GlobalMMCS architecture, which separates media processing from delivery for scalability. Performance tests showed a single broker could support 1500 audio or 400 video participants, and distributed brokers improved scalability further.
The document discusses streaming media technology which enables real-time distribution of audio, video, and multimedia over the Internet. Streamed data is transmitted by a server and received/displayed by client applications. There are advantages like steady service and on-demand access, but also disadvantages like difficulty maintaining steady service with low bandwidth and high maintenance costs for streaming servers. The document then describes different types of streaming including stored, live, and interactive audio/video. It also outlines four approaches to streaming using web servers and media servers with protocols like HTTP, RTSP, and RTP.
This document provides an addendum to the user's guide for WhatsUp software version 2.5. It describes new features such as IPX monitoring and enhanced auto scan options. It also lists changes to system requirements for IPX support and notes documentation updates and corrections to the original user's guide.
The document provides system requirements for InMotion Web Conferencing software. The host and presenter software requires Internet Explorer 5.0 or later and there are no other client applications needed. The server requires Windows 2000 or later, 1.2GHz processor, 256MB RAM, and port 80 available. All users need IP access to the server and the server needs a static IP address. Bandwidth needs vary depending on the features used, with video conferencing requiring the most bandwidth at 256kbps.
The document provides an introduction and overview of transcoding including:
- Transcoding converts media formats to facilitate distribution across different platforms and ecosystems.
- Codecs, profiles, containers, and platforms are key terminology. H.264 is a widely used and patented codec.
- Formats combine containers and codecs with parameters for playback.
- Transcoding allows content to be optimized and customized for different destinations and viewer requirements.
SIP is a protocol for establishing multimedia sessions over IP networks. It originated from work in the 1990s on protocols like SCIP and SIP drafts. SIP eventually became standardized as RFC 3261 and is now widely used for voice and video calling. Cisco supports SIP in products like Cisco Unified Communications Manager, Cisco Unified Border Element, and Cisco Unified Presence to enable VoIP calling and integration between SIP and other protocols. The future of SIP includes more peer-to-peer implementations and using presence as a foundation for new services.
Barry Hesk: Cisco Unified Communications Manager training deck 1Barry Hesk
Cisco Unified Communications Manager (CUCM) training will cover CUCM basics and advanced configurations over three days. The instructor will use a demo environment including two CUCM servers and a Cisco Unity Connection voicemail server. Topics will include CUCM architecture, installation, upgrades, backups/restores, protocols, phones, gateways, and more. The goal is to explain how CUCM works from the perspective of a customer.
This presentation given by Zhaolin Wang, VIA Embedded Software Manager, provides an overview of the Android multimedia framework and then moves on to tackle critical areas such as enabling full support for RTSP and HTTP protocols, and optimizing the Android multimedia framework to match streaming server parameters, as well as shortening processing time (from start to initial playback) to ensure smooth playback, how to solve problems of clock drift (out of sync audio and video), and improving fault tolerance to avoid crashes and freezes due to corrupt data.
Session Initiation Protocol (SIP) is an application layer protocol for setting up and managing multimedia communication sessions over IP networks. It allows users to initiate, modify and terminate multimedia sessions that include voice, video and messaging applications. SIP supports mobility through proxy servers that can forward calls to a user's current location. Common security threats to SIP include registration hijacking, message modification and denial of service attacks. Recommended security mechanisms include TLS for hop-by-hop security, S/MIME for end-to-end encryption, and digest authentication.
Review of video over IP testing tools including: video syntax analyzer, pixel based measurement indexes like PSNR and SSIM and the tools to measure them, IP based video quality testing.
Matthew Kaufman Future Of Communication With Rtmfp Final Revisedstoem
Matthew Kaufman introduced RTMFP, a new protocol developed by Adobe that allows for direct peer-to-peer communication between Flash Players. RTMFP uses UDP instead of TCP to avoid limitations of client-server communication over TCP. It includes security features like encryption and authentication. RTMFP can be used with Flash Media Server for one-to-many streaming, or directly between Flash Players without a server for lower latency one-to-one communication by traversing firewalls and NAT.
This presentation is devoted to the architecture of streaming services, special features of adaptive streaming, benefits and disadvantages of various streaming technologies and specific issues of media streaming apps development.
This presentation by Nazariy Mamrokha, GlobalLogic expert, was delivered at GlobalLogic Lviv C++ TechTalk on September 15, 2016. Learn more here: https://www.globallogic.com/ua/gl_news/globallogic-lviv-c-techtalk-summary/
The document discusses multimedia requirements and techniques for streaming audio and video over the internet. It covers three classes of multimedia applications: streaming, unidirectional real-time, and interactive real-time. Streaming applications can tolerate some delay but not packet loss. Real-time applications have strict delay requirements to avoid jitter. The document discusses protocols like RTP and RTSP that are used for multimedia streaming and techniques for recovering from jitter and packet loss like buffering, FEC, and interleaving.
Sara and jacqui's steaming video projectsaralorenz17
The document discusses streaming video and audio. It defines streaming media as content sent compressed over the internet and displayed in real time, as opposed to downloading where content is saved. It then explains how streaming works, involving a server sending compressed video/audio files to clients through specialized streaming applications. There are two options for streaming - operating your own server or using a hosted streaming plan. Advantages of streaming include handling traffic loads and live events, while disadvantages include potential issues with playback quality and short videos. The document then details various streaming technologies, formats, and services.
The document discusses video streaming, including definitions and concepts. It covers topics such as the difference between streaming and downloading, common streaming categories like live and on-demand, protocols used for streaming like RTSP and RTP, and the development process for creating streaming video including content planning, capturing, editing, encoding, and integrating with servers.
All the content of this website is informative and non-commercial, does not imply a commitment to develop, launch or schedule delivery of any feature or functionality, should not rely on it in making decisions, incorporate or take it as a reference in a contract or academic matters. Likewise, the use, distribution and reproduction by any means, in whole or in part, without the authorization of the author and / or third-party copyright holders, as applicable, is prohibited.
Apple's HTTP Live Streaming system avoids the need for specialized streaming servers by using standard HTTP to deliver video streams. It has four main components: a server that encodes and segments the video into small clips; a segmenter that stores the clips as separate files and creates an index file; a regular HTTP server that distributes the files; and a client that retrieves the files and continuously refreshes the index to receive a live, moving window of video segments or access all segments for video on demand. The system supports delivering multiple bitrate streams to different devices by providing multiple index files that clients can select from.
This document discusses streaming video to HTML. It begins with introducing the speaker and their background in building video applications. The rest of the document covers: an overview of video traffic on the internet; how HTTP adaptive streaming works; the different streaming protocols like HLS, Smooth Streaming, and DASH; browser support for streaming; MediaSource Extensions; what DASH is and how it works; building a DASH player using dash.js; and resources for learning more about streaming video.
MPEG-DASH is an adaptive bitrate streaming technology that allows high quality streaming of media content over HTTP. It works by breaking content into small HTTP-based file segments that can be played at different bitrates depending on network conditions. The standard has gained widespread industry support and adoption with major players supporting it in browsers, media players, and streaming platforms. It provides advantages over proprietary protocols by allowing interoperability between servers and clients from different vendors through its open standard.
RTSP is a protocol for controlling streaming media. It uses RTP for data delivery and allows VCR-like controls like play, pause, etc. RTSP sits on top of RTP and handles session setup and control, while RTP handles actual media data transport. RTSP messages are similar to HTTP but maintain state between requests and responses unlike stateless HTTP. RTSP enables streaming media playback by requesting media from a server and receiving it as a continuous stream.
Building video application on windows 8 with Windows Azure Media ServicesMingfei Yan
Microsoft offers media services and a player framework for adaptive streaming video on various platforms. This includes support for Smooth Streaming, Closed Captioning, and advertising through standards like VAST and VMAP. The player framework is open source and supports features like DVR controls, analytics and DRM. HTML5 may provide a cross-browser solution but lacks support for digital rights management currently.
The document provides information about a multimedia streaming module, including:
- The module code, title, level, and credit value
- Assessment requirements including creating a live/on-demand streaming media station and accompanying website
- An overview of topics covered in the module like media encoding, streaming servers, planning live broadcasts, and streaming non-audio/video content
- Considerations for streaming like file sizes, frame rates, formats, and computer hardware requirements
This paper proposes a new Hybrid Adaptive Mobile Streaming (HAMS) method for efficient video streaming on mobile devices using cloud services. HAMS uses Scalable Video Coding (SVC) to encode video files into base and enhancement layers. It first progressively downloads base layers, then adaptively streams enhancement layers to avoid video freezing while providing high video quality. An experiment compares HAMS to existing methods and finds HAMS achieves zero video freezes for up to three users, shorter freeze times, and better average video quality. The paper concludes HAMS is an effective method for mobile video streaming without freezing issues or reduced quality.
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.
International Journal of Computational Engineering Research(IJCER) ijceronline
nternational Journal of Computational Engineering Research (IJCER) is dedicated to protecting personal information and will make every reasonable effort to handle collected information appropriately. All information collected, as well as related requests, will be handled as carefully and efficiently as possible in accordance with IJCER standards for integrity and objectivity.
This document describes a video conferencing software project that allows users to conduct video calls over an intranet. It uses Java Media Framework for multimedia capabilities, Java Swing for the user interface, and Java networking for peer-to-peer communication. The software captures audio and video using JMF, processes it, then sends the media over the network using RTP protocols. This allows two computers connected via intranet to conduct real-time video calls with synchronized audio and video streams.
IRJET- Protected Adaptive Bitrate Streaming using HTTP Live Streaming and...IRJET Journal
1) The document discusses protected adaptive bitrate streaming using HTTP Live Streaming (HLS) and cryptographic techniques to protect digital video assets.
2) HLS uses MPEG-TS segments and playlist files (M3U8) to provide adaptive bitrate streaming over HTTP. The video is encrypted and segmented into small chunks to adapt to changing network conditions.
3) The document outlines the HLS file format including master and variant M3U8 playlists and TS video chunks. Encryption is applied to the video at the chunk level before storage and delivery.
Unique info about videostreaming compression in iOS from our the best iOS specialist Vladimir Predko. He's ready to answer all your questions! Go ahead!
In order to simplify and consolidate HLS installations, Visionary Solutions created the PackeTV® Mobile HLS, a single device that performs content preparation and delivery. This integrated file server can support hundreds of users, eliminating the need for content delivery network (CDN) services.
Equipped with two gigabit Ethernet ports that provide a substantial amount of network bandwidth, the unit can store hundreds of hours of pre-recorded content. The entry-level system is housed in a single, compact 1RU chassis that fits perfectly into any standard 20-inch deep AV rack.
The PackeTV® Mobile HLS server supports all of the functions needed to accept H.264 video streams (real-time or file-based) and deliver HLS streams, including content
preparation, file storage, and content delivery. All of this functionality is contained within a server that has been specifically designed to optimize throughput and ease of use. PackeTV® Mobile HLS dramatically lowers operational costs compared with systems that use traditional streaming CDNs to simultaneously distribute video to multiple clients. System ownership ensures seamless, around-the-clock availability of the video streams. Each video stream can be published once and made available to all viewers with a simple set of user commands. Occasional users will find that this single, integrated system eliminates much of the configuration complexity that normally occurs when multiple subsystems from different manufacturers and service providers need to be integrated to form a complete solution. Heavy users will appreciate the flexibility that is available within the device configuration menus, which allow system operations to be customized to accommodate a wide range of bit rates, signal formats, and target devices. Also, because standard HTTP
Web-server technologies are used for content delivery, the added fees required for high throughput streaming service providers such as CDNs are eliminated, saving the content
provider money. With an on-site PackeTV® Mobile HLS, content asset management can be greatly simplified and centralized on a single server. A single video file can be created and delivered to an organization’s internal and external viewers, eliminating the need to manage multiple streams on different servers. Network bandwidth is also used more efficiently, as chunks can be downloaded quickly, and each client device only consumes as much data as the quality of their network connection allows More than half of all overall video consumption is consumed on a Wi-Fi device, according to Streaming Media, so content providers must make mobile content delivery an immediate priority.
Streaming media over HTTP has become widespread due to its efficiency and ability to cache content globally using CDNs. This allows media to be streamed far and wide without buffering issues. While early solutions like Move Networks and Apple HLS paved the way starting in 2008, technologies like Microsoft Smooth Streaming, Netflix's own solution, Adobe HDS, and MPEG-DASH soon followed to make adaptive streaming over HTTP the dominant approach.
This document discusses podcasting and serving multimedia. It defines podcasting as a digital medium consisting of episodic audio, video or other files that are subscribed to and downloaded. Podcasters create content while podcatchers listen to content. Popular podcatching clients include Juice Receiver. The document provides an example of a simple RSS feed for podcasting and discusses validating, advertising and finding podcasts. It also covers issues with serving multimedia like file sizes and formats, and describes common container and encoding formats like MP3, Ogg, AVI, ASF and MPEG. Finally, it discusses protocols for multimedia like multicast using RTP and unicast using HTTP streaming.
An Enhanced Independent Component-Based Human Facial Expression Recognition ...أحلام انصارى
This document presents a facial expression recognition system that uses enhanced independent component analysis and fisher linear discriminant analysis (EICA-FLDA) for feature extraction from video frames, and hidden Markov models (HMM) for expression recognition. The system is tested on the Cohn-Kanade facial expression database and achieves a mean recognition rate of 93.23% for six universal expressions (anger, joy, sad, disgust, fear, surprise). Facial expression recognition has applications in human-computer interaction domains like online gaming.
Intention recognition for dynamic role exchange in hapticأحلام انصارى
1) The paper summarizes an experimental study that investigated the utility of a role exchange mechanism in human-computer collaboration through haptic interaction.
2) In this framework, a human dynamically interacts with a computer partner by communicating through the haptic channel to trade control levels on a shared task.
3) The study examined conditions of equal control, role exchange, and visuo-haptic cues, analyzing effects on task performance, efficiency, and subjective measures.
Noise Adaptive Training for Robust Automatic Speech Recognitionأحلام انصارى
This document provides information about speech recognition software and its potential benefits for students. It discusses how speech recognition works by translating spoken words to text. Using dictation could improve writing for students with learning disabilities by removing barriers from mechanical issues. Speech recognition may support writing quality by removing motor demands and increasing productivity. It also allows access for students unable to use a keyboard. While consistency is important, speech does not need to be perfect. Speech recognition requires cognitive skills and works best for students reading at a third grade level or higher. The document reviews various speech recognition programs and considerations for training and use.
Human behaviour analysis based on New motion descriptorأحلام انصارى
Human behavior analysis is an important area of research that involves detecting, tracking, and understanding people's physical behaviors. This paper introduces a new motion descriptor for human behavior analysis with applications in interactive virtual reality systems, video storage and retrieval, sports analysis, and real-time video surveillance. The framework presented analyzes surveillance camera video streams in real-time to monitor people's actions in security-sensitive areas.
Multimodal Biometric Human Recognition for Perceptual Human–Computer Interactionأحلام انصارى
The document discusses the benefits of exercise for mental health. Regular physical activity can help reduce anxiety and depression and improve mood and cognitive functioning. Exercise causes chemical changes in the brain that may help protect against mental illness and improve symptoms.
A cloud database stores data on remote servers accessed via the internet rather than a local physical server. There are two common deployment models - running a database independently on a virtual machine in the cloud, or purchasing access to a database service maintained by a cloud provider. Using a cloud database offers benefits like low cost, easy access to data from anywhere, and automatic data backup, but also security risks that must be addressed.
Html5 offers 5 times better ways to hijack the website أحلام انصارى
Html5 offers 5 times better ways to hijack websites than previous web technologies. It introduces new capabilities like WebSQL, canvas, web workers and messaging that can be exploited by attackers. The document discusses how HTML5 enables information extraction through web storage and DOM manipulation, injection of third party widgets and scripts by poisoning the browser cache, using web workers for payload delivery if DOM-based XSS exists, enabling new CSRF and clickjacking attacks, and aiding SQL injection. In conclusion, while HTML5 provides new features, application security depends on developers implementing them carefully to prevent exploitation.
Honey pots can be implemented in cloud computing to improve security. There are several components, including a cloud controller, cluster controller, honey controller, and log storage system. Low interaction honey pots like Honeyd emulate services to detect attacks, while high interaction honey pots like Honeynets allow more flexibility for attackers but carefully control outbound traffic. Honey pots can be offered as a service for cloud customers, providing logs and statistics to help secure resources against future attacks.
This document discusses grid authentication technologies such as public key infrastructure (PKI) and digital certificates. It provides an overview of symmetric and public key cryptography. Digital certificates bind a user's identity to their public key and are signed by a certification authority. The grid security infrastructure is based on PKI and uses certificates for authentication and encryption through mutual authentication. Cross-certification allows different certification authorities to validate each other's certificates, enabling authentication across security domains.
Security As A Service In Cloud(SECaaS)أحلام انصارى
This document discusses security as a service (SECaaS) in cloud computing. It begins by explaining other common cloud service models like SaaS, PaaS, IaaS, and STaaS. It then defines SECaaS as a business model where large service providers integrate security services like authentication, antivirus, intrusion detection, and security event management into a corporate infrastructure on a subscription basis. The document lists the top 10 cloud service providers and reasons why cloud-based security is required. It outlines common areas covered by SECaaS like identity and access management, data loss prevention, and network security. Finally, it provides examples of specific SECaaS products and services offered by vendors.
The document discusses service oriented architecture (SOA). It defines SOA as an approach where applications use services available over a network to perform business functions. Key components of SOA include services, service types, layers, and governance patterns. SOA uses XML-based standards like SOAP, WSDL, and UDDI to allow services to communicate over different platforms. The presentation covers the basics of SOA, its benefits, and technical concepts like web services.
This document discusses role-based access control (RBAC). It defines the core components of RBAC, including users, roles, operations, objects, and permissions. It also describes hierarchical RBAC and how roles can inherit permissions and users from other roles. Finally, it covers separation of duties, both static and dynamic, which place constraints on role assignments to prevent conflicts of interest. RBAC aims to simplify security administration by defining permissions based on roles rather than individual users.
The document discusses various techniques for cracking passwords, including dictionary attacks, brute force attacks, and exploiting weaknesses in password hashing algorithms. Default passwords, social engineering through phishing emails, and the use of tools like Cain and Abel, John the Ripper, and THC Hydra are also covered as effective cracking methods. Common password mistakes that can enable cracking are also listed.
Operating system vulnerability and control أحلام انصارى
Vulnerabilities exist in operating systems like Linux, UNIX and Windows. A vulnerability is a weakness that allows an attacker to compromise a system's security. Vulnerabilities occur at the intersection of a system flaw, an attacker's access to the flaw, and their ability to exploit it. Common UNIX vulnerabilities include setuid problems, trojan horses and terminal troubles. Windows is vulnerable to password issues, peer-to-peer file sharing exploits, and Outlook/Outlook Express bugs. Linux has flaws like missing permission checks, uninitialized data, and memory mismanagement. Control is important for operating systems to balance robustness, predictability and efficiency. The trusted computing base (TCB) aims to enforce security by containing all elements
This document discusses network security and introduces several common network security tools. It begins by defining network security and explaining the importance of securing a network. It then profiles six security tools: Snort is an open-source network intrusion detection and prevention system that can detect attacks and probes. Ettercap is used for network protocol analysis and can intercept traffic, capture passwords, and conduct eavesdropping. Sam Spade is a network tool suite that finds public information about IP and DNS addresses. BackTrack is a Linux distribution focused on penetration testing. Hydra performs rapid dictionary attacks against network protocols. Deep Freeze makes computer configurations indestructible by wiping out any changes made during a session.
Digital image forgery can be categorized into three main types: image retouching, image splicing, and copy-move attack. Image retouching makes minor enhancements without significantly altering the image. Image splicing combines two or more images to create a composite fake image. Copy-move attack modifies an image by copying and moving a region within the same original image, such as duplicating smoke to conceal details or tamper with the image. Effective and low-cost ways to help secure images and prevent misuse include adding copyright text to images, optimizing image size and compression, slicing images, using mouseover image swaps, and setting images as table backgrounds with transparent GIFs.
Image-based authentication (IBA) uses a set of user-selected images rather than a password for authentication. The IBA system displays an image set including key images mixed with other images. The user is authenticated by correctly identifying their key images. The document discusses IBA in detail, including potential vulnerabilities and methods to counter threats like observation attacks, brute force attacks, and frequency analysis attacks. It also covers the use of CAPTCHAs to distinguish humans and machines.
The document discusses demilitarized zones (DMZs) in computer networks. A DMZ is a small subnetwork located between a company's private network and the outside public network. It contains devices like web, FTP, and email servers that are accessible to internet traffic but isolated from the internal network. DMZs provide enhanced security by separating internal and external networks, and only allowing specific services that need to be accessed from the outside. The document outlines common DMZ architectures, security considerations, and the types of servers and services typically located in a DMZ.
Cryptography is the practice of hiding information and involves techniques like secret key cryptography, public key cryptography, and hash functions. New trends in cryptography include elliptic curve cryptography, which uses points on elliptic curves to securely exchange keys, quantum cryptography which uses quantum effects for secure communication, and pairing based cryptography which pairs elements of groups to construct hybrid cryptosystems. The document discusses different encryption techniques, their drawbacks, and the need for new techniques that are more secure against attacks.
A Strategic Approach: GenAI in EducationPeter Windle
Artificial Intelligence (AI) technologies such as Generative AI, Image Generators and Large Language Models have had a dramatic impact on teaching, learning and assessment over the past 18 months. The most immediate threat AI posed was to Academic Integrity with Higher Education Institutes (HEIs) focusing their efforts on combating the use of GenAI in assessment. Guidelines were developed for staff and students, policies put in place too. Innovative educators have forged paths in the use of Generative AI for teaching, learning and assessments leading to pockets of transformation springing up across HEIs, often with little or no top-down guidance, support or direction.
This Gasta posits a strategic approach to integrating AI into HEIs to prepare staff, students and the curriculum for an evolving world and workplace. We will highlight the advantages of working with these technologies beyond the realm of teaching, learning and assessment by considering prompt engineering skills, industry impact, curriculum changes, and the need for staff upskilling. In contrast, not engaging strategically with Generative AI poses risks, including falling behind peers, missed opportunities and failing to ensure our graduates remain employable. The rapid evolution of AI technologies necessitates a proactive and strategic approach if we are to remain relevant.
ISO/IEC 27001, ISO/IEC 42001, and GDPR: Best Practices for Implementation and...PECB
Denis is a dynamic and results-driven Chief Information Officer (CIO) with a distinguished career spanning information systems analysis and technical project management. With a proven track record of spearheading the design and delivery of cutting-edge Information Management solutions, he has consistently elevated business operations, streamlined reporting functions, and maximized process efficiency.
Certified as an ISO/IEC 27001: Information Security Management Systems (ISMS) Lead Implementer, Data Protection Officer, and Cyber Risks Analyst, Denis brings a heightened focus on data security, privacy, and cyber resilience to every endeavor.
His expertise extends across a diverse spectrum of reporting, database, and web development applications, underpinned by an exceptional grasp of data storage and virtualization technologies. His proficiency in application testing, database administration, and data cleansing ensures seamless execution of complex projects.
What sets Denis apart is his comprehensive understanding of Business and Systems Analysis technologies, honed through involvement in all phases of the Software Development Lifecycle (SDLC). From meticulous requirements gathering to precise analysis, innovative design, rigorous development, thorough testing, and successful implementation, he has consistently delivered exceptional results.
Throughout his career, he has taken on multifaceted roles, from leading technical project management teams to owning solutions that drive operational excellence. His conscientious and proactive approach is unwavering, whether he is working independently or collaboratively within a team. His ability to connect with colleagues on a personal level underscores his commitment to fostering a harmonious and productive workplace environment.
Date: May 29, 2024
Tags: Information Security, ISO/IEC 27001, ISO/IEC 42001, Artificial Intelligence, GDPR
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A review of the growth of the Israel Genealogy Research Association Database Collection for the last 12 months. Our collection is now passed the 3 million mark and still growing. See which archives have contributed the most. See the different types of records we have, and which years have had records added. You can also see what we have for the future.
it describes the bony anatomy including the femoral head , acetabulum, labrum . also discusses the capsule , ligaments . muscle that act on the hip joint and the range of motion are outlined. factors affecting hip joint stability and weight transmission through the joint are summarized.
How to Manage Your Lost Opportunities in Odoo 17 CRMCeline George
Odoo 17 CRM allows us to track why we lose sales opportunities with "Lost Reasons." This helps analyze our sales process and identify areas for improvement. Here's how to configure lost reasons in Odoo 17 CRM
Executive Directors Chat Leveraging AI for Diversity, Equity, and InclusionTechSoup
Let’s explore the intersection of technology and equity in the final session of our DEI series. Discover how AI tools, like ChatGPT, can be used to support and enhance your nonprofit's DEI initiatives. Participants will gain insights into practical AI applications and get tips for leveraging technology to advance their DEI goals.
বাংলাদেশের অর্থনৈতিক সমীক্ষা ২০২৪ [Bangladesh Economic Review 2024 Bangla.pdf] কম্পিউটার , ট্যাব ও স্মার্ট ফোন ভার্সন সহ সম্পূর্ণ বাংলা ই-বুক বা pdf বই " সুচিপত্র ...বুকমার্ক মেনু 🔖 ও হাইপার লিংক মেনু 📝👆 যুক্ত ..
আমাদের সবার জন্য খুব খুব গুরুত্বপূর্ণ একটি বই ..বিসিএস, ব্যাংক, ইউনিভার্সিটি ভর্তি ও যে কোন প্রতিযোগিতা মূলক পরীক্ষার জন্য এর খুব ইম্পরট্যান্ট একটি বিষয় ...তাছাড়া বাংলাদেশের সাম্প্রতিক যে কোন ডাটা বা তথ্য এই বইতে পাবেন ...
তাই একজন নাগরিক হিসাবে এই তথ্য গুলো আপনার জানা প্রয়োজন ...।
বিসিএস ও ব্যাংক এর লিখিত পরীক্ষা ...+এছাড়া মাধ্যমিক ও উচ্চমাধ্যমিকের স্টুডেন্টদের জন্য অনেক কাজে আসবে ...
How to Fix the Import Error in the Odoo 17Celine George
An import error occurs when a program fails to import a module or library, disrupting its execution. In languages like Python, this issue arises when the specified module cannot be found or accessed, hindering the program's functionality. Resolving import errors is crucial for maintaining smooth software operation and uninterrupted development processes.
Macroeconomics- Movie Location
This will be used as part of your Personal Professional Portfolio once graded.
Objective:
Prepare a presentation or a paper using research, basic comparative analysis, data organization and application of economic information. You will make an informed assessment of an economic climate outside of the United States to accomplish an entertainment industry objective.
Assessment and Planning in Educational technology.pptxKavitha Krishnan
In an education system, it is understood that assessment is only for the students, but on the other hand, the Assessment of teachers is also an important aspect of the education system that ensures teachers are providing high-quality instruction to students. The assessment process can be used to provide feedback and support for professional development, to inform decisions about teacher retention or promotion, or to evaluate teacher effectiveness for accountability purposes.
This slide is special for master students (MIBS & MIFB) in UUM. Also useful for readers who are interested in the topic of contemporary Islamic banking.
Main Java[All of the Base Concepts}.docxadhitya5119
This is part 1 of my Java Learning Journey. This Contains Custom methods, classes, constructors, packages, multithreading , try- catch block, finally block and more.
2. ABSTRACT
•In today’s world streaming media plays a vital role, with the advanced
technologies recently developed in the areas of high-speed networks and
multimedia, video-on- demand service (VOD) is considered as the emerging
trend in home entertainment, as well as in education, banking, home
shopping, and interactive games. Better quality video, as well as efficient way
to send and receive them is necessary. Streaming media is multimedia that is
constantly received by and presented to an end-user while being delivered by
a streaming provider. With streaming, the client browser or plug-in can start
displaying the data before the entire file has been transmitted. Quality of
video, streaming time of video and the streaming protocols that we use for
make a lot of difference. Hence we present two different types of streaming
media protocols, HTTP Live Streaming (HLS) and Real Time Streaming Protocol
(RTSP) and their comparison, a DoS resilient self healing video architecture to
provide a better quality video and a survey on different VoD Batching
techniques.
.
Ahlam Ansari 2
3. INTRODUCTION
•HTTP Live Streaming (HLS) is an HTTP-based media streaming communications
protocol implemented by Apple Inc.
It works by breaking the overall stream into a sequence of small HTTP-based file
downloads, each download loading one short chunk of an overall potentially
unbounded transport stream.
Ahlam Ansari 3
4. The Real Time Streaming Protocol (RTSP) is a network control protocol designed
for use in entertainment and communications systems to control streaming media
servers.
RTSP was developed by the Multiparty Multimedia Session Control Working
Group (MMUSIC WG) of the Internet Engineering Task Force (IETF)
Ahlam Ansari 4
5. Video request is delayed for a period of time so that more requests for the same
video are collected. The batch of requests is then served by a multicast video
stream.
The major drawback of this scheme is that the customers have to wait for a
batching interval until the video is started to play. Hence, it may increase the
customer dissatisfaction if the waiting interval is too long.
Ahlam Ansari 5
8. 1. Media Encoder:
The Encoder encodes the video/audio. Videos are compressed into H.264/MPEG-4
format and the audio is encoded using advance audio coding. The video or audio
format is encapsulated in MPEG 2.
Transport Stream is specified in MPEG-2 Part 1, Systems (formally known as
ISO/IEC standard 13818-1 or ITU-T Rec. H.222.0) [3][4][5]. Transport stream
specifies a container format encapsulating packetized elementary streams, with
error correction and stream synchronization features for maintaining transmission
integrity when the signal is degraded.
Program stream is a container format for storage applications specified in MPEG-1
Systems and MPEG-2 Part 1. A program stream contains only one content channel
and is suited to authoring and storage not broadcasting. Contrast with transport
stream, which is designed to address the delivery of the content.
Ahlam Ansari 8
9. 2. Stream Segmenter
The Segmenter segments the video/audio in small duration segmentation. All the
segments are placed in separate files. For protection purpose, the Segmenter
might encrypt the segments and create a key file.
Ahlam Ansari 9
10. 3. Distribution System
Distributed System is a regular HTTP Server. It could be Apache or any embedded
server. Segments are stored with .ts extension and the index files are stored
with .m3u8 extension. The file format of index is as follows:
Ahlam Ansari 10
11. 4. Client:
The client gets all the videos or audios through a HTTP connection. HTTP creates a session
between the client and the file at the DS. The session can be either a Live Stream Broadcast
session or a Video on Demand Session.
A) Live Stream Broadcast Session
LSD contains an index file that is dynamic in type and is continuously
updated and refreshed. It includes a moving window of segments.
B) Video on Demand Session
VoD contains an index file that is static in type and it contains all the
segments of the file. [2]
Ahlam Ansari 11
12. Deploying HTTP Live Streaming
Step 1: Creating a HTML Page
Step 2: Configuring a Web Server
Step 3: Validating the Streams
Step 4: downloade video / audio at client side
Ahlam Ansari 12
13. Step 1: Creating a HTML Page
The easiest way to distribute HTTP Live Streaming media is to create a webpage that
includes the HTML5 <video> tag, using an .M3U8 playlist file as the video source. An
example is shown below:
<html>
<head>
<title>HTTP Live Streaming Example</title>
</head>
<body>
<video
src="http://devimages.apple.com/iphone/samples/bipbop/bipbopall.m3u8"
height="300" width="400" >
</video>
</body>
</html>
Ahlam Ansari 13
14. Step 2: Configuring a Web Server
•HTTP Live Streaming can be served from an ordinary web server; no special
configuration is necessary, apart from associating the MIME types of the files
being served with their file extensions.
Configure the following MIME types for HTTP Live Streaming:
File Extension MIME Type
.M3U8 application/x-mpegURL or vnd.apple.mpegURL
.ts video/MP2T
Ahlam Ansari 14
15. Step 3: Validating the Streams
•The media stream validator tool is a command-line utility for validating HTTP Live
Streaming streams and servers . The media stream validator simulates an HTTP
Live Streaming session and verifies that the index file and media segments
conform to the HTTP Live Streaming specification. It performs several checks to
ensure reliable streaming. If any errors or problems are found, a detailed
diagnostic report is displayed.
You should always run the validator prior to serving a new stream or alternate
stream set. The media stream validator shows a listing of the streams you provide,
followed by the timing results for each of those streams. (It may take a few
minutes to calculate the actual timing.) An example of validator output follows.
Ahlam Ansari 15
16. •Validating
http://devimages.apple.com/iphone/samples/bipbop/gear3/prog_index.m3u8
against iPhone OS 3.1.0
Average segment duration: 8.77 seconds
Average segment bitrate: 510.05 kbit/s
Average segment structural overhead: 96.37 kbit/s (18.89 %)
Video codec: avc1
Video resolution: 480x360 pixels
Video frame rate: 29.97 fps
Average video bitrate: 407.76 kbit/s
H.264 profile: Baseline
H.264 level: 2.1
Audio codec: aac
Audio sample rate: 22050 Hz
Average audio bitrate: 5.93 kbit/s Ahlam Ansari 16
17. Step 4: Download video/audio at client side
How actually the video / audio is downloaded at client side [8]
1. Download the wowza URL (playlist)
2. Display the contents of the file
3. Download the first/second playlist url
4. Display the contents
5. wget the key URL
6. wget the ts file sequence
If the file is encrypted
7. Hexdump the key
8. Using calc get the HEXADECIMAL Value of "MEDIA-SEQUENCE", to get IV -> 95 =
5F using openssl decrypt the file
9. Now play the file "decr_tmp.ts" using "VLC media player"
Ahlam Ansari 17
18. RTSP vs. HTTP
Interoperability
RTSP is an interoperability disaster. None of the three big commercial
implementations seem to abide by the standard (Microsoft Windows Media,
Apple/Darwin and RealMedia), and writing a client stack that can handle all of
them, plus the open-source servers properly, is tough to say the least.
HLS is widely interoperable. It has countless implementations both on the server
and client sides. It has Apache as a de facto reference implementation that every
client can interoperate with.
Ahlam Ansari 18
19. •Transport Layer Protocol
RTSP uses UDP
HTTP uses TCP
•Seeking
RTSP, the server takes care of it
With HLS, client will probably be the one
Ahlam Ansari 19
20. • Play and Pause
RTSP does have built-in PLAY and PAUSE commands, which namely respectively
play and pause the media.
There are several ways you can pause a stream with HTTP:
. Stop de-queuing data from the TCP session, and let the TCP stacks handle pause
with TCP congestion control, until you want to resume,
. Shutdown/reset the TCP session to pause, and start a new one to resume
Ahlam Ansari 20
21. •Congestion
In case of network congestion or packet loss, RTSP will cause part of the stream to be lost.
In case of network congestion or packet loss, HLS will never cause part of the stream to be
lost.
•You Tube
RTSP is used in YouTube Mobile.
HLS is used on You Tube main site.
Ahlam Ansari 21
22. •Reliability
RTSP only carries the signaling reliably.
HTTP conveys both signaling and payload reliably.
•NAT Traversal
RTSP just does not work, without ugly TCP encapsulation that makes it look like HTTP.
HLS just works through Network address translation
Ahlam Ansari 22
23. •Firewall traversal
Not so many firewalls handle or allow RTSP. RTSP has theoretical firewall traversal
capability.
HLS goes through almost any firewall. Even from tightly restricted Intranets, you
can always find an outgoing HTTP proxy to use.
Ahlam Ansari 23
26. •I n the basic architecture of a VoD system consists of:
Video Encoder
Video Server
Directory Server
Client
Network
The Network connects all the other VoD system components together as they are
distributed geographically. The encoder accepts the video input and then uploads it on the
video server after compressing/encoding the video stream received. The video server is a
server with huge amount of disk space to accommodate day by day increasing videos in
segments, not as a single file. The directory server keeps the index file which points to all
the segments of those videos in video server, and these videos are published to the client so
that they can access it. The figure below shows the architecture of basic VoD system.
Ahlam Ansari 26
27. Types of Streaming
1. Realtime
The data have a pre-determined sequence and time of presentation.
For example, video and audio.
2. Non-Realtime
The data does not have presentation time requirement.
For example, progressive JPEG.
Ahlam Ansari 27
28. Types of Video Services
1. Broadcast / Multicast Video
2. Near-Video-on-Demand
3. True Video-on-Demand
Ahlam Ansari 28
30. •The multicast facility of modern communication networks [12] offers an efficient
means of one-to-many data transmission. The basic idea is to avoid retransmitting
the same packet more than once on each link of the network by having branch
routers duplicate and then send the packet over multiple downstream branches.
Multicast can significantly improve the VoD performance.
Passive receive with no control except selecting the channels. One channel is
needed per movie / programme.
Ahlam Ansari 30
32. •Near video on demand is a video delivery service that allows a customer to select
from a limited number of broadcast video channels when they are broadcast.
NVOD channels have pre-designated schedule times and are used for pay-per-view
services. Passive receive with limited controls. System response time inversely
proportional to number of required channels [12].
Ahlam Ansari 32
34. •True Video-on-Demand system uses one dedicated channel for each service
request, offering the client the best TVoD service. However, such a system incurs
very high costs, especially in terms of storage-I/O and network bandwidths. Full
interactive controls, like pause/resume, seeking, fast forward, etc .
Ahlam Ansari 34
35. Comparisons
Broadcast Video True Video-on-
Near-Video-on-Demand Demand
(Pay-Per-View)
Yes, but limited to a Yes, but limited to a few
Select video? Yes
few channels programmes
Select time to
No Yes (limited to fixed time slots) Anytime
watch?
Interactive? No None or very little VCR-like control
No. of Viewers Unlimited Unlimited Limited
Cost / Viewer Low Medium High
Ahlam Ansari 35
36. LITERATURE SURVERY
Some authors have worked on reducing the start-up delay in
multicast delivery systems as the major problem with the batching
approach is that it introduces a significant start-up delay to the
customer, which in fact contradicts the idea of on-demand service.
Ahlam Ansari 36
37. Batching policy for video-on-demand in multicast environment
W.-F. Poon, K.-T. Lo and J. Feng Says:
•Piggybacking approach
It merges different requests together by adjusting the play-out rate of the videos
so as to reduce the start-up delay of the system. The greedy policy attempts to
merge the requests as many times as possible during the entire video session in
order to save more bandwidth.
If the play-out rate is adjusted in real time, a specialized hardware is required to
keep up with the demand. If a replica of the video is generated in advance, since it
is possible that requests can be merged during the entire video session, very large
disk storage is required at the server side.
Ahlam Ansari 37
38. •Double-rate (DR) batching policy
Considering the waiting time (conventional batching approach) of the customers
and the complexity (piggybacking approach) of the VoD system, we develop a
double-rate (DR) batching scheme that attempts to merge the customer requests
in an improvised manner by means of buffering. Instead of changing the play-out
rate of the video, we double the transmission rate so that the trailing customers
are able to catch up with the leading customer. Once the transmitting frames of
the customers are the same, they are grouped together and served by a multicast
stream.
A new batching scheme has been proposed to reduce the start-up delay of VoD
services in a multicast delivery system. When a new customer arrives, a unicast
stream is immediately opened for the new request so that the customer does not
have to wait. The transmission rate is then doubled so that the customer can catch
up with the previous multicast stream.
Ahlam Ansari 38
39. Advantage:
A new batching scheme has been proposed to reduce the start-up delay of VoD
services in a multicast delivery system. When a new customer arrives, a unicast
stream is immediately
opened for the new request so that the customer does not have to wait. The
transmission rate is then doubled so that the customer can catch up with the
previous multicast stream.
Drawback:
It is shown that the DR batching scheme outperforms the greedy policy of the
piggybacking approach only in terms of the bandwidth requirement. As it has fixed
batching time, popularity of the movies affects the performance of the VoD
system.
Ahlam Ansari 39
40. Adaptive Batching Scheme for Multicast Video-On-Demand Systems
W.-F. Poon, K.-T. Lo, Member, IEEE, and J. Feng, Member, IEEE says:
•An adaptive algorithm is developed for providing true video on demand (VoD)
services in multicast environment. If the batching time is wrongly estimated, the
performance of the system will be greatly degraded. This algorithm tries to
dynamically find the optimal batching time by the newly updated arrival rate so as
to minimize the bandwidth requirement.
Adaptive algorithm enhances the double-rate batching policy so that the
popularity of the movies will not affect the performance of the VoD system as the
batching time can be adjusted.
Ahlam Ansari 40
41. •Advantage:
Adaptive algorithm enhances the double-rate batching policy so that the
popularity of the movies will not affect the performance of the VoD system as the
batching time can be adjusted.
Drawback:
If the batching time is wrongly estimated, the performance of the system will be
greatly degraded.
Ahlam Ansari 41
42. Virtual Batching: A New Scheduling Technique for Video-On-Demand Servers
Simon Sheu and Kien A. Hua and Ta-Hsiung Hu says:
•In a video-on-demand (VOD) environment, batching of requests is often applied
to reduce the I/O demand and increase the availability of VOD services. However,
batching unfairly forces first comers to wait for subscribers arriving late at the
batch. Some of these victims may become impatient and decide to renege. To
address this issue, we introduce a chaining technique which allows a client station
being served to forward its video data to other client stations, which requested
the same video, at a slightly later time.
Thus, chaining enjoys the benefit of batching without its side effect of causing long
access latencies. By combining batching and chaining, we also design a scheme
called Virtual Batching.
This property reduces the service latencies and therefore improves the reneging
probability. This characteristic addresses the network-I/O bottleneck problem
facing today’s media servers.
Ahlam Ansari 42
43. •Advantage:
First comers no longer have to wait for requests arriving late at the batch. This
property reduces the service latencies and therefore improves the reneging
probability. This characteristic addresses the network-I/O bottleneck problem
facing today’s media servers.
Drawback:
This policy makes use of Batching as well as Chaining hence it is more complex.
The clients should also have chaining and forwarding mechanism in order to
implement Virtual Batching policy.
Ahlam Ansari 43
46. •The self-healing architecture comprises of four main components: performance
analyzer, repair planner, sensors and effectors. The performance analyzer takes:
(1) server performance parameters collected by sensors to detect performance
degradation; and (2) service parameters collected from synthetic sessions
established with the server. It detects performance degradation and pinpoints the
resources responsible for these anomalous states. The repair planner takes
notifications issued by the performance analyzer, decides the best repair action to
be executed (e.g., session migration and protocol-level redirection of sessions) and
plans the repair to be executed by effectors.
Ahlam Ansari 46
47. •Sensors are responsible for collecting performance parameters. Two types of
sensors are devised server-side: (1) pace sensors; and (2) resource sensors. Pace
sensors capture delays on accomplishment of scheduling of video content. The
unit of scheduled work depends on server implementations and streaming
protocols. In RTSP streaming the packet is the unit of work to be scheduled and
transmitted. Other video standards require coarse-grain performance metrics at
server: HTTP Streaming and HTTP Adaptive streaming. They fragment video
objects into slices that are selectively requested by clients and downloaded
similarly to other web objects.
Ahlam Ansari 47
48. Conclusion
•YouTube uses HTTP/TCP to buffer video into the flash player on its main site. The
video is stored on Google Video's content distribution network. It's not streamed
as much as just sent as fast as possible to your computer. However, for 3g mobile
handsets, m.youtube.com uses RTSP to stream video. So, YouTube uses both
transfer methods [10]. Another thing to consider is that YouTube is popular
because "it just works". RTSP isn't always supported well through routers, which
would prevent that for use on the desktop. Also, m.youtube.com doesn't stream
the same video file as youtube.com. Most YouTube videos are not streamed, but
progressively download over HTTP. A small number of partner videos are securely
streamed, using RTMPE.
Ahlam Ansari 48
49. •DR batching scheme reduce the start-up delay of VoD services in a multicast delivery
system. When a new customer arrives, a unicast stream is immediately opened for the new
request so that the customer does not have to wait. The transmission rate is then doubled
so that the customer can catch up with the previous multicast stream.
Adaptive batching scheme enhances the double-rate batching policy so that the popularity
of the movies will not affect the performance of the VoD system as the batching time can
be adjusted.
Virtual Batching scheme generalize the conventional batching methods to allow requests
arriving at a virtual batch to receive immediate services. First comers no longer have to wait
for requests arriving late at the batch. This property reduces the service latencies and
therefore improves the reneging probability. This characteristic addresses the network-I/O
bottleneck problem facing today’s media servers.
Ahlam Ansari 49