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Sequence number The value of this 16-bit number increments by one for each packet. It is used by the player to detect packet loss and then to sequence the packets in the correct order. The initial number for a stream session is chosen at random. Timestamp This is a sampling instance derived from a reference clock to allow for synchronization and jitter calculations. It is monotonic and linear in time. Source identifiers CSRC is a unique identifier for the synchronization of the RTP stream. One or more CSRCs exist when the RTP stream is carrying information for multiple media sources. This could be the case for a video mix between two sources or for embedded content. RTP usually runs on UDP, and uses its multiplexing and checksum features. Note that RTP does not provide any control of the quality of service or reservation of network resources. IP networks and telecommunications 19 other header info Timestamp Sequence number SSRC CSRC CSRC CSRC CSRC RTP header Timing references Source Identifiers Figure 2.2 RTP header. Proprietary private data formats also are used for data transport between the media server and the browser client. An example is RealNetworks Real Data Transport (RDT). Real-Time Control Protocol (RTCP) RTCP is used in conjunction with RTP. It gives feedback to each participant in an RTP session that can be used to control the session. The messages include reception reports, including number of packets lost and jitter statistics (early or late arrivals). This information potentially can be used by higher layer applications to modify the transmission. For example, the bit rate of a stream could be

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Real-time protocols A number of different protocols have been developed to facilitate real-time streaming of multimedia content. Streaming means that the mean frame rate of the video viewed at the player is dictated by the transmitted frame rate. The delivery rate has to be controlled so that the video data arrives just before it is required for display on the player. The associated audio track or tracks must also remain synchronized to the video. IP data transmission is not a synchronous process and delivery is by best effort. To achieve synchronism, timing references have to be embedded in the stream. 18 The Technology of Video and Audio Streaming Table 2.3 Summary of Protocols Used for Multimedia Sessions Abbreviation Title Notes RFC number RSVP Resource Reservation Protocol Protocol specification 2205 RSVP applicability statement Guide to deployment 2208 Message processing rules 2209 RTCP Real-Time Control Protocol Part of RTP 1889 RTSP Real-Time Streaming Protocol 2326 RTP Real-Time Protocol 1889 SDP Session Description Protocol 2327 UDP User Datagram Protocol 768 The Internet Engineering Task Force issues Request For Comment documents (RFC) that become the de facto protocols. Intimately linked to real-time delivery is the quality of service (QoS). To ensure the reliable delivery of packets, the network bandwidth would have to be reserved for the stream. This generally is not the case with the Internet. One protocol that allows resources to be reserved by a client is Resource Reservation Protocol (RSVP). It allows the client to negotiate with routers in the path for bandwidth, but does not actually deliver the data. RSVP is not widely supported. Transport protocol for real-time applications (RTP) Real-Time Protocol (RTP) is a transport protocol that was developed for streaming data. RTP includes extra data fields not present in TCP. It provides a timestamp and sequence number to facilitate the data transport timing, and allows control of the media server so that the video stream is served at the correct rate for real-time display. The media player then uses these RTP fields to assemble the received packets into the correct order and playback rate.

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col for several application-layer protocols, notably the Network File System (NFS), Simple Network Management Protocol (SNMP), and the Domain Name System (DNS). UDP has neither the error correction nor the flow control of TCP, so this task has to be handled by an application at a higher layer in the stack. It does, however, carry a checksum of the payload data. The media players can often mask video data errors. IP version 6 Most of the Internet uses IP version 4. This protocol has been around since 1981, but is showing its age as use of the Internet has spiraled. It now has many shortcomings, so IP version 6 is offering to solve many of the problem issues. The first problem is lack of addresses. As more and more users connect to the Internet, version 4 addresses are going to run out. The use of always-on broadband connections means that the dynamic sharing of IP addresses (used with dial-up modems) can no longer be used to advantage. One solution to better utilization of the existing address ranges is to move from the fixed number groups of the A, B, and C classes to classless addressing or CIDR (classless inter-domain routing). The class D addresses reserved for multicast are particularly limited in number. If multicasting is to be exploited to save network congestion, many more addresses will be needed. IP version 6 solves the address space issue by increasing from 32-bit address space to 128 bits. This gives 6 1023 IP addresses per square meter of the Earth s surface. This may seem to be a ridiculous overkill, but it allows far more freedom for multilevel hierarchies of address allocation. This is the same as telephone numbers, with the hierarchy of area codes. The big advantage of this hierarchy is that the tables in the network routers can be simplified, so the fine-grain routing need only be done at the destination router. Many other improvements have been incorporated into the version 6 protocol, including a simplified packet header, again to improve router throughput. The advantages specific to streaming will be two-fold: the increased address space and the opportunity to manage quality of service (QoS). One header field is the traffic flow identification, which will allow routers to distinguish real-time data from mail and file transfer (FTP). MPEG-4 potentially could take advantage of this packet priority. The scalable coding option provides a baseline low-resolution image, with helper packets to add detail to an image for higher resolution, albeit requiring a higher bandwidth. The low-resolution image could be allocated a higher priority than the high-resolution helper signals. So if the network becomes congested the resolution degrades gracefully as packets are dropped, rather than the stalling that we see with conventional codecs. Routers compliant with IP version 6 support multicasting as a standard facility. IP networks and telecommunications 17

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a specified time period they are retransmitted. This feature of TCP allows devices to detect lost packets and request a retransmission. The repeated transmission will add to the communication latency, but that is not normally an issue with data exchange. TCP also provides flow control of the data. With audio and video, the viewer requires a continuous stream to view the source in real-time. Retransmission of data is going to add delays; retransmission also uses up bandwidth in the data channel. Ultimately, high levels of network transmission errors will empty the receive buffer in the media player. The interruption to the stream will ultimately will lead to interruptions to the video playback. The alternative is to ignore lost packets. This may cause loss or distortion of a single video frame, but that is a transient event that will be ignored by the viewer. So for real-time applications, timely delivery is more important than error-free transmission. User Datagram Protocol (UDP) Streaming needs a transmission protocol that can ignore data errors. Such a protocol is the User Datagram Protocol (UDP). It is used as a transport proto- 16 The Technology of Video and Audio Streaming Table 2.1 TCP versus UDP TCP UDP Connection oriented Connectionless Reliable Unreliable Controls data flow No flow control Table 2.2 Popular Internet Applications and Their Underlying Transport Protocols Application Application-layer Protocol Typical Transport Protocol E-mail SMTP TCP Remote terminal access Telnet TCP Web HTTP TCP File transfer FTP TCP Remote file server NFS UDP Streaming media RTSP or proprietary UDP Voice-over IP Proprietary UDP Network management SNMP UDP Routing protocol RIP UDP Name translation DNS UDP

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Internet Protocol Internet Protocol is the main network (layer 3) communication protocol. The other protocols at layer 3 are used for control of the network routers to set up the connections. IP has a drawback, however; it is an unreliable delivery system. There is variable network latency. The packets can arrive in a different order from transmission. Packets can be lost. These potential problems are corrected by the higher layer protocols and applications. The most well-known protocol is at the transport layer, Transport Control Protocol (TCP). This is used together with Internet Protocol the ubiquitous TCP/IP. One of the great strengths of TCP is its reliability. The builtin error protection of TCP makes it an excellent protocol for the delivery of general purpose data, but the way this is implemented proves to be a disadvantage for streaming applications. TCP sequences the data bytes with a forwarding acknowledgement number that indicates to the destination the next byte the source expects to receive. If bytes are not acknowledged within IP networks and telecommunications 15 Layer 1 Layer 2 Layer 3 Layer 4 Layer 5 Layer 6 Layer 7 Physical Link Network Transport Session Presentation Application ISO open system interface model TCP UDP IP ARP not specified Internet stack Network Access Internet Host-to-host Process DARPA Figure 2.1 Multilayer network model.

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streaming the intervening network has a major impact on the delivered quality of the video and audio. Most streaming files are delivered over a data network. For internal corporate communications it may be the local network or, for an enterprise with widely dispersed sites, a wide-area network. For business-to-business and consumer streaming, the Internet is a likely carrier. The Internet has become ubiquitous for data communications, from simple e-mail to complex electronic commerce applications. The Internet needs a physical layer, the fiber and copper that carry the data. For this we turn to the telcos. It may be your incumbent telephony supplier, or one of the new wideband fiber networks. In all probability, an end-to-end Internet connection will use a combination of many carriers and networks. This chapter gives an overview of the connections that carry the stream. The first section is about data networks. The second is about telecommunications, with the focus on the last mile to your browser. This final link includes the two popular broadband consumer products: DSL and the cable modem. Network layers The concept of interconnected networking, or the Internet, has its origins in the quest by the U.S. military to connect research institutions over a packetswitched network. In the 1970s the U.S. Department of Defense DARPA project developed the multilayer model of network protocols that evolved into today s Internet. The International Standards Organization later augmented the communication protocols, which evolved into the Open Systems Interface model (the ISO OSI). The Internet does not wholly adhere to the open systems interface; Figure 2.1 shows the relationship, but note that the principles are similar. Later protocols do adhere more closely to the ISO seven-layer model. The DARPA model defined four layers: Network access layer Internet layer Host-to-host layer Process layer The network access layer includes a number of protocols that deliver the data over the physical network (the copper and fiber). The protocol chosen depends upon the type of physical network. The Internet layer delivers the data packets from the source to the destination address. It masks the network technology in use from other layers. The host-to-host layer handles the connection rendezvous and manages the data flow. The process layer is for the implementation of the user application, like e-mail and file transfer. 14 The Technology of Video and Audio Streaming

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2 IP networks and telecommunications Introduction Why is a network drawn as a cloud? Is it because most are unaware of the processes happening as data tranverses the network? In all but the smallest of enterprises, the IT department maintains the computer network infrastructure. How many of us are concerned with what happens beyond the RJ-45 jack on the wall? It says a lot for computer networks that we often forget the network is there. We haul files across the world at the click of a mouse, and they arrive seconds later. With video media, things are different; we place many more demands on the network. So it helps to understand a little more about the data network, and the telecommunications infrastructure that underpins it. The first thing that is different about the delivery of multimedia streams is that usually they do not use the universal TCP/IP (Transport Control Protocol over Internet Protocol). Second, the media files are very large compared with the average e-mail message or web page. Third, delivery in real-time is a prerequisite for smooth playback of video and audio. A new set of network protocols has been developed to support multimedia streaming. As an example, advances in Internet protocols now support multicasting, where one media stream serves hundreds or thousands of players. This is a handy facility for optimizing network resources if you want to webcast live to large audiences. The media files are streamed over the general telecommunications network. Again, this is something we rarely think about, unless your company wants a new telephone switch. Communications channels become an issue as soon as you start to encode. The codec (compression/decompression) configuration menu will offer a number of compression choices: dial-up modem, dual-ISDN, DSL, T-1. So it helps to understand the pipes through which the media is delivered. Streaming is not like the web page where the content arrives after a short delay, and how it reached the browser is of little concern to the user. With

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peer-to-peer distribution has driven the need to deploy digital rights management systems to protect the intellectual property of the content creators and owners. Streaming technology is very fast-moving. New versions of codecs are released every year. New technologies obsolete the incumbent, so any streaming content creation and management system must be designed to be flexible and extensible. Some of the newer applications like mobile and wireless are likely to be more stable. The phone manufacturers prefer fixed standards to ensure reliable operation and low manufacturing cost. Perhaps the greatest advance that benefits the content creator is the recent emergence of tools to aid the production processes. Just as the word processor brought basic DTP to every desktop, these tools will allow the small business and corporate user to deploy streaming without the need to outsource. The streaming production shop will be freed to concentrate on the more creative content creation. 12 The Technology of Video and Audio Streaming Information technology Web development Teleproduction Streaming Media Figure 1.5 The production team.

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The development of streaming has benefited from a very wide range of disciplines. We can thank the neurophysiologists for the research in understanding the psychoacoustics of human hearing that has been so vital to the design of audio compression algorithms. Similar work has led to advances in video compression. The information technology engineers constantly are improving content delivery within the framework of the existing Web infrastructure. We must not forget the creativity of the multimedia developer in exploiting the technologies to produce visually stimulating content. And a final word for Napster; Introduction 11 Capture Streaming Media Video & Audio Encode Distribute Play Chapter 3 World Wide Web Chapter 2 IP Networks and Telecommunications Chapter 4 Video Formats Section 1 Basics Chapter 5 Video Compression Chapter 11 Stream Serving Chapter 10 Preprocessing Chapter 8 Video Encoding Chapter 9 Audio Encoding Chapter 12 Live Webcasting Chapter 6 Audio Compression Chapter 13 Media Players Section 2 Streaming Figure 1.4 The chapter content.

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standards, with the Bell hierarchy in the United States giving a base broadband rate of 1.5 Mbit/s (T-1), and the 2 Mbit/s (E-1) of the Europe Telecommunications Standards Institute (ETSI). It is relatively easy to convert from one to another, so the differing standards are not an obstacle to international media delivery. The production team Much like web design, streaming media production requires a multidisciplinary team. A web site requires content authors, graphic designers, and web developers. The site also needs IT staff to run the servers and security systems. To utilize streaming you will have to add the video production team to this group of people. This is the same as a television production team, but the videographer should understand the limitations of the medium. Streaming media players are not high-definition television. If you are producing rich media, many of the skills should already be present in your web team. These include the design skills plus the ability to write the SMIL and TIME scripts used to synchronize the many elements of an interactive production. So, with luck, you may not need to add to your web production team to incorporate streaming. How this book is organized This book is divided into three sections. The first is a background to telecommunications and audio/video compression. The second section contains the core chapters on streaming. The final section covers associated technologies and some applications for streaming media. The book is not intended to replace the operation and installation manuals provided by the vendors of streaming architectures. Those will give much more detail on the specifics of setting up their products. Summary Streaming media presents the professional communicator with a whole new way to deliver information, messages, and entertainment. By leveraging the Internet, distribution costs can be much lower than the traditional media. The successful webcaster will need to assemble a multiskilled and creative team to produce high-quality streaming media content. The Web audience is unforgiving, so content has to be compelling to receive worthwhile viewing figures that will give a return on the investment in streaming. 10 The Technology of Video and Audio Streaming

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