the audio over ip instant expert guide - the codec...

The Audio over IP Instant Expert Guide

Version 1.1January, 2010

Tieline Pty. Ltd. © 2010

Audio over IP Instant Expert Guide


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Table of ContentsPart I Introduction 4

1 .............................................................................. 5What exactly is IP?

Part II 10 Great Reasons to Broadcast Audio overIP 8

Part III Broadcast Applications 9

Part IV Types of IP Connections 11

Part V Selecting a Network 16

Part VI Important IP Network Considerations 21

1 .............................................................................. 21Audio over IP Transport Protocols

2 .............................................................................. 23Choosing an Algorithm

3 .............................................................................. 25Concealing Packet Loss

4 .............................................................................. 28Managing Jitter (Latency)

Part VII Dialing over IP Networks 30

1 .............................................................................. 32NAT and Port Forwarding

Part VIII Planning IP Network Installation 34

1 .............................................................................. 34Regional Factors Affecting IP Connectivity

2 .............................................................................. 35IP Network Suitability and Reliability

3 .............................................................................. 38Selecting a Data Plan

4 .............................................................................. 41Redundancy Considerations

5 .............................................................................. 41IP Interoperability

6 .............................................................................. 43Checklist for IP Connections

7 .............................................................................. 45Testing a Network

8 .............................................................................. 47Assessing Hardware Requirements

Part IX Glossary of Terms 49

Part X Trademarks and Credit Notices 51

Contents


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Part XI Appendix 1: IP Protocols 51

4 Audio over IP Instant Expert Guide


1 Introduction

Audio-over-IP has proved itself to be the broadcast network infrastructure fortoday and into the future. As a consequence, increasing numbers ofbroadcasters are migrating to low-cost wired and wireless IP networks frommore costly analog leased line, microwave and synchronous datatechnologies like ISDN and X.21.

Broadcasters now clearly recognise that IP networks are more flexible,cheaper to upgrade and just as reliable as older network technologies. As aresult, broadcasters are using IP audio codecs to design and operate moreadaptable broadcast networks with streamlined work flows, reducedoperating costs and the ability to remote control them from anywhere in theworld.

For many years Tieline Technology has recognised that the future ofbroadcasting is in packet-switched networks supporting audio over IP, andas a member of the Audio-via-IP Experts Group, Tieline has been at theforefront of determining the direction of broadcasting audio over IP. Tielinehas assisted thousands of broadcasters to seamlessly transition audiodistribution, studio-to-transmitter link and remote broadcast infrastructureinto IP technologies.

The information in this guide is useful to users of all brands of audio codecsand is supplemented by more detailed information in Tieline's IP and 3GIPStreaming Reference Manual, which is available for download at www.tieline.com/transports/Audio-over-IP. You can also contact Tieline support [email protected] to find out more if you have any further questions orrequests.

How to use this Guide

The Audio over IP Instant Expert Guide is an invaluable resource forbroadcasters new to IP and is a useful reference tool for thosebroadcasters familiar with IP concepts. It dispels myths such as:

1. IP is not reliable enough to broadcast over.2. Broadcasting over IP is complicated.3. You need to be an expert in IT to broadcast over IP.

http://www.tieline.com/transports/Audio-over-IP

http://www.tieline.com/transports/Audio-over-IP

mailto:[email protected]

Introduction


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None of this is true, and after reading this guide broadcasters shouldfeel confident that they have sufficient knowledge to configure, run andmonitor broadcast audio connections over IP. The guide providesinformation about audio over IP in a logical sequence and will provide:

1. An introduction to IP.2. A description of the differences between IP networks and

traditional analog leased line and synchronous leased line datanetworks.

3. An overview of how audio over IP can be used in differentapplications and over different networks.

4. Detailed IP network information and considerations.5. Recommendations of how to plan your IP network installation

and assess your IP network requirements.

In addition to this guide, you can become a part of a community ofbroadcasters who interact regularly to discuss topics relating tobroadcasting audio over IP. Tieline runs online forums at http://forums.tieline.com/, where you can ask any IP broadcast related question.

1.1 What exactly is IP?

Some Background on Networks

When you broadcast over IP you are essentially connecting like acomputer would over a Local Area Network (LAN) or Wide Area Network(WAN). A LAN is a network covering a small local area and a WANcovers a much wider area, e.g. the internet. LANs and WANs can bewired or wireless.

Some networks like wireless WiMAX networks are called MetropolitanArea Networks (MANs) and typically these cover a city. A MAN is largerthan a LAN but smaller than a WAN. There are a plethora of wired andwireless IP networks that interconnect with each other and can be usedto broadcast high quality audio.

What is IP?

IP stands for Internet Protocol, which is a protocol used to send dataacross packet-switched networks. Packet-switching is used by

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computer networks and telecommunications devices (e.g. IP audiocodecs and 3G cell-phones). Data packets are individually routedbetween two devices over Local Area Networks (LANs) or Wide AreaNetworks (WANs).

What do you need to send high quality audio over IP?

Using IP you can make connections between two IP audio codecs, orbetween an IP codec and other compatible devices connected to smallprivate LANs or large public WANs like the internet. These codecs anddevices can connect using hard-wired Ethernet connections (like thoseused by a PC to connect to the internet), wireless connections, or acombination of hard-wired and wireless connections.

Example of IP Codecs using Wired and Wireless IP NetworkConnections

Wireless IP connections can be made over 3G and 4G wireless cellphone networks, public or private WiMAX wireless IP networks andBGAN satellite connections.

What are the Differences between IP and otherSynchronous Digital Data Connections?

Circuit switching, used in synchronous digital data networks like ISDNand X.21 and wireless GSM CSD and HSCSD networks, creates adedicated connection between two end points in order to send datapackets exclusively between two devices.

Introduction


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Packet-switched networks are more efficient and optimize the use ofbandwidth over computer and wireless networks by dividing datastreams into packets with destination addresses embedded withinthem. In this way packets can travel through different routers to theirdestination in order to find the fastest way to their destinations.

IP versus ISDN, POTS and X.21 Networks - An HistoricalPerspective

In the past synchronous data networks have been preferred for studio-to-transmitter links (STLs) and audio distribution within broadcast networksbecause of their guaranteed data rates and reliability, commonlyreferred to as QoS, or Quality of Service.

IP came along with the promise of more efficient use of bandwidth overcomputer and wireless networks, but this came at a cost - well twocosts to be exact. The two key factors you need to understand tomanage network reliability are network 'jitter' and packet loss. Jitterrelates to the amount of time required for an audio codec to receive allthe data packets sent to it, then reorder them and play them out insequence and reliably stream audio without any audio interruption.

Packet loss relates to data packets sent from one codec to another thatare lost. Lost packets can potentially cause 'artifacts' or glitches inquality when streaming audio, unless you have the right equipment tomanage it. We will discuss these factors in detail later, but the keything to remember is that software developments and improvements tobroadband network infrastructure have mitigated the effects of jitter andpacket loss to a large extent in most situations.

Despite the potential pitfalls of IP, broadcasters are moving inexorablytowards the technology because of the cost advantages and flexibility.The transition into IP network infrastructure is also gathering pace asolder analog and digital synchronous networks like ISDN are phased outand shut down.



2 10 Great Reasons to BroadcastAudio over IP

1. Broadcasting over IP is cost-effectiveIP network infrastructure is cheaper because you can distributebroadcast quality audio over existing broadband networks such asDSL/ADSLIP network broadband costs are generally much cheaper than analogleased lines and synchronous data networks like ISDN and V.35/X.21

2. The hardware required for IP broadcasting is cheaper

A single IP audio codec can send multiple streams of audio tomultiple points, so less hardware is required than over traditionalsynchronous networks like ISDN and X.21

3. Broadcasting over IP is more flexible

Routing audio over IP is much more flexible because a single IP audiocodec can deliver a choice of unicast, multicast and multiple unicastIP streams for network audio distribution

4. IP networks can be scaled to suit individual installations

Broadband Internet Service Providers and Telcos offer a range ofcompetitively priced data plans that provide flexible connectionbandwidth to suit each installation - minimising data costs andmaximising network efficiencyIt is possible to incrementally increase available network databandwidth as demands increase over time

5. Wireless IP networks deliver flexible broadcast connections fromanywhere at anytime

A range of wireless networks are available to broadcast audio over IP,including:

o Wireless 3G networks (EVDO/UMTS/HSDPA/HSUPA)o Long-range WiMAX wireless IP networks (2-100kms)o Wireless BGAN satellite connections

6. IP Networks are Widely Available

Wireless broadband networks are widely available in most regions ofthe world

10 Great Reasons to Broadcast Audio over IP


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Wired broadband connections are widely available in most regions ofthe world and at major sporting venues (etc).

7. Setting up remote IP broadcasts is extremely simple

There is no longer any need for remote vans and cumbersomemicrowave linksA single codec can be preprogrammed to connect to a wired orwireless broadband network very simply

8. Audio over IP integrates seamlessly with new broadcasttechnologies

Packet-based audio over IP integrates seamlessly when broadcastingaudio streams over the internet and a wide range of digital radioformats

9. Integration of audio over IP into large radio networks createseconomies of scale

Opportunities to consolidate and centralise the distribution of audioaround radio networks and affiliates is facilitated by the flexible andscalable nature of IP codec hardware and broadband networkinfrastructure

10.Audio over IP is the future of broadcasting

Major networks around the world are migrating to IPAnalog leased line and synchronous network infrastructure like ISDNis being phased out in most regions of the worldAudio over IP has the flexibility to adapt to meet the changing needsof technologyRegular DSL/ADSL data plans are sufficient to deliver 22kHz audioquality for audio distribution or studio-to-transmitter link applications

3 Broadcast Applications

IP audio codecs deliver a range of flexible solutions to broadcasters.

Remote Broadcasts

IP codecs are suitable for many different wired or wireless remotebroadcast applications.



Live sports.Live news reports.Live radio and television shows.Live concerts.

Studio-to-Transmitter Links (STLs)

IP codecs can be used to send program audio from the studio to thetransmitter site over a range of different IP networks.

Public Internet Connections (WANs). Private LAN Connections. Dedicated or Shared Fiber Connections. Public or private WiMAX networks. A mix of the above-mentioned services.

Broadcast Applications


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Audio Distribution

With the advent of digital radio broadcasting there has been exponentialgrowth in audio distribution using IP. Multichannel digital radio hasopened the door to new networking and narrowcast opportunities forradio networks and IP audio distribution delivers a cost-effective andflexible solution for:

Distributing programming between network affiliates or studios. Sending program inserts to studios or affiliates. Distributing audio from one point to multiple end points. Sending voice tracks from remote studios, affiliates and otherlocations.

4 Types of IP Connections

IP offers the ability to create much more flexible broadcast networks for amuch lower investment than traditional analog and synchronous digitalnetworks. Next we outline the three basic audio codec application conceptsimportant to understanding the capability of broadcasting audio over IP -unicasting, multicasting and multiple unicasting.

IP Information: An IP address is a unique number that allowsdevices to communicate between each other over IP networks usingthe Internet Protocol standard. There are two types of IP addresses –public and private (see Dialing over IP Networks).



What is Unicasting?

In computer networking a unicast transmission is defined as thesending of data packets to a single end point or node. A similarprincipal is employed in audio over IP broadcasting and a unicastconnection is a one-to-one connection between transmit and receiveaudio codecs.

Unicast Applications for Broadcasters

Unicasting over IP provides full-featured connections with highquality bidirectional stereo audio capabilities, as well as full duplexcommunications. It is useful for:

STLs between a studio and a single transmitter site.Broadcasts from a remote site to a single destination.Simple audio distribution between two points.

Example of a Unicast IP Connection

What is Multicasting

IP multicasting is used by broadcasters to deliver a single audio streamto many recipients. In some ways it is a lot like traditional radiobroadcasting where you transmit a single signal over a wide area andanyone with a radio can tune in. When multicasting, the audio streamsent from the transmitting codec is distributed over the IP network toother codecs and only a minimal amount of bandwidth is required to

Types of IP Connections


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transmit the original program audio.

Multicast routers over the IP network replicate packets on demand asrequired. They are then forwarded to the group of codecs that hasexpressed an interest in receiving the transmissions.

Multicast Applications for Broadcasters

Multicasting is an effective way to distribute audio to manylocations with minimal IP configuration. It does not require a largeamount of bandwidth at the codec transmitting the broadcast audioand it is particularly useful to broadcasters over private LANs thatsupport multicast audio distribution within a network. Multicastsare ideal for:

Distributing high quality audio over broadcast LANs.Distributing audio to multiple zones within a broadcast ornon-broadcast network.Distributing broadcast quality audio throughoutenvironments like large buildings, airports, hotels and retailoutlets.

Multicasting can also be a good way to set up permanent STLconnections to affiliate codecs across LANs that support multicastconnections.

Multicast Broadcast Example



Key Multicasting Concepts

Like broadcasting generally, with multicasting it is not necessaryfor the transmitting codec to know all the recipients of atransmission.

Multicast transmissions are sent using a dedicated IP multicastaddress that looks similar to a regular IP address and multicastsubscribers request transmissions from this address. This uniqueaddress allows multicast routers to identify multicast requests froma group of codecs interested in a particular transmission andpackets are replicated depending on demand. This can createlarge demands on network bandwidth if the multicast group issignificant in size.

Only small sections of the internet are multicast enabled and manyInternet Service Providers (ISPs) block multicast traffic over widearea networks like the public internet. This restricts most multicastbroadcasts to private local area networks.

Some ISPs provide quality of service (QoS) priority to multicaststreams for an increased service charge. Some also offer QoS tobroadcasters if the broadcast transmissions are delivered as aservice to the ISPs subscribers.

The important multicast concepts to remember are:

Multicast streams are not automatically allowed over WANsand are usually difficult and more expensive over thesenetworks.The network path must include multicast-enabled routersand switches.Bandwidth required at the transmitting codec is minimal.The total bandwidth of all transmissions over a network canbe significant if the multicast group is large.Codec streams are unidirectional (receive only) for themulticast group subscribed to a broadcast.

Types of IP Connections


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What are Multiple Unicasts?

Multiple unicasting (multi-unicasting) technologies expand the conceptof unicasting by creating multiple connections from one broadcastcodec to a specific selection of other codecs. The transmitting codecmust specify exactly which codecs will receive individual audio streamsand dial them directly. This differs from multicasting, where thetransmitting codec sends a single stream into the network and thenetwork replicates the streams.

Multiple unicasts can be performed over either LANs or WANs and aremost suited to broadcasting over the internet when compared withmulticasting. Multiple unicasting is limited only by the number ofconnections the codec is able to dial and the bandwidth available at thetransmitting codec.

The total bandwidth of each connection is the bandwidth required tosuccessfully broadcast all the individual IP streams. For example, if youcreate ten 100Kbps connections, you will need at least 1Mbps ofbandwidth capacity for program content at the codec broadcasting themultiple unicast audio streams.

Multiple Unicast Applications for Broadcasters

Multiple unicast technologies provide broadcasters withopportunities to deliver high quality broadcast audio streams tomultiple codecs from a single codec. Compared to multicasting,unicast streams are more capable of traversing wide area networkslike the internet and are more secure. Multiple unicasts are idealfor:

Distributing multiple streams of program audio to radionetwork affiliates.Sending multiple STL signals to different transmitter sites. Monitoring STL connections at several locations.Distributing network audio for local program inserts.Sending remote broadcast audio to several affiliates within anetwork.

They are a great way to send multiple feeds from any broadcast



location. As long as the network connection sending the audio hasthe bandwidth required, connections can be made over WANsquite quickly and simply.

Multiple Unicast Example

Key Multi-Unicast Concepts

The important multiple unicast concepts to remember are:

Multiple unicast connections can be sent over WANs orLANs quite simply by dialing each individual connection.Bandwidth required at the transmitting codec is directlyproportional to the number of connections being used.Different codecs have different multi-unicast capabilities andsome can provide a return signal path for confidencemonitoring of audio.

5 Selecting a Network

IP networks come in various shapes and sizes and the network that is mostsuitable for your requirements depends on your broadcast application (e.g.remote broadcast, audio distribution or STL). In this section we explain thedifferent types of networks and suggest which ones can be used to performstudio-to-transmitter links, audio distribution and remote broadcasts.

Selecting a Network


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Wired LANs/WANs/MANs

Ethernet connections to LANs, WANs, MANs are used extensively forwired IP connections over local, metropolitan and wide area networks.Wired networks are capable of high data transfer rates and are morereliable than wireless networks. Depending on data requirements, fiber-optic cabling is used increasingly for high-bandwidth data networks,particularly over local area networks. Depending on the networkinfrastructure available over private LANs, higher data rates may providethe opportunity to send uncompressed digital audio at very high bit-rates.

Wired IP is the ideal solution for:Dedicated studio-to-transmitter links between studios, includingmulticast and multiple unicast applications.IP audio distribution across broadcast networks, includingmulticast and multiple unicast applications.Remote broadcasts.

Wireless 3G Networks

There are basically two different types of 3G networks; UMTS/HSDPA/HSPA+ and EV-DO. Speeds vary from network to network and are also affected by the hardware used (i.e. type of antenna) and environmentalfactors. The data bandwidth provided by 3G wireless broadbandnetworks is often sufficient to send up to two channels of high quality20kHz audio. Wireless 3G networks provide low-delay connections withtypical latency of around 100 to 200 milliseconds.

Wireless 3G is the Ideal Solution for:Wireless remote broadcasts from wherever a 3G signal isavailable.Backup connections when a primary broadcast connection fails.



A Typical Wireless Remote Broadcast

UMTS/HSDPA/HSPA+

W-CDMA is the technology behind the UMTS (Universal MobileTelecommunications System), HSDPA (High-Speed DownlinkPacket Access ) and HSPA+ (also known as HSPA Evolution,Evolved HSPA, I-HSPA or Internet HSPA) standards for 3G.

HSDPA is commonly referred to as 3.5G and extends UTMStechnology to provide higher data uplink and downlink bit-rates thantraditional W-CDMA. Maximum network download speeds of up to14.4Mbps and upload speeds of up to 384Kbps can be achievedover HSDPA networks. HSPA+ provides even higher data rates ofup to 42 Mbit/s on the downlink and 22 Mbit/s on the uplink.

These networks are the most suitable for streaming audio over IPand are typically found in Europe, the Middle East, Africa andAustralia (AT&T in the USA).

EV-DO

EV-DO (Evolution Data Optimised) was evolved from CDMA2000®standards and EVDO Rev 0 can potentially deliver 400 - 1000Kbpson the downlink and 50 - 100Kbps on the uplink. EVDO Rev Adelivers 600Kbps - 1,400Kbps downlink and 500Kbps-800Kbpsuplink. These networks are typically found in the USA (e.g.Verizon, Sprint, Alltell).

Unsuitable Wireless Networks

Edge, GPRS and 1xRTT are not suitable for live streamingbecause the bit-rates are too low for continuous live streaming.

Selecting a Network


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Wireless 4G WiMAX Networks

It is possible to broadcast over either public WiMAX Metropolitan AreaNetworks (MANs) or portable point-to-point and multipoint WiMAXconfigurations. WiMAX is short for Worldwide Interoperability forMicrowave Access and WiMAX IP links effectively create reliable, highspeed, long range broadband IP connections at up to 70 megabits persecond between two points or multiple points.

WiMAX operates using the IEEE 802.16 wireless standard and it hasbeen developed primarily for medium to long-range outdoor transmissionhops. WiMAX is more efficient than Wi-Fi connections and it has higherdata rates and a greater range.

WiMAX is the ideal solution for:Studio-to-transmitter links in remote locations where wired orwireless telecommunications infrastructure is unavailable. Remote broadcasting where 3G networks are unavailable, orwhere large amounts of data bandwidth are required. Audio distribution within regions where good line-of-sight can beachieved over long distances.

Dedicated Private WiMAX Networks

Instead of leasing a dedicated link from a Telco it is possible tocreate your own private long-range LAN. Portable low-cost WiMAXsystems deliver dedicated full-duplex, high-speed data connectionsbetween two points or between the studio and multiple remotelocations, providing cost-effective bi-directional transmission pathsfor audio distribution, remote broadcasting or studio-to-transmitterlinks. Once these systems have been purchased there are noongoing data costs.

Portable systems generally consist of a base station and areceiver that can operate at distances of between 2km and 100km,depending on the line of sight available, the antenna arrangementused and whether repeaters are added. Portable WiMAX links areideal for roof-top or rural deployments because of their small sizeand low power requirements. They can operate in unlicensed RFbands and be used by broadcasters to deploy WiMAX solutions



easily and cost effectively.

A Typical Portable WiMAX Broadcast

Metropolitan WiMAX Networks

Metropolitan 4G WiMAX networks have a range of up to 30 miles(50kms) and are becoming more prevalent in cities around theglobe. These 4G wireless broadband networks provide high-speeddata connections for broadcasting high quality audio from withinlarge MANs. Visit http://www.wimaxmaps.org/ to view globaldeployments of WiMAX networks.

A Typical Metropolitan WiMAX Network Broadcast

Satellite IP

Satellite IP connections are a dependable way to send broadcast audioto the studio from very remote locations where other wireless networkinfrastructure is unavailable. Using a BGAN satellite terminal it ispossible to send one or two channels of studio FM quality audio from aremote location.

Satellite IP is the ideal solution for:Broadcasts from very remote locations where 3G wireless orwired IP connections are not available.

http://www.wimaxmaps.org/

Selecting a Network


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A Typical Satellite IP Broadcast

6 Important IP NetworkConsiderations

Packet switching optimizes the use of bandwidth over computer andwireless networks by dividing data streams into packets with destinationaddresses embedded within them. In this way packets are routed throughISP routing tables to find the best route to their destinations.

The exact form of a packet is determined by the protocol (see Audio over IPTransport Protocols) a network is using and this affects the actual size ofthe packet. Packets are generally split into three parts which include:

A Header: This section contains instructions about the data containedwithin the packet; The Payload: This contains the actual data that is being sent to thedestination; andA Trailer (Footer): This tells the receiving device that it has receivedthe entire packet and it may also contain error checking information(used to send a packet resend request if a packet is corrupted).

6.1 Audio over IP Transport Protocols

A number of protocols are used in creating connections over IP.These protocols are used to:

Create IP packetsProvide statistics and feedback about IP streamsEstablish connections.



TCP versus UDP

TCP (Transmission Control Protocol) is an internet transport protocolmost commonly used for many of the internet’s applications such asemail and the World Wide Web and it is what most codecs use forestablishing a connection. The TCP protocol ensures reliable in-orderdelivery of data packets between a sender and a receiver. Its twofunctions include controlling the transmission rate of data and ensuringreliable transmission occurs. TCP is generally not well-suited tostreaming live audio. Broadcasting audio packets over TCP connectionswill typically deliver more latency than UDP connections. This isbecause more buffering is employed to ensure data packets arereceived in order.

UDP (User Datagram Protocol) is the protocol used most commonly forsending internet audio and video streams and the EuropeanBroadcasting Union (EBU) standard for audio over IP recommends usingRTP over UDP rather than TCP. The UDP protocol is different to theTCP protocol in that it sends datagram packets. These packets includeinformation which allows them to travel independently of previous orfuture packets in a data stream. In general, UDP is a much faster andmore efficient method of sending audio over IP and RTP over UDPsometimes has a higher priority than TCP in internet and networkrouters. Tieline has written special Forward Error Correction software(FEC) for UDP data streams, which significantly increases the stabilityof a connection over IP.

SIP (Session Initiation Protocol)

SIP is a signaling protocol used to connect, monitor and disconnect amyriad of different connections over the internet such as telephonecalls, conferencing and multimedia distribution. It provides multi-user/device sessions and connections without regard for the particular deviceor the media content that is delivered and is the protocol, along withSDP, used to provide codec compatibility and interoperability accordingto EBU N/ACIP Tech 3326 (the audio over IP standard used for providingcompatibility between different brands of codecs).

SIP works with a myriad of other protocols to establish connections with

Important IP Network Considerations


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other devices over the internet and carries SDP messages. It is used tofind call participants and devices even when they move from place-to-place and is the method used by most broadcast codecs to connect tocompeting brands of codec for interoperability. SIP and SDP combine tonegotiate the type of audio coding that can be used over a connection.

Other Protocols

Other important IP protocols are listen in Appendix 1 of this document.

6.2 Choosing an AlgorithmBefore you send audio over your IP network you need to select whether youwill be sending the data uncompressed or compressed. To senduncompressed data requires very high rates of data, therefore it is bettersuited to a private LAN or WAN. In most situations you will need to select acompression algorithm.

Most audio codecs allow you to select your preferred compression algorithmusing software menus. The algorithm you select will depend on how muchbandwidth you have available and it will affect not only the quality of thebroadcast, but also contribute to the amount of latency or delay introduced.For example, if MPEG Layer 2 algorithms are used, program delays will bemuch longer than when using Tieline Music, MusicPLUS, aptX or AACalgorithms. This is due to the additional inherent encoding delays involvedwhen using MP2 algorithms. This can be a major consideration for liveapplications where you need bidirectional communications.

The algorithm you choose to connect with will also depend upon:

The codecs you are connecting to (Tieline versus non-Tieline)Whether you are creating point-to-point (unicast), multicast ormultiple unicast connections.Whether you are connecting using SIP or not (some algorithms arenot commonly used over SIP). The uplink bandwidth capability of your broadband connection.

It is a good idea to listen to the quality of your program signal using eachalgorithm and to see how it sounds when it is sent at different connectionbit-rates (as well as different FEC and jitter-buffer millisecond settings). Thiswill assist you to determine what the best algorithm is for the connection



you are setting up.

Algor-

ithm

AudioBand-

width

Algor-ithmicDelay

IP bit-rateperchannel

IP over-head

Recommendedconnection for on-air use

Linear(Uncom-pressed)

16/24bit upto96kHz

0ms samplerate x bitsper samplex no.channels

80Kbps Extremely highquality uncompressedaudio distribution andSTLs

TielineMusic

Up to15kHz

20ms 24 Kbps(minimum)

16Kbps High quality low bit-rate remotes, STLsand audio distribution

TielineMusic-PLUS

Up to22kHz

20ms 48 Kbps(minimum)

16Kbps Very high quality lowbit-rate remotes,STLs and audiodistribution

G.711 3kHz 1ms 64Kbps(minimum)

80Kbps Voice qualityconnections to otherbrands of audio codec

G.722 7kHz 1ms 64Kbps(minimum)

80Kbps Voice qualityconnections to otherbrands of audio codec

MPEGLayer 2

Up to22kHz

24 to36ms

64Kbps(minimum)

8.5 -13.3Kbps

Very high qualityaudio connectionsbetween Tieline orother brands ofcodec.

MPEGLayer 3

Up to15kHz

100ms 64Kbps High quality low bit-rate remotes, STLsand audio distribution

AAC-LC Up to15kHz

64ms 64Kbps 15Kbps High quality low bit-rate remotes, STLsand audio distribution

AAC-HE v.1 Up to15kHz

128ms 32-48Kbps 7.4Kbps High quality low bit-rate remotes, STLsand audio distribution

AAC-HE v.2 Up to15kHz

128ms 16-24Kbps 7.4Kbps DAB+ radiostreaming and high



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quality low bit-rateremotes, STLs andaudio distribution

aptXEnhanced

24kHz 2ms 384Kbps(Stereo)

STLs and audiodistribution

6.3 Concealing Packet LossWhen broadcasting using audio over IP it is critical that the codec you usehas very good packet loss and jitter buffer management software, as well as error concealment and Forward Error Correction (FEC) strategies.

Packet Loss

Packet loss in IP networks can be caused by:

Signal degradation over the IP link.Network congestion, i.e. buffer overruns in IP routers.Corrupted packets.Faulty hardware.

The amount of audio degradation caused by lost packets will depend onthe number and size of the packets lost during transmission andreception. Audio artifacts become evident if many packets are lost or iflarge packets containing a lot of data are lost. IP codecs can detect theintegrity of every packet because the UDP and TCP protocols used in IPdata packets verify the integrity of every packet received by a device.

If you select broadcast audio codecs that provide packet deliverystatistics then you will be able to assess network congestion andpacket delivery reliability. This allows you to reliably adjust yourconnection bandwidth or other settings like the jitter buffer or ForwardError Correction (FEC) to maximise connection stability. This maysound complicated but in practice it is quite simple to do.

Concealment

Network protocols like TCP provide for reliable delivery of packets byasking for retransmission of lost packets. This can be inefficient andlead to the connection bit-rate being higher than expected if manypackets are lost. Packet loss concealment can also be used to mask



the effects of lost or discarded packets during an IP broadcast. Lossconcealment methods include:

Reproduction of the packet received prior to the lost packet.Estimation of the value of each dropped packet by interpolationand insertion of these artificially generated packets into the bit-stream.

These methods can be useful in disguising a few dropped packets hereand there, but if several packets are lost in a row audio quality willbecome noticeably impaired.

Forward Error Correction (FEC)

Forward Error Correction (FEC) is a method designed to increase thestability of UDP/IP connections. FEC works by sending a secondarystream of audio packets so that if your primary audio packets are lost orcorrupted, then packets from the secondary stream can be substitutedto correct the primary stream.

The amount of FEC that you require will depend on how many datapackets are being lost over the network connection and it can only beused over networks where bandwidth congestion is not an issue. Welldesigned codecs let you to manually adjust the FEC setting usingsoftware.

A high quality broadcast codec should provide statistics that allow youto view how many packets are being lost over the network. This let's yougauge the amount of FEC that you require to maximise connectionquality and stability. For example, if you are losing one packet in everyfive that is sent, and you have a FEC setting of 20%, the lost packetswill be replaced by FEC to maintain the quality of the connection. If youare losing more packets than this, say one in three, it will be necessaryto increase the FEC setting to 33% to compensate.

Why not use 100% FEC every time?

The answer is because you need twice the data rate or bit-rate toachieve full redundancy and depending on link conditions, this couldcause more dropouts because of network congestion than it fixes. Here



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is a simple rule to remember: Your maximum uplink speed is all thebandwidth you have to play with. As a rule of thumb, try not to exceedmore than 50% of your maximum bandwidth. If your link is shared, beeven more conservative.

You should also consider the remote end too. What is the remotecodec's maximum upload speed? Is the connection shared at eitherend? Your bit-rates, FEC settings and buffer rates must be pre-configured at both ends before you connect, so it's always better to setyour connection speed and balance your FEC according to the availableuplink bandwidth at each end for best performance.

Conserving Bandwidth with FEC

There is a trade-off between the quality and the reliability of an IPconnection particularly when FEC is activated on your codecs.However, it is possible in certain situations to set different FEC on eachcodec to match connection bandwidth requirements at either end of thelink, conserve bandwidth and create more stable IP connections.

For example, if your broadcast is a one-way broadcast from a remotesite, i.e. you are not using the return path from the studio, or only usingit for communications purposes, it is possible to reduce or turn off FECat the studio codec. This effectively reduces the bandwidth required overthe return link (communications channel) and increases the overallbandwidth available for the incoming broadcast signal from the remotesite. This could be particularly useful if you have limited uplinkbandwidth at the remote location.

Keep in mind that as you move from local to national to internationalconnections, you should be more conservative with your FEC andconnection bit-rates. As a general recommendation, choose a codecthat shows you how much data you are using per second in aconnection and never exceed 50 percent of your total upload bandwidthat each end of your link - especially over the internet.



6.4 Managing Jitter (Latency)

Jitter

Jitter, (also known as latency or delay), is the amount of time it takesfor a packet of data to get from one point to another. Over packet-switched networks delay is variable, depending on the path packetstake from their source to their destination. Latency is an important issuewhen using packet-switched networks particularly when broadcastingaudio in live situations. Latency over packet-switched networks iscreated by:

Network transmission delay.Physical processing delay over the network via switchers androuters etc.Packet delay, including algorithm compression delays.

Packet jitter occurs when data packets sent over a network do not arrivein regular intervals. This occurs because packets can travel over anyroute to their destination despite being sent in regular time intervals.The random delays that occur, and the severity and frequency of thesedelays, will be different for every connection. The combination of factorscontributing to the total latency over a network mean that a temporarybuffer is required to ensure reliable play-out of audio streams whenbroadcasting.

What is a Jitter Buffer?

A jitter buffer is a temporary storage buffer in codec software used tocapture incoming data packets to ensure the continuity of audiostreams is maintained. Data packets travel independently and arrivaltimes can vary greatly depending on network congestion and the type ofnetwork used, i.e. LAN versus wireless networks.

In a way, a jitter-buffer can be looked upon as a pre-programmed delayinsurance for packets not turning up in time. The trade-off, or cost ofincreasing the jitter-buffer is increased latency in the overall connection.The greater the jitter-buffer delay programmed, the greater the programdelay. Packets are retrieved from the jitter buffer at regular intervals by adevice’s decoder in order to provide a smooth and regular play-out of



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audio streams. The concept of jitter buffering is displayed visually in thefollowing image.

If a jitter buffer delay setting is not high enough then it is likely thatinterruptions to streams will occur as a result of late packets. If the timevalue or ’depth’ of the jitter buffer is set at a point larger than the longestexperienced jitter delay, then all packets received by a device will bedelivered to the decoder and the best possible audio quality isrecreated.

Unfortunately there are two problems with this scenario:

1. There is no way to predict for sure what the longest jitter delaywill be, and

2. The larger a jitter buffer is (to increase the chance of catching alllate packets) the longer the end-to-end and round trip delay ofdata becomes. (In extreme circumstances this can becomeunacceptable for bidirectional audio applications that need lowdelay)

Tieline has developed an automated jitter buffer solution that analyzesthe history of observed jitter over a connection and then set the jitterbuffer depth automatically based on this result. This is dynamicallyadjusted over time automatically and compensates for observed networkcongestion. Packet delivery statistics are provided that allow you tooptimise the jitter buffer setting on your codec to accurately suitprevailing IP network conditions.



7 Dialing over IP Networks

Private versus Public IP Networks

Public IP networks are operated by the various Telcos and InternetService Providers and they provide a range of different data transmissionservices for businesses and the public. These Telco networks generallyprovide connections to WANs like the internet as well as MANs andLANs and they facilitate sending data between users from a wide varietyof different networks. Customers can subscribe to various different dataplans tailored to suit their individual requirements.

Examples of private IP networks include company Intranet and Wikiservices, portable WiMAX systems and private home computernetworks that are not accessible to users outside of the network. Publicnetworks provide interconnections between these private networks viathe internet.

Public versus Private IP Addresses

An IP address is a unique number that allows devices to communicateover networks and the internet using the Internet Protocol standard.There are two types of IP addresses public and private and theseaddresses can be static (fixed) or dynamic (assigned from a pool of IPaddresses). As examples, a private IP address might look like192.168.0.100, and a public address might look like 203.36.205.133 or74.76.21.72.

Certain IP address ranges have been allocated for private use and theseprivate addresses help to create secure private networks. Privateaddresses can be used by anyone on a private LAN but computers ordevices using these numbers are unable to connect directly over theinternet without using Network Address Translation (NAT) and a publicIP address.

Conceptually, public and private IP addresses operate similarly to publicphone numbers and private phone extensions because an IP numbercan be public or private. For example, a standard PBX telephonesystem allows people to call you on a single public telephone numberand performs the translation and routing of the public number into aparticular private PBX extension. Private and public IP addresses

Dialing over IP Networks


31

operate in a similar way to private and public phone numbers - sosimilar dialing principles apply.

If you want to dial a codec with a private IP address you will requireNetwork Address Translation (NAT). NAT allows a single device, suchas a broadband router, to act as an agent between the public internetand a local private LAN. Usually this will be set up at the studio end soyou can dial into the studio from the remote codec.

You can think of NAT as if it was the receptionist in an office. Whensomeone calls the main office number looking for you, the receptionistlooks up your number and routes the call to your private extension. NATworks in the same way by forwarding data packets to codecs withprivate IP addresses.

Don't get too hung up on IP addresses and NAT because although itmay seem confusing at first, it is really quite straightforward to programwith some simple instructions and your IT administrator can assist youwith this sort of programming. Following is a table describing thedifferent types of IP addresses you may encounter and how they impacton broadcasting over IP.

Type of IP Address

How the IPAddress isAllocated

Description

Public StaticPublic IP Address

InternetServiceProviders(ISPs)

ISPs allocate a static public IPaddress to allow network devices tocommunicate with each other over theinternet. It works like a publictelephone number and will allow yourremote codec to call your studio codecover the internet.



Dynamically AssignedPublic IPAddress

InternetServiceProviders(ISPs)

ISPs usually allocate dynamically(automatically) assigned public IPaddresses to allow network devices tocommunicate with each other over theinternet. (Not recommended for studioinstallations because each time youconnect to your ISP the IP address canchange).

Private Dynamic-ally AssignedPrivate IPAddress

DHCP Serverfrom yourown privateLAN network.

A DHCP server-allocated IP addressthat is automatically assigned to adevice on a LAN to allow it tocommunicate with other devices andthe internet. This address can changeeach time a device connects.

StaticPrivate IP Address

LANAdministra-tor

A network administrator-allocatedstatic address which is programmedinto a device to allow it to connect to aLAN. Often a security measure to onlyallow access to devices approved by anetwork administrator.

7.1 NAT and Port ForwardingWe have mentioned how Network Address Translation (NAT) is used toconnect codecs with private IP addresses with devices that have public IPaddresses. Computers and other devices that connect over IP also havesoftware ports that are used to sort different types of network traffic.

In TCP and UDP IP networks the codec port is the endpoint of yourconnection. Software network ports are in a sense doorways for systems tocommunicate with each other. For example, several codecs in your studiomay use the same public static IP address. Therefore it is necessary toallocate port numbers to these codecs so that when an incoming callcomes in, the network knows which codec to send the call to.

Picture a house and imagine the front door is the entry point represented byan IP address. You want to get to several codecs in different rooms of thesame house and the doors to each of those rooms are represented bydifferent port numbers. In principle this is how port addressing works. Whena studio with a designated public IP address receives data from several

Dialing over IP Networks


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different remote codecs, port addressing information is extracted from theincoming data packets to ensure the correct packets are sent to the rightstudio codecs. This process is performed by Port Address Translation(PAT), which is a feature of Network Address Translation. Visit http://en.wikipedia.org/wiki/Port_address_translation to learn more about theseprinciples.

Managing Port Forwarding

By default Tieline codecs use a TCP session port (9002 or 9012) tosend session data and can use either a TCP or UDP (9000 and 9010)port to send audio. UDP is best for streaming audio and the reason thesession port always uses the TCP protocol is that TCP is the mostlikely protocol to get through firewalls ensuring critical session data(including dial, connect and hang-up data) will be received reliably.

When dialing other brands of codecs using SIP, codec manufacturersuse UDP port 5060 to send session data and UDP port 5004 is used tosend audio.

Codec manufacturers let you program port forwarding using softwareapplications. The following example shows Tieline's web-GUI codecprogramming application with default TCP 9002 session port and UDP9000 audio port settings for an IP connection.

If there is a need to change your codec's port settings, in mostsituations you should consult your organization’s resident ITprofessional and they can assist you with this over your network.

http://en.wikipedia.org/wiki/Port_address_translation

http://en.wikipedia.org/wiki/Port_address_translation



8 Planning IP Network Installation

There are several factors that you need to consider when choosing theequipment that is most appropriate for your requirements. The typicalquestions you may face include:

1. Network Reliability: how reliable are connections over the IP networkthat I want to broadcast over?

2. Which IP network is most suitable for remote broadcasting, STLs andaudio distribution.

3. Data costs: what is my return on investment for IP broadcastingcompared to traditional leased line networks like ISDN?

4. What codec and algorithm will I use to broadcast and what sort ofdata plan will I need?

5. What level of redundancy do I require?6. Hardware costs: how do I assess my hardware requirements based

on my broadcast requirements?

8.1 Regional Factors Affecting IPConnectivity

Connection reliability will vary from region to region and country to country.However, as a rule of thumb, it is possible to apply some generalassumptions about local, national and international IP connections. Thefollowing information is a guide only, because networks are always beingupgraded and depending on the network you are connecting to you canachieve great results over local, national and international connections.

A local IP connection will usually:

Route data using the same service providerAchieve higher bit-rates and better quality audio connectionsRequire low rates of FEC or none at allRequire low jitter buffer delaysBe most reliable

A national IP connection will usually:

Require data to be routed through more internet router pointsAchieve good bit-rates and good quality audio connections

Planning IP Network Installation


35

Require low to medium rates of FECRequire low to medium jitter buffer delay settingsBe reliable

An international IP connection will usually:

Require data to be routed through many internet router points andmany service providersAchieve lower bit-rates and hence lower quality audio connectionsRequire medium to high rates of FECRequire the highest jitter buffer delay settingBe less reliable

An awareness of these factors when you are setting up your IP connectionwill assist you to configure each IP connection successfully, and obtain thebest performance.

8.2 IP Network Suitability and ReliabilityOther factors that affect the stability of an IP connection include whether itis:

Over the public internet or a managed IP network with QoS (Quality ofService)A wired or wireless connection.Shared with other devices like computers.

Whenever possible use wired IP connections that are not being shared withother devices.

Quality of Service (QoS) Networks

It is necessary to make a distinction between managed IP networks andthe internet, which is essentially a public unmanaged IP network. Thehighest reliability is achieved by broadcasting over managedconnections provided by Telcos and some Internet Service Providers(ISP). These can provide Quality of Service (QoS), meaning that prioritycan be given to different users or data flows across their IP network.This generally requires a Service Level Agreement (SLA) with the Telcoor ISP to provide consistent data flow at all times. This is not possiblewith unregulated wide area network internet connections.



SLAs are normally associated with dedicated 24/7studio-to-transmitterlinks and audio distribution requiring guaranteed reliability. They aremore expensive than standard DSL/ADSL connections but usually lesscostly than synchronous data links like ISDN.

Broadcasting over the Internet

Advances in codec technology have led to audio codecs being usedwidely over the public internet for remote broadcasts, STLs and audiodistribution. Part of the problem with broadcasting over the internet isthe unpredictability of how congested the network will be at any point intime. To a large extent these factors can be dealt with by software likeTieline's QoS Performance Engine software, which automatically adaptsto the prevailing conditions of the internet and adjusts automatically tocompensate for increases in packet arrival latency - ensuring audiocontinuity is maintained over time.

This can be problematic in some situations if congestion causeslatency to be severe and bi-directional communications is required.However, to a large extent advances in coding and networkmanagement technologies have led to most of these latency issuesbecoming manageable in most situations.

IP Network Alternatives

There are a range of common wired IP networks available forbroadcasting audio over IP

IPNetworkInterface

Description Recommen-dation

DSL/ADSL(DigitalSubscriberLine)

Common and transmits bi-directionaldigital data over the public internet usinga POTS/PSTN line. Typically uses mostof the data channel bandwidth todownload data to a subscriber and willonly transmit data as fast as the DSL/ADSL data uplink will provide. The

Point to PointSTL/AudioDistribution

Point-to-PointRemoteBroadcastsMulticasting



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outbound data rate can vary greatly, socheck with your Internet Service Providerto discover the speed of their connection.

SHDSL(SymmetricHigh-speedDigitalSubscriberLine)

SDSL/SHDSL connections sendsymmetrical data (i.e. 512 kbps downlinkand 512 Kbps uplink) as opposed toDSL/ADSL connections which sendasymmetrical data (i.e. 512 Kbpsdownlink and 256 Kbps uplink).Symmetrical data normally delivershigher uplink speeds than DSL/ADSLconnections - increasing the stability andquality of your connections. Unlike DSL/ADSL, SDSL and SHDSL cannot betransported on top of a POTS line.

Point-to-PointSTL/AudioDistribution(MPEGAlgorithms)Multicasting

MPLSCompliantInterfaces

Multi-protocol Label Switching is a high-performance data carrying mechanismused to send multiple types of data traffic- including IP packets, ATM, SONET(fiber) and Ethernet frames. MPLS tagsdata packets with a 'header' to define thepath of the packets across the network.The protocol supports bandwidthreservation and delivers QoS guarantees,so is ideal for STLs and audiodistribution.

Point-to-PointSTL/AudioDistribution(MPEGAlgorithms)Multiple UnicastSTL/AudioDistribution/RemoteBroadcasts(MPEGAlgorithms)

Wireless3G

Different wireless 3G networks likeUMTS, HSDPA and EV-DO deliverwireless broadband IP connections overwide areas of most countries.

Wireless remotebroadcasts

WiMAXMetropol-itan AreaNetwork(4G)

Like wireless 3G networks, metropolitan4G WiMAX networks have a range of upto 30 miles (50kms) and provideconnectivity to the internet wirelessly.Bandwidth is generally greater thanstandard 3G wireless networks.


PortableWiMAX

Portable WiMAX systems deliverdedicated full-duplex, high-speed data




wirelesslinks

connections between two points orbetween the studio and multiplelocations, providing cost-effective bi-directional transmission paths for audiodistribution, remote broadcasting orstudio-to-transmitter links. Operatingdistances of between 2-100 kms arepossible.

Point-to-PointSTL/AudioDistribution(MPEGAlgorithms)Multicasting

8.3 Selecting a Data PlanIP network data costs vary depending on the network you are connecting toand the number of channels you need to broadcast. However, in general IPnetworks are much cheaper to operate than synchronous data networks likeISDN. There is a wide range of IP networks to choose from whenbroadcasting over IP and some of the factors that affect the selection of anetwork to broadcast over include:

Your program content: Are you performing a simple remote broadcastor distributing high bandwidth audio around a network, i.e. STL oraudio distribution.The number of audio channels you are sending: Do you need a simplepoint-to-point IP audio connection, are you multicasting, or do youneed to send multiple unicast IP audio streams to different studios?Your broadcasting region: Depending on where you are situated, youmay have access to different infrastructure like DSL/ADSL or fiber;similarly, you may have access to different wireless networks likeUMTS/HSDPA, EV-DO or WiMAX.Your budget: A community radio station may be looking for a costeffective hardware and data solution, whereas a large network may belooking to integrate flexible and high quality hardware with innovativesoftware management solutions.Contact Tieline to receive a spreadsheet that will tell you how muchdata your codec will consume per hour of broadcasting to help youdecide what plan to buy.

Data Plan Suggestions

1. Always use the best quality Internet Service Provider (ISP). Tier 1service providers are best as their infrastructure actually makes upthe internet ‘backbone’.



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2. You will get the best quality connection if both the local (studio) andremote codecs use the same Internet Service Provider. This cansubstantially increase reliability, audio bandwidth and reduce audiodelay. Using the same service provider nationally can give betterresults than using different local service providers. This is especiallytrue if one of the service providers is a cheap, low-end domesticservice provider, which buys its bandwidth from other ISPs. Secondand third tier providers sub-lease bandwidth from first tier providersand can result in connection reliability issues due to multiple switchhops. We also highly recommend using Tier 1 ISPs if connecting twocodecs in different countries.

3. Sign up for a business plan that provides better performance thandomestic or residential plans. Business plans typically have a fixeddata limit per month with an additional cost for data beyond that limit.In addition, Service Level Agreements (SLA) will often provide bettersupport and response times in the event of a connection failure.Domestic plans are often speed-limited or 'shaped' when usageexceeds a predefined limit. These plans are cheap but they aredangerous for streaming broadcast audio.

4. Ensure that the speed of the connection for both codecs is adequatefor the job. The minimum upload speed recommended is 256 Kbps fora studio codec and 64 Kbps for a field unit connection.

5. Use a managed IP network connection or a dedicated DSL/ADSL linefor your codecs. Do not share a connection with PCs or otherdevices. The only exception to this rule is if an organisation hasnetwork equipment and engineers that can implement and managequality of service (QoS) across its network.

How to Order the Right Plan for your Wireless IPService

There are many 3G data services offered by Telcos, e.g. UMTS/HSDPAand EV-DO Rev A. When using wireless data services choose reliableTelcos in your region that offer the highest bit-rates and therefore thebest opportunity for delivering stable high quality audio.



One of the most expensive mistakes you can make is borrowing a 3GSIM card for a broadcast that will last a couple of hours. It is likely thatthis type of 3G plan is optimized for voice and not IP data. Don’t find outthe hard way it could be an expensive mistake! We recommend youpurchase a plan that includes unlimited data for a fixed price per month.Then you can broadcast for as long as you need for a fixed price permonth. If this type of plan is not available, estimate the number of remotebroadcast minutes/hours you need per month and buy a plan thatbundles large blocks of data for one price. Some Telcos also offer‘timed’ or ‘minutes’ plans, which offer unlimited data for fixed amounts oftime.

Warnings: Some 3G network providers prohibit streamingmultimedia of any kind on certain accounts. Also, some planscharge very high rates for data, or may ‘throttle’ or ‘shape’ youravailable bandwidth after a certain amount of data has beentransferred. Check these factors with your Telco beforesubscribing to a plan.

Calculating Data Requirements and Costs

To calculate your total IP data requirements you need to:

Determine how many channels you are sending: is yourconnection mono, stereo, multicast or multiple unicast?Calculate the bit-rate requirement per channel; this will dependon the compression algorithm you select and need to includepacket overhead data requirements.

As a general rule of thumb, when connecting using UDP to send audioensure the total bit-rate (audio bit-rate plus header bit-rate) is no morethan 50% of the ISP connection rate. For example, with a 48 Kbpsaudio bit-rate when using the Tieline MusicPLUS algorithm, add 16Kbpsfor the packet overheads and multiply by 2 (48 + 16 x 2 = 128Kbps).

Once you have calculated the total connection bit-rate (64Kbps) andhow high the ISP connection bit-rate needs to be (128Kbps = twice theconnection bit-rate), you can shop around for the most suitable and



41

competitive data plan to suit your needs.

8.4 Redundancy ConsiderationsIn studio-to-transmitter link applications it is a good idea to have a strategyfor backing up your program audio in the event of hardware failure or the lossof an IP link. Some of the methods used by professional audio codecs toprotect against lost audio over a connection in critical broadcastapplications include:

Automatic silence detection.Dual-redundant power supplies for hardware.Fail-safe audio program backups using either on-board or externalaudio storage media.Failover to a second IP connection, or to an alternative audio transportlike POTS/PSTN.

The methods employed depend on the hardware being used and theconnections both supported by the codec, and available at the studio andtransmitter sites.

8.5 IP InteroperabilityIn the past, audio codec manufacturers have designed codecs that havelargely been incompatible with each other in many different situations due tothe use of:

Proprietary session data protocols (used to establish andmaintain codec connections)Different proprietary audio algorithms.Different control data.

As a result, universal compatibility between manufacturers was difficult toachieve. In the early stages of broadcast audio over IP development, Tielineand other partners in the Audio-via-IP Experts Group lobbied for standardsthat manufacturers should adhere to in order to make IP compatibilitybetween different brands a reality. As a result, the EBU has publishedstandards in EBU N/ACIP Tech 3326 that manufacturers should comply within order to deliver compatibility of their codec with other brands over IP.



All Tieline codecs are EBU N/ACIP Tech 3326 compatible over IPand the company is committed to developing new IP and 3GIPapplications that take advantage of emerging networkinfrastructures around the globe.

SIP (Session Initiation Protocol)

SIP is central to codec compatibility because it allows different devicesto communicate with each other and codecs need to be SIP-compatibleto comply with EBU N/ACIP Tech 3326.

There are two very distinct parts to a call when dialing over IP. The initialstage is the call setup stage and this is what SIP is used for. Thesecond stage is when data transference occurs and this is left to theother protocols used by a codec (i.e. using UDP to send audio data).SIP can also be used for other elements of a call but it is important toremember that SIP only defines the way in which a communicationsession between devices should be managed. It does not define thetype of communication session that is established.

SIP leverages on the use of web architectures like DNS, and SIPaddresses are similar in appearance to email addresses. A device usingSIP can dial another device’s SIP address to find its location. This taskis performed by SIP servers, which communicate between registeredSIP-compliant devices to set up a call.



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Mandatory Algorithms Decreed under EBU N/ACIP Tech3329 for Broadcasting Audio over IP

Mandatory algorithms decreed under EBU N/ACIP Tech 3326 includeG.711, G.722, MPEG Layer II and PCM (pulse code modulation) andmust be present in codecs for them to comply with the specification.

Optional algorithms include AAC-LD, AAC-HE v.2, Enhanced APT-X,AMR-WB+ and Dolby AC-3.

8.6 Checklist for IP ConnectionsConnection reliability can be improved through the use of:

IP connection management software (i.e. Tieline QoS PerformanceEngine for managing IP audio connections)Low bit-rate, low delay algorithms optimised for use over wireless IPnetworks (e.g. Tieline Music, Tieline MusicPLUS, AAC)

The following checklist can be used to further improve reliability whenconnecting over IP. Aim for a score of at least 8 out of 10 before going live.



Check Result

1 Connecting using a reputable Tier1 ISP that’s part ofInternet backbone.

2 The same ISP is being used for both codecconnections.

3 The ISP data plan is a Business Plan or equivalent.

4 The ISP connection speed is adequate (e.g. higherthan audio bit-rate plus packet overheads).

5 Equipment is high quality and suitable for mediastreaming.

6 The ISP connection speed has been tested.

7 The ISP connection is not shared with PCs or otherdevices.

8 UDP is being used as the audio transport protocol.

9 Only 50% of ISP connection uplink bandwidth is beingused.

10 There are no wireless connections being used.

Wireless Network Reliability

It is very difficult to guarantee connection quality when there is no wayof knowing how many people are sharing the same wireless connectionat any point in time. For example, wireless 3G broadband IPconnections can easily become congested and result in packet lossand audio drop-outs, particularly when using cell-phone connections atspecial events where thousands of people have mobile phones. This canresult in poor quality connections and audio drop-outs if cell-phone basestations are overloaded.

Wireless network reliability can be improved through the use ofdedicated portable WiMAX wireless links. Audio codecs should also becapable of using automated reconnection features to redial and IPconnection immediately if an IP connection is lost.



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8.7 Testing a NetworkFinally, there are a few very simple tools that you can use to test whether acodec can be reached over an IP network.

Ping the Codec

A ping test can be used to test whether it is possible to reach a codecor any device over an IP network. A ping test measures:

The round-trip time of packets.Any packet loss.

There are two types of ping tests:

1. Short test: sends 4 packets and delivers statistics. i. Point to the start menu on your PC and click once.ii. Use your mouse pointer to select Run.iii. Type CMD in the text box and click OK.iv. Type ping and the IP address of the codec you are pinging (i.e.

ping 192.168.0.159) and press the Enter key on your keyboard.v. The round trip time of the packets is displayed, as well as any

packet loss.

2. Long test: sends packets continuously until stopped.i. Point to the start menu on your PC and click once.ii. Use your mouse pointer to select Run.



iii. Type CMD in the text box and click OK.iv. Type ping, the IP address of the codec you are pinging, and

then -t (i.e. ping 203.36.205.163 -t) and press the Enter key onyour keyboard.

v. Let the test run for several minutes and then press CTRL C.vi. The round trip time of the packets is displayed, as well as any

packet loss for the period of time that the test occurred.

Use Telnet to Verify Ports

Telnet on your PC can be used to verify that the TCP ports are availableon the codec you are dialing. This lets you know that:

The codec is available to call.The port being used to send session data when connecting isopen.

The process for testing is similar to the ping test.

i. Point to the start menu on your PC and click once.ii. Use your mouse pointer to select Run.iii. Type CMD in the text box and click OK.iv. Type telnet, the IP address of the codec you are contacting,

and then the port number (i.e. telnet 203.36.205.163 9002) andpress the Enter key on your keyboard.

If the test is successful then a row of different characters are displayed.If it is unsuccessful an error message will be displayed saying that theport was not available.

Trace the Route of Packets

Another utility available on your PC is traceroute. This tool can be useto determine the route and number of hops that data packets are takingto their destination (codec). This is useful because the more routers thatpackets traverse, the more latency your connection will have, and theless reliable it will be.

i. Point to the start menu on your PC and click once.ii. Use your mouse pointer to select Run.



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iii. Type CMD in the text box and click OK.iv. Type tracert, the IP address of the codec you are contacting (i.

e. tracert 203.36.205.163) and press the Enter key on yourkeyboard.

8.8 Assessing Hardware RequirementsUltimately the hardware you require will be determined by the broadcast youwant to perform. DSP-based codecs are generally the most reliable over allIP connections and have greater stability compared to PC systems.

There is a large range of codecs available that are suitable for differentbroadcast situations. A sample of these products follows and they can allconnect over 3G wireless broadband networks, wired and wireless LANs,WANs, the internet, satellite IP, WiMAX and Wi-Fi..

Tieline’s Bridge-IT is a low-cost, high-performance, point-to-point or multi-pointstereo IP audio codec solution for broadcastand professional applications.

2 input analog or AES/EBU withsimultaneous analog and digital outputsIdeal for STL and audio distributionapplicationsIP Multicasting and multiple unicastingSimple remote broadcast linksMultiple codec installations (2 codecs fitin 1 x 19” rack unit)

The i-Mix G3 is an advanced IP codec forradio and television, combining six essentiallive remote broadcast products into onelightweight 16" x 9" box, replacing tens ofthousands of dollars of expensiveequipment.

A wireless-capable 6 input digital mixerwith a cross point digital matrix routerBi-directional audio & simultaneouscommunications circuits with 4



headphone controls/outputsOn-board PA output control with a built-intelephone hybridWired and wireless IP and POTS codecswith wireless 3G/3.5GIP, ISDN, X.21,GSM and Satellite Codec capabilityOn-board relays and RS-232 with fullstudio remote control

The wireless-capable Commander G3 is apowerful and reliable remote broadcast IPcodec.

3 input stereo mixer with 2 headphonecontrols/outputsConnect over wired IP or use twointerchangeable module slots to connectover wireless 3G/3.5G, POTS/PSTN,ISDN, X.21, GSM and B-GAN satellitenetworks.

On-board relays and RS-232 with fullstudio remote control

The 2RU Commander G3 is the ideal STLand audio distribution codec, or can be usedto receive IP audio from i-Mix, Commanderfield or Bridge-IT.

2 balanced XLR inputs with front and rearpanel headphone outputs and mic inputs. 4 front panel PPM meters displaying yourchoice of send, return or channel audiolevelsInsert an analog XLR 2 input/output cardor an AES/EBU XLR 2 input/output cardInternal or external AES/EBU clockAutomatic failover to any compatible audiotransport

The TLG3 GUI software controller emulatesthe hardware front panel of the 2RU



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Commander G3. It can be used to control1RU or 2RU codecs using USB or RS-232serial or LAN connections. This advancedsoftware GUI can control all normal codecfunctions such as dialing, menu navigation,audio monitoring and level controls

9 Glossary of Terms

AES/EBU Digital audio standard used to carry digital audio signalsbetween devices.

AES3 Official term for the audio standard referred to often as AES/EBU.

DNS The Domain Name System (DNS) is used to assign domainnames to IP addresses over the World-Wide Web.

Failover Method of switching to an alternative audio stream if theprimary connection is lost.

GUI Acronym for Graphic User Interface

ISP Internet Service Providers (ISPs) are companies that offercustomers access to the internet

IP Internet Protocol; used for sending data across packet-switched networks.

Latency Delay associated with IP networks and caused byalgorithmic, transport and buffering delays.

Multicast Efficient one to many streaming of IP audio using multicastIP addressing.

Narrowcast Transmitting a signal or data to a specific recipient orrecipients.

NetworkAddressTranslation(NAT)

A system for forwarding data packets to different private IPnetwork addresses that reside behind a single public IPaddress.

Packet A formatted unit of data carried over packet-switchednetworks.

Port AddressTranslation(PAT)

Related to NAT; a feature of a network device that allows IPpackets to be routed to specific ports of devicescommunicating between public and private IP networks.



QoS (Qualityof Service)

Priority given to different users or data flows acrossmanaged IP networks. This generally requires a ServiceLevel Agreement (SLA) with a Telco or ISP.

Redundancy Choosing an alternative audio stream to use if a primaryaudio connection is lost.

RTP A standardized packet format for sending audio and videodata streams and ensures consistency in the delivery orderof voice data packets.

SDP SDP (Session Description Protocol) defines the type ofaudio coding used within an RTP media stream. It workswith a number of other protocols to establishes a device’slocation, determines its availability, negotiates call featuresand participants and adjusts session management features.

SIP SIP (Session Initiation Protocol) works with a myriad of otherprotocols to establish connections with other devices. It isused to find call participants and devices and is the methodused by most broadcast codecs to connect to competingbrands of codec for interoperability.

SLA Service Level Agreements (SLAs) a contractual agreementbetween an ISP and a customer defining expectedperformance levels over a network

STL Studio to transmitter link for program audio feeds.

TCP TCP (Transmission Control Protocol) ensures reliable in-order delivery of data packets between a sender and areceiver. Its two functions include controlling thetransmission rate of data and ensuring reliable transmissionoccurs. Generally not well-suited to streaming live audiobecause buffering (latency) is employed to ensure datapackets are received in order

UDP UDP (User Datagram Protocol) most commonly used forsending internet audio and video streams. UDP packetsinclude information which allows them to travelindependently of previous or future packets in a data stream.In general, UDP is a much faster and more efficient methodof sending audio over IP.

Unicast Broadcasting of a single stream of data between two points.

Trademarks and Credit Notices


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10 Trademarks and Credit Notices1. Windows is a registered trademark of Microsoft Corporation in the United

States and/or other countries.2. Other product names mentioned within this document may be trademarks

or registered trademarks, or a trade name of their respective owner.

Disclaimer

Whilst every effort has been made to ensure the reliability and accuracyof the information contained in this guide, Tieline is not responsible forany errors or omissions within it, and the guide should not be reliedupon solely when designing, purchasing and installing broadcast IPnetworks. Always consult a qualified and experienced IP broadcastnetwork professional for advice or to undertake appropriate training priorto purchasing and installing equipment for use over IP networks.

11 Appendix 1: IP Protocols

Additional IP transport protocols that can affect sending audio over IPinclude the following:

RTP (Real-time Transport Protocol)RTP has been designed to transport real-time multimedia streams overIP networks. It is a standardized packet format for sending audio andvideo data streams and ensures consistency in the delivery order ofvoice data packets.

RTCP (RTP Control Protocol)

RTCP is a sister protocol of RTP and it gathers statistics and providesfeedback on the quality of a streaming media connection. The type ofinformation distributed includes packet counts, lost packet counts, jitterand round-trip delay times.

SDP (Session Description Protocol)

SDP defines the type of audio coding used within an RTP mediastream. It works with a number of other protocols to:



Establishes a codec’s location.Determines the availability of a codec.Negotiate the features to be used during a call, i.e. the algorithmand bit-rate.Provide call management of participants.Adjust session management features while a call is in progress(i.e. termination and transfer of calls etc).

RTSP (Real Time Streaming Protocol)

The Real Time Streaming Protocol is a control protocol used toestablish and control streaming media servers and is typically used inconjunction with RTP, which controls the transport of streaming dataitself.

SAP (Session Announcement Protocol)

SAP is an announcement protocol used to advertise multicast sessionsand communicate setup information to prospective broadcastparticipants.

SNMP (Simple Network Management Protocol)

This UDP-based network protocol is used primarily in networkmanagement systems to monitor devices attached to a network.

Index


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Index

- 3 -3G 16

- A -Algorithms 23

Audio distribution 9

- C -Credit notices 51

- D -Data Costs 38

Data Plans 38

Data Requirements 38

Disclaimer 51

- E -EBU N/ACIP Tech 3329 41

Error Concealment 25

- F -FEC 25

Forward Error Correction About 25

Conserving bandwidth 25

- G -Glossary 49

- I -Internet Broadcasting 35

Interoperability 41

Introduction 4

IP

10 reasons to broadcast over IP 8

Addresses 30

Background on IP networks 5

Connection types 11

Interoperability 41

IP versus POTS, ISDN and X.21 5

IP versus synchronous data 5

Jitter 28

Jitter Buffering 28

Latency 28

MANs/WANs/LANs 16

NAT 32

Network Address Translation 32

Network considerations 21

Network types 16

Planning Installation 34

Port Forwarding 32

Private and public networks 30

Redundancy 41

Regional factors 34

Testing connections 45

Transport protocols 21

What is IP 5

Wireless 3G and WiMAX 16

IP Addresses 30

IP Codecs 47

IP Hardware 47

IP LANs 16

IP MANs 16

IP Protocols 21Appendix 51

IP WANs 16

- J -Jitter 28

Jitter Buffering 28

Audio over IP Instant Expert Guide


54

- L -Latency 28

- M -Multicasting

About 11

Applications 11

Multiple Unicasts About 11

Applications 11

- N -NAT 32

Network Address Translation 32

Network Types 35

Networks Considerations 21

- P -Packet Loss 25

Planning Installation 34

Port Forwarding 32

Private IP Networks 30

Public IP Networks 30

- Q -Quality of Service (QoS) 35

- R -Redundancy 41

Reliability Checks 43

Remote broadcasts 9

RTCP 51

RTP 51

RTSP 51

- S -SAP 51

Satellite IP 16

SDP 51

SIP 21

SIP, how it works 41

SNMP 51

STLs 9

Studio to transmitter links 9

- T -TCP and UDP 21

Testing IP Networks 45

Trademarks 51

- U -Unicasting

About 11

Applications 11

- W -WiMAX 16

Wireless 3G 16

EV-DO 16

Satellite 16

UMTS/HSDPA/HSPA+ 16

WiMAX 16

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