d1vol2.doc.doc

Project P913-GI

Real-time services on the InternetDeliverable 1

Road-map for current real-time services over the Internet

Volume 2 of 2: Annexes

Suggested readers:

Participants in P913-GI Staff of Shareholders responsible for or involved in the provision of real-time

Internet services Staff of Shareholders who are involved in offering services over the Internet with

real-time and guaranteed bandwidth requirements R&D engineers working on or interested in advanced Internet services Managers at Shareholders involved in Internet services

For full publication

May 1999

EURESCOM PARTICIPANTS in Project P913-GI are:

BT

Deutsche Telekom AG

France Télécom

Portugal Telecom S.A.

This document contains material which is the copyright of certain EURESCOM PARTICIPANTS, and may not be reproduced or copied without permission.

All PARTICIPANTS have agreed to full publication of this document.

The commercial use of any information contained in this document may require a license from the proprietor of that information.

Neither the PARTICIPANTS nor EURESCOM warrant that the information contained in the report is capable of use, or that use of the information is free from risk, and accept no liability for loss or damage suffered by any person using this information.

This document has been approved by EURESCOM Board of Governors for distribution to all EURESCOM Shareholders.

ã 1999 EURESCOM Participants in Project P913-GI

Deliverable 1 Volume 2: Annexes

Preface

(Prepared by the EURESCOM Permanent Staff)

Real-time services over the Internet are gaining increasing importance and attention from customers, who expect reduced costs and from operators who need to open new markets. The Internet, designed as a connection-less transport technology poses, however, considerable challenges if used over and as replacement of connection-oriented network systems, especially when real-time and high quality are main requirements. Real-time conversational services, such as Internet telephony are currently being deployed and it is expected that they will rapidly gain a share of the market. More demanding real-time services, such as audio/video streaming or multi-party conversational services still suffer from many technical problems which need to be resolved before realistic success can be expected.

The Project will investigate both AV-streaming and multi-party conversational services and will test how these services will perform in best-effort Internet and with Internet solutions which offer some bandwidth control. From these results possible scenarios, recommendations on protocols and solutions for access networks, Internet and Internet Gateways will be produced.

As key results the Project will produce:

A Roadmap for key issues on real-time streaming services over the Internet

Knowledge about the abilities and constraints of IP based real-time technology today.

Recommendations about solutions, possible configurations and protocols to be used for the deployment of real-time streaming services over Internet in Europe.

The Project started its activities on January 1999 and will be finished in December 1999. The Project is led by Hans-Detlef Schulz (Deutsche Telekom, MTG). BT, DT, FT and PT are participating in the Project.

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Volume 2: Annexes Deliverable 1

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List of Authors

Kevin Wilson BT

Andrew Auchterlonie Deutsche Telekom

Hans-Detlef Schulz Deutsche Telekom

David Girault France Télécom

Stephane Pallavicini France Télécom

João Paulo Firmeza Portugal Telecom

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Table of Contents

Preface............................................................................................................. i

List of Authors...............................................................................................iii

Table of Contents...........................................................................................iv

Abbreviations.................................................................................................vi

Annex A: Review of Internet Standards.......................................................1A.1 Standardisation Bodies...................................................................1

A.1.1 International Telecommunications Union (ITU)..................1A.1.2 The Internet Engineering Task Force (IETF).......................1A.1.3 European Telecommunications Standards Institute (ETSI). .1A.1.4 International Multimedia Teleconferencing Consortium

(IMTC)..............................................................................1A.1.5 International Standardisation Organisation (ISO) &

ISO/MPEG.........................................................................1A.1.6 DAVIC..............................................................................2

A.2 The Internet Suite of Protocols........................................................2A.2.1 Internet Protocol (IP)..........................................................2A.2.2 Transmission Control Protocol (TCP).................................5A.2.3 User Datagram Protocol (UDP)..........................................6A.2.4 Hypertext Transfer Protocol (HTTP) Streaming..................6

A.3 Real-time services specific Protocols..............................................7A.3.1 Real Time Transport Protocol (RTP) and Real Time Control

Protocol (RTCP)................................................................7A.3.2 Real Time Streaming Protocol (RTSP)..............................7A.3.3 Reservation Protocol (RSVP).............................................8A.3.4 Session Description Protocol (SDP)....................................9A.3.5 Session Initiation Protocol (SIP).........................................9

A.4 Audio and Video Standards...........................................................10A.4.1 H.323...............................................................................10A.4.2 MPEG..............................................................................12A.4.3 T.120................................................................................15A.4.4 Other standards.................................................................17

A.5 Proprietary Encoding Schemes......................................................18A.5.1 RealNetwork’s RealSystem G2.........................................18A.5.2 Microsoft NetShow and Active Streaming Format............18A.5.3 Apple QuickTime.............................................................19

A.6 Coding Techniques Description....................................................19A.6.1 Layered Coding................................................................19A.6.2 Redundant Coding............................................................19A.6.3 Transcoding......................................................................20

Annex B: Review on tools & commercial products.....................................21B.1 Applications audio/video streaming..............................................21

B.1.1 RealAudio/Real Video......................................................21B.1.2 Microsoft NetShow Services.............................................23B.1.3 Xing Tech. Streamworks..................................................24B.1.4 VDO Live........................................................................25B.1.5 Apple Quicktime..............................................................26B.1.6 Cisco IP/TV.....................................................................27

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B.1.7 DT/MTG Music on Demand application...........................28B.1.8 Oracle iTV.......................................................................29B.1.9 Feature “at-a-glance“ comparison form.............................32

B.2 Applications for telephony/conferencing.......................................33B.2.1 Microsoft NetMeeting......................................................33B.2.2 CU-SeeMe.......................................................................34B.2.3 WhitePine’s MeetingPoint tools........................................34B.2.4 VocalTec Internet Phone Lite............................................35B.2.5 VDO Phone......................................................................36B.2.6 Isabel................................................................................37B.2.7 NetSpeak WebPhone........................................................38B.2.8 Netscape CoolTalk...........................................................38B.2.9 VIC/VAT - Video Conferencing Tool...............................39B.2.10 Intel Internet Video Phone................................................40B.2.11 Hardware based Internet conferencing solutions................40B.2.12 Feature “at-a-glance“ comparison form.............................44

Annex C: Market Trends............................................................................46C.1 Introduction..................................................................................46C.2 Market Trends in different Business Segments..............................46

C.2.1 Internet “Core Technology”..............................................46C.2.2 Trends in Software Technology........................................48C.2.3 Internet Content oriented Applications..............................51C.2.4 Offers from Network Operators........................................52C.2.5 Internet Service Providers.................................................53

C.3 Market Trends in different Global Regions....................................54C.3.1 North America..................................................................54C.3.2 Europe..............................................................................55C.3.3 Asian Pacific region..........................................................55

C.4 Trends in the Behaviour of Users..................................................56C.4.1 Private Users....................................................................56C.4.2 Public Users.....................................................................56C.4.3 Corporate Users................................................................57

C.5 Market Trends on Quality-Requirements.......................................57C.6 Figures about the market development..........................................59C.7 Conclusions about Market Trends.................................................60

Abbreviations

A/V Audio/Video

ADSL Asymmetric Digital Subscriber Line

AF Assured Forwarding

AoD Audio-on-Demand

AQUAVIT Assessment of QUality for Audio-Visual signals over Internet and UMTS

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ARP Address Resolution Protocol

ATM Asynchronous Transfer Mode

BGP Border Gateway Protocol

BRFQ Bit Round Fair Queuing

CBQ Class Based Queuing

CLIP Classical IP

CPU Central Processing Unit

DAVIC Digital AudioVIsual Council

DHCP Dynamic Host Configuration Protocol

DiffServ Differenciated Services

DNS Domain Name Service/Server

DTMF Dual Tone Multi Frequency

EBONE European backBONE

EF Expedited Forwarding

ETSI European Telecommunications Standards Institute

FAQ Frequently Asked Questions

FDDI Fiber Distributed Data Interface

FIFO First-In-First-Out

FQ Fair Queuing

FQDN Fully Qualified Domain Name

FTP File Transfer Protocol

GPS Global Positioning System

HDLC High level Data Link Control

HTML Hypertext Markup Language

HTTP HyperText Transfer Protocol

IAB Internet Architecture Board

IAP Internet Access Provider

ICMP Internet Control Messaging Protocol

IGMP Internet Group Management Protocol

IGP Internal Gateway Protocol

IGRP Interior Gateway Routing Protocol

IETF Internet Engineering Task Force

IntServ Integrated Services

IP Internet Protocol

IPCE InterProcess Communication Environment

IRC Internet Relay Chat

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IRTF Internet Research Task Force

ISDN Integrated Services Digital Network

ISO International Standard Organisation

ISP Internet Service Provider

ITU-T International Telecommunication Union - Telecommunications

JUPITER Joint Usability, Performability and Interoperability Trials in Europe

L2TP Layer 2 Tunneling Protocol

LAN Local Area Network

LLC Logic Link Control

MBONE Multicast backBONE

MIME Multipurpose Internet Mail Extensions

MoD Music-on-Demand

MP3 MPEG1 Audio – Layer 3

MPEG Moving Pictures Expert Group

MPOA Multi Protocol Over ATM

NAS Network Access Server/Service

NBMA Non Broadcast Multiple Access

NIC National Information Center

NNTP Network News Transfer Protocol

NoD News-on-Demand

OSI Open Systems Interconnect

OSPF Open Shortest Path First

PC Personal Computer

PHB Per Hop Behavior

PIN Personal Identification Number

PPP Point-to-Point Protocol

PPTP Point-to-Point Tunneling Protocol

PQ Priority Queuing

PSK Phase-Shift-Keying

PSTN Public Switched Telephone Network

QoS Quality of Service

RFC Request For Comments

RIP Routing Information Protocol

RSVP Resource reSerVation Protocol

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RTCP Real Time Control Protocol

RTP Real time Transport Protocol

RTSP Real Time Streaming Protocol

SAP Session Announcement Protocol

SDH Synchronous Digital Hierarchy

SDLC Synchronous Data Link Control

SDP Session Description Protocol

SIP Session Initiation Protocol

SMIL Synchronized Multimedia Integration Language

SMTP Simple Mail Transfer Protocol

SOCRATES Streaming and Online Conversational Real Time Services

TCP Transmission Control Protocol

UDP User Datagram Protocol

URI Uniform Resource Identifier

URL Uniform Resource Locator

VCR Video Cassette Recorder

VoD Video-on-Demand

W3C World Wide Web Consortium

WAN Wide Area Network

WDM Wavelength Division Multiplexing

WFQ Weighted Fair Queuing

WRED Weighted Random Early Detection

WWW World Wide Web

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Annex A: Review of Internet Standards

A number of standards have been developed for the Internet by a number of international standardisation bodies and independent organisations. This section identifies and discusses the most important standardisation bodies which exist and the main standards which are in use.

A.1 Standardisation Bodies

A.1.1 International Telecommunications Union (ITU)

A.1.2 The Internet Engineering Task Force (IETF)

The IETF defines the standards for use on the Internet. It is strongly supported by industrial and academic institutions. The IETF, as with the ITU, operates by forming working groups which research and standardise various aspects of Internet technology. In general, the IETF is less bureaucratic that the ITU-T and is more fast-moving in its deliberations.

A.1.3 European Telecommunications Standards Institute (ETSI)

ETSI is responsible for developing European standards for telecommunications and does this in close co-operation with other bodies such as the ITU-T, as well as with industrial and academic institutions. ETSI is currently operating the TIPHON project, a world-wide project which is concerned with the interoperability of services between IP networks and non-IP networks (e.g. PSTN, ISDN, GSM and UMTS).

A.1.4 International Multimedia Teleconferencing Consortium (IMTC)

The IMTC aims to promote the development and implementation of interoperable multimedia solutions based on international standards, and has over 140 members and affiliates from across the globe. The IMTC organises interoperability test sessions which are attended by vendors of standards-based conferencing products and services. The IMTC also tries to educate businesses and residential customers about the potential value and benefits to be gleaned from the technologies and resultant applications. IMTC members provide submissions and feedback to the standardisation bodies in an attempt to help improve the interoperability and usability of multimedia teleconferencing products.

A.1.5 International Standardisation Organisation (ISO) & ISO/MPEG

The ISO was founded in 1947 with the aim of promoting the development of standards to enable global exchanging of goods and services and to encourage co-operation in the areas of intellectual, scientific and commercial activity. In terms of the Internet, the ISO and the IETF communicate with each other to aid the development of internationally acceptable standards.

The Moving Picture Coding Experts Group (MPEG) was established in January 1988 with the mandate to develop international standards for compression, decompression, processing, and coded representation of moving pictures, audio and their combination. It operates in the framework of the Joint ISO/IEC Technical Committee (JTC 1) on

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Information Technology and is formally WG11 of SC29. MPEG usually holds three meetings a year, attended by some 300 experts from 20 countries. So far the MPEG family of standards includes MPEG-1, MPEG-2 and MPEG-4, formally known as ISO/IEC-11172, ISO/IEC-13818 and ISO/IEC-14496.

A.1.6 DAVIC

DAVIC was formed in 1994 in Switzerland and has a membership which includes over 157 companies from a diverse range of countries. This membership includes companies from many different sections of the audio/visual industry. The main aims of DAVIC are to promote the success of interactive digital audio/visual applications and services and to develop industry standards for end-to-end interoperability of broadcast and interactive digital audio/visual and multimedia communication. To date DAVIC specifications 1.0 to 1.4 have been published. DAVIC is working to extend the standards for TV-anytime and TV-anywhere into IP-based applications and services.

A.2 The Internet Suite of Protocols

Many different protocols are used over the Internet. Each of the protocols have their own characteristics, strengths and weaknesses and are used for different types of applications which require different Qualitys of Service (QoS). The entire set of Internet Protocols is referred to as the IP Suite of Protocols. This section briefly examines each of the protocols, and the standards by which they are defined.

A.2.1 Internet Protocol (IP)

Internet Protocol (IP) was originally developed by the US Defence Advanced Research Projects Agency (DARPA) in a strategic attempt to create a computer network that was distributed over a very wide area to reduce the potential risk of information loss in the event of a nuclear strike on a US city. It evolved from the use of multiple interconnected X.25 WANs and X.75-based gateways which were unsuitable for internet-wide deployment due to X.25 packets having a very large header, resulting in a low packet throughput in the networks. The number of IP networks in use has grown considerably, and these networks are interconnected to form the global Internet. IP is an internet-wide protocol which enables two transport protocol entities resident on different hosts to exchange message units in a transparent way. IP is only one of the many protocols in the complete protocol suite. There are several versions of IP in commercial use. The version which is in general use at the moment is IPv4. However a new version, IPv6 is currently in use in a number of research institutions, and is likely to be used more widely in the early years of the next millennium. This section discusses the properties of both versions, and highlights the similarities and differences between them.

A.2.1.1 IPv4

As with all packet-based protocols, IPv4 datagrams are packets of data which contain some header information and the data payload. Figure 1 shows the format of IPv4 datagrams.

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Version

Time to Live

Padding

IHL Type of Service Total Length

Identification Flags Fragment Offset

Protocol Header Checksum

Source Address

Destination Address

Options

Datagram payload<65'536 bytes

Figure 1: IP Packet structure (IPv4)

Below is a list and description of the field types in the IP datagram :

Version: This field indicates what version of IP was used to create the datagram.

Header length: This indicates the actual length of the datagram in multiples of 32-bit words.

Type of Service: This field allows an application to specify the preferred attributes of the route which the packet will take. Each gateway and router uses this information to forward the packet across the most suitable link.

Total Length: This indicates the total length of the datagram including the header and payload

Identification: Allows one user message to be sent in a number of different datagrams.

Flag bits: Three bits which are used to indicate various requirements to the IP gateways and routers.

Time to Live: specifies a maximum number of hops for which a datagram can be in transit across the Internet. If the field has a zero value, the packet is discarded.

Protocol: Allows the destination IP to pass the datagram to the relevant protocol.

Header Checksum: Helps to avoid corruption of the header by performing 1’s compliment on the header and then inverting this at the receiver

Destination/Source address: IP addresses of the source and destination hosts.

Options: This field carries additional information regarding security, source routing, route recording, stream identification and time-stamping.

IPv4 is based upon open standards which were developed by a number of educational and research institution, including DARPA itself. These standards were initially published in 1981 and have been modified only slightly since then. The main IPv4



standard is RFC 7911. A new version of IP, IPv6, has been developed to help overcome the restrictions of IPv4.

A.2.1.2 IPv6

While IPv4 has been the main protocol which has been used on the Internet over the last twenty years or so, it has a number of weaknesses in terms of scalability, security, management and performance. The addressing fields of the IPv4 packet header can only accommodate 32-bit addresses, meaning that there will be an insufficient number of IP addresses to cope with the continuing popularity and increasing ubiquity of the Internet. An additional problem with IPv4 was that the packet-routing tables which are required to be stored at each exterior gateway were becoming very large due to the almost random way in which IP addresses were allocated. IPv6 was developed in order to alleviate these problems. IPv6 provides enhanced support for multicasting and is capable of supporting ‘anycast’ applications whereby a packet is forwarded on to any one person from a group of people. It also enables users to transmit packets more securely and to authenticate the source of an incoming packet. By introducing a simple hierarchical addressing structure, and by simplifying the standard header of packets, network performance should improve. In addition, IPv6 has a means of specifying a QoS at the network level, meaning that different traffic types can be handled more appropriately.

The header of IPv6 packets is shown in Figure 2.

Figure 2: Structure of IPv6 Packet

The header of IPv6 packets has been simplified relative to IPv4 to enable faster processing of packets and higher throughput at routers. The extension headers are only included in the packet if the packet requires any special services or processing at each router. In general, the fields in the IPv6 header perform similar functions to those in the IPv4 header. There are, however, some new fields in the IPv6 header. The flow label identifies the type of information being carried by the packet. The maximum size of the payload is 65536 octets as before, although larger payload sizes can be accommodated in IPv6 by setting the Payload Length field to zero and using an

1 This document (RFC791) can be viewed at http://www.ietf.org/rfc/rfc0791.txt


http://www.ietf.org/rfc/rfc0791.txt


optional header to specify the actual size of the payload. The time-to-live field contains a number which indicates the maximum number of links over which a packet is allowed to pass before it is discarded. This prevents it from circulating endlessly on the networks. The source and destination address fields both contain 128-bit hierarchical addresses. This address space is sufficient to allow everyone on the planet to have numerous IP addresses. While IPv4-based exterior gateways use netid and subnetid addresses, IPv6-based exterior gateways use an additional address called the cluster address. This address identifies the topological region in which the network (and hence the host) is located.

The complete standard for IPv6 defined in IETF document RFC18831.

A.2.2 Transmission Control Protocol (TCP)

Since the maximum payload size of an IP datagram is 65536 octets, it is often necessary to break up larger user messages into many packets and send them individually across the network. When the packets are received at the destination, the original message is re-assembled. This process is achieved by using Transmission Control Protocol (TCP). TCP is responsible for establishing the end to end connection and allocating input and output ports to be used for the duration of the communication. TCP breaks up the message into fixed-size packets and adds the TCP header, which is shown in Figure 3 below. These TCP packets are then passed to the IP which places them inside the IP datagrams and adds the IP header.

Figure 3: TCP Header structure

The source and destination ports contain information regarding which TCP ports are to be used by the communicating devices. The sequence number ensures that the receiving terminal can assemble the incoming packets into the correct order, and the acknowledgement number indicates what acknowledgement packet number should be sent from the receiver. The total number of octets in the packet is counted and the result is placed into the Checksum field. This helps the routing equipment and the receiver check for errors or corrupted packets.

Once the TCP packet is assembled, it is passed to the IP. The IP is not concerned with what is in the TCP payload, or the TCP header. IP's function is to find a route for the

1 This document can be viewed at http://www.ietf.org/rfc/rfc1183.txt




packet across the network. TCP is responsible for arranging retransmission of packets in the event of packet loss or corruption. This ability to deal with lost and corrupted packets means that TCP can provide a reliable stream of data.

The TCP standard is defined in IETF document RFC7931.

A.2.3 User Datagram Protocol (UDP)

Unlike TCP, the User Datagram Protocol (UDP) does not require that a connection be established with another program in order to exchange information. Data is exchanged in discrete units called datagrams, which are similar to IP datagrams. In fact, the only features that UDP offers over raw IP datagrams are the inclusion of port numbers and an optional checksum. In UDP, there is no scope for retransmitting packets which have been lost or corrupted. As such, UDP is sometimes referred to as an ‘unreliable’ protocol because when a program sends a UDP datagram over the network, there is no method of confirming its arrival at the destination. This means that the sender and receiver must typically implement their own application-layer protocol to deal with packet loss and data corruption. Much of the work that TCP does transparently (such as generating checksums, acknowledging the receipt of packets, arranging retransmission of lost packets and so on) must be performed by the application itself.

With the limitations of UDP, one might wonder why it is used at all. UDP has an advantage over TCP in two critical areas: speed and packet overhead. Because TCP is a reliable protocol, it goes to great lengths to insure that data arrives at its destination intact, and as a result it exchanges a fairly high number of packets over the network. UDP doesn't have this overhead, and is considerably faster than TCP. In those situations where speed is paramount, or the number of packets sent over the network must be kept to a minimum, UDP can be a solution.

The UDP standard is defined in IETF document RFC7682.

A.2.4 Hypertext Transfer Protocol (HTTP) Streaming

Hypertext Transfer Protocol (HTTP) is the protocol which is used by most servers for communications with remote hosts and enables the remote host to gain access to the files and drives on another host (the server) which could be in another network on another continent. For the remote host to be able to get access to the required file on the other machine, the other machine must be running a web server program. The remote host sends a request to the server, which then retrieves the file from the appropriate location and returns it to the host via the relevant protocol (e.g. TCP/IP). The server also indicates to the host what type of file is being returned, so that the host can launch the necessary application(s). HTTP only specifies what the host and the server say to each other, and not what means they use to say it. The TCP/IP protocol suite is used for this.

The standard for HTTP 1.0 is defined in IETF document RFC19453 and the standard for HTTP 1.1 is defined in IETF document RFC20684.

1 This document can be viewed at http://www.ietf.org/rfc/rfc0793.txt2 This document can be viewed at http://www.ietf.org/rfc/rfc0768.txt3 This document can be viewed at http://www.ietf.org/rfc/rfc1945.txt4 This document can be viewed at http://www.ietf.org/rfc/rfc2068.txt







A.3 Real-time services specific Protocols

In the purpose of real-time services such as audio-video streaming or video-conferencing, many new protocols have been standardised in the last few years. These protocols have the goal of compensating for the lack of features proposed by the classic Internet suite of protocols in regards with real-time services specificity.

A.3.1 Real Time Transport Protocol (RTP) and Real Time Control Protocol (RTCP)

Real Time Transport Protocol (RTP) is an IP-based protocol which enables real-time streams such as video and audio to be transported across the Internet. RTP provides services for the transported media such as time-reconstruction, loss detection, security and content identification. It was designed mainly for use with multicast, but can also be used with unicast. RTP provides a solution to the problem of trying to carry synchronous multimedia traffic over an asynchronous shared datagram network. It provides time-stamping of packets to enable the receiver to play back the media at the correct rate. The time stamping also enables the application layer to synchronise different streams, such as audio and video data in MPEG. The packets also require to be put in order, as one video frame may be split into a number of packets. Sequence numbers are added to the packet overhead for this purpose. In a multimedia session, each medium is carried in a separate RTP session and synchronisation of the two media sessions must be performed by the application.

RTP works in unison with the Real Time Control Protocol (RTCP) to get feedback on the QoS as well as information regarding who is participating in the on-going session. RTCP also provides information on source identification, inter-media synchronisation and control information scaling. Both RTP and RTCP are usually transmitted over UDP, although efforts have been made to allow them to be transmitted over the Connectionless Network Protocol (CLNP), Internetwork Packet Exchange (IPX) and ATM networks. Some extensions to RTP and RTCP headers for specific audio-video codecs are defined in specific internet-drafts or RFCs

The standards for RTP and RTCP are defined in IETF document RFC18891.

A.3.2 Real Time Streaming Protocol (RTSP)

The Real Time Streaming Protocol, or RTSP, is an application-level protocol for control over the delivery of data with real-time properties. RTSP provides an extensible framework to enable controlled, on-demand delivery of real-time data, such as audio and video. Sources of data can include both live data feeds and stored clips. This protocol is intended to control multiple data delivery sessions, provide a means for choosing delivery channels such as UDP, multicast UDP and TCP, and provide a means for choosing delivery mechanisms based upon RTP.

The Real-Time Streaming Protocol (RTSP) establishes and controls either a single or several time-synchronised streams of continuous media such as audio and video. It does not typically deliver the continuous streams itself, although interleaving of the continuous media stream with the control stream is possible. In other words, RTSP acts as a "network remote control" for multimedia servers.

There is no notion of an RTSP connection; instead, a server maintains a session labelled by an identifier. An RTSP session is in no way tied to a transport-level





connection such as a TCP connection. The streams controlled by RTSP may use RTP, but the operation of RTSP does not depend on the transport mechanism used to carry continuous media. The protocol is intentionally similar in syntax and operation to HTTP/1. Therefore it has some overlap in functionality with HTTP. It also may interact with HTTP in that the initial contact with streaming content is often to be made through a web page. However, RTSP differs fundamentally from HTTP in that data delivery takes place out-of-band in a different protocol.

The protocol supports the following operations: retrieval of media from media server, invitation of a media server to a conference, addition of media to an existing presentation.

The "rtsp" schemes are used to refer to network resources via the RTSP protocol. The scheme-specific syntax and semantics for RTSP URLs are as follows :

rtsp_URL = rtsp://host:port[abs_path]

For example, the RTSP URL rtsp://media.example.com:554/twister/audiotrack, identifies the audio stream within the presentation "twister", which can be controlled via RTSP requests issued over a TCP connection to port 1554 of host media.example.com.

The standard for RTSP is defined in the IETF document RFC23261.

A.3.3 Reservation Protocol (RSVP)

The Reservation Protocol (RSVP) is a control and signalling protocol which allows the data receiver to request an end-to-end QoS for its data flows. The real time application uses RSVP to reserve space in the buffers of nodes along the transmission path so that the necessary bandwidth can be available when the transmission actually takes place. The RSVP negotiates the connection parameters and maintains router and host states so that the requested service can be provided. Each router may deal with RSVP requests in different ways and may use different means of reserving bandwidth on the channel. The reservation process does not actually provide the guaranteed QoS for the application, but it does guarantee that the network resources required will be available when transmission actually takes place. The processing capability and task scheduling of the computer they are using can also affect the QoS perceived by the end users. RSVP does not scale well, and as a result much research activity has gone into finding other ways of improving QoS. The most notable outcomes of this research so far include Integrated and Differential Services. These technologies are discussed in Section 2.5.2.

The standards for RSVP are defined in the IETF documents RFC22052, RFC22063, RFC22074, RFC22085, RFC22096.

1 This document can be viewed at http://www.ietf.org/rfc/rfc2326.txt2 This document can be viewed at http://www.ietf.org/rfc/rfc2205.txt3 This document can be viewed at http://www.ietf.org/rfc/rfc2206.txt4 This document can be viewed at http://www.ietf.org/rfc/rfc2207.txt5 This document can be viewed at http://www.ietf.org/rfc/rfc2208.txt6 This document can be viewed at http://www.ietf.org/rfc/rfc2209.txt









A.3.4 Session Description Protocol (SDP)

The Session Description Protocol, SDP, is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. It has been standardised by the Multiparty Multimedia Session Control (MMUSIC) working group of the IETF.

This protocol has the purpose of advertising multimedia conferences and communicate the conference addresses and conference tool-specific information necessary for participation, on the Internet multicast backbone (Mbone.)

The Mbone is the part of the internet that supports IP multicast, and thus permits efficient many-to-many communication. It is used extensively for multimedia conferencing. Such conferences usually have the property that tight co-ordination of conference membership is not necessary; to receive a conference, a user at an Mbone site only has to know the conference's multicast group address and the UDP ports for the conference data streams. Session directories assist the advertisement of conference sessions and communicate the relevant conference set-up information to prospective participants. SDP is designed to convey such information to recipients. SDP is purely a format for session description – it does not incorporate a transport protocol, and is intended to use different transport protocols as appropriate including the Session Announcement Protocol, Session Initiation Protocol, Real-Time Streaming Protocol, electronic mail using the MIME extensions, and the Hypertext Transport Protocol.

SDP is not intended for negotiation of media encodings. The purpose of SDP is to convey information about media streams in multimedia sessions to allow the recipients of a session description to participate in the session. A multimedia session, for these purposes, is defined as a set of media streams that exist for some duration of time. Media streams can be many-to-many. The times during which the session is active need not be continuous.

In such an environment, SDP serves two primary purposes. It is a means to communicate the existence of a session, and is a means to convey sufficient information to enable joining and participating in the session. In a unicast environment, only the latter purpose is likely to be relevant.

The standards for SDP is defined in IETF document RFC23271.

A.3.5 Session Initiation Protocol (SIP)

The Session Initiation Protocol can be used to initiate sessions as well as invite members to sessions that have been advertised and established by other means. Sessions can be advertised using multicast protocols such as SAP, electronic mail, news groups, web pages or directories (LDAP), among others.

SIP supports five facets of establishing and terminating multimedia communications:

User location: determination of the end system to be used for communication;

User capabilities: determination of the media and media parameters to be used;

User availability: determination of the willingness of the called party to engage in communications;





Call setup: "ringing", establishment of call parameters at both called and calling party;

Call handling: including transfer and termination of calls.

SIP can also initiate multi-party calls using a multipoint control unit (MCU) or fully-meshed interconnection instead of multicast. Internet telephony gateways that connect Public Switched Telephone Network (PSTN) parties can also use SIP to set up calls between them.

SIP can also be used in conjunction with other call setup and signaling protocols. In that mode, an end system uses SIP exchanges to determine the appropriate end system address and protocol from a given address that is protocol-independent. For example, SIP could be used to determine that the party can be reached via H.323, obtain the H.245 gateway and user address and then use H.225.0 to establish the call.

In another example, SIP might be used to determine that the callee is reachable via the PSTN and indicate the phone number to be called, possibly suggesting an Internet-to-PSTN gateway to be used.

SIP does not offer conference control services such as floor control or voting and does not prescribe how a conference is to be managed, but SIP can be used to introduce conference control protocols. SIP does not allocate multicast addresses.

SIP can invite users to sessions with and without resource reservation. SIP does not reserve resources, but can convey to the invited system the information necessary to do this.

The standard for SIP is defined in IETF document RFC25431.

A.4 Audio and Video Standards

Real time applications such as video and audio must use some sort of compression/encoding scheme to keep the required bandwidth to a minimum without significantly reducing the perceived quality of the media. This section discusses some of the main internationally standardised encoding schemes used for real-time communications on the Internet.

A.4.1 H.323

The H.3232 standard provides a foundation for audio, video and data communications across IP-based networks, including the Internet. By complying with H.323, multimedia products and applications from multiple vendors can interoperate, allowing users to communicate without concern for compatibility conflicts.

H.323 is an umbrella recommendation from the ITU that sets standards for multimedia communications over networks that do not provide a guaranteed QoS. Thus, H.323 standards are important building blocks for a new range of collaborative applications for multimedia communications.

The H.323 specification was approved by ITU Study Group 16 in 1996, and version 2 was approved in January 1998. The standard has a very broad scope, dealing with issues relevant to stand-alone devices, embedded PC technology and point-to-point

1 This document can be viewed at http://www.ietf.org/rfc/rfc2543.txt2 This document can be viewed at http://www.itu.int. A license is required to gain access to ITU documents.


http://www.itu.int/



and multipoint conferencing. H.323 also deals with call control, multimedia management, bandwidth management and interfacing between LANs and other networks. H.323 is a subset of the H.32X series of standards which enable videoconferencing across a range of networks. H.323 is a comprehensive yet flexible standard which can be applied to voice-only handsets as well as full multimedia videoconferencing applications.

H.323 establishes standards for compression and decompression of audio and video data streams, ensuring equipment and software from different vendors will have some common support. The standard also enables the communicating parties to exchange information regarding their processing capabilities. H.323 is designed to run on top of common network architectures. Thus H.323 applications will be able to evolve to take advantage of enhanced network capabilities. The standard provides bandwidth management to prevent the network from being saturated with multimedia traffic. The network manager can limit the total number of simultaneous H.323 connections and can also limit the amount of bandwidth available to H.323 applications. Multipoint conferences can take advantage of H.323‘s multicast capability to make more efficient use of the bandwidth available. H.323 has a means of linking LAN-based desktop systems with ISDN-based group systems to enable conferencing from LANs to remote sites. H.323 uses common codec technology from various videoconferencing standards to minimise transcoding delays to provide optimum performance.

The H.323 standard is based on the IETF real-time protocol (RTP) and real-time control protocol (RTCP), with additional protocols for call signalling and data and audio-visual communications.

H.323 provides a variety of options for audio and video coding. Two codecs, G.711 for audio and H.261 for video, are required by the H.323 specification. H.323 terminals must be able to send and receive G.711 encoded audio (ITU-T A-law and µ-law – as used in standard PCM telephony). There is additional support for audio and video codecs provided in the standard. This gives a variety of standard bit rate, delay, and quality options which are suitable for a range of different networks and applications. H.323 also enables products to negotiate non-standard audio and video codecs.

The list below1 describes the audio and video codecs (G.711 and H.261) which are required for H.323, as well as other codecs which can be included. G.723 and H.263 offer the low-bit rate connections necessary for audio and video transmission over the Internet. :

G.711. The G.711 codec can transmit audio at 48, 56 or 64 kb/s. This codec is appropriate for audio over higher speed connections.

G.723. The G.723 audio codec specifies the format and algorithm used to send and receive voice communications over the network. G.723 transmits audio at 5.3 and 6.3 kb/s, which is adequate for Internet telephony over low-bit-rate connections.

H.261. The H.261 codec transmits video images at 64 kb/s. This codec is appropriate for video over higher speed connections.

H.263. The H.263 video codec specifies the format and algorithm used to send and receive video images over the network. This codec supports CIF, QCIF, and

1 The complete standards of G.711, G.723, H.261 and H.263 can be viewed at http://www.itu.net. A licence is required to gain access to these documents.


http://www.itu.net/


SQCIF picture formats and is adequate for Internet transmission over low-bit-rate connections, such as a 28.8 kb/s modem.

The architecture of H.323 is shown in Figure 4.

Figure 4: Architecture of H.323

H.263 is an ITU standard for video-conferencing communication and is optimised for relatively low data rates and low-motion video. The standard is an advancement on the H.261 and M-PEG video standards. It was designed to produce better picture qualities at data rates below 64 kb/s. H.263 is supported by many architectures including Apple’s QuickTime, Microsoft NetShow and Video for Windows. H263 offers relatively good quality at low data rates, and offers better performance than H.261 in general. However, H.263 requires very powerful processors, and low-to-medium specification processors may have trouble dealing with bit-rates in excess of 50 kb/s. The standard is most suited to encoding low-motion video.

A.4.2 MPEG

MPEG (Motion Pictures Expert Group) is an ISO study group which has developed a series of standards for the compression and transmission of digital video and audio streams and files. The standards define a compressed bit stream, which implies the use of a decompresser. The actual compression algorithm implemented is dependent on the vendor. The underlying philosophy of the MPEG standards is to capitalise on the nature of the human visual and audio system to reduce the amount of information required to digitally represent a moving picture or sound wave.

A.4.2.1 MPEG-video

Like it’s sister standard JPEG, MPEG-video utilises the discrete cosine transform (DCT) and zig-zag coding to discard the high spatial frequencies which are not essential requirements of the human visual system. The visual information which remains after the DCT has been performed is quantised and encoded to remove any statistical redundancy in the bit stream. For compression of video streams and files, MPEG uses adjacent frames of the sequence, due to the fact that there is typically a very high correspondence between adjacent frames, and it would be wasteful to



encode each frame independently when adjacent frames have such similarity. Figure 5 shows how adjacent frames are used to encode the present frame for MPEG-1.

I P2B4B3P1B2B1

Figure 5: MPEG Inter-frame relations

I-frame (Intra-frame): Encoded separately (using DCT) and add error resilience at the expense of a large data rate.

P-frame (Inter-frame): Uses DCT encoding with motion compensation from preceding frames.

B-frame (Inter-frame): Uses DCT encoding with both forward and backward motion compensation.

While the use of adjacent frames in the sequence can help to significantly reduce the amount of data required to represent the video, this advantage is partially offset by the delay introduced by requiring a number of adjacent frames to be cached to enable decoding to take place.

A.4.2.2 MPEG-audio

MPEG-audio principle is based on the human perception ear. The MPEG Audio coders, aimed for generic audio, i.e. all types of speech and music signals, are perceptual audio coders, rather

than so-called 'waveform coders' : its goal is to ensure that the output signal sounds the same to a human listener. It uses a psychoacoustic effect called the 'auditory masking', where parts of a signal are not audible due to the function of the human auditory system. In order to remove the irrelevant part of a signal, the encoder contains a psychoacoustic model, which analyses the input signals within consecutive time blocks and determines for each block the spectral components of the input audio signal by applying a frequency transform. Then it estimates the just noticeable noise-level. In parallel, the input signal is fed through a time-to-frequency mapping, resulting in spectrum components for subsequent coding. In its quantisation and coding stage, the encoder tries to allocate the available number of data bits in a way that meets both the bitrate and masking requirements.

There are 3 layers in MPEG-Audio standard, considering the complexity of the encoder and decoder, the encoder/decoder delay, and the coding efficiency

Layer I has the lowest complexity and is specifically suitable for applications where also the encoder complexity plays an important role. HiFi quality is reached with 384 Kbit/s in stereo mode.

Layer II requires a more complex encoder and a slightly more complex. Compared to Layer I, Layer II is able to remove more of the signal redundancy



and to apply the psychoacoustic threshold more efficiently. HiFi quality is reached with 256 Kbit/s in stereo mode.

Layer III is again more complex and is directed towards lower bit rate applications due to the additional redundancy and irrelevancy extraction from enhanced frequency resolution in its filterbank. HiFi quality is reached with 128 Kbit/s in stereo mode.

A.4.2.3 MPEG-system

MPEG-Systems is a standard that defines the syntax and semantics of bitstreams in which digital audio and visual data are multiplexed. Such bitstreams are said to be MPEG Systems compliant. This specification does not mandate, however, how equipment that produces, transmits, or decodes such bitstreams should be designed. As a result, the specification can be used in a diverse array of environments, including local storage, broadcast (terrestrial and satellite), as well as interactive environments.

A.4.2.4 MPEG levels

There are three levels of MPEG compression, MPEG-1, MPEG-2 and MPEG-4. A fourth standard, MPEG-7 ("Multimedia Content Description Interface") is being developed at the time of writing, and has the purpose to extend the limited capabilities of proprietary solutions in identifying content on multimedia databases.

MPEG-11 is for use over connections with a data capacity of up to 1.5 Mbit/s. It can compress a 352x240x30Hz video to around 1.25 Mbit/s, and stereo audio to around 240 kbit/s. It uses non-interlaced video, and has been optimised for CD-ROM applications.

MPEG-22 is for use over higher bandwidth connections for broadcast applications like digital television (up to 15 Mbit/s for standard TV, 60 Mbit/s for HDTV). It can deal with a variety of frame sizes and up to 5 audio channels simultaneously (enabling HDTV and surround sound to be implemented), and can also cope with interlaced video.

MPEG-43 is building on the proven success of three fields: digital television, interactive graphics applications (synthetic content) and interactive multimedia (World Wide Web, distribution of and access to content). It will provide the standardised technological elements enabling the integration of the production, distribution and content access paradigms of the three fields. MPEG-4 is being developed for very low bandwidth connections, such as dial-up modems. The standard video frame size is 176x144 at 10 frames per second (it is optimised for video-conferencing). MPEG-4 is still under definition.

A.4.3 T.120

The T.120 standard4 was approved by the ITU in 1995 and is aimed at the distributed sharing of documents and other forms of data, including virtual reality, gaming and

1 The document relating to the MPEG-1 standard has the ISO/IEC document number: 11172 (Nov 92)2 The document relating to the MPEG-2 standard has the ISO/IEC document number: 13818 (Nov 94)3 The document relating to the MPEG-4 standard has the ISO/IEC document number: 14496 (version 1 : Oct 98)



real-time news subscription feeds. Document conferencing allows many users to make simultaneous changes to the one document without having to download the document to a local storage medium. The T.120 standard will enable products from different vendors to interoperate, meaning that distributed document sharing can be achieved more readily. Document conferencing can be a very useful tool to use in conjunction with video and audio-conferencing, enabling users to discuss and modify documents online and in real-time. Document and application sharing is more sophisticated than whiteboard technology as it allows changes made by any of the users to be exported immediately to all hosts in the document conference.

T.120 enables developers to easily create a multipoint domain and deliver data in real time with ease. Since T.120 is based on open standards, endpoint applications from multiple vendors can communicate freely. T.120 also specifies how applications may communicate via a number of network bridging products and services which support the T.120 standard.

In multicast-capable networks, T.120 can deliver both reliable and unreliable data streams. The infrastructure of T.120 allows multicast and unicast to be used simultaneously, allowing a flexible solution to be developed for mixed unicast and multicast networks. The T.120 applications have complete transparency from the data transport mechanisms being used, and can run on a broad range of transport options, including PSTN, ISDN and IP. More importantly, these network transport types can co-exist in the same multipoint conference. T.120 applications can be used on almost all standard operating systems, including Microsoft Windows, UNIX, MAC-OS and OS-2 as well as a number of proprietary real-time operating systems. The implementation of the standard is not dependent upon the use of specific hardware, meaning that the standard can scale very well.

Multipoint conferences can be set up without limitation on the logical network topology. Star topologies are likely to be most suitable, although many others could be more appropriate in certain circumstances. In complex multipoint conferences, the network topology which is adopted for the conference can have a significant effect on the performance and efficiency of the communication.

T.120 can work alone or in unison with other ITU standards, such as the H.32x family of videoconferencing standards as well as the V-series of communications standards, and can also be extended to support new standards and transport types.

The architecture of T.120 is multi-layered with well-defined protocols between layers. It should be noted that not many commercially available products are fully T.120 compliant. While many vendors claim to be T.120 compliant, there are still many interworking problems. The overview of the architecture is shown in Figure 5.

4 The document which defines the standard for T.120 can be viewed at http://www.itu.int. A license is required to gain access to the document.


http://www.itu.int/


Figure 6: Architecture of T.1201

Layers T.122, T.123, T.124 and T.125 specify an application-independent mechanism for providing multipoint data communication to services which support the facilities. Layers T.126 and T.127 define protocols for specific conferencing applications.

A.4.4 Other standards2

There are a number of other standards which are directly or indirectly relevant to real-time services on the Internet. One of the most important technologies for transmitting audio data over low-bandwidth links is Adaptive Differential Pulse Code Modulation (ADPCM). ADPCM utilises the fact that there is usually only a very small amplitude difference between consecutive samples of an audio wave, assuming the wave is sampled at the correct rate. Instead of assigning 8-bit codes to each sample individually, it is possible to assign an 8-bit code to one sample and then encode the amplitude difference between this sample and the next sample(s). This can result in significant reductions in the amount of data requiring to be sent.

1 Image taken from http://gw.databeam.com/ccts/t120primer.html2 Content adapted directly from

http://www.itu.int/itudoc/itu-t/rec/g/g700-799/s_g727_e_38919.html

and

http://www.itu.int/itudoc/itu-t/rec/g/g700-799/s_g726_e_38918.html.




http://gw.databeam.com/ccts/t120primer.html


The G.726 standard published by the ITU sets out the characteristics for the conversion of a 64 kbit/s A-law or µ-law pulse code modulation (PCM) channel to and from a 40, 32, 24 or 16 kbit/s ADPCM channel. The conversion is applied to the PCM bit stream using an ADPCM transcoding technique. Real-time applications which are bandwidth restricted can use 16, 20 and 40 kbit/s channels for carrying voice at different levels of quality.

ITU Recommendation G.726 provides an outline description of the ADPCM transcoding algorithm, and provides the principles and functional descriptions of the ADPCM encoding and decoding algorithms respectively. Networking aspects and digital test sequences are also addressed by Recommendation G.726.

ITU standard G.727 contains the specification of ADPCM algorithms with 5-, 4-, 3- and 2-bits per sample (i.e., at rates of 40, 32, 24 and 16 kbit/s). The characteristics indicated are recommended for the conversion of 64 kbit/s A-law or µ-law PCM channels to/ from variable rate-embedded ADPCM channels.

Recommendation G.727 defines the transcoding law when the source signal is a pulse-code modulation signal at a pulse rate of 64 kbit/s developed from analogue voice frequency signals (i.e. normal PCM telephony). Applications where the encoder is aware and the decoder is not aware of the way in which the ADPCM codeword bits have been altered, or when both the encoder and decoder are aware of the ways the codewords are altered, or where neither the encoder nor the decoder are aware of the ways in which the bits have been altered can benefit from other embedded ADPCM algorithms. The embedded ADPCM algorithms specified in Recommendation G.727 are extensions of the ADPCM algorithms defined in Recommendation G.726 and are recommended for use in packetised speech systems operating according to the Packetised Voice Protocol (PVP) specified in draft Recommendation G.764. PVP is able to relieve congestion by modifying the size of a speech packet when the need arises. Utilising the embedded property of the algorithm described here, the least significant bit(s) of each codeword can be disregarded at packetisation points and/or intermediate nodes to relieve congestion. This provides for significantly better performance than by dropping packets during congestion.

Recommendation G.727 first outlines a description of the ADPCM transcoding algorithm. It then provides the principles and functional descriptions of the ADPCM encoding and decoding algorithms, respectively. It contains also the computational details of the algorithm.

A.5 Proprietary Encoding Schemes

While many applications adopt internationally standardised encoding standards, such as MPEG and H.323, there are many other applications which use encoding schemes which are unique to the producer of the application. Many of these encoding schemes are very widely used. This section discusses some of these schemes and the standards that define them.

A.5.1 RealNetwork’s RealSystem G21

RealSystem G2 is a streaming solution that was developed by RealNetworks. It offers the ability to stream media to users over most types of networks, and over all bandwidths. It is an open, standards based platform which can deal with many media types simultaneously. The codecs used by G2 can offer good quality at all bit rates,

1 See Annex B.1.1 for description about this product



even under lossy network conditions. RealVideo and RealAudio files can automatically scale to bandwidths between 14.4 kb/s and 56 kb/s, and the bit rate is dynamically adjusted to match the available bandwidth using RealSystems SureStream technology.

RealSystem G2 implements RTSP, the client/server protocol which is used to provide services such as pausing, fast-forward and rewind of streaming media. Synchronised Multimedia Integration Language (SMIL), a language developed for laying out and choreographing multimedia presentations is also implemented.

RealSystem supports RealNetworks proprietary file formats as well as many standard and third party file formats including VRML, MIDI, MPEG, AVI, WAV, ASF, VIVO, and AU among others.

A.5.2 Microsoft NetShow and Active Streaming Format1

Microsoft’s NetShow can give very good delivery of multimedia over a range of networks, from low bit-rate modems up to very high-speed LANs. NetShow supports a very wide range of audio and video codecs. NetShow itself refers to the software which usually resides on the media server. The files used by NetShow are Active Streaming Files (ASF) and these files are played back using Microsoft’s Windows Media Player. NetShow supports true media streaming. If the performance of the network degenerates, the NetShow compensates by automatically sending less video data to ensure continuous media playback (albeit at a lower quality level). Video playback may cease completely in circumstances of severe network congestion, leaving only the audio stream playing back.

NetShow supports codecs for G.723, MPEG-1 layer 3, MPEG-4, H.263 among many others.

A.5.3 Apple QuickTime2

QuickTime is a multi-platform industry-standard multimedia software architecture which was developed by Apple Computers. It can be used to develop and publish synchronised graphics, video, text, music, virtual reality and 3-dimensional media. QuickTime supports a range of media delivery mechanisms, from the Internet to DVD-ROM. Some of the key components in QuickTime’s success are its ease-of-use, wide availability and the fact that it can be downloaded and installed free of charge. Also, any FTP or HTTP web-server can support QuickTime’s progressive download mechanism. At present, there is no dedicated QuickTime server software to enable QuickTime to support true media streaming or live webcasting.

QuickTime supports many codecs and file formats for video, audio, text, still images, animation and 3D.

A.6 Coding Techniques Description

A.6.1 Layered Coding

Layered coding of video or audio streams creates a compressed bitstream with a base layer and one or more associated enhancement layers. The base layer bitstream is

1 See Annex B.1.2 for description about this product2 See Annex B.1.5 for description about this product



decoded separately and the enhancement layers can be decoded in conjunction with the base layer to provide an increase in the perceived quality of the video or audio presentation. Layered coding enables video and audio streams to be scaled well.

A client can receive as many layers as it wants, depending on the available bandwidth. A client with a low-capacity connection can opt to receive only the base layer while a client with a high-speed connection can choose to receive some additional layers and receive a higher quality video/audio presentation. The server can therefore meet the requirements of many types of clients without needing to know each client’s specific capabilities.

Many encoding standards are capable of performing layered coding. H.263 for example allows for 15 layers.

A.6.2 Redundant Coding

Modern data compression techniques can exploit the redundancy in a bit stream to reduce the raw data rate by a significant factor. However, when a data stream is compressed to it‘s maximum level it becomes very difficult to detect when errors have occurred. In the English language, errors in normal sentences can usually be detected and compensated for quite easily due to the redundancy of the language. If this redundancy is removed (for instance by using abbreviations or anagrams) the ability to detect errors is significantly reduced. For example take the term Syncfronous Digitel Hierarcy. There are three errors in this phrase, but it can still be interpreted correctly. However, had we used the three-letter abbreviation for the phrase (SDH), a single error could have made us completely misinterpret the message (PDH, SDU etc). Redundancy coding uses the inherent redundancy in digital data streams to help the receiving client recover from transmission errors in the data stream which it receives. Hamming Coding1 and Cyclic Redundancy Checking are both examples of redundant coding schemes. With Hamming Coding, the bitstream is split up into codewords, and extra check-bits are added into the codeword to enable detection and correction of a certain number of errors. The number of errors which can be detected and corrected is dependent upon the number of check bits which are added to the codeword. The use of these check bits requires more overall bandwidth to transmit the message, but means that the receiver can recover from both single and burst errors which occur during transmission. For real-time applications where retransmissions can not be made, redundancy coding is an ideal way to help ensure error resilience.

A.6.3 Transcoding

A transcoder can be thought of as a device which can change the type of encoding used and/or the data rate between the input bitstream and the output bitstream. The input can be encoded with one compression algorithm, and the transcoder can convert it into another compression format, or can output the bitstream with the same format. Similarly, the input can have one data rate and the transcoder can convert it to a higher or a lower data rate. Conversion to lower bit rates is more common, as it is difficult to recover information after it has been discarded.

Transcoders can be used at network boundaries to reduce the delivery rate of video or audio data to prevent overwhelming the network or the receiving client. Transcoders can also be used when there is not sufficient network resource to transmit a video or 1 Hamming Coding and Cyclic Redundancy Coding are quite complex algorithms. Refer to Digital Communications: Fundamentals and Applications, by Bernard Sklar, Prentice Hall International, ISBN 0-13-212713-X.



audio stream to a section of the network. If the data rate of the media cannot be reduced, the only other option is to lose data, since real-time data cannot be delayed significantly without serious degradation to the media presentation. Thus, transcoders can play a crucial role in ensuring real-time data streams can continue even when the network is congested, or when there is insufficient bandwidth available.



Annex B: Review on tools & commercial products

In the last few years, driven by the huge development of Internet’s use, lots of audio/video streaming and telephony/conferencing applications have appeared on the market. This section does a review of several commercial and non-commercial products for both best effort and bandwidth controlled environments.

This review is organised on two sections, focusing streaming and conversational applications. As a result, at the end, this section will present feature tables for each of the application group analysis.

B.1 Applications audio/video streaming

Several commercial and freeware/shareware audio/video streaming products are available nowadays, presenting lots features and a great variety of authoring, diagnostic and content administration tools.

On the negative side, the general lack of use of standards and the number of proprietary solutions represents a big question to be addressed to users and service providers on which is the best solution, since most of them are not interoperable.

This review includes a summarised description of several applications and tools, and presents for each one of them a list of features such as:

Audio, video or A/V capability

Operating systems supported

Audio/video codecs supported

Minimum bandwidth required for audio/video

Delivery mechanisms (from application layer to IP layer)

Multicast support

Firewall support

Monitoring tools

Server features, including scalability, number of clients, input media sources supported etc

Client features (standalone, browser plug in, integration with other Internet services)

Content handling tools

Level of compliance with IETF Standards

Price range

B.1.1 RealAudio/Real Video

http://www.real.com

RealAudio/RealVideo (currently on version G2) is a cross platform audio-video solution built on the industry standard protocol for streaming media, RTSP and on the file format SMIL (Synchronised Multimedia Integration Language). RealVideo introduces some compression technologies for video and audio offering a higher


http://www.real.com/


quality than its predecessors (Real Video 5 and below). The full-motion video codecs are scaleable for all bit rates and optimised for the three most prevalent (US) Internet connection speeds of 28.8 Kbps, 56 Kbps and 112 Kbps. Real Video's codec-independent open architecture supports installable compression algorithms as add-ons called "plug-ins" to its native codecs: RealVideo Standard (developed by RealNetworks) and RealVideo Fractal (using Clear Video technology from Iterated Systems, Inc (http://www.iterated.com / )).

These "plug-ins" may be proposed by companies (e.g. DigitalBitCasting for MPEG1, MPEG2 and MP3 "rendering" plug-in on the player side and "File Format" plug-in on the server side) or be developed with the SDK.

Clickable Video Maps are another feature that allow the user to interact with a video presentation. A mouse click on a region within the video image can cause a new clip to be played, seek within the current clip, or send a URL to the Web browser through Synchronised Multimedia. Regions are made interactive using an extension of the image-map standard with the added dimension of time.

The RealPlayer also supports interactivity through embedded ActiveX and Netscape Plug-In controls that support JavaScript and VBScript.

A proprietary mechanism called "adaptive stream management" allows the RealServer to identify the streaming rate consistent with the available bandwidth between the Server and the Player. In the case of files encoded with the proprietary "sure stream" mechanism, the server switches from one rate to another in a transparent way from the Player.

Concerning the server administration, an HTTP session on a given port is established between the WEB server and the RealServer. Some Monitoring tools are also available for analysing the log reports of the RealServer.

Stream Types Audio & VideoSupported OS Windows (95/98 and NT), MacOS and UNIX (Linux and Solaris

2.5/2.6)Codecs Video: Proprietary RealVideo (standard and fractal) + MPEG-1

Audio: RealAudio proprietary + MPEG-1Possibility to extend the list of supported codecs adding the "rendering" (on the player side) and "file format" (on the server side) plug-ins relative to a given AV encoding. For live encoding, it may also be necessary to upgrade the RealProducer (or to define a new one).

Min. bandwidth Video+Audio: 20 KbpsAudio: 6.5-8.5 Kbps for voice and 8-12 Kbps for music quality

Delivery methods

HTTP, TCP, UDP+RDP (proprietary) and UDP+RTP Multicast mechanism supported (incl. MBONE)1

RTSP for unicast and "back-channel" multicast flow controlPossibility to endlessly stream a "pre-recorded" file on an infinite loop using the g2slta mechanism

Firewall support Yes

1 2 Multicast modes are implemented :

- The "Back-channel" mode where a TCP session is maintained between the player and the server all along the streaming session.

- The "Scaleable" mode which is the standardised Multicast mode implementing the SAP/SDP and RTP/RTCP standards protocols. In this mode, the player requests the SDP file of the multicast session from the RealServer using HTTP.


http://www.iterated.com/


Client features Standalone player + browser plug in + ActiveX controlStream feeds From server stored files or from real time encoding serverSteaming text Possibility to stream text (e.g. slides) using the RealText format.Bandwidth adaptation

"sure stream" mechanism

Server administration

Via a WEB server

Program guide SDPCompliance with standards

IP Level: Compliant with HTTP, UDP, IP Multicast and RTP/RTCP standardsCodec Level: Not compliant, except for MPEG-1Application level : Compliant with SMIL, RTSP

Price Real Player G2: freeStandard Basic Server Plus G2, including 40 simultaneous streams, Real Server G2 Administrator, Real Server G2 Reports, Real Producer Plus G2, CD-Rom & documentations: US$ (500-1000)

B.1.2 Microsoft NetShow Services

http://www.microsoft.com/netshow

NetShow Services enable Internet providers and organisations to deliver audio and video at every bandwidth across the Internet or enterprise networks. One of the streaming media components of the Windows® Media Technologies, NetShow Services consist of server and tools components for delivering audio, video, illustrated audio, animations and other multimedia types over networks. Users play streaming media content with the new Windows Media Player, a universal player that plays most local and streamed media file types, including Advanced Streaming Format (ASF), the native file format delivered by NetShow Services.

NetShow Services incorporate intelligent streaming technology which automatically delivers the best quality audio and video by dynamically optimising stream quality based on network conditions. This technology comprises enhancements in the encoder, the server, and the client. The technology is designed to guarantee optimum audio or video quality (at low bit rates [28.8 and 56 Kbps]) even when network congestion is present.

The latest release of NetShow Services, version 3.0, features enhanced video and audio quality, simplified set-up, configuration, and administration mechanisms of NetShow Services and tools give a good platform for hosting large amounts of content. And tighter integration with other Microsoft products makes it easy for companies to deploy streaming solutions.

Another product is Netshow Theater Server, which is also based on Windows Media technology. This server is providing full-screen MPEG-1 & 2 streaming video content across high-bandwidth networks or dedicated video LANs, up to broadcast quality.


http://www.microsoft.com/netshow


Stream Types Audio & VideoSupported OS Server: Windows NT Clients: Windows (95/98 and NT), MacOS

and UNIXCodecs Audio: Voxware MetaSound & Voxware MetaVoice, FhG MPEG

Layer-3, Lernout & Hauspie CELP 4.8 Kbps, Microsoft G.723.1 and Sipro Lab Telecom ACELP.netMPEG-1 & 2 layers I, II, III with Theater ServerVideo: Microsoft MPEG-4 v2, Duck TrueMotion RT, Intel H.263, Indeo Video Interactive R4.1, Indeo Video R3.2, Iterated Systems ClearVideo and VDOnet VDOwaveMPEG-1 & 2 with Theater Server

Min. bandwidth Video+Audio: 20 KbpsAudio: min 2.4 Kbps, 6.5-8.5 Kbps for voice and 8-12 Kbps for music quality

Delivery methods HTTP, TCP and UDP. Multicast supported (incl. MBONE)Firewall support YesClient features Standalone player + browser plug in + ActiveX controlStream feeds From server stored files or from real time encoding server

Supports both ‚on-demand‘ and ‚ TV broadcast like‘ contentsBandwidth adaptation

“intelligent streaming technology”

Compliance with standards

IP Level: Compliant with IETF’s HTTP, UDP, TCP, IP MulticastCodec Level: Compliant with G.723.1, MPEG 1 & 2, H.263, other are nativeInteroperability Level: Compliant with ITU-T‘s H.323

Price NetSHow Services Components: freeNetshow Theater Server, including 5 CALs client pack: US$ (2000-3000)

B.1.3 Xing Tech. Streamworks

http://www.xingtech.com

Xing’s streaming media technology includes StreamWorks Player that allows users to receive and playback audio and video data simultaneously as it is being broadcast over existing network infrastructures, with no wait for long file downloads or transfers. The StreamWorks Player uses MPEG-1 as its core compression technology. The Player also displays full-screen, full-motion video over enterprise networks and remote serving applications, such as corporate training or distance education. The StreamWorks Player is available for download from Xing's website at no cost, and can be used as both a stand-alone application or as a plug-in to popular Web browsers like Netscape Navigator, Netscape Communicator or Microsoft’s Internet Explorer.


http://www.xing.com/


Stream Types Audio & VideoSupported OS Player: Windows (3.1, 95/98 and NT), MacOS, SGI Irix, Solaris and

LinuxServer : Windows NT, Linux, Solaris, SGI/Irix, BSDI, HP-UX

Codecs Audio: MPEG-1 layer 1,2, and 3 audio, MPEG-2 layer 2 and 3 and LBR audio streamsVideo: MPEG-1

Min. bandwidth MPEG-1 bandwidth requirements (i.e. 32 Kbps for Audio, about 300 Kbps for video)

Delivery methods IP unicast and multicast, UDPFirewall support YesClient features Standalone player + browser plug in + ActiveX controlStream feeds From server stored files

Supports both 'on-demand' and 'multicast live' contentsCompliance with standards

IP Level: Compliant with IETF’s UDPCodec Level: MPEG Compliant

Price Xing Streamworks Player: freeXingTech Streamworks studio kit, including streamworks Server for 50 clients, streamworks encoder, streamworks Player: US$ (2500-3500)MPEGLive! Encoder : US$ (7000-8000)

B.1.4 VDO Live

http://www.vdo.net/vdostore

VDOLive is a video streaming technology that takes web sites beyond static text and images and into Internet Video Broadcasting.

To accomplish this, the VDOLive product line uses two core technologies. The first is a scaleable compression algorithm that not only shrinks video down small enough to deliver at low bandwidth, but also dynamically scales the video transmission up or down depending on the user's current available bandwidth.

The second core technology is a communications protocol which maintains the integrity of the video as it makes its way through the vagaries of the Internet. The benefit of these combined technologies is that motion video can be transmitted to virtually anyone, regardless of their connection.

VDOLive servers are capable of both on-demand video delivery and live broadcasts.

The VDOLive On-Demand Server transmits video clips over both the Internet and Intranets, using dynamic bandwidth scaling to maximise video quality throughout each connection.

The VDOLive Broadcast Station features real-time capture, compression and transmission to a VDOLive Broadcast Server for streaming of live events.

The VDOLive Player provides real-time playback over the Internet and Intranets at modem connection rates and higher. Web designers can create HTML pages to play VDOLive content using the Player as a helper application, as a Netscape Navigator Plug-In, or as an ActiveX control for Internet Explorer.

The VDOLive Tools are made up of 3 programs. VDO Capture is used, along with a video capture card, to convert analogue audio and video source material (such as


http://www.vdo.net/vdostore


video and audio tapes) into digital format. VDO Clip is used to compress video and audio into VDO format. VDO Producer uses a Windows wizard interface to guide you through all aspects of content creation.

Stream Types Audio & VideoSupported OS VDOLive Player: Windows 95/98 Windows NT and MacOS 7.1.2 or

laterVDOLive Server: Windows NT 4.0 and Solaris2.4 or higherVDOLive Tools: Windows 95/98; Windows NT; Mac OS

Codecs Audio: G.723, G.728 and G.711Video: H.263 and H.261

Min. Bandwidth Max Bandwidth

14.4 kbps512 kbps

Delivery methods UDP, HTTPFirewall support YesClient features Standalone player + browser plug in + ActiveX controlStream feeds From server stored files

Supports broadcast live streams and direct-to-disk streamingCompliance with standards

IP Level: Compliant with IETF’s UDP and HTTPCodec Level: Compliant with G.723, G.728, G.711, H.263 and H.261

Price VDO Player: freeVDO Server: US$ (2000-3000)VDO Tools: included with VDO Server

B.1.5 Apple Quicktime

http://www.apple.com

QuickTime is Apple Computer’s software architecture that makes it possible to create, integrate, and publish several types of digital media. QuickTime is composed of three distinct elements—the QuickTime Movie file format, the QuickTime Media Abstraction Layer, and the QuickTime media services.

The QuickTime Movie file format specifies a standard means of storing digital media compositions. QuickTime also specifies a comprehensive software architecture known as the QuickTime Media Abstraction Layer. This abstraction layer specifies how software tools and applications access the rich set of media support services built into QuickTime. It also specifies how hardware can accelerate performance critical portions of the QuickTime system. Finally, the QuickTime Media Abstraction Layer outlines the means by which component software developers can extend and enhance the media services accessible using QuickTime, such as the Internet plug-in.

QuickTime is not a product specially developed for Internet use, but its browser plug-in can be use to retrieve and view several types of media stream formats from the Internet.


http://www.apple.com/


Stream Types Audio & Video. Supports other non-streaming media typesSupported OS Internet plug-in: Windows (95/98 and NT) and MacOS (7 and 8)Codecs Audio: A-Law 2:1, -Law 2:1, IMA 4:1, MACE 3:1, MACE 6:1, MS

ADPCM, Qdesign Music, and Qualcomm PureVoiceVideo: Apple Video, Cinepak, H.263, Microsoft Video 1, Motion JPEG 1&2, Sorenson Video

Min. bandwidth Not describedDelivery methods HTTP via browserFirewall support YesClient features Browser plug in (Internet Explorer 3.0/4.0 and Netscape Navigator

3.0/4.0)Stream feeds From server stored filesCompliance with standards

IP Level: Compliant with IETF’s HTTPCodec Level: Compliant with G.711, H.263

Price Player: free

B.1.6 Cisco IP/TV

http://www.cisco.com/warp/public/732/net_enabled/iptv/

IP/TV is a comprehensive client/server software application aimed at delivering business communications to the enterprise market. It combines TV-quality streaming video with application and management features and has some scalability and bandwidth efficiencies usually needed in enterprise deployment.

IP/TV captures, stores, and transmits digital video and audio to desktops over IP networks, supporting three modes of video distribution: live, scheduled and on demand.

IP/TV has three components--IP/TV Content Manager, IP/TV Server, and IP/TV Viewer—that work together to provide easy scheduling, transmission and desktop viewing of program content. The heart of the application is IP/TV's Content Manager, which allows users to manage content, schedule broadcasts and optimise network performance across distributed servers.

StreamWatch is a tool associated with IPTV which allows monitoring of the number of clients having subscribed to each session provided by servers. In particular, it enables checking to see if RTCP messages are properly transmitted between different members of a multicast session, or between a server and viewer.

Moreover, IPTV offers SmallCast mode designated to forward a multimedia session across non-multicast-enabled routers, as a unicast session (kind of tunnel). In this way, even viewers connected on a remote segment are able to receive multicast sessions.

SlideCast is a special mode which allows to mix slides, audio and video ; it’s particularly interesting for distance learning applications.

Cisco is also providing IP/TV 3400 Series Servers : those servers are currently pre-configured with IP/TV 2.0 client-server software and pre-installed video store, video cards, network interface cards and device drivers. The customer has only to rely on server performances guaranteed by Cisco System performances (for example, specification of number of cumulative requests supported by servers for each type of service - VoD or scheduled) to carry out its own client/server architecture depending


http://www.cisco.com/warp/public/732/net_enabled/iptv/


on its specific needs. In that way, IP/TV 3400 Series Servers perfectly fit the needs of enterprise customers with a tuned solution which is easy to install and deploy.

Stream Types Audio-Video. Supports other non-streaming media typesSupported OS Windows NTCodecs Audio: -Law 2:1, GSM, DVI, MPEG-1, MSN Audio, DSP Group

Truespeech and Lernout & Hauspie.Video: H.261, Indeo 4.1, MPEG-1, Vxtreme and Apple QuickTime

Min. bandwidth 56 KbpsDelivery methods RTP, RTCP and RTSP. IP multicast supported (incl. MBONE)Firewall support Yes (Security functionalities related to content access)Client features Standalone player and browser plug inStream feeds Live, Scheduled and On DemandProgram guide SDPCompliance with standards

IP Level: Compliant with IETF’s UDP, RTP/RTCP, RTSP, IP Multicast, SDP formatCodec Level: Compliant with G.711, GSM, MPEG-1, H.261

Price Cisco IP/TV 2.0 Starter Pak, including Server, Content Manager, StreamWatch, 20 Viewers: US$ (5000-10000)IPTV-3410-CTRL Cisco IP/TV 3400 Series Control Server: US$ 10000IPTV-3420-BCAST Cisco IP/TV 3400 Series Broadcast Server: US$ 22000IPTV-3430-ARCH Cisco IP/TV 3400 Series Archive Server: US$ 24000

B.1.7 DT/MTG Music on Demand application

http://www.audio-on-demand.de/

Music on Demand provides streaming audio with the best hi-fi quality from a server to the client PC. The clients can listen to tracks or transfer the compressed music data over direct ISDN, TCP, or other communication networks to their own PC.

The basic client device is the MoD Software, which is provided free, for the communication between the client multimedia PC and the audio server system from Deutsche Telekom. This MoD software, which controls the network access device, can be downloaded from the MOD server.

Client multimedia requirements for MoD are Windows 95 or Windows 98 operating systems and a 16-bit sound card for Windows. The speaker system at client site should be of sufficiently high performance according to the quality of the music data provided by Music on Demand.

The system offers a huge database where the client can select from their PC any kind of music title listed by title name, CD-name, artist and composer. This is done over the Internet from a web address (http://www.audio-on-demand.de / ) through a friendly search utility. It is also possible to search for music according to different musical styles and music recording companies. A hit list with the music that has been downloaded most and a search function for new music events are other utilities of this application.

The application allows storing all the music obtained during different searches for a full download. The clients can download a free 20 seconds pre-listening to evaluate the music before downloading and paying for the full song. The music can be listened





to online, if requested, while it is being downloaded if enough bandwidth is available, or saved onto the PC.

A thumbnail bitmap of the CD cover is transmitted, if available, together with the audio stream.

The data transmission in the commercial service is done through ISDN, according to the contracts with the music industry. Here, normally both B-channels are used during the download of a track, to guarantee real-time. In non commercial use of the Music-on-Demand platform, the data transmission is currently also done in other network scenarios e.g., TCP/IP. The data transmission includes dynamic bandwidth allocation, according to the delivered content.

The music tracks are stored in MPEG 1-Layer 3 (MP3) format that provides a 1:12 reduction. The MP3 Player is provided together with the free MoD software. This guarantees a high quality of the music tracks and a download in real time.

Stream Types AudioSupported OS Server: Solaris

Client: Windows 95 or Windows 98Codecs Audio: MPEG-1 Layer 3 (MP3)Min. bandwidth 64 KbpsDelivery methods Proprietary, ongoing patent procedureFirewall support Yes (interaction with a CA process)Client features Standalone player plus databaseStream feeds server stored filesCompliance with standards

Codec Level: Compliant with MPEG-1 Layer 3 (MP3)

Price Client: free. Different prices depend on tracks. Fee comes with the telephone bill

B.1.8 Oracle iTV

http://www.oracle.com/itv/

The Oracle Interactive TV system is an integrated technology for streaming digital video from a server to multiple clients. This system’s benefits include real-time delivery, minimal client storage requirements, elimination of bulky media, VCR-style control over content playback, and support for value-added services such as near video-on-demand. Another way to make streamed digital video available to clients is to schedule broadcasts at regular intervals. An administrator can schedule delivery of a specific video at a specific time on a specific channel. Near Video-on-Demand.

(NVOD) is an application of scheduling in which the administrator schedules delivery of a video to begin automatically at regularly scheduled intervals on a different channel.

Scalability on the Oracle Video Server (OVS) regarding the number of simultaneous users is possible because the data is stored as a series of stripes, typically of 32 KB or 64 KB each, with the stripes distributed over multiple disks on the OVS.

The iTV system is fully compliant with Oracle’s Network Computing Architecture (NCA), which consists of three tiers:

Client

Application Server


http://www.oracle.com/itv/


Data Server

The NCA also defines three layers:

Server Layer: provides the basic functionality for each tier

Cartridge Layer: provides programmatic functionality for the Server Layer of each tier

Communication Layer: provides communication between servers, cartridges, and tiers

There are two client components provided as part of the iTV system: The Oracle Video Client and the Oracle Video Server Manager.

The Oracle Video Client (OVC) is responsible for obtaining, decoding and displaying the video stored on the server. The OVC provides multiple interfaces that allow you to incorporate streaming audio and video from OVS into your own applications.

The Oracle Video Server Manager (VSM) is a Java application that provides a management environment for OVS services, clients, and content.

The Application Server itself consists of the Oracle Video Server and the VSM application logic. The OVS functions as the application server in this environment, receiving and processing requests for digital video, then delivering the digital video content to the client device. The VSM application uses both HTTP and CORBA (Common Object Request Broker Architecture) as the network layer.

In terms of scalability iTV is scalable across multiple dimensions:

Concurrent Streams - If demand for a server (such as an OVS service) increases, you can run additional instances of the server to better handle the load. Oracle Media Net can distribute server access request across the available instances to balance the load. If one instance stops, Oracle’s Media Net redistributes new requests across the remaining instances.

Bit rates - In networked media systems, quality improves linearly as bandwidth increases. OVS enables scalability across the widest range of bit-rates. A single architecture spans rates from 28.8Kb/sec modems to ATM, 100 Mbps LANs, and digital satellite delivery systems.

Hardware platforms - OVS is truly multi-platform; from a single-processor server to the latest Symmetric Multi-Processor (SMP) clusters to Massively Parallel (MPP) media servers such as nCube. OVS is optimised for each architecture by Oracle product lines and by leveraging our strategic partnerships with the hardware vendors.



Stream Types Audio/Video.Supported OS Server: nCube Transit, Windows NT, Solaris, Digital UNIX and

SGI/IRIXClient: Windows 95/98/NT and several UNIXes. A JAVA client is also available

Codecs Video: MPEG-1 or MPEG-2, H.263, AVI (using Iterated Systems ClearVideo, Radius CinePak and Intel Indeo)Audio: MPEG-1 or MPEG-2, AVI (using Voxware MetaSound and MetaVoice)

Min. bandwidth 28.8 KbpsDelivery methods TCP, UDP. Multicast supported (incl. MBONE)Firewall support YesClient features Standalone player and browser plug in (JAVA and ActiveX)Stream feeds Live, Scheduled, On DemandCompliance with standards

IP Level: Compliant with IETF’s UDP, TCP, RTSP, IP MulticastCodec Level: Compliant with G.711, MPEG-1, MPEG-2/DVB, H.263

Price NA



B.1.9 Feature “at-a-glance“ comparison form

Product Stream Types

Supported OS Audio Codecs Video Codecs Min. Bandwidth Delivery Methods Bandwidth Adaptation Bandwidth Control

Firewall Support

Standard Compliance Price Range1

Real Audio/Real Video audio+video Windows 95/98, Windows NT, MacOS, Linux, Solaris 2.5/2.6

Real Audio Proprietary Real Video Proprietary 20Kbps (A/V) 6.5-12Kbps (Audio)

HTTP, TCP, UDP+RDP (proprietary) and UDP+RTP.

Multicast supported

RTP, RSTP for unicast and 'backchannel' for multicast

NA Yes HTTP, UDP, IP Multicast, RTP/RTCP

Player: 1 Server: 3

Microsoft NetShow Services

audio+video Server: Windows NT, Clients: Windows 95/98/NT, MacOS, UNIX

Voxware MetaSound & Voxware MetaVoice, FhG MPEG Layer-3, Lernout & Hauspie CELP 4.8

Kbps, Microsoft G.723.1 and Sipro Lab Telecom ACELP.net

MPEG-1 & 2 layers I, II, III with Theater Server

Microsoft MPEG-4 v2, Duck TrueMotion RT, Intel H.263, Indeo Video Interactive R4.1, Indeo Video R3.2, Iterated Systems

ClearVideo and VDOnet VDOwaveMPEG-1 & 2 with Theater Server

20Kbps (A/V) 2.4-12Kbps (Audio)

HTTP, UDP, IP Multicast RTP/RTCP NA Yes IP Level: HTTP, UDP, IP Multicast and RTP/RTCP

Codec Level:G.723.1, MPEG 1 & 2

Standard Services: 1 Theater Server 4

Xing Technology Streamworks

audio+video Windows (3.1, 95/98 and NT), MacOS, SGI Irix, Solaris and Linux

MPEG-1 layer 1,2, and 3 audio, MPEG-2 layer 2 and 3 and LBR

MPEG-1 32Kbps (audio) 300Kbps (video)

UDP unicast and multicast NA NA Yes IP Level:UDP Codec Level: MPEG-1

1

VDO Live audio+video VDOLive Player: Windows 95/98 Windows NT and MacOS 7.1.2 or later

VDOLive Server: Windows NT 4.0 and Solaris2.4 or higher VDOLive Tools: Windows 95/98; Windows NT; Mac OS

G.723, G.728 and G.711 H.263 and H.261 14.4Kbps HTTP, UDP NA NA Yes IP Level:UDP and HTTPCodec Level: G.723, G.728,

G.711, H.263 and H.261

Player 1

Apple QuickTime audio+video Windows (95/98 and NT) and MacOS (7 and 8)

A-Law 2:1, -Law 2:1, IMA 4:1, MACE 3:1, MACE 6:1, MS ADPCM, Qdesign Music, and

Qualcomm PureVoice

Apple Video, Cinepak, H.263, Microsoft Video 1, Motion JPEG 1&2, Sorenson

Video

NA HTTP (via browser) NA NA Yes IP Level: HTTPCodec Level: G.711, H.263

1

Cisco IP/TV audio+video Windows NT -Law 2:1, GSM, DVI, MPEG-1, MSN Audio, DSP Group Truespeech and Lernout & Hauspie

H.261, Indeo 4.1, MPEG-1, Vxtreme and Apple QuickTime

56Kbps UDP, IP multicast RTP, RTCP and RTSP NA Yes IP Level: UDP, RTP/RTCP, RTSP, IP Multicast Codec

Level: G.711, GSM, MPEG-1, H.261

6

DT/MTG Music on Demand

audio Server: SolarisClient: Windows 95 or Windows 98

MPEG-1 Layer 3 (MP3) NA 64Kbps Proprietary NA Proprietary Yes Codec Level: MPEG-1 Layer 3 (MP3)

1

Oracle iTV audio+video Server: nCube Transit, Windows NT, Solaris, Digital UNIX and SGI/IRIX

Client: Windows 95 and several UNIXes. A JAVA client is also available

MPEG-1 or MPEG-2, AVI (using Voxware MetaSound and MetaVoice)

MPEG-1 or MPEG-2, H.263, AVI (using Iterated Systems ClearVideo,Radius

CinePak and Intel Indeo)

28.8Kbps TCP, UDP, , IP multicast RTSP NA Yes IP Level:UDP, RTSP, IP Multicast

Codec Level: G.711, MPEG-1, H.263

6

1 Price Ranges (in US$): (1) Free, (2) <=500, (3) 501-1000, (4) 1001-3000, (5) 3001-5000, (6) 5001-10000, (7) 10001-15000



B.2 Applications for telephony/conferencing

As with streaming applications, several commercial and freeware/shareware conferencing products are available nowadays, presenting different approaches to addressing the telephony and conferencing concepts. Some of these applications also include CSCW capabilities by providing some data sharing functionalities.

Although developers and software houses seem to agree that H.323 compatibility is a big step towards addressing this kind of application, providing the mechanisms to allow transparent interoperability between different vendors, and also the interoperability with H.323 gateways and gatekeepers, some of the reviewed applications are not H.323 compatible. This results in poor or non-existent interoperability when using them.

The use of some QoS control or bandwidth allocation mechanism is also another important aspect that is differently addressed by the applications under review.

Like on the previous analysis, this review provides a summarised description of several applications and tools, and presents for each one of them a list of features such as:

Brief product overview & description for each application

Audio, Video or both

Operating systems supported

Audio/video Codecs supported

Delivery mechanisms (from application layer to IP layer)

Point-to-Point or Point-to-Multipoint

Multicast support

H.323 compliance

Compliance with standards (IETF, ITU-T or DAVIC)

Firewall support

Monitoring tools

Client features (standalone, browser plug in, integration with other Internet services)

Price range

B.2.1 Microsoft NetMeeting

http://www.microsoft.com/netmeeting

Microsoft NetMeeting is an integrated tool that enables real-time communications and collaboration over the Internet or corporate intranet, providing audio, video, and multipoint data conferencing support. From a Windows 95 or Windows NT 4.0 desktop, users can communicate over the network with real-time voice and video technology. It allows them to share data and information with many people through true application sharing, electronic whiteboard, text-based chat, and file transfer features.


http://www.microsoft.com/netmeeting


In its initial version, 1.0, this product was one of the first to introduce multipoint data conferencing capabilities based on the ITU T.120 standard. In its current version, 2.1, NetMeeting supports H.323 communications, making it interoperable with several other H.323 enabled conferencing applications, by using the same protocols both on the transport issues (e.g. RTP, RTCP) and also on the codecs supported.

Application Type Multipoint conferencing with data sharing (A/V is only supported in a point-to-point fashion)

Supported OS Windows 95/98 and Windows NT 4.0Codecs & Standards Communications: H.323. Supports LDAP based directories.

Audio: G.711, G.723, Lernout & Hauspie SBCVideo: H.261, H.263Data Sharing: T.120

Delivery methods IP unicast; RTP/RTCP, RSVP (on version 2.11 Beta)Firewall support Yes, if several UDP and TCP ports are kept open on the firewallClient features Single applications integrating audio, video, chat, whiteboard,

directory listing and data collaboration tools,H.323 compliance Compliant. Don’t support the RAS MechanismPrice Free

B.2.2 CU-SeeMe

http://cu-seeme.cornell.edu/

CU-SeeMe desktop videoconferencing software was developed by Cornell University. The application uses peer-to-peer and client/server technology that allows interactive multiparty video and voice communication over the Internet or any local TCP/IP-based network. Beginning as a collaboration with the Global Schoolhouse in 1993, the goal of this project was to provide very low cost face-to-face audio and video connectivity to the largest possible number of users and enable a social experiment of global proportions. Running on popular desktop platforms with full cross-platform connectivity and no proprietary hardware, CU- desktop videoconferencing technology established a new price-point in cost and a new paradigm of availability, quickly becoming one of the most popular videoconferencing platforms.

CU-SeeMe requires at least a 14.4 baud data rate, preferably 28.8 - or higher. Initially it only supported B&W video, but now a colour version for the PC platform is available.

Application Type Multipoint conferencingSupported OS Windows 3.xx/9x, Windows NT 4.0, MacOS, Linux, Amiga and

OS/2Codecs & Standards Communications: CU-SeeMe proprietary

Audio: G.723.1, Digitalk, DeltaMod, Intel DVIVideo: H.263 and M-JPEG 1.1

Delivery methods IP unicast, UDPFirewall support Yes, if several UDPClient features Single application integrating audio, video and chat.H.323 Compliance Not compliantPrice Free

B.2.3 WhitePine’s MeetingPoint tools

http://www.wpine.com/products/MeetingPoint

MeetingPoint consists of a set of tools to manage the participation in virtual conference rooms: Several conferencing systems can be used as clients; White Pine's


http://www.wpine.com/products/MeetingPoint

http://cu-seeme.cornell.edu/


CU-SeeMe®, Intel® ProShare® or TeamStation™, Microsoft® NetMeeting™ or PictureTel's LiveLAN.

It has some interesting features as:

Possibility to choose from MeetingPoint's conference list to find a particular meeting.

The MeetingPlanner™ to schedule meetings and notify participants of time and location via automatic e-mail invitation.

Control who you want to see and hear with MeetingPoint's VideoSwitcher.

A built-in gatekeeper allows you to set bandwidth limits per conference and per user. The gatekeeper also controls who can access the conference by providing authentication services. MeetingPoint can send audio and video data in multicast format so that one stream of data is sent over the network and shared by multiple users. MeetingPoint Servers can be linked together so that conferences can be shared and the conference manager can load-balance the traffic on the network. This system also supports a set of Web-based administration tools and contains an embedded AAA system.

Application Type Multipoint AV conferencing with data sharingSupported OS Windows NT 4.0 and Sun Solaris 2.5.1/2.6Codecs & Standards Communications: H.323, CU-SeeMe

Audio: G.711, G.723 Video: H.261, H.263 Data: T.120

Delivery methods IP unicast and multicastFirewall support YesClient features Uses CU-See Me or H.323 clientH.323 Compliance CompliantPrice 10 users: US$ (5000-10000)

25 users: US$ (10000-15000)

B.2.4 VocalTec Internet Phone Lite

http://www.vocaltec.com/products/iphone5

Internet Phone Lite allows users to conduct telephone conversations using a computer instead of a regular telephone. The dialler users this application to connect to a gateway (such as VocalTec Telephony Gateway) via the Internet or another type of public/private IP network. The gateway, by using H.323, then connects to a phone number using the PSTN network, enabling voice conversation.

This way, Internet Phone Lite is a light, flexible tool for making calls from a PC to a telephone. This tools takes only a small amount of disk space and has a very intuitive usage, just like a regular telephone.

Application Type Audio telephony applicationSupported OS Windows 95/98 & Windows NT 4.0Codecs & Standards Communications: H.323

Audio: G.711, G.723.1, G.729Delivery methods IP unicast (TCP, UDP, RTP/RTCP)Firewall support Yes, if several UDP and TCP ports are kept open on the firewallClient features Phone-like simple H.323 clientH.323 Compliance Fully H.323v2 compliantPrice Free


http://www.vocaltec.com/products/iphone5


B.2.5 VDO Phone

http://www.vdo.net/

VDOPhone Internet is a videotelephony application that implements point-to-point conversation with other VDOPhone Internet users with full-colour video and audio over connection speeds of 14.4 Kbps to 128 Kbps. VDOPhone (Internet version) is optimised for the Internet, and has the ability to automatically adjust video quality as the available bandwidth changes. In addition, finding each other on the Internet is easy with the use of VDOnet's Online Directory.

In terms of performance issues, VDOPhone presents the following features:

Dynamic Bandwidth Scaling: Intelligently adjusts the compression rate in order to provide optimal video quality during Internet bandwidth fluctuations.

Echo Cancellation: blocks out the annoying echo to permit hands-free toll quality speakerphone conversations.

MMX optimisation: VDOPhone takes advantage of the multimedia extensions available with the latest processors from both Intel and AMD.

In addiction, the product also has some personalisation features, (create a greeting message, thumbnail picture, and an electronic "Virtual business card"), a Personal Photo Album allowing to create a photo album by collecting a directory of snapshots of the dialled people.

Another interesting feature of this application is the data integration concept: It is possible to place calls using the email addresses and telephone numbers already entered in the Microsoft Exchange address book or any other MAPI compliant database.

Application Type Videotelephony applicationSupported OS Windows 95/98Codecs & Standards Audio: G.723, G.728 and G.711

Video: H.263 and H.261Delivery methods IP unicast. Uses a proprietary application level protocol.Firewall support No information is availableClient features Videotelephone-like Windows applicationMin/Max Bandwidth Min: 14.4 Kbps Max: 128 KbpsH.323 Compliance Not compliant.other technical specifications

- IP connections: Full duplex real time transmission and reception of audio, motion colour video, and data. - Video window size: SQCIF (128 x 96) to CIF (352 x 288). Enlargement up to full screen with Direct Draw enabled graphics cards. - Video Frame Rate on Premium IP Networks: 8-15 fps @ >128 Kbps IP access connection at CIF resolution (15-24 fps at QCIF resolution). - Video Frame Rate on standard IP networks: 4-10 fps @ 28.8 Kbps, 8-20 fps @ 128 Kbps IP access connection at QCIF resolution.

Price VDOPhone Internet US$49.00


http://www.vdo.net/vdostore/


B.2.6 Isabel

http://isabel.dit.upm.es/

ISABEL is a customisable multiconferencing application which provides advanced video/audio/data conferencing features as well as CSCW (Computer Supported Collaborative Work) support. It is designed to produce remote collaboration services in the areas of Teleeducation, Telework, Telemeeting and Teleconference. ISABEL has been extensively used in real service trials, especially in the European Broadband Programs, such as the RACE/ACTS Summer Schools or Global Events which are probably the largest interactive collaborations made worldwide, with coverage of North America, Europe and Asia. ISABEL has been developed with three main goals:

To connect a large number of interactive endpoints. Conferences with up to 20 interactive sites have been done.

To connect sites through heterogeneous networks in an event, such us ATM, ISDN, Internet or MBONE.

To achieve integrated event management to enable an effective control of a distributed collaboration.

ISABEL has been designed as an integration of a set of components each of them devoted to solving the interaction with one media stream inside of a group. There are components to support digital video (based on M-JPEG compression), digital audio (based on 16 bits linear with silence and echo detection), some group and graphical components and a sophisticated kernel to manage from the central control site the whole or part of the conference flow control.

The multipoint issues are handled by a piece of software called irouter (isabel router). The ISABEL irouter is not a router in the proper sense of the word but a transport agent for the ISABEL application providing an additional layer in the software structure of the application. None of the ISABEL components send it's data directly to the network, instead they all rely on having the irouter do it for them.

The irouter is implemented as a separate UNIX process in each host taking part in the conference and in some network nodes devoted to support the distribution of the multiconference.

Application Type Multipoint Audio/video/data conferencingSupported OS SunOS and LinuxCodecs & Standards Video: MJPEG

Audio: G721, GSM, IMADelivery methods IP (UDP for audio and video, and TCP for general data transfer)

Multicast (MBONE) compatibleFirewall support No informationMin. Bandwidth 3MbpsClient featuresH.323 Compliance Not compliant.Price free


http://isabel.dit.upm.es/


B.2.7 NetSpeak WebPhone

http://www.netspeak.com/product/webphone/

NetSpeeak WebPhone is an H.323v2 compatible point-to-point videotelephony application.

Its main features are:

Point-to-point voice and video over the Internet or any TCP/IP-based network

Real-time, full-duplex voice communications

H.323 Version 2.0 support; communicate with any H.323-compliant Internet telephone

Audio Set-up Wizard for configuring speakers and microphone

Voice Auto Detection which automatically detects and adjusts to your voice, improving audio quality

Online/offline integrated voice mail system

Video phone support using the H.263 standard

Interactive party-specific TextChat

Personal directory to store frequently called parties

Audio CODECS: TrueSpeechTM, G.723.1, G.711 and GSM voice compression

Application Type Point-to-point videotelephony applicationSupported OS Windows 95/98 and Windows NTCodecs & Standards Communications: H.323v2

Video: H.263Audio: G.723.1, G.711, GSM and TrueSpeech

Delivery methods IP,UDP, RTP/RTCPFirewall support YesMin. Bandwidth 28.8 KbpsClient features Videotelephony style application, including directory and text chatH.323 Compliance Compliant H.323v2Price Standard version: US$49.95 Lite version: US$19.95

B.2.8 Netscape CoolTalk

http://www.netscape.com/navigator/v3.0/cooltalk.html

CoolTalk is an Internet telephone tool included with Netscape Navigator that provides audio conferencing over IP, a whiteboard, and text-based communications using the chat tool. With CoolTalk it is possible to talk and work collaboratively. And because CoolTalk works seamlessly with Netscape Navigator, it is possible to send and receive calls directly from any Web page.

Application Type Audio conference applicationSupported OS Windows 95/98, Windows NT, Windows 3.1, MacOS, SunOS,

Solaris, HP-UX, Digital Unix, and IRIX.Codecs & Standards Audio: ProprietaryDelivery methods IP unicastFirewall support YesClient features Standalone simple audio-conference applicationH.323 Compliance Not compliant


http://www.netscape.com/navigator/v3.0/cooltalk.html

http://www.netspeak.com/product/webphone/


Price Free

B.2.9 VIC/VAT - Video Conferencing Tool

http://www-nrg.ee.lbl.gov/vic/

The UCB/LBNL video tool, Vic, is a real-time, multimedia application for video conferencing over the Internet. Vic was designed with a flexible and extensible architecture to support heterogeneous environments and configurations. For example, in high bandwidth settings, multi-megabit full-motion JPEG streams can be sourced using hardware assisted compression, while in low bandwidth environments like the Internet, aggressive low bit-rate coding can be carried out in software.

Vic is based on the IETF’s Real-time Transport Protocol (RTP) standard. RTP is an application-level protocol implemented entirely within vic -- you need no special system enhancements to run RTP. Although Vic can be run point-to-point using standard unicast IP addresses, it is primarily intended as a multiparty conferencing application. To make use of the conferencing capabilities, the user system must support IP Multicast, and ideally, the network should be connected to the IP Multicast Backbone (MBone). Vic also runs over RTIP, the experimental real-time networking protocols from U.C. Berkeley's Tenet group and over ATM using Fore's SPANS API.

Vic provides only the video portion of a multimedia conference; audio, whiteboard, and session control tools are implemented as separate applications. The audio tool is called Rat and our whiteboard tool wb. UCL developed the session directory tool sdr. Other related applications include ISI's Multimedia Conference Control, mmcc, the Xerox PARC Network Video tool, nv and the INRIA Video-conferencing System, ivs. Vic is backward compatible with RTPv1 and can interoperate with both nv (v3.3) and ivs (v3.3)

The main vic features consist in:

an ``Intra-H.261'' video encoder,

voice switched viewing windows,

multiple dithering algorithms,

interactive ``title generation'', and

routing of decoded video to external video ports.

The Intra-H.261 encoder combines the advantages of nv's block-based conditional replenishment scheme (i.e., robustness to loss) with those of H.261 (i.e., higher compression gain and compatibility with hardware codecs). This is achieved by coding only ``intra-mode'' macroblocks and using macroblock skip codes to replenish only the blocks that change. For a fixed bit rate, the H.261 coder achieves frame rates typically 2-4 times that of the nv coding format.


http://www-nrg.ee.lbl.gov/vic/


Application Type Multipoint Audio/video/data conferencingSupported OS SunOS, Linux, NetBSD, FreeBSD, Ultrix, UP-UX, OSF,

SGI/IRIX and AIXCodecs & Standards Video: H.261, JPEG

Audio: rat codecsMin. Bandwidth 14.4 KbpsDelivery methods UDP. RTP/RTCP and proprietary RTIP

Multicast (MBONE) compatibleFirewall support No informationH.323 Compliance Not compliant.Price Free

B.2.10 Intel Internet Video Phone

http://www.intel.com/createshare/Videophone

The Intel Internet Video Phone trial application allows to make either audio-only or audio and video phone calls over the Internet (with a PC camera). Because these are calls over the Internet, their quality will vary depending on the speed of the modem and the Internet traffic.

The full-featured version of Intel Video Phone software is provided with the Intel Create & Share Camera Pack. It includes a PC camera and the Intel Video Phone application, plus loads of other software : fun e-mail Postcards (including video or still images, as well as text and sound ), Intel Movie Builder for home page or sending, , Intel Home Page Builder, virtual Games.

The purpose of this tool is mainly family calls over standard telephone lines or through the Internet connection, with data sharing, and family entertainment.

Application Type Audio/Video telephony applicationSupported OS Windows95/98Codecs & Standards Communications: H.323, H.324

Audio: G.723, Intel Indeo AudioVideo: i263, Intel Indeo Video 5.0

Min. Bandwidth 28.8 KbpsDelivery methods No informationFirewall support No informationClient features Video phone calls style, including postcards, still pictures,etc…H.323 Compliance “Compatible”Price Intel Create&Share Camera Pack : US$ (100-200)

B.2.11 Hardware based Internet conferencing solutions

This section also makes a brief review of the 3 most popular hardware based internet conferencing solutions: VCON Escort/Cruiser series, PictureTel LiveLan Conferencing, and Intel ProShare.


http://www.intel.com/createshare/Videophone


B.2.11.1 VCON Escort25Pro Desktop System

VCon Escort25Pro is a desktop system that is based on a PCI hardware codec board, a camera and handset, and conferencing software and tools.

This system supports H323 LAN-based videoconferencing from 64kbit/s up to 768 kbit/s.

Escort’s main features are:

Desktop videoconferencing software for LANs via Windows 95 and Windows NT

Microsoft NetMeeting application included, providing full application sharing, file transfer and electronic whiteboard collaboration via T.120 protocol

H.323/H.320 Gateway support

Frame grabber

Image phonebook

Self-view video window

Picture-in-picture

Scaleable video window up to full screen

Multiple audio device selection

End-user diagnostics

Support for dual video input (NTSC or PAL) camera with analogue interface

Application Type Desktop videoconferencing hardware and softwareSupported OS Windows 95/98 and Windows NT 4.0Codecs & Standards Communications: H.323, H.320, T.120

Video: H.261Audio: G.711

Delivery methods ISDNIP unicast (TCP, UDP. RTP/RTCP)

Firewall support No informationBandwidth Range 64 Kbps to 768 KbpsH.323 Compliance Compliant.Price US$ (500-1000)

B.2.11.2 Intel ProShare Video System 500

Intel ProShare Video System 500 is a video conferencing solution which includes a single PCI ISDN, Audio and Video board, a colour camera, an headset with microphone and conferencing software such as Microsoft NetMeeting.

In terms of conferencing features, ProShare include the following:

Audio/Video:

Video up to 30fps CIF (353x288 pixels) and 30fps QCIF (176x144 pixels)

Deblocking filters for smoother video (requires 400 MHz Pentium® II processor or higher)

Video inputs: composite video and S-video (using the dual video input cable)



Video formats: NTSC/PAL auto detect Snapshot of local or remote video windows

Sizeable local and remote video windows

Full-duplex audio with built-in echo cancellation

External speakerphone support

Standards compliance:

ITU H.320 video conferencing on ISDN

ITU H.323 video conferencing on the LAN

ITU T.120 data conferencing

Audio codecs: G.711, G.723, G.728 Video codecs: H.261 FCIF/QCIF, H.263 FCIF/QCIF

LAN/Internet:

LAN connections up to 800 Kbps (400 Kbps send/400 Kbps receive)

H.323 bandwidth selectable from 56 Kbps to 800 Kbps (bi-directional)

H.323 gatekeeper support

VideoServer and RADVision H.323/H.320 gateway support

H.323 proxy/firewall support

Internet Locator Server (ILS) support

Voice Over IP support

LAN physical layer independent (can use to conference over Ethernet, Token Ring, FDDI, T-1, Frame Relay, ATM, xDSL, cable modems or any LAN that supports Windows 95/98/NT IP networking).

Application Type Desktop videoconferencing hardware and softwareSupported OS Windows 95/98 and Windows NT 4.0Codecs & Standards Communications: H.323, H.320, T.120

Video: H.261, H.263Audio: G.711, G.723, G.728

Delivery methods ISDNIP unicast (TCP, UDP. RTP/RTCP)

Firewall support Yes. H.323 proxy/firewall supportedBandwidth Range 56 Kbps to 800 KbpsH.323 Compliance Compliant. Supports other vendor H.323 gatewaysPrice US$ (1000-3000)

B.2.11.3 PictureTel LiveLan

http://www.picturetel.com/products/livelanp.htm

LiveLAN is a TCP/IP-based visual collaboration solution for Windows desktop PC that is compliant with the H.323 Terminal specifications. It is a T.120 compliant multipoint data collaboration tool, including features such as bi-directional application sharing, a shared Whiteboard, T.127 file transfer, a shared clipboard, messaging, password-protected remote control.


http://www.picturetel.com/products/livelanp.htm


It is provided with some components as a PCI Video/Audio codec board, a color composite camera, multimedia speakers, a microphone, and headset for privacy. It is provided with LiveLAN videoconferencing software on CD-ROM including T.120 data collaboration software, online documentation and optional FlipCam.

LiveLan include the following features in terms of conferencing :

Video:

Encoding : H.261

Format : CIF (352 x 288), QCIF (176 x 144)

Input : Composite camera

Analog input : NTSC, PAL - auto detect

Audio:

Modes : G.711 (16 Kbps), G.7281 (64 Kbps) for narrowband, G.722 (64 Kbps ), PT7242 (24 Kbps) for wideband

Features : echo cancellation, noise suppression, automatic gain control, integrated dynamic echo canceller (IDEC) II, full duplex

Application Type Desktop videoconferencing hardware and softwareSupported OS Windows 95/98Codecs & Standards Communications: H.323, T.120

Video: H.261Audio: G.711, G.722, G.728, PT724

Bandwidth Range Transmission Bandwidth Options : 64 Kbps, 174 Kbps, 384 Kbps, 768 Kbps

Delivery methods IP unicast (TCP, UDP. RTP/RTCP)H.323 Compliance Compliant.Price US$ 1000

1 G.728 is used during LiveGateway (H.323/H.320 gateway) calls to non-PictureTel H.320 systems2 PT724 is used during LiveGateway calls to PictureTel H.320 systems



B.2.12 Feature “at-a-glance“ comparison form

Product Application Type Supported OS Audio Codecs Video Codecs BW Range H.323 Comp.

Delivery Methods

Bandwidth Adaptation

Bandwidth Control

Firewall Support

Conferencing Standards Comp.

Price Range1

Microsoft NetMeeting Multipoint conferencing with data sharing (A/V

point-to-point only)

Windows 95/98, Windows NT 4.0 G.711, G.723, Lernout & Hauspie

SBC

H.261, H.263 28.8Kbps to 2Mbps

Yes. RAS not

supported

TCP, UDP (unicast)

RTP/RTCP RSVP Yes H.323, T.120 1

CU-See Me (Univ. Cornell)

Multipoint A/V conferencing

Windows 3.xx/9x, Windows NT 4.0, MacOS, Linux, Amiga and

OS/2

G.723.1, Digitalk, DeltaMod, Intel DVI

H.263 and M-JPEG 1.1

14.4Kbps to 256Kbps

No UDP (unicast) NA NA Yes CU-See Me proprietary

1

White Pine's Meeting Point Tools

Multipoint AV conferencing with data

sharing

Windows NT 4.0 and Sun Solaris 2.5.1/2.6

G.711, G.723 H.261, H.263 28.8Kbps to LAN

Yes IP (unicast and multicast)

NA NA Yes H.323, T.120, CU-See Me proprietary

10 Users: 6 25 Users: 7

VocalTec Internet Phone Lite

Audio Telephony Windows 95/98 & Windows NT 4.0 G.711, G.723.1, G.729

NA 14.4Kbps Yes. Fully H.323v2 compliant

TCP, UDP (unicast)

RTP/RTCP NA Yes H.323v2 1

VDO Phone Audio/Video Telephony Windows 95/98 G.723, G.728, G.711 H.261, H.263 14.4Kbps to 128Kbps

No IP Unicast Proprietary NA NA 2

Isabel Multipoint Audio/video/data

conferencing

SunOS and Linux G721, GSM, IMA M-JPEG 3Mbps No IP (UDP, TCP) IP multicast supported

NA NA NA 1

NetSpeak WebPhone Video Telephony point-to-point

Windows 95/98, Windows NT 4.0 G.723.1, G.711, GSM and TrueSpeech

H.263 28.8Kbps to 256Kbps

H.323v2 UDP, TCP (unicast)

RTP/RTCP NA Yes H.323v2 2

Netscape CoolTalk Audio conference Windows 95/98, Windows NT, Windows 3.1, MacOS, SunOS,

Solaris, HP-UX, Digital Unix, IRIX

Proprietary NA 14.4 to 64Kbps No UDP, TCP (unicast)

NA NA Yes 1

vic/vat - Video conferencing Tool

Multipoing AV conferencing with data

sharing

SunOS, Linux, NetBSD, FreeBSD, Ultrix, UP-UX, OSF, SGI/IRIX and

AIX

rat codecs H.261, JPEG 14.4Kbps to LAN

No UDP unicast and Multicast

(MBONE compatible)

RTP/RTCP, RTIP (proprietary)

NA NA 1

1 Price Ranges (in US$): (1) Free, (2) <=500, (3) 501-1000, (4) 1001-3000, (5) 3001-5000, (6) 5001-10000, (7) 10001-15000



Intel Internet Video Phone

Audio/Video telephony application

Windows 95/98 G.723, Intel Indeo Audio

i263, Intel Indeo Video 5.0

28.8Kbps to 64Kbps

“Compatible”

? NA NA NA H.323, H.324 2

VCON Escort25Pro Desktop System

Desktop videoconferencing harware and software

Windows 95/98 and Windows NT 4.0

G.711 H.261 64Kbps to 768Kbps

Yes UDP/TCP (unicast)

RTP/RTCP NA NA H.323, T.120 3

Intel ProShare Video System 500

Desktop videoconferencing harware and software

Windows 95/98 and Windows NT 4.0

G.711, G.723, G.728 H.261, H.263 56Kbps to 800Kbps

Yes UDP/TCP (unicast)

RTP/RTCP NA Yes. H.323 proxy/firewall

supported

H.323, T.120 4

PictureTel LiveLan Desktop videoconferencing harware and software

Windows 95/98 G.711, G.722, G.728, PT724

H.261 64 Kbps to 768 Kbps

Yes TCP, UDP. (unicast)

RTP/RTCP NA NA H.323, T.120 3



Annex C: Market Trends

The main purpose of this chapter is to focus on the current situation and the next evolution of the real-time applications over Internet considering both the available technology and products. In fact, what it is happening today in this market and what will be there tomorrow.

C.1 Introduction

The previous chapters of this roadmap have focused on all the technological features of real-time applications over IP and the Internet itself. In addition, Annex B has presented a review of tools & commercial products providing these services through available technologies.

The next sections will describe the market situation today, what can be expected in the future and how and why it will evolve in that way.

The market trends will not be defined by one only feature. A mix of research advances and technological possibilities, user demands, network operators and ISP strategic decisions will define the trends of this market in advance. The following sections describe how all these issues will take an important role and why all these aspects should be taken into account.

A year goes quickly. Definite trends cannot be predicted at the outset and, of course, the evolution of this project itself could modify the current views that are presented here by the end of the year. However, the following descriptions of the situation should be considered as one part of the basis, to focus the trials of P913-GI in the right way.

C.2 Market Trends in different Business Segments

Key questions for different business segments as analysed in this section are: How big is the market now? Is there enough technology available or there is still a lack of it? In which way are the products being developed? What is the strategy of the leading manufacturers, developers or content providers?

C.2.1 Internet “Core Technology”

C.2.1.1 Introduction

There is currently a major theme in Internet technology companies: Convergence. A wave of mergers and acquisitions is spreading among the major inter-networking vendors.

For instance, Cisco has acquired 28 companies since September 1993. Even Cabletron, the least acquisitive of the big inter-networking crowd, bought four companies during 1998. But even in this climate, Northern Telecom's summer 1998 acquisition of Bay Networks created a new blueprint for convergence in the industry. In addition, Lucent has assembled pieces of the IP puzzle with important acquisitions like Agile, Prominent Corp., Yurie Systems Inc., Lannet Corp, Quadritek Systems, Kenan Systems and more recently, Ascend Communications.



A tremendous consolidation is coming in the industry, where only a handful of significant competitors will remain. The search for growth is endless and the big are getting bigger. Cisco has 64 percent of the overall router market, while its nearest competitor, Bay, has only 11 percent of all router sales.

In fact, a new wave of mergers is predicted for this year. Resistance seems to be futile and more companies are prepared to be assimilated.

Meanwhile, these companies are using different ways to promote their products and make their presence available to all consumers. Banners, conferences, advertisements in newspapers and technical magazines are being used to expand on the users´ knowledge of the companies in a global world.

For instance, Nortel Networks has put a big panel in Heathrow Airport to make the new company strategy known, after the Bay merger, to all people landing in and taking off from London.

Most of the bigger companies interested in promoting their dominance over this market sponsor new conferences and fora, which are created to explode it. Cisco Systems, NetCom systems, Nortel Networks, Newbridge, IBM, HP and Sun are using this system to promote their company.

But apart from delivering new products and technology in a general way, companies are approaching different strategies depending on their technological know-how and their point of view.

C.2.1.2 Cisco

Cisco is widely announcing its IP/TV solutions. IP/TV, a comprehensive network video solution, delivers TV-quality video programming over enterprise data networks to desktop PCs using IP/TV software running on the IP/TV 3400 Series Servers. IP/TV uses standard protocols running on existing IP networks.

IOS IP version 6 support is currently under development by Cisco. However, Cisco's Beta release of IPv6 has been made generally available and offered to allow customers to gain experience with IPv6. The current IOS implementation includes support for IPv6 Neighbour Discovery, RIPv6, TCP/UDP, BGP4, IPv6, IPv4 Address Translation etc. and is available for the Cisco C1000, C1600, C2500, C36x0, C4x00 and C7x00 platforms.

Cisco will play a major role with its Cisco Assure policy-based framework, which is based on the QoS software developed by Class Data Systems (an Israeli company that Cisco acquired last year). Cisco has its own technology but nevertheless announced to migrate to the Diff-Serv standard. Cisco also announced it is to begin to use MPLS.

C.2.1.3 3Com

3Com is claiming to support RSVP on both the NetBuilderII router family and on the CoreBuilder switch family. NetBuilder II is a family of multi-protocol routers that provides differentiated levels of QoS for multi-media applications. It supports both Integrated Service/RSVP, offering Controlled Load service, and Differentiated Service models. 3Com will also use an autonomous policy-based management system called PolicyPowered Networking. In addition to the use of IP TOS at Level 3, the company will use IEEE 802.1p in switches at level 2.



C.2.1.4 Ascend

Ascend announced some time ago its GRF architecture that combines its Layer-3 switch with intelligent IP Forwarding Media Cards to deliver scalable performance. It provides a broad range of QoS features, including RSVP. It supports both Integrated Service/RSVP, offering Controlled Load and Guaranteed services, and Differentiated Service models.

C.2.1.5 Bay Networks

Bay Networks is promoting its fully featured multi-protocol routers that provide differentiated levels of QoS for multi-media applications. They support both Integrated Service/RSVP, offering Controlled Load service, and Differentiated Service models. The policy-based networking strategy from Bay Networks will be its management network system called Optivity Policy Services.

C.2.1.6 Intel

Intel advertises its Express Router 81xx family supporting RSVP.

C.2.1.7 Nortel

In a different sense, Nortel is promoting its ISP partnership program. It is clear that players of all business segments are prepared to collaborate with the others.

C.2.1.8 General recognition on supported protocols

While RSVP is being support by most of the companies, IPv6 is still not so widely supported. There are still companies doubting whether IPv6 will establish itself as the new protocol for the coming years. Anyway, companies are working on it. For instance, Cisco is engineering an implementation of IPv6 as part of the Cisco IOS software functionality, and 3Com is saying it will provide IPv6 functionality across all its relevant products and platforms.

C.2.2 Trends in Software Technology

C.2.2.1 Market Competition

NetShow (Microsoft) and RealVideo (RealNetworks) seem to be the preferred real-time applications over Internet. Apple's QuickTime is playing a much smaller role. When goals turn to higher-quality enterprise-wide multicast presentations, vendors such as Cisco, Starlight Networks and Xing Technologies come to the fore.

The two biggest players, in the daily evolving streaming-media market, Microsoft and RealNetworks, find themselves engaged simultaneously in cut-throat competition and hand-in-hand co-operation. RealNetworks has 85% of the market, and it is a market that Microsoft is making an aggressive attempt to own. There is a global trend from the both to target higher bandwidth networks. On one hand last April 13th, RealNetworks Inc. Has signed a definitive agreement to acquire privately-held Xing Technology Corporation, the leading developer and provider of MP3 software. On the other hand Microsoft expands its media services with Netshow Theater server.



The first RealPlayer was released in 1995, and to date over 50 million unique users have been registered. That's up from about 15 million at the end of 1997. The average download rate, provided by RealNetworks, now exceeds 175,000 per day, an increase of more than 270% since the beginning of 1997.

Every week, over 145,000 hours of live sports, music, news and entertainment are broadcast over the Internet using the RealSystem technology. There are also hundreds of thousands of hours of content available on-demand.

On the other hand, it is established from Microsoft that consumer adoption of Microsoft Windows Media Technologies is increasing, with more and more users of top sites choosing the Windows Media Player over other players to experience content. More than 17 million consumers now have the Windows Media Player, with copies being distributed at an average rate of 130,000 each day – up nearly 86 percent from September 1998.

Use of Microsoft's own web events site, an audio and video events site featuring links to streaming media content using the Windows Media Technologies, has soared 700 percent from 50,000 page views per day in October to over 400,000 per day. Increasingly, the Windows Media Technologies are playing a key role in enabling the Internet’s top sites to expand their reach to new Internet consumers with daily live content.

C.2.2.2 RealNetworks1

In fact, RealNetworks has collaborated in an "Anyone But Microsoft" (ABM) strategy. These include agreements with Netscape to bundle Real Player audio and video software with its free Web browser and Lotus to distribute RealNetworks´ technology to its 25 million Domino and Notes users. In addition, America Online is to embed Real Networks´ technology in its online access software, while RealPlayer becomes the default media player in AOL software, and RealNetworks is to license Intel´s data streaming technology.

RealNetworks is using a classic lock-in strategy in a competitive market. Its business model is to give away the media players, but make up the money on servers. If it sounds familiar, it's what Microsoft has been doing to the industry for 20 years.

There is a decision between two companies that have a double monopoly. But, nobody wants to feel locked in as a prisoner of Microsoft or RealNetworks. The solution comes if at the end of the road we get open standards. Both Microsoft and RealNetworks provide ways to enhance media streams and create synchronised presentations with HTML-like scripting. RealNetworks uses the SMIL format, a W3C standard, while Microsoft uses its own ASF and ASX formats.

C.2.2.3 Microsoft2

Microsoft is promoting its NetMeeting product that supports standards-based T.120 data-communication with H.323 audio calls over the Internet with optional video for face-to-face communication by adding a video camera. It combines the ability to share data in an audio/video conference with people in real-time.

Microsoft is also announcing its NetShow services version 3.0 that features enhanced video and audio quality delivering the best user experience. Simplified set-up,

1 See Annex B.1.1 for a description of Real Network tools2 See Annex B.1.2 & B.2.1 for a description of Microsoft tools



configuration, and administration of NetShow Services and tools give Internet service providers a reliable and cost-effective platform for hosting large amounts of content. It is claimed that NetShow Services provide the most complete, easy, and cost-effective streaming media solution that allows for the creation, storage, and streaming of live and on-demand content.

In addition, from a technical point of view, Microsoft focuses on its Windows Sockets 2 to provide a protocol-independent interface capable of supporting real-time multimedia communications. Windows sockets 2 supports Generic QoS, RSVP service provider and traffic control.

On the other hand, RealNetworks offers its tools, servers and players in the area of real-time support. The company announced its advanced tools for creating RealMedia and new RealSystem G2 presentations, the professional solution for creating streaming audio and video Web sites, with timeline-based features for building synchronised multimedia, SMIL and HTML templates and advanced compression control. Its RealSystem G2 Server is the next generation in web media delivery that incorporates revolutionary advances developed by RealNetworks and leading industry hardware and software partners such as Intel Corporation. But basically, RealNetworks focus is its advertisements in the RealPlayer Plus G2 player.

C.2.2.4 Cisco1

A new player is getting into the streaming market : Cisco, who bought Precept Software for their IP/TV solution. IP/TV is a comprehensive client-server software application that transmits video programs, both live and pre-recorded, to desktop PCs over enterprise IP networks.

As opposed to Real Networks and Microsoft, Cisco targets the enterprise data network, and not the public Internet network. What differentiates IP/TV from other network video products is that it focuses on the large-scale communication needs of major corporations, government agencies and academic institutions: corporate communications, employee training, distance learning and business TV. With this focus, IP/TV is very good in combining TV-quality streaming video with the application and management features, scalability and bandwidth efficiencies required for enterprise deployment.

Cisco has based its advertisement on the fact that IP/TV streamlines delivery of business communications, empowering an entire workforce with a common knowledge base: "well-trained, well-informed employees dramatically improve a company's productivity, bottom line and competitive edge".

Moreover, Cisco is very proud of announcing that IP/TV is the only solution to use fully standard protocols for all layers: transport layer standards, application layer standards, standard encoding techniques. Last but not least, an affordable network video system, IP/TV runs on existing IP networks and is backed by Cisco's superior service and support.

C.2.2.5 Audio and Videoconferencing

There are a wide range of telephony and teleconferencing applications in the market. Microsoft NetMeeting and WhitePine´s MeetingPoint tools are at the forefront of this sector. All these applications support the H.323 protocol and T.120 data conferencing standard.

1 See Annex B.1.6 for a description of Cisco IP/TV tools



For Instance, White Pine CU-SEEME PRO Version 4.0, announced in March 1999, integrates T.120 data collaboration via Microsoft NetMeeting, and provides the option of running in H.323 mode, which allows direct interoperability with other H.323 clients such as Intel ProShare.

Collaborations and strategies together with content providers will also define the future of all these software products.

IDT Net2Phone, for instance, has entered recently into an agreement with RealNetworks whereby RealNetworks will market Net2Phone. In addition, White Pine and WIN, the regional Walloon IP Network provider in Belgium, have introduced an integrated desktop videoconferencing service offering to be made available to the WIN customer base.

C.2.3 Internet Content oriented Applications

Day by day, the number of content providers offering information and entertainment to their users is increasing. Also, they are also accomplices to the success or defeat of the different software products according to their decision to use them.

Some content providers make both RealPlayer and MediaPlayer available for use. Other tools are not much promoted from content providers web pages. This is the case of CNN's videoselect service, which is the central source for streaming video at CNN, and audioselect services, NASA Real-time data service and National Geographic. They allow download of these products from their web page.

Most of the other content providers have chosen between offering their products through one of these two products. A smaller number do it through QuickTime. Both, Media Player and RealPlayer only provide links in their web pages to the content providers that use their product.

Thousands of streaming audio/video content providers can be reached through links from the same RealNetworks web page. All these providers only make use of RealAudio and RealVideo products.

ABCnews, Wall of Sound, Mr.Showbiz and Disney, under the Go Network make use of RealNetworks.

Rapidly growing adoption of the Windows Media Technologies among high-profile Internet sites shows that News broadcast sites such as CBS.com, NBC’s Videoseeker, CNN.com, Fox News Online, Fox Sports Online, broadcast.com, Bloomberg L.P.,CNBC/Dow Jones Business Video, and MSNBC, as well as sports and entertainment sites, including Warner Bros. Online and Musicvideos.com, are serving content to millions of consumers each month.

All content providers are set to attend a conference aimed at interactive media industry senior executives, Spotlight '99, that will address the "what" and "how" of investing in new media, getting to strategies and models that define and create new marketplace value. This will take place next July 1999 in California.

In the same way, the Interactive Newspapers´99 conference took place this February 1999 in Atlanta. The new technologies, and the way business is done, have been discussed. The new applications and investments that newspapers should develop were other issues treated. One of the major points was to design a business model that serves multiple media streams.

For instance, Audible Inc. and The New York Times announced this February that they have created a daily audio digest of The New York Times. It is available to



subscribers from Audible's Web site and includes news stories from the paper, as well as the editorial and OpEd pages of The New York Times. Subscribers can listen to audio either at their PC or by downloading the files to an Audible-ready handheld player.

The PointCast Network is a free Internet news service that takes the work out of staying informed by broadcasting personalised news and information directly to the computer screen. It is becoming very popular, so people get just the news they’re interested in from trusted sources like CNN, The Wall Street Journal, The New York Times and others, presented in a way that’s easy-to-scan, easy-to-read and easy-to-digest.

A different strategy is the Network Multimedia Connection which is a collaboration of three world-wide information technology leaders - Cisco, Intel, and Microsoft - coming together to accelerate and simplify adoption of multimedia applications.

Other companies, like Source Media, make advertiser-supported audio programs available to web sites. These companies offer Web sites the opportunity to keep users for a longer time with their audio reports.

Most of the newspapers, radios and TV channels are constantly offering the possibility to acquire real-time audio and video streams from their web sites. They normally provide a short audio pre-listening or video pre-view. Apart from offering their content, they provide links to different audio and video players. In some cases they only refer to a business partner player. They are also promoting conferences and fora to merge with the technological possibilities available from the software and hardware manufacturers. They are forced to work together.

C.2.4 Offers from Network Operators

Network operators are betting hard in this area. Some are working alone, others will establish collaborations with both content providers and Internet service providers.

Carriers are pushing vendors to develop faster and more robust routers and technologies to put into the backbone. In turn, customers are pushing carriers for even higher bandwidth for business critical applications.

Telstra, the Australian network operator, is working hard in this way. Its Telstra Big Pond range of Internet services, is designed to meet the needs of everyone from families to multinational corporations.

MCI WorldCom is launching a new consumer Internet service through an alliance with America Online´s Compuserve unit. The service, called MCI WorldCom Internet, replaces the business that MCI was forced to sell off last year to gain regulatory approval of its merger with WorldCom.

MCI´s return to the Net depends largely on the company's UUNet business, which will provide the backbone for the new ISP. It is another step, followed by both network operators and ISPs, of a long race in a market where no-one wants to lose their opportunity.

According to analysis by Ovum, network operators will focus their efforts in these areas due to the growth of IP services. Audio and video real time transmission will increase.

The biggest US phone company, AT&T, said last October it would set up an IP laboratory to develop new equipment and technological standards for the industry. In February 1999, AT&T announced that three service providers and seven network



equipment manufacturers (Cisco Systems, Clarent, Ericsson, Lucent Technologies, Siemens, 3COM and VocalTec) will be the first to join this industry testbed to promote IP telephony and other advanced IP services.

Deutsche Telekom is offering a content delivery platform “ODS” (On Demand Services) which supports different applications such as Music on Demand, Video-Clips on Demand, Software on Demand, Photoshop etc. The Music-on-Demand application is commercially introduced in co-operation with the major music labels. It currently offers real-time sales of approximately 30.000 music CDs and already has more than 20.000 customers.

AT&T and Time Warner have announced a joint venture recently. Nobody is likely to find a cure in the foreseeable future, but as time passes another wave of merges, joint-ventures and in-depth collaborations will take place between content providers and network operators to offer real time applications over Internet.

Network operators are focusing their efforts in offering new technological possibilities to be able to broadcast all the new multimedia content. They are all focused on offering bandwidth, bandwidth and more bandwidth.

To sum up, network operators are desperate to find new media applications and content to create new revenue streams to break the financial bottleneck of the Internet. The opportunities available to content providers work together with network operators.

C.2.5 Internet Service Providers

IP-focused service providers are in a tough competitive position. Competition is often based on price and new customer wins are costly. Margin improvement is a top priority. They are eager to deploy new services to their enterprise customers, but the limitations of today's networks and costs make this difficult to do profitably.

Cheaper and bigger networks suitable for everything from traditional telephony to videoconferencing is where Internet technology is going. Between now and the year 2000, researchers estimate that the volume of data traffic running through the world’s networks will surpass voice.

There are a number of strategic opportunities that emerge for a group of bandwidth-related products for ISPs and the overall forecast for the ISP industry is very positive. The question is if ISPs will have enough bandwidth to support the applications that companies are requiring.

An example is UUnet who are already offering multicasting through their UUcast network with different monthly rates for text applications and for audio and video.

UUNET's strategic focus is on the business market. Its primary objective is to develop a broad portfolio of products and services to enable businesses to develop applications for the Internet. UUNET has teamed up with the industry's leading Steaming Media experts to offer a world class solution. It is prepared to use both Microsoft's Media Player or Real Video/Audio.

Another example is AboveNet, the only Internet Provider that displays real-time internal and external network connectivity status to the public. AboveNet has developed a proprietary real-time monitoring system that allows AboveNet staff and customers to view customers’ servers real-time bandwidth usage. This system also ensures that links are not overloaded and that adequate bandwidth is provided at all times to their customers.



IP Multicast deployment by ISPs is happening. Between 1986 and 1999 they have learned many lessons about building internets and what is required of new routing protocols. They need well tested standards-based protocols, management tools and a compelling business reason for change. Today they have all three for IP multicast. In 1999, ISPs will announce the inter-domain work that has been done to provide customers with an IP multicast-enabled internet. 1999 could be the year of commercial IP Multicast services.

The growing attention being paid to IP multicast results from the increasing use of multimedia content on the Web. Most of these companies that are under pressure to use multimedia have contributed this February 1999 to the IP Multicast Summit in California. There is a cost of implementing IP multicast, but every day there is a growing audience for real-time events. As it was said in that conference, everyone is going to work on it.

There are about 15,000 active Internet Service Providers (ISPs) in the world. The competition in this market is very tough and profits are eroding constantly. ISPs are searching for new services for their subscribers. Real-time applications are an ideal service for these companies, offering a significant added value to ISPs.

ISPs, like AOL, are already offering content information and links to software applications from their web pages. They are also supporting real-time applications from their web pages to promote themselves. The trend is to work together and business partnerships are increasing. Without losing its real business, other market segments grow nearer every day.

C.3 Market Trends in different Global Regions

Apart from the major players presented in the previous section, another important key-point is the attitude of the respective governments towards this market.

C.3.1 North America

The Office of Advanced Scientific Computing Research (OASCR) in the Office of Science (SC), U.S. Department of Energy (DOE), is working intensively in promoting applications for the Next Generation Internet-Research in Basic Technologies program. The Next Generation Internet (NGI) is a multi-agency federal research and development program to develop, test, and demonstrate advanced networking technologies and applications.

This department of the U.S. Government is collaborating in developing different real-time applications such as real-time environmental data and real-time telemedicine. Most of these developments are done in conjunction with U.S. Universities.

The NGI initiative is part of an ongoing multiagency R&D programme. It is a key component of the activities of the Large Scale Networking (LSN) working group of the Subcommittee on Computing, Information, and Communications (CIC) R&D. This Subcommittee reports to the Committee on Technology of the White House National Science and Technology Council.

Due to the evolution of this market, new conferences are taking place and new fora are being created. We are now at the crest of a breaking wave of Internet bandwidth management technologies that promise to provide applications with QoS.

A QoS Forum has been recently founded under the auspices of Stardust Forums with Cisco, Netcom, Nortel, Lucent, Microsoft, Newbridge within its charter members.



The QoS Forum has been created as an international, industry forum accelerating the adoption of standards-based QoS technologies. Utilising a comprehensive set of education, marketing and testing services, the goal of the QoS Forum is to educate the market and increase demand for QoS-enabled IP products and services.

Similarly, more national and international conferences are taking place involving all the people with responsibilities in this market. Some instances are:

the Internet bandwidth Management Summit (iBand´98), November 1998 in California;

the IP Multicast Summit, February 1999 in California;

the MultimediaCom, March 1999 in California;

the RealNetworks Conference & Exhibition, May 1999 in San Francisco;

the MultimediaCom, September 1999 in Boston;

C.3.2 Europe

In Europe, the main initiatives on key technologies or applications usually come from the European Commission. Currently, a new Framework Programme (FP) is being launched, the fifth one, although the fourth one is still running. It is planned to finish in year 2000, then overlapping the 5th FP.

A specific area on real time services over the Internet does not exist in the 4th nor in the 5th FP. However, several projects covering some related aspects are, or have been, funded by the Commission, mainly to validate developing standards.

In the 5th FP, Key Action 4 on essential technologies and infrastructures, will probably receive proposals in this area. The same could happen, but probably to a lesser extent, in Key Action 3 on multimedia content and tools.

On the other hand, and for the moment, fora and initiatives similar to those started in the USA have not had the same importance in Europe.

C.3.3 Asian Pacific region

China has specific behaviour due to political reasons. China's Ministry of Information Industry, which was created last year, has announced that it plans to split the strongest telecommunications provider in the market, China Telecom, into three separate entities responsible for paging, mobile and fixed line networks. The agency indicated that it would not tolerate competition from foreign companies or Internet telephony providers because it would affect China Telecom's large profits, which are used to develop telecommunications facilities in poor inland areas. The Chinese government has also indicated that it will now require ITSPs to be fully licensed.

The Singapore government expressed a desire to use multimedia as a means of creating an information society dedicated to regional and cultural changes in the distribution of information. The success of Singapore ONE in Asia indicates that the entertainment value of multimedia services has given way to a more educational and professional purpose.

Australia is in the wave of the market. The Australian government is helping Telstra, a key player in the development of the global Internet and Australia´s No. 1 Internet carrier.



C.4 Trends in the Behaviour of Users

This section will review the trends in the behaviour of the users. They provide different reactions according to their capabilities.

C.4.1 Private Users

Private users are the last step in the chain of Internet use. Normally, they have tested applications within their companies or business, before they acquire them for their private use.

Audio/Video streams are becoming to be a real demand from private users in the Internet. A new market for selling these new products is already available now. A lot of new companies, like Goodnoise and Liveconcerts, are having a great success selling audio streams through Internet.

The largest increase by private users is the use of conversational applications. Away from their business, these applications provide not only a technological advantage to users but also an economic benefit.

At the moment, streaming applications are more popular between private users than any others. Private users require content basically for entertainment. They are less demanding for quality than in business environments.

Another item is IP telephony, which seems to be becoming popular for private users.

C.4.2 Public Users

Video-conferencing ability is widely in use nowadays within public authorities. Different departments of the same institution placed in different buildings or cities are making its daily use necessary.

Retrieval of audio/video databases is also becoming more popular. For instance, the MediaStream company are focusing their effort in providing technology-based solutions to assist different kinds of users to acquire content from leading sources, publish, store and retrieve that content and resell it through Mediastream´s distribution channels. The company is constantly adding newspapers to its service.

This use is becoming very popular in hospitals. Both applications are being used to progress the diagnosis and cure of some illnesses. Real-time telemedicine provides a means of remote medical consultations through the use of real-time analysis of medical diagnostic procedures.

Public users are focusing their requirements on the quality of video-conferencing applications. This is becoming a key-decision factor for buying one or other product.

C.4.3 Corporate Users

It is well established that there is wide daily use of real-time applications within corporations around the world. Multinational companies with departments and buildings in the five continents are taking advantage of these applications. Overall, they are promoting training courses and meetings through video conferencing between employees situated far apart.In addition, some of these companies with manufacturing facilities have placed video kiosks at all its sites. These kiosks give employees up-to-date reference information on product assembly and quality control.



Not only the telecommunication companies but all corporations are at the forefront of their market. For instance, an advertising department works with advertising agencies around the world. This team now has interactive, on-demand access to the company's video assets to enable faster approval of new advertisements and better evaluation of ad performance in different markets.

Some companies consider information access as the key to competitive position. TV news from around the world is scanned and pertinent news is streamed to each employee. Indeed, in a professional sports team, coaches and players can review games on-demand and easily locate specific plays or analyse their opponent's tactics.

All these companies are increasing their use of real-time applications day by day. The use of the Internet and these applications is becoming more important in all competitive markets where time, efficiency and costs are crucial.

Corporate users are commonly those who first test most of the applications. After work, the same people become private users at home. But in their different roles, they have different requirements in terms of applications and quality.

Whenever corporate users are using real-time applications, basically multiconference applications or on-demand access to any kind of information, they are normally in a hurry, and they require quick applications that provide content without delays. Quality is decisive in this environment and major efforts of the industry are focused to increase it in every new product.

C.5 Market Trends on Quality-Requirements

The IETF meeting last August 1998 was expected to lead to the end of RSVP, the primary initiative to assure QoS over IP networks. However, it turned out that RSVP, with the help of Diff-Serv and MPLS, was not only reconsidered again but moved forward.

Instead of focusing each QoS technique individually, it was possible for the three technologies to work together. The objective was to use RSVP in the domains and Diff-Serv and MPLS to control traffic through the sub-domains. The IETF´s action was to answer the requirements. Not only the corporations´ requirements but the ISPs' that should satisfy their users´ demands to offer differential services in order to use certain applications with a required level of quality. Now, RSVP seems to be the key piece in the QoS puzzle in IP.

As a result of the tremendous growth of the Internet, IP´s weakness is becoming obvious. Increasing the available bandwidth to avoid congested Internet links may be the simplest solution. But the problem is more than a simple capacity issue. The issue is that not only has traffic increased in volume, it has also changed in nature. There are many new types of traffic, from many new IP-based applications, and they vary tremendously in their operational requirements.

To address this point, some innovative companies have set up electronic exchanges that treat bandwidth literally as a commodity that can be bought, sold, and traded.

A new school of thinking is emerging that says most corporate networks do not need QoS; — instead, selectively adding bandwidth to problem areas makes QoS obsolete in the campus.

The main argument is, that if there is enough bandwidth for all traffic, then there is no congestion. If there is no congestion, there is no need to assign network traffic any kind of priority, or level of QoS, because everything will get through the network at



the fastest possible speed. But even in the light of this arguments, that doesn´t mean that QoS doesn´t have any value. It may play a key role in networks where bandwidth is still a scarce resource.

Another way of looking on quality, is to consider the requirements of new kinds of traffic, such as voice and video. QoS is gaining importance as customers increasingly ask for guaranteed response times and network that can handle the issues of jitter and latency that can affect the quality of audio and video playback.

The new applications, especially those that are delay sensitive, are forcing the development of new IP QoS levels. As these new applications demand more bandwidth, less delay and packet loss, QoS capacities are becoming a real purchase key factor.

A policy-based bandwidth control was a dream some years ago, but may become reality now. All issues like IP Precedence or IP Type Of Service (TOS), Differentiated Services, IEEE 802.1, Resource Reservation Protocol (RSVP), Common Open Policy Service (COPS), Random Early Detection (RED), Multiprotocol Label Switching (MPLS) and others will add this quality of service. But all should pay attention to the mix between standards and the proprietary implementations of the different manufacturers like Cisco and Bay Networks to avoid future problems. The IETF and the recently created QoS Forum will play a role.

In addition, service providers are also looking for ways to manage networks to better fulfil service-level agreements. At this point, economic benefits are becoming important. A carrier can offer not only homogeneous service to its customers, but it will offer valuable services at a higher price.

It is a common feeling today that bandwidth is necessary but not sufficient. It is not the only thing that is necessary; it depends how you use it. When the pool of available bandwidth increases so will the demand for bandwidth.

But the demand to control this bandwidth is also increasing. So, different approaches are possible. It is difficult to establish the relation between bandwidth and quality of service, but it seems that a general rule cannot be applied. Different users and different applications require different types of QoS and bandwidth capacity. All options should be possible. So, it is highly desirable to manage both QoS and bandwidth. Currently, most Web video is unicast, serving a separate video stream to each viewer. For high-demand live content, unicasting can be a big drain on server resources. IP multicasting sends out a single stream and let routers replicate it, potentially far away from the server. Unfortunately, while virtually all new routers and switches include multicast capabilities, most default configurations do not have the feature enabled. Worse, existing ISP and corporate equipment is often too old to support multicast. Nonetheless, enabling multicast is a necessary precursor to widespread user acceptance of live streaming video.

For radio and television networks, the convergence on IP has started but still has a way to go. Broadcasters currently serve millions of customers simultaneously, and unicast on the Internet could possibly never scale to these levels. Hence, multicast deployment is necessary to enable convergence of television and radio networks to IP.

For IP telephony, and other real time applications, the timing requirements are much more significant that the bandwidth requirements. Dropouts and delays are noticeable and distracting. Delivery delays above 0.5 second can make them unusable.



C.6 Figures about the market development

The following table gives an overview of the expected growth of data-transfer over the Internet in US-Dollars.

Reference: Frankfurter Allgemeine Zeitung, 18. January 1999.

Figure 7: Expected growth of data-transfer over the Internet

C.7 Conclusions about Market Trends

The big technology issue for 1999 seems to be the demand for bandwidth.

The biggest challenge for the bandwidth industry heading into 1999 will be concentrating more traffic through existing access lines. Whoever develops the capacity to quickly and efficiently handle voice, data, and video communications will become the major players.

The telecommunications industry started with a specific application (i.e., telephony) and built a network to suit it. The Internet, on the other hand, started in exactly the opposite way: it started with a new network technology and explored, successfully, new applications that were able to use the undefined (best-effort) service.

All business segments are betting on this market in all global regions. All users are demanding these applications. As the offer in technology increases, so will the users demands.

The Internet is network of networks, a mesh of various transmission media, with a wide range of bandwidth capacity and latency characteristics. QoS does not create



bandwidth. It only manages bandwidth according to demands. So, the demand of QoS also becomes an important key factor.

As the Internet grows in global size and bandwidth, and as computer technology increases in speed and drops in price, real-time applications over the Internet will become increasingly more feasible. We are witnessing a world-wide explosion in the demand for bandwidth.

Considering IP multicast, we are at the threshold of the point where the problems of unicast become huge, and the benefits of multicast solve them.

All players including software and hardware companies, content providers, network operators, ISPs and authorities have decided to do the best to obtain their respective rewards.

There is no doubt that real-time applications over the Internet will improve, and be widely used.


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