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Department of Science and Technology Institutionen för teknik och naturvetenskap

LITH-ITN-KTS-EX--06/004--SE

IP TV påverkan och anpassning

av IP-nätet hos en

datakomoperatör.

Jessica Eriksson

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IP TV påverkan och anpassning

av IP-nätet hos en

datakomoperatör.

Examensarbete utfört i kommunikation- och transportsystem

vid Linköpings Tekniska Högskola, Campus

Norrköping

Jessica Eriksson

Handledare Monika Gullin

Examinator Di Yuan

Norrköping 2006-02-03

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Rapporttyp Report category Examensarbete B-uppsats C-uppsats D-uppsats _ ________________ Språk Language Svenska/Swedish Engelska/English _ ________________ Titel Title Författare Author Sammanfattning Abstract ISBN _____________________________________________________ ISRN _________________________________________________________________

Serietitel och serienummer ISSN

Title of series, numbering ___________________________________

Nyckelord

Keyword

URL för elektronisk version

Institutionen för teknik och naturvetenskap Department of Science and Technology

2006-02-03

x

x

LITH-ITN-KTS-EX--06/004--SE

IP TV påverkan och anpassning av IP-nätet hos en datakomoperatör.

Jessica Eriksson

TDC Song has just entered the private market with Internet Access and VoIP services. To stay

competitive on the market in the future, the company will be required to offer the customers video- and TV- services and have a complete triple play packet.

The purpose of this masters thesis is to determine which techniques that exist within IPTV, analyze TDC Songs network and see what actions the company needs to take in network design and network quality in order to distribute a premium packet with up to 50 channels when using Moving Picture Experts Group phase 2 (MPEG-2) to compress the video.

A literature study was made to get knowledge about IPTV and the available standards to compress, transmit and display the video. Furthermore, tests were set up and performed in both laboratory and real life network environments.

The theoretical study showed that the network should be configured with Protocol Independent

Multicast Sparse Mode (PIM-SM) as the intra-domain routing protocol and Multicast Source Discovery Protocol (MSDP) as the inter-domain protocol to connect the PIM-SM domains together. The switches that can handle layer 3 routing should also be configured with PIM-SM and the other switches should use Internet Group Management Protocol Snooping to enhance their performance.

In the laboratory tests the Extreme Summit48i switch couldnt act as Rendezvous Point (RP), but has no problem with handling multicast traffic up to 1 Gbps. No multicast traffic ought to be transmitted in the parts of the network where the Cisco routers 7206 and 3640 are situated, since their performance decrease and they will drop packets. The downlink speed from the network to the Digital Subscriber Line Access Multiplexer (DSLAMs) should be upgraded from 100 Mbps to 1 Gbps to be able to handle the IPTV, VoIP and Internet traffic with as little congestion as possible.

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Introducing IPTV

Impacts and adjustments to the distribution and

backbone network of a network operator

Master Thesis project performed at TDC Song AB

by

Jessica Eriksson

Tutor at TDC Song AB: Monika Gullin

Tutor at Linköping University: David Gundlegård

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Abstract

TDC Song has just entered the private market with Internet Access and VoIP services. To stay competitive on the market in the future, the company will be required to offer the customers video- and TV- services and have a complete triple play packet.

The purpose of this master’s thesis is to determine which techniques that exist within IPTV, analyze TDC Song’s network and see what actions the company needs to take in network design and network quality in order to distribute a premium packet with up to 50 channels when using Moving Picture Experts Group phase 2 (MPEG-2) to compress the video. A literature study was made to get knowledge about IPTV and the available standards to compress, transmit and display the video. Furthermore, tests were set up and performed in both laboratory and real life network environments.

The theoretical study showed that the network should be configured with Protocol Independent Multicast Sparse Mode (PIM-SM) as the intra-domain routing protocol and Multicast Source Discovery Protocol (MSDP) as the inter-domain protocol to connect the PIM-SM domains together. The switches that can handle layer 3 routing should also be configured with PIM-SM and the other switches should use Internet Group Management Protocol Snooping to enhance their performance.

In the laboratory tests the Extreme Summit48i switch couldn’t act as Rendezvous Point (RP), but has no problem with handling multicast traffic up to 1 Gbps. No multicast traffic ought to be transmitted in the parts of the network where the Cisco routers 7206 and 3640 are situated, since their performance decrease and they will drop packets. The downlink speed from the network to the Digital Subscriber Line Access Multiplexer (DSLAMs) should be upgraded from 100 Mbps to 1 Gbps to be able to handle the IPTV, VoIP and Internet traffic with as little congestion as possible.

A buffer in the Set-Top Box should be enough to eliminate any jitter the network may cause. To be able to handle packet losses and the limited bandwidth, the network should be designed with a separate VLAN per service. This will make it possible for the company to prioritize IPTV by using VLAN tags to give Quality of Service.

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Acknowledgements

I would like to thank the employees at TDC Song that have been involved by helping me whenever I have had a question. Furthermore, I want to give a special thanks to my tutor Monika Gullin for her support and Mikael Abrahamsson for his great networking knowledge and help during the tests.

From the University I would like to thank my tutor David Gundlegård. His broad knowledge in data- and telecommunication has given me valuable inputs and reflections on my work. I would also like to thank my opponents Mattias Bruhn and Anna Garde for their comments and suggestions that have improved my rapport.

Finally, I want to thank my family and friends for the great support I have been given. Thank you for listening when I needed it the most.

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Contents

1 INTRODUCTION 7 1.1 BACKGROUND 7 1.2 PURPOSE 7 1.3 OBJECTIVE 7 1.4 SCOPE 8 1.5 METHODOLOGY 8 1.6 OUTLINE 8

2 THE FUNDAMENTALS OF IPTV 9

2.1 THE INTERNET PROTOCOL 9

2.2 TELEVISION 10

2.3 SYSTEM COMPONENTS 11

2.4 VIDEO COMPRESSION 13

2.5 MEDIA STREAMING PROTOCOLS AND STANDARDS 14

2.5.1 TRANSPORT CONTROL PROTOCOL AND USER DATAGRAM PROTOCOL 15

2.5.2 REAL-TIME TRANSPORT PROTOCOL AND REAL-TIME CONTROL PROTOCOL 15

2.5.3 REAL-TIME STREAMING PROTOCOL AND SESSION INITIATION PROTOCOL 16

2.5.4 SESSION DESCRIPTION PROTOCOL 16

2.6 QUALITY OF SERVICE 16

2.6.1 JITTER 17

2.6.2 PACKET LOSS 18

2.6.3 BANDWIDTH 20

2.6.4 TRAFFIC ISOLATION 20

2.6.5 SCHEDULING AND POLICING 22

3 MULTICAST 25 3.1 INTERNET GROUP MANAGEMENT PROTOCOL 26

3.1.1 IGMPV1 27

3.1.2 IGMPV2 27

3.1.3 IGMPV3 28

3.2 MULTICAST ROUTING ALGORITHMS 29

3.2.1 FLOODING 29

3.2.2 MULTICAST DISTRIBUTION TREES 29

3.2.3 REVERSE PATH FORWARDING 31

3.3 MULTICAST ROUTING PROTOCOLS 32

3.3.1 DISTANCE VECTOR MULTICAST ROUTING PROTOCOL 32

3.3.2 MULTICAST OPEN SHORTEST PATH FIRST 33

3.3.3 PROTOCOL INDEPENDENT MULTICAST - DENSE MODE 34

3.3.4 PROTOCOL INDEPENDENT MULTICAST - SPARSE MODE 37

3.3.5 CORE-BASED TREES 40

3.4 INTER-DOMAIN MULTICAST ROUTING 42

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3.4.2 MULTICAST SOURCE DISCOVERY PROTOCOL 43

3.4.3 BORDER GATEWAY MULTICAST PROTOCOL 45

3.4.4 MULTICAST ADDRESS SET-CLAIM 46

3.5 MULTICAST AT LAYER 2 46

3.5.1 LAN SWITCHES 47

3.5.2 INTERNET GROUP MANAGEMENT PROTOCOL SNOOPING 48

3.5.3 CISCO GROUP MANAGEMENT PROTOCOL 50

4 TDC SONG’S NETWORK 52 4.1 CORE NETWORK 52 4.2 DISTRIBUTION NETWORK 53 4.2.1 STOCKHOLM 53 4.2.2 GÖTEBORG 54 4.3 ACCESS NETWORK 55 4.4 TRAFFIC 56 5 TEST ENVIRONMENT 57 5.1 LABORATORY TEST 57 5.1.1 TEST 1 57 5.1.2 TEST 2 58 5.1.3 TEST 3 58 5.1.4 TEST 4 59 5.1.5 TEST 5 59 5.1.6 TEST 6 60 5.1.7 TEST 7 60 5.1.8 TEST 8 61

5.2 LIVE NETWORK TEST 62

5.2.1 TEST 9 62

6 RESULT AND DISCUSSION 64

6.1 TEST 1, 2 AND 3 64 6.2 TEST 4 64 6.3 TEST 5 65 6.4 TEST 6 66 6.5 TEST 7 AND 8 67 6.6 TEST 9 70 7 CONCLUSION 71 7.1 FURTHER WORK 72

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APPENDIX A - NETWORKING FUNDAMENTALS 76 APPENDIX B - NETWORK AND TEST EQUIPMENT 77

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1 Introduction

1.1 Background

TDC Song is a Nordic company that delivers Asymmetric Digital Subscriber Line (ADSL, see appendix A) access to wholesale customers, i.e. companies that offer the service down the chain to the private market. In 2005, the services consist of Internet Access and Voice over Internet Protocol (VoIP). Since the end of 2005 TDC Song also offers these services directly to private customers.

To stay competitive on the market the company will in the near future be required to offer the customers video-and TV- services and have a complete triple play packet. This means that the customers will have the opportunity to get Internet, telephone and TV on the same copper wire at a lower price.

When Internet Protocol Television (IPTV) arrives as a service, apart from Internet and VoIP, new demands will be required on TDC Songs network infrastructure in the form of quality, capacity and equipment. The quality because the transmissions are in real-time and the customers expects a TV picture with good quality. The capacity since the company wants to offer a large amount of different channels. It also requires equipment that can handle the multimedia format and the demand for compression. Another interesting aspect to consider is the potential to store already sent programs so the customer can watch them when it is

convenient for them.

This master thesis is based on the fact that TDC Song wants to know what is required to deliver the quality the customer demands when they are watching TV. It is also based on the interest of examine how and if the network design for the backbone- and distribution-network will have to change in order to deliver this TV services.

1.2 Purpose

The purpose of this master’s thesis is to determine which techniques exist within IPTV, analyze the network and the actions needed in network design and network quality given a premium packet with up to 50 channels when using Moving Picture Experts Group phase 2 (MPEG-2) to compress the video, see section 2.4.

1.3 Objective

The main goal of the master’s thesis is to recommend suitable networking technologies for distributing IPTV that is appropriate to implemented within TDC Song’s network and to locate weaknesses in TDC Song’s present network design.

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1.4 Scope

A complete IPTV solution consists of many elements and TDC Song is faced with the challenge of making numerous decisions that in the end will impact the customers IPTV experience. Due to the limited amount of time, the scope of this thesis will be restricted to the network that the company will use to transmit the service IPTV. It will not handle subjects like content rights, location of the streaming source in the network, security aspects or any specific IPTV equipment like set-up boxes or servers.

1.5 Methodology

The first step was to get knowledge about IPTV and the standards that exists to compress, transmit and display the video. To obtain this information a literature study was made. From that, conclusions was drawn about what multicast protocol to use for the tests. After that the tests were set up and performed in both laboratory and real life network environments. From this recommendations and conclusions was drawn.

1.6 Outline

This master thesis is written to a audience with a basic knowledge about data- and telecommunication. Chapter 2 gives a theoritical framework about IPTV and the quality demands required on the network to distribute IPTV services. Chapter 3 introduces multicast and describes the most common protocols used to communicate via multicast. Chapter 4 displays TDC Song’s network structure and equipment. Chapter 5 shows the setup for the tests. Chapter 6 includes results and discussions for each test. In chapter 7, conclusions can be found together with further work that would be interest to do in the area.

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2 The fundamentals of IPTV

Today there exist several techniques to get TV transmitted to your home. It can for instant be distributed via satellite, cableTV network, terresterial or through a telecom operators IP network, which is the case with IPTV. TDC Song has decided to enter the private market under the name TDC and is today offering Internet and VoIP services to the customers. The company also wants to include IPTV to be able to offer a complete triple play packet, which is necessary to compete on the market.

The competition on the market is one of many factors that drive the development of IPTV forward. Other examples are mentioned below:

Interactivity – for example voting on your favorite candidate in a TV program with a click on the TV remote instead of making a phone call.

Individualized channel content – the possibility for customers to pay for the channels they want to see and not be forced to pay for an entire packet to get one particular channel.

On-demand – offer the customer the possibility to see their favorite program when it suits them, without recording it first. For example watching the six o’clock news at eight o’clock. This is done by storing the programs on a server.

Cost-effective – three services in one network.

Channel space – the technology to offer more channels, e.g. local channels [Internet academy, 2005].

IPTV comes with advantages, like the ones described above, on-demand, high capacity, many channels, modern technique and interactivity. But like every other new technique IPTV also has some disadvantages. Because it is new it has not yet been standardized and has a

complicated end-to-end delivery compared with for example satellite that only has transmitter → satellite → end customer. It will involve a big initial investment for the telecom operators with few customers at the beginning [Internet academy, 2005].

As the term IPTV suggests the technique consists of two parts, the Internet Protocol (IP) and Television (TV).

2.1 The Internet Protocol

The Internet Protocol (IP) is defined in Request For Comments (RFC) 791 maintained by Internet Engineering Task Force (IETF). IP specifies the packets format and addressing in the network and resides at the network layer of the Open Systems Interconnection (OSI) model, see appendix A. It offers an unreliable connectionless service for transporting data from source to destination in an interconnected network, which means that there is no guarantee that the delivery will be successful [Broadband Services Forum, 2005].

Every node in the network has its own IP address that is used to communicate with each other. It consists of a network number/identifier (netid) and a host number/identifier (hostid). To send packets of data from one host to another, it is IP with the help of other protocols that performs the routing and other harmonization functions. For example routing protocols are used, to set up a routing table. When a router receives a packet, it reads the destination netid from the packet header and uses its routing table to forward the packet on the correct route. If

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the size of the packet, that was received, is larger than the maximum frame size of the destination network, the IP in the destination gateway divides the block of data into smaller blocks. Each block is then forwarded in a separate packet to the IP address in the destination host, where they are reassembled to the original block again [Halsall, 2005].

There exist two versions of IP, IPv4 and IPv6. Version number four is still the dominating version of the Internet. The upgrades in version six consists of significantly more available addresses and improvements in areas of routing and auto configuration. IPv6 is forecasted to be the protocol used in the future [Ahlin, 2003].

2.2 Television

Television (TV) specifies the medium of the communication, which in this case is the

transmission of pictures and sounds to the end-users located at the end of the access networks [Broadband Services Forum, 2005]. The TV technology is based on many scientist’s

inventions and discoveries through the years, but John Logie Baird (1888-1946) from Scotland is often called the inventor of television [Utbildningsradion, 2005].

In Sweden, TV was first shown 1930 at Röda Kvarn in Stockholm and during the 40ssome test broadcasts was performed at Kungliga Tekniska Högskolan (KTH), but the official start of TV was in September 1956. In 1962, the first live broadcast was sent over the Atlantic via a satellite and at the end of the 60s the color TV was introduced. Until the middle of the 80s there had only been two channels, SVT1 and SVT2, but 1987 TV3 started to broadcast through satellite and today there exist more than 30 channels that transmit digital and/or analog TV [Teracom, 2005].

Today analogue transmission is the most common technique used to transmit TV and Sweden, like most of Europe, transmits according to the Phase Alternating Lines system (PAL). Digital TV, on the other hand, sends according to another standard called Digital Video Broadcasting (DVB). The signal can be transmitted via satellite (DVB-S), cable (DVB-C) or terrestrial (DVB-T). With digital TV, compared to analog, more channels can be transmitted on equal amount of bandwidth and another advantage with the digital TV is the quality. The digital signal eliminates analog broadcasting artifacts like “snow” and static noise in audio, which results in a better quality. The Swedish government has taken a decision to go from analog towards digital TV and the conversion is planned to be finished in 2008 [Internet academy, 2005].

The government’s decision to go from analogue to digital TV opens up a new market for the telecom operators when the customers change.The telecom operators will compete for the customers against the other distributors of digital TV. As mentioned before, IPTV has not yet been standardized and the network is designed to transport data traffic and not sensitive media traffic. These are disadvantages compared to digital TV, but some attempts are done called

transport of DVB services over IP (DVB-IPI). There are also advantages with IPTV compared

to digital TV. IPTV has, theoretically, unlimited amount of channel capacity because one channel, at the time, is received. This is a advantage compared with digital TV where all channels are delivered in a bundle, at the same time, to the receiver. This reduces the local access bandwidth demand. The new High Definition Television (HDTV) will give better picture quality and digital sound. The technique requires more bandwidth per channel and is

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Access network

therefore easier to adapt to IPTV systems since the channels are distributed individually. Traditional digital TV system transmits the channels bundled together [Internet academy, 2005].

At the same time as operators invest in IPTV, distributors of cable/digital TV are investing in both Internet and Telephony. These two markets are now merging into one with more

competitors.

2.3 System components

An IPTV system is made up of four major elements: Video head end

Core/Distribution IP network Access network

Home network

Figure 2-1 is a simplified model to get an overview of the elements that is involved in distributing IPTV.

Figure 2-1 A simplified distribution model for IPTV [Broadband Services Forum, 2005].

At the video head end the TV content is received from a content provider through for example a satellite (see figure 2-2) and formatted for transmission over the network [Broadband

Services Forum, 2005]. The channels are encoded into a digital video format, like MPEG-2 or MPEG-4, at the encoder and one Multiple Program Transport Stream (MPTS) consisting of several channels is sent over the asynchronous serial interface (ASI) to the IP streamer [Paulsen, 2003].

Figure 2-2 Receiver to streaming server [Internet academy, 2005].

Receiver/Encoder

Streaming server

Content provider

Video head Core/Distribution IP network Home network

ASI, MPTS is transmitted to the IP streamer

Core network networkAccess Streaming

Server

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ASI is not a video standard-based signal. It is only an interface, which means that it is the format for how the data is carried. It is often referred to as the digital video broadcasting-asynchronous serial interface (DVB-ASI) and is designed to transport MPEG-2 streams in today’s IPTV solutions. Figure 2-3 shows an example of four compressed program streams that are multiplexed into a 35 Mbps MPTS and transmitted from the ASI interface to a 270 Mbps link [Paulsen, 2003].

Figure 2-3 Example of a MPTS carried out of an ASI interface [Paulsen, 2003]

The MPTS stream reaches the IP streamer, which is in charge of separating the MPTS stream to individual Single Program Transport Streams (SPTS), one for each channel. The channels are then encapsulated into IP and sent out through the IP interface. These channels are often multicast streams (se figure 2-4), but may also be unicast. Multicast and unicast is described in chapter 3. The last device before the channels are transported out to the network is the Crypto gateway, where the channels are encrypted to increase the security [Internet academy, 2005].

Figure 2-4 IP streamer [Internet academy, 2005].

The encoded video streams are then transported over the service provider’s (in this case TDC Song’s) IP network. At the end of the distribution network, the IP network is connected to the access network, which is the link from the service provider to the individual household. Telecom providers are usually using Digital Subscriber Line (DSL) technology, see appendix A, to serve individual households and some have also started to use fiber. The home network

A MPTS (ASI) that includes for example SVT1, SVT2 and TV4

These different channels are transmitted as individual channels belonging to unique multicast groups (SPTS)

MPEG-2 encoder

MPEG-2 encoder MPEG-2 encoder

MUX INTERFACEASI

12 Mbps program 6 Mbps program 3 Mbps program 35 Mbps MPTS 270 Mbps ASI carrier

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distributes the IPTV service and at the end point, the set-top box (STB) is connected to the television set. In the STB, the channel is decrypted to be displayed on the TV.

It is easier for the service provider to ensure Quality of Service (QoS) for the consumers, because IPTV services often operate over a private network and not on the public Internet. This is an important factor to be able to offer the needed video quality [Broadband Services Forum, 2005]. QoS is described further in section 2.7. The focus of this thesis is on the core and access network of the model showed in figure 2-1.

2.4 Video compression

Video compression is accomplished by making use of the similarities or redundancies that exist in a video signal. An additional goal is to reduce the irrelevant data in the signal, which means that only the features that are important is coded. In this way valuable bits will not be wasted. A video sequence consists of a series of video frames or images. Each frame can be coded as a separate image, but because neighboring video frames are typically very similar, a higher compression can be achieved by using that similarity [Apostolopouos et al., 2002]. There exist some video compression standards that offer a number of benefits, like for example, interoperability between encoders and decoders made by different companies. The Moving Pictures Expert Group (MPEG) is a standard that was established by the International Organization for Standardization (ISO) in 1988. It was created to develop a standard for compressing moving pictures (video) and related audio. The first standard, MPEG-1, was finalized in 1991 and gives a Video Home System (VHS) quality video and audio at 1,5 Mbps. After further work an extension was released called MPEG-2. It was developed for applications toward digital television and is the compression standard used in this thesis. A third standard, MPEG-3 was originally developed for higher bit rate applications like HDTV, but it was acknowledged that those applications could be addressed into MPEG-2, so there exist no MPEG-3. The latest standard, MPEG-4, was designed to provide improved

compression efficiency and error resilience features and will probably gain wide acceptance in the future [Apostolopouos et al., 2002].

The video compression is performed in two ways, movement and color. If a frame has fields with similar colours, the colours are replaced with one color and if a sequence of frames has very little movement the still parts are equal. A compressed video consists of three different frames (see figure 2-5) [Ydrenius, 2000].

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Figure 2-5 I-, P- and B-pictures and their predication reference [Ydrenius, 2000]. I-frames = real pictures, encoded independently of other frames.

P- frames = predictive coded pictures, dependent on the previous picture (either I- or P-frame).

B-frames = bidirectional pictures, resembles the P-frame but is independent on both the preceding and the succeeding I- or P-frame [Ydrenius, 2000].

When an IPTV user change a channel, an I-frame has to be received at the end-user before the whole video picture can be displayed. Therefore, the time it takes to change the channel (zapping) can never be less than the time between two I-frames in the stream. In most of the video streams, an I-frame is transmitted every other second [Internet academy, 2005]. The output bit-rate of an MPEG-2 encoder can be constant or variable. Most real-time MPEG-2 encoders are designed to execute in a constant-bit-rate (CBR) mode. These streams are often good-quality sequences. Many MPEG-2 encoding applications are real-time and the video signal has to be encoded with no significant look ahead. These applications are encoded with a constant bit rate. However, there exist non-real-time applications where the video sequence first can be analyzed and the results can be used to optimize the encoding process. One example is Digital Video Disk (DVD), which is encoded to a variable-bit-rate (VBR) output stream. This means that the MPEG-2 encoder can produce a video sequence with a constant visual quality over time. In test nine, both a CBR encoded MPEG-2 stream as well as a VBR encoded were transmitted. See chapter five and six for more information and results [Gonzales et al., 1999].

Today, MPEG-2 is the standard compression method for transmitting video, but the

development of the new standard 4 is moving quickly and is expected to take MPEG-2’s place as soon as the content providers approve the quality. MPEG-4’s main advantage is that it requires less bandwidth per channel.

2.5 Media Streaming Protocols and Standards

Above IP in the OSI model (described earlier in this chapter) reside the end-to-end transport protocols, where Transport Control Protocol (TCP) and User Datagram Protocol (UDP) are the most important [Apostolopouos et al., 2002]. Later in this section, the media delivery and the control protocols situated above the transport protocol are explained.

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2.5.1 Transport Control Protocol and User Datagram Protocol

The Transport Control Protocol (TCP) offers a reliable transport service, where it guarantees delivery via the use of retransmissions and acknowledgements. The User Datagram Protocol (UDP) on the other hand is only a user interface to IP, which means that it is unreliable and connectionless. It uses checksum and port numbering for demultiplexing traffic sent to the same destination [Apostolopouos et al., 2002].

According to (Apostolopoulos John, 2002), there are some differences between TCP and UDP that affects streaming applications:

TCP operates on a byte stream while UDP is packet oriented.

TCP delivery is guaranteed via retransmissions, but these retransmissions causes further delays. UDP does not guarantee delivery, but for the packets that are delivered the delay is less and easier to predict.

TCP provides flow control and congestion control. UDP provides neither, which provides more flexibility for the application to determine the appropriate flow control and congestion control procedures.

TCP requires a back channel for the acknowledgements. UDP does not require a back channel.

Web, e-mail and file transfers are in the most cases delivered using TCP/IP since guaranteed delivery is more important than delay or jitter (described in 2.6.1), but in the case of media streaming the uncontrolled delay of TCP is undesirable and IPTV is generally transmitted using UDP/IP [Apostolopouos et al., 2002].

2.5.2 Real-time Transport Protocol and Real-Time Control Protocol

The Internet Engineering Task Force (IETF) has developed the Real-time Transport protocol (RTP, RFC 1889) and Real-Time Control Protocol (RTCP, RFC 3550) to support the

transmission of streaming media. These protocols provide the functionalities that support real-time services [Apostolopouos et al., 2002].

RTP was designed for real-time data transfer and does not guarantee QoS (see section 2.7) or reliable delivery. It offers support for applications with time constraints, like for example IPTV, by providing a standardized framework for time stamps, sequence numbering and payload specifications. The protocol can also detect lost packets [Apostolopouos et al., 2002]. The standard way for media streaming, determined by DVB-IPI, is to use RTP/UDP for the data and send the control messages using RTCP/TCP or RTCP/UDP. This standard is not widely used today in commercial systems. The companies use theit own unique solutions to distribute IPTV.

RTCP was developed for the control messages and provides feedback on quality of data delivery. The feedback includes for example number of lost packets, inter-arrival jitter (described in 2.6.1) and delay. RTCP sends periodic feedback packets at least every five seconds and uses only five percent of the total session bandwidth. For instance the source can use the information to adapt its bit rate [Apostolopouos et al., 2002].

RTP gives a transport layer for carrying the multimedia streams, but it does not specify how the data will be organized inside this transport layer. There exist two different solutions today.

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One is inherited from the previous digital broadcasting networks (satellite and cable). Video and audio is encapsulated into Transport Streams (TS). TS is defined in the MPEG-2 systems standard and gives synchronization, signaling and security. The proposed solution consists in encapsulating the TS packets into RTP packets like in an old digital broadcasting network. Present equipment deployed for IPTV, coming from the satellite or cable world using TS encapsulation, decreases the initial costs of deployment. But it also brings an additional overhead to the network, lacks flexibility and scalability (all the information for a stream must be presented in the same TS, for example it would be difficult to have multiple audio tracks for a single movie).

The other solution, which is relatively new, consists in encapsulating the video data directly inside the RTP packet without TS. It consumes less bandwidth, permitting more flexibility and scalability. The disadvantage may be that it requires changes in broadcasting equipment. Both approaches are available and satisfactory. The Digital Video Broadcasting (DVB) forum has issued a standard for the first solution with TS and the Internet Streaming Media Alliance (ISMA) has issued specifications for the other approach without TS [Fleury, 2005].

2.5.3 Real-Time Streaming Protocol and Session Initiation Protocol

There are two session control protocols, Real-time Streaming Protocol (RTSP, RFC 2326) and Session Initiation Protocol (SIP RFC 3261). RTSP is often used for video streaming to establish a session and it supports VCR functions like play, pause and record. It is generally used for sessions in Video on Demand (VoD) systems and most of the VoD servers support RTSP today. SIP is more commonly applied for Voice over IP (VoIP) [Apostolopouos et al., 2002].

2.5.4 Session Description Protocol

The Session Description Protocol (SDP, RFC 2327) provides information describing a session. The information can for example explain whether it is a video or audio, the specific codec and bit rate. SDP is often used by RTSP for content description and it is also used with the Session Announcement Protocol (SAP, RFC 2974) to announce the availability of

multicast programs [Apostolopouos et al., 2002].

2.6 Quality of Service

The demands that exists on a complete IPTV solution is according to Internet Academy 2005):

End-to-end quality of service Effective use of bandwidth High availability

Short time for changing channels (zapping) Authentication

Conditional Access

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Quality of Service and effective use of bandwidth, displayed in the list above, are relevant for this thesis, because the focus lies on the network and not on the entire IPTV solution. Some might also consider zapping to be a matter of the network, however since the zapping time depends on how often an I-frame is transmitted rather than on the delay in the network, it will not be included in this master thesis (see video compression 2.4). It might be possible to implement a solution where a I-frame is transmitted independently of the multicast stream. This might change the users opinion of the zapping time because the user receives a standard picture when they push on the remote control instead of a black screen.

The distributor of IPTV has to know the end consumers vision of quality, which means that IPTV should have a good picture- and sound-quality, be user-friendly and have short response times. This suggests that the telecom provider should:

Receive the TV content from the content provider with good quality Have flawless coding

Have flawless transmission on the IP network [Internet Academy, 2005]

This means for the network provider that parameters like jitter, packet loss and bandwidth is quality parameters that should be controlled for a flawless transmission. That is one of the challenges with IPTV since the Internet Protocol is a best-effort service with no guarantees considering jitter, packet losses or bandwidth. These characteristics are unknown and time varying [Apostolopouos et al., 2002].

2.6.1 Jitter

Packets may experience different end-to-end delays when they are transmitted between identical end-hosts, referred to as jitter. The reason why this is a problem is because the receiver must receive, decode and display frames at a constant rate. Late frames due to jitter can be useless and cause problems in the reconstructed video in form of jerks, which leads to an unacceptable quality [Apostolopouos et al., 2002].

To address this problem it is common to have a few seconds buffer at the end-user before playback starts. The buffer gives some important advantages. It successfully extends the presentation deadlines for the media samples and can therefore almost eliminate playback jerkiness caused by delay jitter. The extended deadline also allows for error recovery through retransmissions and because video is sensitive to errors this can greatly improve the quality. The extra time can also be used to cover up errors using interleaving and error correcting codes. Like everything else the buffer also comes with a cost, it requires more storage space and additional delay before playback can start when for example a user have changed to another channel. Therefore, it is important to consider how large the buffer should be. It is a balance between eliminating jitter jerkiness without increasing the delay too much

[Apostolopouos et al., 2002].

Internet Academy (2005) illustrated an example where they generated a TV-channel with different levels of jitter, transmitted through a STB and displayed it on a TV-set. The levels of jitter where:

1. 0.01 s 2. 0.1 s 3. 0.5 s 4. 1.0 s

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In example one and two the jitter could be eliminated with the help of the buffer and had no affect on the TV picture. In example three and four the picture suffered from a little jerkiness, but it was not disturbing and disappeared quickly. Today, TDC Song delivers VoIP as a service and it has a jitter tolerance range of 0,02 - 0,03 second. This implies that VoIP is more sensitive to jitter compared to IPTV, because IPTV with the help of a buffer, can have jitter up to 0,1 second with good picture quality. Since the company has assured that the network can handle the jitter requirements for VoIP service with good quality, IPTV should not be a problem [Apostolopouos et al., 2002].

2.6.2 Packet loss

Wired packet networks like TDC Song’s are afflicted by packet loss in the sense that an entire packet is erased and these losses can have a very destructive effect on the video quality. These losses are often situated in the access network where many services uses the limited

bandwidth and not in the core where the bandwidth is higher.

Internet Academy (2005) illustrated an example where they generated a TV-channel with different levels of packet loss, transmitted throw a STB and displayed it on a TV-set. The levels of packet loss where:

1. 0.01 % 2. 0.1 % 3. 0.5 % 4. 1.0 %

Already in example one, where the packet loss was only 0.01 percent, the disturbance from the packet loss where visual in small areas of pixels that displayed wrong colours and the picture had an unacceptable quality. In example two the areas just grew in size and returned more frequently. For example three and four, the picture was totally blurry and unusable. The DVB-IPI believes that an end customer should have maximum one visible fault per hour. To further understand what that demand means, consider a typical TV channel that transmits 500 packets/second and one lost packet gives a visible fault [Internet Academy, 2005]. 500 packets · 3600 seconds = 1 800 000 packets per hour.

If the company delivers 40 channels then only one packet in 72 million is allowed to disappear. This example also shows how important it is to have a network of good quality with almost no packet loss.

To eliminate these effects the network is designed with error control. There are four classes for error control: retransmission, forward error correction (FEC), error concealment and

error-resilient video coding [Apostolopouos et al., 2002].

Retransmission involves the use of a back channel to inform the sender of which of the

packets were received correctly and which should be retransmitted. This approach uses the available bandwidth effectively but the additional delay is often not acceptable when it comes to streaming video. There are some schemes to address this problem. Delay-constrained scheme, where packets are only retransmitted if they can arrive before their deadline, or priority-based where important packets are sent before less important. With these approaches

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scheduling problems appear, concerning which packet should be transmitted next, and

additional delay that is not appropriate for real time services like IPTV [Apostolopouos et al., 2002].

In FEC, specialized redundancy is added to recover from errors. If the source transmits N packets where K packets are data and N-K packets are redundant packets. For certain codes, provided that any K of the N packets is correctly received, the original data can be recovered, but on the other hand FEC increases the necessary bandwidth with a factor of N/K. FEC does not require a back-channel and might give a lower delay, but there are some disadvantages too, like the overhead data that is present even when there are no losses and the latency related with the reconstruction of the packets [Apostolopouos et al., 2002].

When it comes to error concealment, the goal is to estimate the lost information in order to conceal the error. Video has a correlation along the spatial and temporal dimensions, which is used for video compression. If the correlation is unexploited it can be used to estimate the lost information by using the correlation and perform some form of spatial or temporal

interpolation. In a packet switched network the entire packet is either lost or received

correctly, hence spatial interpolation can’t be used because there is no spatial information left (all the pixels in the frame are lost). So only temporal information can be used and in the most cases the lost frame is estimated as the last correctly received frame causing the picture to freeze momentarily [Apostolopouos et al., 2002].

Error resilient video coding designs the video compression algorithm and the compressed bit

stream to be resilient to specific sorts of errors. One problem is the loss of bit stream

synchronization and refers to the case when an error causes the decoder to lose track of what bits correspond to what parameter. To overcome this problem, the solution is to provide mechanisms that enable the decoder to quickly isolate the problem and resynchronize to the bit stream. The simplest approach is to use resync markers. Unique and easy to find entry points are placed in the bit stream, therefore if the decoder loses sync, it can look for the next entry point and start again after that point. The resync marker can be placed in different ways. In MPEG-1 and MPEG-2, the markers are placed at strategic locations in the compressed video hierarchy, for example in picture and slice headers. This means that resyncs are placed every fixed number of blocks. MPEG-4 instead provides the capability to place the resync markers periodically after fixed number of bits [Apostolopouos et al., 2002].

Losses in a packet network like TDC Song’s have an important structure that can be exploited. Either a packet is correctly received or it is discarded, which means that the

boundaries for lost information are exactly determined by the packet boundaries. This has led to the design of a packet payload to minimize the effect of the losses. The idea is called Application Level Framing (ALF) and MPEG-4 supports it [Apostolopouos et al., 2002]. To prevent packet losses the network can be implemented with traffic isolation and priority to provide QoS. The video can be coded into different layers, one base layer and one or more enhancement layers. This method is called scalable coding and prioritizes the video data and supports intelligent discarding of the data. For example, the enhancement data can be lost or discarded while still maintaining usable video quality. The different prioritizes can be exploited to enable reliable video delivery by the use of unequal error protection (UEP). It protects important information more than data that can be lost without affecting the video. Therefore, scalable coding is a good match for networks, which support different qualities of service [Apostolopouos et al., 2002].

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2.6.3 Bandwidth

The available bandwidth between two points in the network can vary in time, depending on the load on the network. If the sender transmits faster than the available bandwidth congestion may occur, which leads to packet loss and the quality of the video decreases. On the other hand if the sender transmits slower, the receiver produces sub-optimal video quality [Apostolopouos et al., 2002].

Congestion is rather common in communication networks and happens when the traffic load exceeds the designed limit of the network. It may lead to decreased throughput, packet losses, higher delay and jitter, which all have a negative effect on the video quality. One way to deal with this problem is to increase the bandwidth in the network, but that investment is very expensive so it is more of an ongoing process than a solution. Another method is to implement traffic isolation so that packets that are more important, for example VoIP and IPTV, has priority in the routers and will travel through the network faster and thus with a reduced amount of lost packets. This will be at the cost of less important packets like web browsing on the Internet. There has been some work done to provide QoS support to face these difficulties with streaming media [Apostolopouos et al., 2002].

2.6.4 Traffic isolation

Traffic isolation permits the service provider to offer different classes of quality service to their customers. Today, two main methods are used depending on the type of network service. The most common and simplest service is Layer 2 Ethernet transport, which means that the service provider offers Ethernet as the transport and doesn’t have to test for higher protocols. To provide priority, Virtual Local Area Network (VLAN) tagging is used. If the service provider offers services up to layer 3 (IP) Type of Services (TOS) or Differentiated Services Code Points (DSCP) are used for priority. This is more complex, because layer 3 services involves more parameters that need to be configured, like for example a firewall.

[Demyttenaere and Legault, 2005].

To provide priority on layer 2, the Institute of Electrical and Electronics Engineers (IEEE) has

developed the 802.1 Q/p. It allows a service provider to attach special tags (VLAN IDs) to all the incoming frames. Therefore, the service provider can have many customers that use the same physical circuit, while still maintaining a logical separation between them. The priority field in the VLAN tag is used for priority and by using this field, the service provider can offer different classes of quality to their customers. The IEEE VLAN standard (802.1 Q/p) consists of adding 4 bytes to the Ethernet frame, see figure 2-6. The first two bytes are the type protocol identifier and identifies the frame as a tagged frame. The other two bytes are the VLAN tag and identifies the frame as belonging to a specific group on the network. When the frames go through the Ethernet network, the different switches on the path will read the VLAN tag and determine where the frame should be delivered. The first three bits in the VLAN tag identify the priority of the frame [Demyttenaere and Legault, 2005].

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Figure 2-6 One Ethernet frame and one IEEE 8021p/Q tagged frame [Demyttenaere and Legault, 2005].

Currently, two standards exist to provide priority on layer 3 and as mentioned they are called TOS (RFC 791) and DSCP (RFC 2475). These standards use the same field in the IP packet header to identify the level of service for the packet. ToS was the original standard but it is often replaced by DSCP, because it offers more flexibility in configuring different QoS parameters for customers [Demyttenaere and Legault, 2005].

The Type of Service (TOS) field is an 8-bit field in the IP datagram, see figure 2-7. The first three bits are the Predence field that prioritizes packets within a queue. Packets with the highest priority value are transmitted first. The other five fields, delay, throughput, reliability, cost and future, also act as routing criteria [Demyttenaere and Legault, 2005].

Differentiated service (Diffserv) is a model in which traffic is treated with relative priorities based on the same ToS field in the IP datagram. Diffserv increases the number of definable priority levels by reallocating bits of an IP packet for priority marking. The first six bits of the ToS field are defined as the DSCP and there exist a number of class models for DSCP values explained in RFC 2697, RFC 2698 and RFC 2598. The last two bits in the ToS field are not used for QoS. Instead, the ECN field is used for explicit congestion notification (RFC 3168) [Demyttenaere and Legault, 2005].

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2.6.5 Scheduling and policing

When the traffic is isolated from each other through traffic isolation, scheduling and policing are applied to provide QoS in the networks are scheduling and policing. Scheduling is

concerned with which packet will be transmitted next onto a link. When packets belonging to different network flows reach a router, they are multiplexed and queued for transmission at the output buffers connected with a link. The way that the queued packets are selected for transmission on the link is called the link-scheduling discipline and the most important ones are First-In-First-Out (FIFO), priority queuing, round robin queuing and weighted fair queuing [Kurose and Ross, 2005].

Figure 2-8 displays a FIFO queue in operation. Packets arrive at the link output queue and wait if the link is busy transmitting another packet. If there is not enough buffering space in the queue to hold the arriving packet, the queue’s packet-discarding policy decides if the packet is dropped (lost) or if another packet should be removed from the queue to make space for the arriving packet. FIFO selects to send the packets in the same order as they were

received [Kurose and Ross, 2005].

Figure 2-8 FIFO queue [Kurose and Ross, 2005].

In priority queuing the different packets arriving at the output link are classified into priority classes at the output queue, see figure 2-9 with two priority classes. The packets priority can depend on a marking either carried in its packet header, its source or destination IP address, its destination port number or some other criteria. In figure 2-9, packet 1 arrives and because the link is idle it gets transmitted directly. During the transmission, packets 2 and 3 arrive and are queued into low- and high-priority queues. When packet 1 is transmitted, packet 3 (high-priority) is selected for transmission over packet 2 (low-(high-priority). At the time when packet 2 is transmitted, packet 4 (high-priority) arrives, but once a transmission of a packet has begun it is not interrupted so packet 4 has to queue [Kurose and Ross, 2005].

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The round robin queuing discipline also sorts packets into classes with priority queuing, but instead of having a strict priority of service among classes, a round robin scheduler alternates service among the classes. The simplest form with two priority classes transmits first a class 1 packet, then a class 2 packet, then again a class 1 packet and so on (see figure 2-10) [Kurose and Ross, 2005].

Figure 2-10 Two-class round robin queue [Kurose and Ross, 2005].

Another concept of round robin queuing is Weighted Fair Queuing (WFQ) showed in figure 2-11. Arriving packets are classified and queued in respective right queue depending on service. The feature that differs WFQ from round robin is that each class may have a different amount of service in any interval of time. Each class i is given a weightw . During any i

interval of time as long as there are class i packets to transmit, class i will be guaranteed to receive a fraction of the bandwidth equal to:

i

w /

wj

Where the sum is taken for all classes that also have packets queued for transmission. For a link with a transmission rate R, class i will always have a throughput of at least:

wi wj

R /

Figure 2-11 Weighted fair queuing [Kurose and Ross, 2005].

The other mechanism for QoS is policing, it handles the regulation of the rate at which a flow is permitted to insert packets into the network. There exist three policing criteria: average rate, peak rate and burst size [Kurose and Ross, 2005].

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Average rate gives the network a change to limit the long-term average rate (packets per interval) at which a flow’s packets can be sent. The critical matter in average rate is the interval of time over which the average rate will be policed. For example is an average rate of 100 packets per second more constrained than 6000 packets per minute, even though both have the same average rate over a long enough interval of time. This is because the second constraint could allow a flow to send 1000 packets in a given second-long interval (as long as it don’t send more than 6000 packets per minute), while the first cannot have that behaviour [Kurose and Ross, 2005].

Peak rate limits the maximum number of packets that can be transmitted over a shorter amount of time. From the example described above, the network may police the average rate to 6000 packets per minute, while restricting the flow’s peak rate to 1500 packets per second. The network might also want to limit the maximum number of packets, the burst of packets, which can be transmitted over a very short interval of time. That is called burst size. The burst size restricts the number of packets that can be sent into the network at the same time as the interval length approaches zero [Kurose and Ross, 2005].

The leaky bucket is an example that can be used to explain these policing limits. Figure 2-12 shows a leaky bucket with a bucket that can hold up to b tokens (regulates the burst size). New tokens are generated at a rate of r tokens per second (limits the average rate at which packets can enter the network). If the bucket has less than b tokens new token is added and otherwise it is discarded. A packet must first remove a token from the bucket before it is sent onto the network. If the bucket is empty, the packet will either wait for a token or it can also be dropped. To regulate the peak rate, two buckets are needed [Kurose and Ross, 2005].

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3 Multicast

Users on the network can communicate in different ways like unicast, broadcast or multicast. In unicast communication packets are transmitted to one receiver. Therefore, in the IPTV case, one individual stream would be transmitted to every end-user that wants to watch TV, regardless of where they are located in the network. This mean that packets with the same content might travel on the same links several times consuming unnecessary amount of bandwidth. Broadcast communication is a one-to-all transmission, where the packets are sent to all nodes on the network. This is also not an efficient way since some parts of the network might not want to have the IPTV stream [Forouzan, 2003].

Multicast is a communication form that consists of a one-to-many transmission. The sender transmits the packets to a multicast IP address that users, which want to receive the packets, have to join. This eliminates that duplicate packets travels on the same link and restricts the stream to network segments that want to receive it. That is why multicast is the

communication technique that has been chosen for IPTV and it will be described closer in the following chapter. Figure 3-1 shows the different protocols that can be used for multicast transmission to deliver the IP packets [Forouzan, 2003].

IGMP MSDP Multicast Source DR DVMRP MOSPF PIM-DM PIM-SM CBT D R DR MBGP, BGMP RP BR BR Core Distribution RP DR DR IGMP Snooping, CGMP

Figure 3-1 Overview of the different parts in a multicast solution [Cisco Systems, 2003]. There are two problems that arise when using multicast communication, the first is how to identify the receivers of the multicast packet and the second is how to address the packet to these receivers. In unicast communication the network use the IP address of the receiver to deliver the packet to the correct destination. In broadcast, the packet is delivered to all nodes, no destination address is needed [Kurose and Ross, 2005].

In the case of multicast there are multiple receivers. One alternative is to have the multicast packets carry all the IP addresses of the receivers. That might work with a small number of receivers, but it would not scale well if a group consisted of hundreds or thousands users because the amount of addressing information would be more than the actual data.

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Furthermore, it requires that the sender know the identities and addresses of all the receivers and in some cases this requirement might be undesirable [Kurose and Ross, 2005].

To go around these drawbacks a multicast packet is addressed using address indirection, which means that a single identifier is used for the group of receivers. The packet is copied and delivered to all the multicast receivers related with that group. The IP address used is a class D multicast address and the group of receivers connected with a class D address is called a multicast group [Kurose and Ross, 2005].

It is quite simple to understand the concept of the multicast group, but it raises a lot of questions and some of them are mentioned in Kurose and Ross (2005):

How does a group get started and how does it terminate? How is the group address chosen?

How are new hosts added?

Can anyone join the group or is membership restricted?

Do group members know the identities of other group members?

How do the network nodes interoperate with each other to deliver a multicast packet to all group members?

The answers to all these questions involve the Internet Group Management Protocol (IGMP).

3.1 Internet Group Management Protocol

The Internet Group Management Protocol (IGMP) defines how to join or leave multicast groups and provides information about existing groups. The function of the protocol is to operate between the host and its directly attached router, see figure 3-2 [Kurose and Ross, 2005]. IGMP supplies the ways for a host to inform its attached router that an application, for example IPTV, running on the host wants to join a specific group by sending a Host

Membership Report message. Each group represents in this case one TV channel [Kurose and Ross, 2005]. IGMP is today the most common used protocol for switching channels according to Broadband Services Forum (2005).

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3.1.1 IGMPv1

IGMPv1 is defined in RFC1112 and figure 3-3 displays the message format.

Figure 3-3 IGMPv1 message format [Zheng, 2001].

The version field in the message corresponds to the version of the protocol, in this case version one. The type field can consist of two different types of messages:

Type 1, Host Membership Query (sent by multicast routers) Type 2, Host Membership Report (sent by hosts) [Zheng, 2001].

The unused field must be set to zero and the checksum field includes the checksum for the IGMP message. Hosts use the group address field to report their membership in a specific multicast group and in a query message, the field is all zeros and means nothing [Zheng, 2001].

The multicast router transmits Host Membership Query messages periodically to decide which hosts groups have members on its directly attached networks. The Time-To-Live (TTL) field is set to one so other multicast routers do not forward the message. When a host receives a Query message it responds by transmitting a Host Membership Report for each group to which it belongs. Host Membership Reports suppress redundant messages by using a random back-off timer. The host suppress its response and resets the timer to a new random value, if a host hears another Report for the same group during the timeout period. A

multicast router does not need to know the number of members in a group, only that one exits. This leads to a reduced amount of traffic on the subnetwork [Zheng, 2001].

When a host wants to join a group it immediately transmits a Report for the group instead of waiting for a query message and if a member wants to leave a group it simply stops

responding to the query messages. This means that the leaving process is only determined by a timeout, which leads to a long delay. The timeout for a router that uses IGMPv1 is usually 3 times the polling interval (the time interval between two Host Membership Query messages). Even if the hosts on the subnetwork have left the group, the router will keep on forwarding the data. If there existed a leave group message to inform the router that members are leaving, the leaving delay could be reduced. IGMPv2 was developed to address this problem [Zheng, 2001].

3.1.2 IGMPv2

IGMPv2 (RFC 2236) improves IGMPv1 and is backward compatible at the same time. The message format for IGMPv2 is displayed in figure 3-4. The version field and type field from IGMPv1 are combined into one type field. Some examples of different types are shown in table 3-1.The checksum and Group Address are the same as in IGMPv1.

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Figure 3-4 GMPv2 message format [Zheng, 2001].

Type Ox11 (Group Address is zero) General membership query

Type Ox11 (Group Address ≠ zero) Specific group membership query

Type Ox16 Membership report

Type Ox17 Leave group

Type Ox12 Membership report (IGMPv1)

Table 3-1 Type numbers for different types of messages sent in IGMPv2 [Zheng, 2001]. There are some differences between IGMPv1 and IGMPv2:

IGMPv2 has an explicit leave message that can reduce the delay. The host sends a leave message to the all routers group address, 224.0.0.2. The designated router responds with a Group-Specific Query message to the subnet where the leave message came from and if no Host List Report is generated as a response the subnet is

removed.

IGMPv2 includes a new Group-Specific Query message that the router uses to send query messages to a specific multicast group rather than to all groups on the subnet as in IGMPv1.

When there is several routers on the same subnet, the one with the lowest IP address is automatically selected to be multicast querier, responsible for sending the Host

Membership Query messages. In IGMPv1 it is the multicast routing protocol that decides the querier [Zheng, 2001].

3.1.3 IGMPv3

The main improvement with IGMPv3 (RFC 3376) is that it has support for Group-Source Report messages, which means that a host can select to receive traffic from a specific source of a multicast group. This makes it possible for receivers to control which source is allowed to send to them. There are two types of Group-Source Report. Inclusion Group-Source Report, where the host can specify the IP address of the specific sources it wants to receive data from and Exclusion Group-Source Report that specifies the IP addresses of the sources that the host does not want to receive data from. In IGMPv1 and IGMPv2 is the information from all sources for that group forwarded to the subnetwork when a host joins a multicast group. By selecting sources in IGMPv3, the multicast protocols get some help to conserve bandwidth by not constructing unwanted branches in the multicast tree. The Leave-Group message in IGMPv2 has also been modified to allow a host to specify the IP address of any source-group it wishes to leave [Zheng, 2001].

Today, IGMPv2 is the version that is most commonly used, but most experts believe that IGMPv3 will replace it in the future.

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3.2 Multicast Routing algorithms

IGMP defines how hosts communicate with their designated router, but it does not state how to deliver the multicast traffic between the routers, how the routers exchange membership information or how they make a copy of every data unit to arrive to all group members. That is done by multicast routing protocols. They are in charge of constructing the multicast delivery trees and performing multicast packet forwarding. This part describes the routing algorithms that are applied in multicast routing protocols [Zheng, 2001].

3.2.1 Flooding

Flooding is the simplest algorithm, when a router receives a packet it just forwards it to all interfaces except the one, which it was received from. The router has no routing table to uphold and the algorithm is uncomplicated to implement, but there are some disadvantages. It is very inefficient and not scalable, because it produces many duplicate packets and uses all available paths along the network. To avoid that packets are circling around for an eternity the algorithm uses the TTL (time-to-live) field. Whenever a packet pass a router the TTL field is decreased by one and when it reaches zero the packet is thrown away [Zheng, 2001].

3.2.2 Multicast distribution trees

To be able to understand IP multicast it is important to have knowledge about multicast distribution trees. In unicast, traffic is routed through the network along a single path from the source to the destination, but when it comes to multicast, the source is transmitting to a

changing group of hosts represented by a multicast address. Multicast distribution trees are used to describe the path that the multicast traffic flows through the network to reach all receivers. There are two kinds of multicast trees, source trees and shared trees [Williamson, 2000].

The source tree is the simplest form of distribution tree. The source of the multicast traffic is the root, which branches form a spanning tree through the network to the end-users. This tree applies the shortest path through the network and is therefore also known as a shortest path tree (SPT). In the spanning tree algorithm there are only one active path between two routers. Once the spanning tree has been set up, the packets can travel along the tree and reach all end hosts. One problem with the spanning tree is that it concentrates traffic on the root paths of the tree, which can cause a bottleneck in the case of high traffic. Spanning tree is an

improvement over flooding since it stops uncontrolled multiple forwarding of packets throw the network [Williamson, 2000].

Figure 3-5 displays an SPT for multicast group 224.1.1.1 with the root at the source (Host A). Two receivers are placed at Host B and Host C. The notation (S, G) that is written in the picture specifies a SPT where S is the IP address of the source and G is the multicast address of the group. There exist a separate SPT for every individual source transmitting to each group [Williamson, 2000].

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Figure 3-5 Shortest Path Tree (SPT) from Host A. (S, G) = (192.1.1.1, 224.1.1.1) [Williamson, 2000]. Shared trees are different from source trees that have their root at the source. They use a single root located at a selected point in the network. The root is called Rendezvous Point (RP) or Core depending on the multicast routing protocol. In a shared tree, sources must send its traffic to the root for the traffic to reach all the end-users. Shared trees can be divided into two sorts, bidirectional and unidirectional. In the case of a bidirectional tree, the traffic can flow up and down the tree to reach all end-users. In figure 3-6 the packets, from Host B, flows both up the tree to the root (router D) and down the tree to Host A [Williamson, 2000].

Figure 3-6 Bidirectional shared tree [Williamson, 2000].

Unidirectional-shared trees only let multicast traffic to flow down the shared tree from the root to the end-users. This means that the source needs a way to get the traffic to the root to be forwarded down the tree. One technique is that the root joins a SPT rooted at the source to drag the traffic to the root. This is displayed in figure 3-7. The root has joined a SPT to Host B and drags the traffic down (dashed arrows), which will then be forwarded to the other receivers (solid arrows). Protocol Independent Multicast (PIM), described later in 3.3.3, uses this method. Another way of forwarding the traffic from the source to the root is for the first-hop router to unicast the traffic directly to the root. The CBT multicast routing protocol uses this technique [Williamson, 2000].

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Figure 3-7 Unidirectional shared tree with SPT to get the traffic to the root [Williamson, 2000].

3.2.3 Reverse Path Forwarding

Almost every IP multicast routing protocol applies some form of Reverse Path Forwarding (RPF) or incoming interface check to decide whether to forward or drop an incoming multicast packet [Williamson, 2000]. The multicast router must include a routing table with the shortest path to all destinations. The RPF algorithm uses the packet source address to avoid data from traveling around in loops [Zheng, 2001]. Once a packet arrives at the router, it examines the source address to establish whether the packet arrived from an interface that is on the reverse path back to the source. If this is the case, the RPF check is successful and the packet is forwarded, otherwise it is discarded. Figures 3-8 and 3-9 describe two examples. In figure 3-8 the multicast packet from source 151.10.3.21 is received on interface S0 and when the router check the multicast routing table, the reverse path back to the source is S1, not S0, so the RPF check fails and the packet is discarded. In figure 3-9 the RPF check is a success cause the packet is received on interface S1 and the packet is forwarded to all the other interfaces on the outgoing interface list, which does not have to be all interfaces on the router [Williamson, 2000].

References

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