Audio Over IP Instant Expert Guide v1

The Audio over IP
Instant Expert Guide
Version 1.1
January, 2010
Tieline Pty. Ltd. 2010

2 Audio over IP Instant Expert Guide
Table of Contents
Part I Introduction 4
1 What exactly
..............................................................................
is IP? 5
Part II 10 Great Reasons to Broadcast Audio over
IP 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 Audio over
..............................................................................
IP Transport Protocols 21
2 Choosing
..............................................................................
an Algorithm 23
3 Concealing
..............................................................................
Packet Loss 25
4 Managing
..............................................................................
Jitter (Latency) 28
Part VII Dialing over IP Networks 30
1 NAT and..............................................................................
Port Forwarding 32
Part VIII Planning IP Network Installation 34
1 Regional..............................................................................
Factors Affecting IP Connectivity 34
2 IP Network
..............................................................................
Suitability and Reliability 35
3 Selecting
..............................................................................
a Data Plan 38
4 Redundancy
..............................................................................
Considerations 41
5 IP Interoperability
.............................................................................. 41
6 Checklist..............................................................................
for IP Connections 43
7 Testing a..............................................................................
Network 45
8 Assessing
..............................................................................
Hardware Requirements 47
Part IX Glossary of Terms 49
Part X Trademarks and Credit Notices 51

Contents 3
Part XI Appendix 1: IP Protocols 51

1 Introduction
Audio-over-IP has proved itself to be the broadcast network infrastructure for
today and into the future. As a consequence, increasing numbers of
broadcasters are migrating to low-cost wired and wireless IP networks from
more costly analog leased line, microwave and synchronous data
technologies 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 a
result, broadcasters are using IP audio codecs to design and operate more
adaptable broadcast networks with streamlined work flows, reduced
operating costs and the ability to remote control them from anywhere in the
world.
For many years Tieline Technology has recognised that the future of
broadcasting is in packet-switched networks supporting audio over IP, and
as a member of the Audio-via-IP Experts Group, Tieline has been at the
forefront of determining the direction of broadcasting audio over IP. Tieline
has assisted thousands of broadcasters to seamlessly transition audio
distribution, studio-to-transmitter link and remote broadcast infrastructure
into IP technologies.
The information in this guide is useful to users of all brands of audio codecs
and is supplemented by more detailed information in Tieline's IP and 3GIP
Streaming Reference Manual, which is available for download at www.tieline.
com/transports/Audio-over-IP. You can also contact Tieline support at
support@tieline.com to find out more if you have any further questions or
requests.
How to use this Guide

The Audio over IP Instant Expert Guide is an invaluable resource for
broadcasters new to IP and is a useful reference tool for those
broadcasters 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.

Introduction 5
None of this is true, and after reading this guide broadcasters should
feel confident that they have sufficient knowledge to configure, run and
monitor broadcast audio connections over IP. The guide provides
information 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 data
networks.
3. An overview of how audio over IP can be used in different
applications 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 of

broadcasters who interact regularly to discuss topics relating to
broadcasting 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 a
computer would over a Local Area Network (LAN) or Wide Area Network
(WAN). A LAN is a network covering a small local area and a WAN
covers a much wider area, e.g. the internet. LANs and WANs can be
wired or wireless.
Some networks like wireless WiMAX networks are called Metropolitan

Area Networks (MANs) and typically these cover a city. A MAN is larger
than a LAN but smaller than a WAN. There are a plethora of wired and
wireless IP networks that interconnect with each other and can be used
to broadcast high quality audio.
What is IP?
IP stands for Internet Protocol, which is a protocol used to send data
across packet-switched networks. Packet-switching is used by

computer networks and telecommunications devices (e.g. IP audio

codecs and 3G cell-phones). Data packets are individually routed
between two devices over Local Area Networks (LANs) or Wide Area
Networks (WANs).
What do you need to send high quality audio over IP?

Using IP you can make connections between two IP audio codecs, or
between an IP codec and other compatible devices connected to small
private LANs or large public WANs like the internet. These codecs and
devices can connect using hard-wired Ethernet connections (like those
used by a PC to connect to the internet), wireless connections, or a
combination of hard-wired and wireless connections.
Example of IP Codecs using Wired and Wireless IP Network

Connections
Wireless IP connections can be made over 3G and 4G wireless cell

phone networks, public or private WiMAX wireless IP networks and
BGAN satellite connections.
What are the Differences between IP and other

Synchronous Digital Data Connections?
Circuit switching, used in synchronous digital data networks like ISDN
and X.21 and wireless GSM CSD and HSCSD networks, creates a
dedicated connection between two end points in order to send data
packets exclusively between two devices.

Introduction 7
Packet-switched networks are more efficient and optimize the use of

bandwidth over computer and wireless networks by dividing data
streams into packets with destination addresses embedded within
them. In this way packets can travel through different routers to their
destination in order to find the fastest way to their destinations.
IP versus ISDN, POTS and X.21 Networks - An Historical

Perspective
In the past synchronous data networks have been preferred for studio-to-
transmitter links (STLs) and audio distribution within broadcast networks
because of their guaranteed data rates and reliability, commonly
referred to as QoS, or Quality of Service.
IP came along with the promise of more efficient use of bandwidth over
computer and wireless networks, but this came at a cost - well two
costs to be exact. The two key factors you need to understand to
manage network reliability are network 'jitter' and packet loss. Jitter
relates to the amount of time required for an audio codec to receive all
the data packets sent to it, then reorder them and play them out in
sequence and reliably stream audio without any audio interruption.
Packet loss relates to data packets sent from one codec to another that
are lost. Lost packets can potentially cause 'artifacts' or glitches in
quality when streaming audio, unless you have the right equipment to
manage it. We will discuss these factors in detail later, but the key
thing to remember is that software developments and improvements to
broadband network infrastructure have mitigated the effects of jitter and
packet loss to a large extent in most situations.
Despite the potential pitfalls of IP, broadcasters are moving inexorably

towards the technology because of the cost advantages and flexibility.
The transition into IP network infrastructure is also gathering pace as
older analog and digital synchronous networks like ISDN are phased out
and shut down.

2 10 Great Reasons to Broadcast

Audio over IP
1. Broadcasting over IP is cost-effective
IP network infrastructure is cheaper because you can distribute
broadcast quality audio over existing broadband networks such as
DSL/ADSL
IP network broadband costs are generally much cheaper than analog
leased 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 to
multiple points, so less hardware is required than over traditional
synchronous 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 audio
codec can deliver a choice of unicast, multicast and multiple unicast
IP streams for network audio distribution
4. IP networks can be scaled to suit individual installations

Broadband Internet Service Providers and Telcos offer a range of
competitively priced data plans that provide flexible connection
bandwidth to suit each installation - minimising data costs and
maximising network efficiency
It is possible to incrementally increase available network data
bandwidth as demands increase over time
5. Wireless IP networks deliver flexible broadcast connections from

anywhere 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 of
the world

10 Great Reasons to Broadcast Audio over IP 9
Wired broadband connections are widely available in most regions of

the 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 cumbersome
microwave links
A single codec can be preprogrammed to connect to a wired or
wireless broadband network very simply
8. Audio over IP integrates seamlessly with new broadcast

technologies
Packet-based audio over IP integrates seamlessly when broadcasting
audio streams over the internet and a wide range of digital radio
formats
9. Integration of audio over IP into large radio networks creates

economies of scale
Opportunities to consolidate and centralise the distribution of audio
around radio networks and affiliates is facilitated by the flexible and
scalable nature of IP codec hardware and broadband network
infrastructure
10. Audio over IP is the future of broadcasting

Major networks around the world are migrating to IP
Analog leased line and synchronous network infrastructure like ISDN
is being phased out in most regions of the world
Audio over IP has the flexibility to adapt to meet the changing needs
of technology
Regular DSL/ADSL data plans are sufficient to deliver 22kHz audio
quality 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 remote
broadcast 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 the
transmitter 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 11
Audio Distribution
With the advent of digital radio broadcasting there has been exponential
growth in audio distribution using IP. Multichannel digital radio has
opened the door to new networking and narrowcast opportunities for
radio networks and IP audio distribution delivers a cost-effective and
flexible 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 other
locations.
4 Types of IP Connections
IP offers the ability to create much more flexible broadcast networks for a
much lower investment than traditional analog and synchronous digital
networks. Next we outline the three basic audio codec application concepts
important to understanding the capability of broadcasting audio over IP -
unicasting, multicasting and multiple unicasting.
IP Information: An IP address is a unique number that allows

devices to communicate between each other over IP networks using
the 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 the
sending of data packets to a single end point or node. A similar
principal is employed in audio over IP broadcasting and a unicast
connection is a one-to-one connection between transmit and receive
audio codecs.
Unicast Applications for Broadcasters

Unicasting over IP provides full-featured connections with high
quality bidirectional stereo audio capabilities, as well as full duplex
communications. 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 stream
to many recipients. In some ways it is a lot like traditional radio
broadcasting where you transmit a single signal over a wide area and
anyone with a radio can tune in. When multicasting, the audio stream
sent from the transmitting codec is distributed over the IP network to
other codecs and only a minimal amount of bandwidth is required to

Types of IP Connections 13
transmit the original program audio.
Multicast routers over the IP network replicate packets on demand as

required. They are then forwarded to the group of codecs that has
expressed an interest in receiving the transmissions.
Multicast Applications for Broadcasters

Multicasting is an effective way to distribute audio to many
locations with minimal IP configuration. It does not require a large
amount of bandwidth at the codec transmitting the broadcast audio
and it is particularly useful to broadcasters over private LANs that
support multicast audio distribution within a network. Multicasts
are ideal for:
Distributing high quality audio over broadcast LANs.

Distributing audio to multiple zones within a broadcast or
non-broadcast network.
Distributing broadcast quality audio throughout
environments like large buildings, airports, hotels and retail
outlets.
Multicasting can also be a good way to set up permanent STL

connections to affiliate codecs across LANs that support multicast
connections.
Multicast Broadcast Example

Key Multicasting Concepts

Like broadcasting generally, with multicasting it is not necessary
for the transmitting codec to know all the recipients of a
transmission.
Multicast transmissions are sent using a dedicated IP multicast

address that looks similar to a regular IP address and multicast
subscribers request transmissions from this address. This unique
address allows multicast routers to identify multicast requests from
a group of codecs interested in a particular transmission and
packets are replicated depending on demand. This can create
large demands on network bandwidth if the multicast group is
significant in size.
Only small sections of the internet are multicast enabled and many
Internet Service Providers (ISPs) block multicast traffic over wide
area networks like the public internet. This restricts most multicast
broadcasts to private local area networks.
Some ISPs provide quality of service (QoS) priority to multicast

streams for an increased service charge. Some also offer QoS to
broadcasters if the broadcast transmissions are delivered as a
service to the ISPs subscribers.
The important multicast concepts to remember are:
Multicast streams are not automatically allowed over WANs

and are usually difficult and more expensive over these
networks.
The network path must include multicast-enabled routers
and switches.
Bandwidth required at the transmitting codec is minimal.
The total bandwidth of all transmissions over a network can
be significant if the multicast group is large.
Codec streams are unidirectional (receive only) for the
multicast group subscribed to a broadcast.

Types of IP Connections 15
What are Multiple Unicasts?

Multiple unicasting (multi-unicasting) technologies expand the concept
of unicasting by creating multiple connections from one broadcast
codec to a specific selection of other codecs. The transmitting codec
must specify exactly which codecs will receive individual audio streams
and dial them directly. This differs from multicasting, where the
transmitting codec sends a single stream into the network and the
network replicates the streams.
Multiple unicasts can be performed over either LANs or WANs and are
most suited to broadcasting over the internet when compared with
multicasting. Multiple unicasting is limited only by the number of
connections the codec is able to dial and the bandwidth available at the
transmitting codec.
The total bandwidth of each connection is the bandwidth required to

successfully broadcast all the individual IP streams. For example, if you
create ten 100Kbps connections, you will need at least 1Mbps of
bandwidth capacity for program content at the codec broadcasting the
multiple unicast audio streams.
Multiple Unicast Applications for Broadcasters

Multiple unicast technologies provide broadcasters with
opportunities to deliver high quality broadcast audio streams to
multiple codecs from a single codec. Compared to multicasting,
unicast streams are more capable of traversing wide area networks
like the internet and are more secure. Multiple unicasts are ideal
for:
Distributing multiple streams of program audio to radio

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

location. As long as the network connection sending the audio has

the bandwidth required, connections can be made over WANs
quite 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 or

LANs quite simply by dialing each individual connection.
Bandwidth required at the transmitting codec is directly
proportional to the number of connections being used.
Different codecs have different multi-unicast capabilities and
some can provide a return signal path for confidence
monitoring of audio.
5 Selecting a Network
IP networks come in various shapes and sizes and the network that is most
suitable for your requirements depends on your broadcast application (e.g.
remote broadcast, audio distribution or STL). In this section we explain the
different types of networks and suggest which ones can be used to perform
studio-to-transmitter links, audio distribution and remote broadcasts.

Selecting a Network 17
Wired LANs/WANs/MANs
Ethernet connections to LANs, WANs, MANs are used extensively for
wired IP connections over local, metropolitan and wide area networks.
Wired networks are capable of high data transfer rates and are more
reliable 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 network
infrastructure available over private LANs, higher data rates may provide
the 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, including
multicast and multiple unicast applications.
IP audio distribution across broadcast networks, including
multicast 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 environmental
factors. The data bandwidth provided by 3G wireless broadband
networks is often sufficient to send up to two channels of high quality
20kHz audio. Wireless 3G networks provide low-delay connections with
typical latency of around 100 to 200 milliseconds.
Wireless 3G is the Ideal Solution for:

Wireless remote broadcasts from wherever a 3G signal is
available.
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 Mobile
Telecommunications System), HSDPA (High-Speed Downlink
Packet 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 UTMS

technology to provide higher data uplink and downlink bit-rates than
traditional W-CDMA. Maximum network download speeds of up to
14.4Mbps and upload speeds of up to 384Kbps can be achieved
over HSDPA networks. HSPA+ provides even higher data rates of
up to 42 Mbit/s on the downlink and 22 Mbit/s on the uplink.
These networks are the most suitable for streaming audio over IP
and are typically found in Europe, the Middle East, Africa and
Australia (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 - 1000Kbps
on the downlink and 50 - 100Kbps on the uplink. EVDO Rev A
delivers 600Kbps - 1,400Kbps downlink and 500Kbps-800Kbps
uplink. 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 streaming
because the bit-rates are too low for continuous live streaming.

Wireless 4G WiMAX Networks

It is possible to broadcast over either public WiMAX Metropolitan Area
Networks (MANs) or portable point-to-point and multipoint WiMAX
configurations. WiMAX is short for Worldwide Interoperability for
Microwave Access and WiMAX IP links effectively create reliable, high
speed, long range broadband IP connections at up to 70 megabits per
second between two points or multiple points.
WiMAX operates using the IEEE 802.16 wireless standard and it has
been developed primarily for medium to long-range outdoor transmission
hops. WiMAX is more efficient than Wi-Fi connections and it has higher
data rates and a greater range.
WiMAX is the ideal solution for:

Studio-to-transmitter links in remote locations where wired or
wireless telecommunications infrastructure is unavailable.
Remote broadcasting where 3G networks are unavailable, or
where large amounts of data bandwidth are required.
Audio distribution within regions where good line-of-sight can be
achieved over long distances.
Dedicated Private WiMAX Networks

Instead of leasing a dedicated link from a Telco it is possible to
create your own private long-range LAN. Portable low-cost WiMAX
systems deliver dedicated full-duplex, high-speed data connections
between two points or between the studio and multiple remote
locations, providing cost-effective bi-directional transmission paths
for audio distribution, remote broadcasting or studio-to-transmitter
links. Once these systems have been purchased there are no
ongoing data costs.
Portable systems generally consist of a base station and a

receiver that can operate at distances of between 2km and 100km,
depending on the line of sight available, the antenna arrangement
used and whether repeaters are added. Portable WiMAX links are
ideal for roof-top or rural deployments because of their small size
and low power requirements. They can operate in unlicensed RF
bands 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 the
globe. These 4G wireless broadband networks provide high-speed
data connections for broadcasting high quality audio from within
large MANs. Visit http://www.wimaxmaps.org/ to view global
deployments of WiMAX networks.
A Typical Metropolitan WiMAX Network Broadcast
Satellite IP
Satellite IP connections are a dependable way to send broadcast audio
to the studio from very remote locations where other wireless network
infrastructure is unavailable. Using a BGAN satellite terminal it is
possible to send one or two channels of studio FM quality audio from a
remote location.
Satellite IP is the ideal solution for:

Broadcasts from very remote locations where 3G wireless or
wired IP connections are not available.

A Typical Satellite IP Broadcast
6 Important IP Network
Considerations
Packet switching optimizes the use of bandwidth over computer and
wireless networks by dividing data streams into packets with destination
addresses embedded within them. In this way packets are routed through
ISP routing tables to find the best route to their destinations.
The exact form of a packet is determined by the protocol (see Audio over IP
Transport Protocols) a network is using and this affects the actual size of
the packet. Packets are generally split into three parts which include:
A Header: This section contains instructions about the data contained

within the packet;
The Payload: This contains the actual data that is being sent to the
destination; and
A Trailer (Footer): This tells the receiving device that it has received
the 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 packets
Provide statistics and feedback about IP streams
Establish connections.

TCP versus UDP

TCP (Transmission Control Protocol) is an internet transport protocol
most commonly used for many of the internets applications such as
email and the World Wide Web and it is what most codecs use for
establishing a connection. The TCP protocol ensures reliable in-order
delivery of data packets between a sender and a receiver. Its two
functions include controlling the transmission rate of data and ensuring
reliable transmission occurs. TCP is generally not well-suited to
streaming live audio. Broadcasting audio packets over TCP connections
will typically deliver more latency than UDP connections. This is
because more buffering is employed to ensure data packets are
received in order.
UDP (User Datagram Protocol) is the protocol used most commonly for
sending internet audio and video streams and the European
Broadcasting Union (EBU) standard for audio over IP recommends using
RTP over UDP rather than TCP. The UDP protocol is different to the
TCP protocol in that it sends datagram packets. These packets include
information which allows them to travel independently of previous or
future packets in a data stream. In general, UDP is a much faster and
more efficient method of sending audio over IP and RTP over UDP
sometimes has a higher priority than TCP in internet and network
routers. Tieline has written special Forward Error Correction software
(FEC) for UDP data streams, which significantly increases the stability
of a connection over IP.
SIP (Session Initiation Protocol)

SIP is a signaling protocol used to connect, monitor and disconnect a
myriad of different connections over the internet such as telephone
calls, conferencing and multimedia distribution. It provides multi-user/
device sessions and connections without regard for the particular device
or the media content that is delivered and is the protocol, along with
SDP, used to provide codec compatibility and interoperability according
to EBU N/ACIP Tech 3326 (the audio over IP standard used for providing
compatibility between different brands of codecs).
SIP works with a myriad of other protocols to establish connections with

Important IP Network Considerations 23
other devices over the internet and carries SDP messages. It is used to
find call participants and devices even when they move from place-to-
place and is the method used by most broadcast codecs to connect to
competing brands of codec for interoperability. SIP and SDP combine to
negotiate 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 Algorithm

Before you send audio over your IP network you need to select whether you
will be sending the data uncompressed or compressed. To send
uncompressed data requires very high rates of data, therefore it is better
suited to a private LAN or WAN. In most situations you will need to select a
compression algorithm.
Most audio codecs allow you to select your preferred compression algorithm
using software menus. The algorithm you select will depend on how much
bandwidth you have available and it will affect not only the quality of the
broadcast, but also contribute to the amount of latency or delay introduced.
For example, if MPEG Layer 2 algorithms are used, program delays will be
much longer than when using Tieline Music, MusicPLUS, aptX or AAC
algorithms. This is due to the additional inherent encoding delays involved
when using MP2 algorithms. This can be a major consideration for live
applications 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 or
multiple unicast connections.
Whether you are connecting using SIP or not (some algorithms are
not 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 each
algorithm and to see how it sounds when it is sent at different connection
bit-rates (as well as different FEC and jitter-buffer millisecond settings). This
will assist you to determine what the best algorithm is for the connection

you are setting up.
Algor- Audio Algor- IP bit-rate IP over- Recommended

ithm Band- ithmic per head connection for on-
width Delay channel air use
Linear 16/24 0ms sample 80Kbps Extremely high
(Uncom- bit up rate x bits quality uncompressed
pressed) to per sample audio distribution and
96kHz x no. STLs
channels
Tieline Up to 20ms 24 Kbps 16Kbps
High quality low bit-
Music 15kHz (minimum) rate remotes, STLs
and audio distribution
Tieline Up to 20ms 48 Kbps 16Kbps Very high quality low
Music- 22kHz (minimum) bit-rate remotes,
PLUS STLs and audio
distribution
G.711 3kHz 1ms 64Kbps 80Kbps Voice quality
(minimum) connections to other
brands of audio codec
G.722 7kHz 1ms 64Kbps 80Kbps Voice quality
(minimum) connections to other
brands of audio codec
MPEG Up to 24 to 64Kbps 8.5 - Very high quality
Layer 2 22kHz 36ms (minimum) 13.3Kbp audio connections
s between Tieline or
other brands of
codec.
MPEG Up to 100ms 64Kbps High quality low bit-
Layer 3 15kHz rate remotes, STLs
AAC-LC Up to 64ms 64Kbps 15Kbps High quality low bit-
15kHz rate remotes, STLs
AAC-HE v.1 Up to 128ms 32-48Kbps 7.4Kbps High quality low bit-
15kHz rate remotes, STLs
AAC-HE v.2 Up to 128ms 16-24Kbps 7.4Kbps DAB+ radio
15kHz streaming and high

quality low bit-rate

remotes, STLs and
audio distribution
aptX 24kHz 2ms 384Kbps STLs and audio
Enhanced (Stereo) distribution
6.3 Concealing Packet Loss

When broadcasting using audio over IP it is critical that the codec you use
has 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 on

the number and size of the packets lost during transmission and
reception. Audio artifacts become evident if many packets are lost or if
large packets containing a lot of data are lost. IP codecs can detect the
integrity of every packet because the UDP and TCP protocols used in IP
data packets verify the integrity of every packet received by a device.
If you select broadcast audio codecs that provide packet delivery

statistics then you will be able to assess network congestion and
packet delivery reliability. This allows you to reliably adjust your
connection bandwidth or other settings like the jitter buffer or Forward
Error Correction (FEC) to maximise connection stability. This may
sound complicated but in practice it is quite simple to do.
Concealment
Network protocols like TCP provide for reliable delivery of packets by
asking for retransmission of lost packets. This can be inefficient and
lead to the connection bit-rate being higher than expected if many
packets are lost. Packet loss concealment can also be used to mask

the effects of lost or discarded packets during an IP broadcast. Loss

concealment methods include:
Reproduction of the packet received prior to the lost packet.

Estimation of the value of each dropped packet by interpolation
and insertion of these artificially generated packets into the bit-
stream.
These methods can be useful in disguising a few dropped packets here

and there, but if several packets are lost in a row audio quality will
become noticeably impaired.
Forward Error Correction (FEC)

Forward Error Correction (FEC) is a method designed to increase the
stability of UDP/IP connections. FEC works by sending a secondary
stream of audio packets so that if your primary audio packets are lost or
corrupted, then packets from the secondary stream can be substituted
to correct the primary stream.
The amount of FEC that you require will depend on how many data
packets are being lost over the network connection and it can only be
used over networks where bandwidth congestion is not an issue. Well
designed codecs let you to manually adjust the FEC setting using
software.
A high quality broadcast codec should provide statistics that allow you
to view how many packets are being lost over the network. This let's you
gauge the amount of FEC that you require to maximise connection
quality and stability. For example, if you are losing one packet in every
five that is sent, and you have a FEC setting of 20%, the lost packets
will be replaced by FEC to maintain the quality of the connection. If you
are losing more packets than this, say one in three, it will be necessary
to 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 to
achieve full redundancy and depending on link conditions, this could
cause more dropouts because of network congestion than it fixes. Here

is a simple rule to remember: Your maximum uplink speed is all the

bandwidth you have to play with. As a rule of thumb, try not to exceed
more than 50% of your maximum bandwidth. If your link is shared, be
even more conservative.
You should also consider the remote end too. What is the remote
codec's maximum upload speed? Is the connection shared at either
end? Your bit-rates, FEC settings and buffer rates must be pre-
configured at both ends before you connect, so it's always better to set
your connection speed and balance your FEC according to the available
uplink bandwidth at each end for best performance.
Conserving Bandwidth with FEC

There is a trade-off between the quality and the reliability of an IP
connection particularly when FEC is activated on your codecs.
However, it is possible in certain situations to set different FEC on each
codec to match connection bandwidth requirements at either end of the
link, conserve bandwidth and create more stable IP connections.
For example, if your broadcast is a one-way broadcast from a remote

site, i.e. you are not using the return path from the studio, or only using
it for communications purposes, it is possible to reduce or turn off FEC
at the studio codec. This effectively reduces the bandwidth required over
the return link (communications channel) and increases the overall
bandwidth available for the incoming broadcast signal from the remote
site. This could be particularly useful if you have limited uplink
bandwidth at the remote location.
Keep in mind that as you move from local to national to international

connections, you should be more conservative with your FEC and
connection bit-rates. As a general recommendation, choose a codec
that shows you how much data you are using per second in a
connection and never exceed 50 percent of your total upload bandwidth
at 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 takes
for a packet of data to get from one point to another. Over packet-
switched networks delay is variable, depending on the path packets
take from their source to their destination. Latency is an important issue
when using packet-switched networks particularly when broadcasting
audio in live situations. Latency over packet-switched networks is
created by:
Network transmission delay.

Physical processing delay over the network via switchers and
routers etc.
Packet delay, including algorithm compression delays.
Packet jitter occurs when data packets sent over a network do not arrive
in regular intervals. This occurs because packets can travel over any
route to their destination despite being sent in regular time intervals.
The random delays that occur, and the severity and frequency of these
delays, will be different for every connection. The combination of factors
contributing to the total latency over a network mean that a temporary
buffer is required to ensure reliable play-out of audio streams when
broadcasting.
What is a Jitter Buffer?

A jitter buffer is a temporary storage buffer in codec software used to
capture incoming data packets to ensure the continuity of audio
streams is maintained. Data packets travel independently and arrival
times can vary greatly depending on network congestion and the type of
network used, i.e. LAN versus wireless networks.
In a way, a jitter-buffer can be looked upon as a pre-programmed delay

insurance for packets not turning up in time. The trade-off, or cost of
increasing the jitter-buffer is increased latency in the overall connection.
The greater the jitter-buffer delay programmed, the greater the program
delay. Packets are retrieved from the jitter buffer at regular intervals by a
devices decoder in order to provide a smooth and regular play-out of

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

following image.
If a jitter buffer delay setting is not high enough then it is likely that
interruptions to streams will occur as a result of late packets. If the time
value or depth of the jitter buffer is set at a point larger than the longest
experienced jitter delay, then all packets received by a device will be
delivered to the decoder and the best possible audio quality is
recreated.
Unfortunately there are two problems with this scenario:
1. There is no way to predict for sure what the longest jitter delay
will be, and
2. The larger a jitter buffer is (to increase the chance of catching all
late packets) the longer the end-to-end and round trip delay of
data becomes. (In extreme circumstances this can become
unacceptable for bidirectional audio applications that need low
delay)
Tieline has developed an automated jitter buffer solution that analyzes

the history of observed jitter over a connection and then set the jitter
buffer depth automatically based on this result. This is dynamically
adjusted over time automatically and compensates for observed network
congestion. Packet delivery statistics are provided that allow you to
optimise the jitter buffer setting on your codec to accurately suit
prevailing IP network conditions.

7 Dialing over IP Networks

Private versus Public IP Networks
Public IP networks are operated by the various Telcos and Internet
Service Providers and they provide a range of different data transmission
services for businesses and the public. These Telco networks generally
provide connections to WANs like the internet as well as MANs and
LANs and they facilitate sending data between users from a wide variety
of different networks. Customers can subscribe to various different data
plans tailored to suit their individual requirements.
Examples of private IP networks include company Intranet and Wiki

services, portable WiMAX systems and private home computer
networks that are not accessible to users outside of the network. Public
networks provide interconnections between these private networks via
the internet.
Public versus Private IP Addresses

An IP address is a unique number that allows devices to communicate
over networks and the internet using the Internet Protocol standard.
There are two types of IP addresses public and private and these
addresses can be static (fixed) or dynamic (assigned from a pool of IP
addresses). As examples, a private IP address might look like
192.168.0.100, and a public address might look like 203.36.205.133 or
74.76.21.72.
Certain IP address ranges have been allocated for private use and these
private addresses help to create secure private networks. Private
addresses can be used by anyone on a private LAN but computers or
devices using these numbers are unable to connect directly over the
internet without using Network Address Translation (NAT) and a public
IP address.
Conceptually, public and private IP addresses operate similarly to public

phone numbers and private phone extensions because an IP number
can be public or private. For example, a standard PBX telephone
system allows people to call you on a single public telephone number
and performs the translation and routing of the public number into a
particular private PBX extension. Private and public IP addresses

Dialing over IP Networks 31
operate in a similar way to private and public phone numbers - so

similar dialing principles apply.
If you want to dial a codec with a private IP address you will require
Network Address Translation (NAT). NAT allows a single device, such
as a broadband router, to act as an agent between the public internet
and a local private LAN. Usually this will be set up at the studio end so
you can dial into the studio from the remote codec.
You can think of NAT as if it was the receptionist in an office. When

someone calls the main office number looking for you, the receptionist
looks up your number and routes the call to your private extension. NAT
works in the same way by forwarding data packets to codecs with
private IP addresses.
Don't get too hung up on IP addresses and NAT because although it

may seem confusing at first, it is really quite straightforward to program
with some simple instructions and your IT administrator can assist you
with this sort of programming. Following is a table describing the
different types of IP addresses you may encounter and how they impact
on broadcasting over IP.
Type of IP How the IP Description

Address Address is
Allocated
Public Static Internet ISPs allocate a static public IP
Public Service address to allow network devices to
IP Address Providers communicate with each other over the
(ISPs) internet. It works like a public
telephone number and will allow your
remote codec to call your studio codec
over the internet.

Dynamicall Internet ISPs usually allocate dynamically

y Assigned Service (automatically) assigned public IP
Public IP Providers addresses to allow network devices to
Address (ISPs) communicate with each other over the
internet. (Not recommended for studio
installations because each time you
connect to your ISP the IP address can
change).
Private Dynamic- DHCP Server A DHCP server-allocated IP address
ally from your that is automatically assigned to a
Assigned own private device on a LAN to allow it to
Private IP LAN network. communicate with other devices and
Address the internet. This address can change
each time a device connects.
Static LAN A network administrator-allocated
Private Administra static address which is programmed
IP Address -tor into a device to allow it to connect to a
LAN. Often a security measure to only
allow access to devices approved by a
network administrator.
7.1 NAT and Port Forwarding

We have mentioned how Network Address Translation (NAT) is used to
connect codecs with private IP addresses with devices that have public IP
addresses. Computers and other devices that connect over IP also have
software ports that are used to sort different types of network traffic.
In TCP and UDP IP networks the codec port is the endpoint of your
connection. Software network ports are in a sense doorways for systems to
communicate with each other. For example, several codecs in your studio
may use the same public static IP address. Therefore it is necessary to
allocate port numbers to these codecs so that when an incoming call
comes in, the network knows which codec to send the call to.
Picture a house and imagine the front door is the entry point represented by
an IP address. You want to get to several codecs in different rooms of the
same house and the doors to each of those rooms are represented by
different port numbers. In principle this is how port addressing works. When
a studio with a designated public IP address receives data from several

Dialing over IP Networks 33
different remote codecs, port addressing information is extracted from the

incoming data packets to ensure the correct packets are sent to the right
studio 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 these
principles.
Managing Port Forwarding

By default Tieline codecs use a TCP session port (9002 or 9012) to
send 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 the
session port always uses the TCP protocol is that TCP is the most
likely 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 manufacturers

use UDP port 5060 to send session data and UDP port 5004 is used to
send audio.
Codec manufacturers let you program port forwarding using software

applications. The following example shows Tieline's web-GUI codec
programming application with default TCP 9002 session port and UDP
9000 audio port settings for an IP connection.
If there is a need to change your codec's port settings, in most

situations you should consult your organizations resident IT
professional and they can assist you with this over your network.

8 Planning IP Network Installation

There are several factors that you need to consider when choosing the
equipment that is most appropriate for your requirements. The typical
questions you may face include:
1. Network Reliability: how reliable are connections over the IP network

that I want to broadcast over?
2. Which IP network is most suitable for remote broadcasting, STLs and
audio distribution.
3. Data costs: what is my return on investment for IP broadcasting
compared to traditional leased line networks like ISDN?
4. What codec and algorithm will I use to broadcast and what sort of
data 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 IP

Connectivity
Connection reliability will vary from region to region and country to country.
However, as a rule of thumb, it is possible to apply some general
assumptions about local, national and international IP connections. The
following information is a guide only, because networks are always being
upgraded and depending on the network you are connecting to you can
achieve great results over local, national and international connections.
A local IP connection will usually:
Route data using the same service provider

Achieve higher bit-rates and better quality audio connections
Require low rates of FEC or none at all
Require low jitter buffer delays
Be most reliable
A national IP connection will usually:
Require data to be routed through more internet router points

Achieve good bit-rates and good quality audio connections

Planning IP Network Installation 35
Require low to medium rates of FEC

Require low to medium jitter buffer delay settings
Be reliable
An international IP connection will usually:
Require data to be routed through many internet router points and

many service providers
Achieve lower bit-rates and hence lower quality audio connections
Require medium to high rates of FEC
Require the highest jitter buffer delay setting
Be less reliable
An awareness of these factors when you are setting up your IP connection

will assist you to configure each IP connection successfully, and obtain the
best performance.
8.2 IP Network Suitability and Reliability

Other factors that affect the stability of an IP connection include whether it
is:
Over the public internet or a managed IP network with QoS (Quality of

Service)
A wired or wireless connection.
Shared with other devices like computers.
Whenever possible use wired IP connections that are not being shared with
other devices.
Quality of Service (QoS) Networks

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

SLAs are normally associated with dedicated 24/7studio-to-transmitter

links and audio distribution requiring guaranteed reliability. They are
more expensive than standard DSL/ADSL connections but usually less
costly than synchronous data links like ISDN.
Broadcasting over the Internet

Advances in codec technology have led to audio codecs being used
widely over the public internet for remote broadcasts, STLs and audio
distribution. Part of the problem with broadcasting over the internet is
the unpredictability of how congested the network will be at any point in
time. To a large extent these factors can be dealt with by software like
Tieline's QoS Performance Engine software, which automatically adapts
to the prevailing conditions of the internet and adjusts automatically to
compensate for increases in packet arrival latency - ensuring audio
continuity is maintained over time.
This can be problematic in some situations if congestion causes

latency to be severe and bi-directional communications is required.
However, to a large extent advances in coding and network
management technologies have led to most of these latency issues
becoming manageable in most situations.
IP Network Alternatives
There are a range of common wired IP networks available for
broadcasting audio over IP
IP Description Recommen-
Network dation
Interface
DSL/ADSL Common and transmits bi-directional Point to Point
(Digital digital data over the public internet using STL/Audio
Subscriber a POTS/PSTN line. Typically uses most Distribution
Line) of the data channel bandwidth to Point-to-Point
download data to a subscriber and will Remote
only transmit data as fast as the DSL/ Broadcasts
ADSL data uplink will provide. The Multicasting

outbound data rate can vary greatly, so

check with your Internet Service Provider
to discover the speed of their connection.
SHDSL SDSL/SHDSL connections send Point-to-Point
(Symmetric symmetrical data (i.e. 512 kbps downlink STL/Audio
High-speed and 512 Kbps uplink) as opposed to Distribution
Digital DSL/ADSL connections which send (MPEG
Subscriber asymmetrical data (i.e. 512 Kbps Algorithms)
Line) downlink and 256 Kbps uplink). Multicasting
Symmetrical data normally delivers
higher uplink speeds than DSL/ADSL
connections - increasing the stability and
quality of your connections. Unlike DSL/
ADSL, SDSL and SHDSL cannot be
transported on top of a POTS line.
MPLS Multi-protocol Label Switching is a high- Point-to-Point
Compliant performance data carrying mechanism STL/Audio
Interfaces used to send multiple types of data traffic Distribution
- including IP packets, ATM, SONET (MPEG
(fiber) and Ethernet frames. MPLS tags Algorithms)
data packets with a 'header' to define the Multiple Unicast
path of the packets across the network. STL/Audio
The protocol supports bandwidth Distribution/
reservation and delivers QoS guarantees, Remote
so is ideal for STLs and audio Broadcasts
distribution. (MPEG
Algorithms)
Wireless Different wireless 3G networks like Wireless remote
3G UMTS, HSDPA and EV-DO deliver broadcasts
wireless broadband IP connections over
wide areas of most countries.
WiMAX Like wireless 3G networks, metropolitan Wireless remote
Metropol- 4G WiMAX networks have a range of up broadcasts
itan Area to 30 miles (50kms) and provide
Network connectivity to the internet wirelessly.
(4G) Bandwidth is generally greater than
standard 3G wireless networks.
Portable Portable WiMAX systems deliver Wireless remote

WiMAX dedicated full-duplex, high-speed data broadcasts

wireless connections between two points or Point-to-Point

links between the studio and multiple STL/Audio
locations, providing cost-effective bi- Distribution
directional transmission paths for audio (MPEG
distribution, remote broadcasting or Algorithms)
studio-to-transmitter links. Operating Multicasting
distances of between 2-100 kms are
possible.
8.3 Selecting a Data Plan

IP network data costs vary depending on the network you are connecting to
and the number of channels you need to broadcast. However, in general IP
networks are much cheaper to operate than synchronous data networks like
ISDN. There is a wide range of IP networks to choose from when
broadcasting over IP and some of the factors that affect the selection of a
network to broadcast over include:
Your program content: Are you performing a simple remote broadcast

or distributing high bandwidth audio around a network, i.e. STL or
audio distribution.
The number of audio channels you are sending: Do you need a simple
point-to-point IP audio connection, are you multicasting, or do you
need to send multiple unicast IP audio streams to different studios?
Your broadcasting region: Depending on where you are situated, you
may have access to different infrastructure like DSL/ADSL or fiber;
similarly, you may have access to different wireless networks like
UMTS/HSDPA, EV-DO or WiMAX.
Your budget: A community radio station may be looking for a cost
effective hardware and data solution, whereas a large network may be
looking to integrate flexible and high quality hardware with innovative
software management solutions.
Contact Tieline to receive a spreadsheet that will tell you how much
data your codec will consume per hour of broadcasting to help you
decide what plan to buy.
Data Plan Suggestions

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

2. You will get the best quality connection if both the local (studio) and
remote codecs use the same Internet Service Provider. This can
substantially increase reliability, audio bandwidth and reduce audio
delay. Using the same service provider nationally can give better
results than using different local service providers. This is especially
true if one of the service providers is a cheap, low-end domestic
service provider, which buys its bandwidth from other ISPs. Second
and third tier providers sub-lease bandwidth from first tier providers
and can result in connection reliability issues due to multiple switch
hops. We also highly recommend using Tier 1 ISPs if connecting two
codecs in different countries.
3. Sign up for a business plan that provides better performance than

domestic or residential plans. Business plans typically have a fixed
data limit per month with an additional cost for data beyond that limit.
In addition, Service Level Agreements (SLA) will often provide better
support and response times in the event of a connection failure.
Domestic plans are often speed-limited or 'shaped' when usage
exceeds a predefined limit. These plans are cheap but they are
dangerous for streaming broadcast audio.
4. Ensure that the speed of the connection for both codecs is adequate
for the job. The minimum upload speed recommended is 256 Kbps for
a studio codec and 64 Kbps for a field unit connection.
5. Use a managed IP network connection or a dedicated DSL/ADSL line

for your codecs. Do not share a connection with PCs or other
devices. The only exception to this rule is if an organisation has
network equipment and engineers that can implement and manage
quality of service (QoS) across its network.
How to Order the Right Plan for your Wireless IP

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

One of the most expensive mistakes you can make is borrowing a 3G

SIM card for a broadcast that will last a couple of hours. It is likely that
this type of 3G plan is optimized for voice and not IP data. Dont find out
the hard way it could be an expensive mistake! We recommend you
purchase 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 per
month.
If this type of plan is not available, estimate the number of remote

broadcast minutes/hours you need per month and buy a plan that
bundles large blocks of data for one price. Some Telcos also offer
timed or minutes plans, which offer unlimited data for fixed amounts of
time.
Warnings: Some 3G network providers prohibit streaming

multimedia of any kind on certain accounts. Also, some plans
charge very high rates for data, or may throttle or shape your
available bandwidth after a certain amount of data has been
transferred. Check these factors with your Telco before
subscribing 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 your

connection mono, stereo, multicast or multiple unicast?
Calculate the bit-rate requirement per channel; this will depend
on the compression algorithm you select and need to include
packet overhead data requirements.
As a general rule of thumb, when connecting using UDP to send audio

ensure the total bit-rate (audio bit-rate plus header bit-rate) is no more
than 50% of the ISP connection rate. For example, with a 48 Kbps
audio bit-rate when using the Tieline MusicPLUS algorithm, add 16Kbps
for the packet overheads and multiply by 2 (48 + 16 x 2 = 128Kbps).
Once you have calculated the total connection bit-rate (64Kbps) and
how high the ISP connection bit-rate needs to be (128Kbps = twice the
connection bit-rate), you can shop around for the most suitable and

competitive data plan to suit your needs.
8.4 Redundancy Considerations

In studio-to-transmitter link applications it is a good idea to have a strategy
for backing up your program audio in the event of hardware failure or the loss
of an IP link. Some of the methods used by professional audio codecs to
protect against lost audio over a connection in critical broadcast
applications include:
Automatic silence detection.

Dual-redundant power supplies for hardware.
Fail-safe audio program backups using either on-board or external
audio storage media.
Failover to a second IP connection, or to an alternative audio transport
like POTS/PSTN.
The methods employed depend on the hardware being used and the
connections both supported by the codec, and available at the studio and
transmitter sites.
8.5 IP Interoperability
In the past, audio codec manufacturers have designed codecs that have
largely been incompatible with each other in many different situations due to
the use of:
Proprietary session data protocols (used to establish and

maintain codec connections)
Different proprietary audio algorithms.
Different control data.
As a result, universal compatibility between manufacturers was difficult to

achieve. In the early stages of broadcast audio over IP development, Tieline
and other partners in the Audio-via-IP Experts Group lobbied for standards
that manufacturers should adhere to in order to make IP compatibility
between different brands a reality. As a result, the EBU has published
standards in EBU N/ACIP Tech 3326 that manufacturers should comply with
in order to deliver compatibility of their codec with other brands over IP.

All Tieline codecs are EBU N/ACIP Tech 3326 compatible over IP
and the company is committed to developing new IP and 3GIP
applications that take advantage of emerging network
infrastructures around the globe.
SIP (Session Initiation Protocol)

SIP is central to codec compatibility because it allows different devices
to communicate with each other and codecs need to be SIP-compatible
to comply with EBU N/ACIP Tech 3326.
There are two very distinct parts to a call when dialing over IP. The initial
stage is the call setup stage and this is what SIP is used for. The
second stage is when data transference occurs and this is left to the
other 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 to
remember that SIP only defines the way in which a communication
session between devices should be managed. It does not define the
type of communication session that is established.
SIP leverages on the use of web architectures like DNS, and SIP
addresses are similar in appearance to email addresses. A device using
SIP can dial another devices SIP address to find its location. This task
is performed by SIP servers, which communicate between registered
SIP-compliant devices to set up a call.

Mandatory Algorithms Decreed under EBU N/ACIP Tech

3329 for Broadcasting Audio over IP
Mandatory algorithms decreed under EBU N/ACIP Tech 3326 include
G.711, G.722, MPEG Layer II and PCM (pulse code modulation) and
must 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 Connections

Connection reliability can be improved through the use of:
IP connection management software (i.e. Tieline QoS Performance

Engine for managing IP audio connections)
Low bit-rate, low delay algorithms optimised for use over wireless IP
networks (e.g. Tieline Music, Tieline MusicPLUS, AAC)
The following checklist can be used to further improve reliability when

connecting 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 thats part of
Internet backbone.
2 The same ISP is being used for both codec
connections.
3 The ISP data plan is a Business Plan or equivalent.
4 The ISP connection speed is adequate (e.g. higher
than audio bit-rate plus packet overheads).
5 Equipment is high quality and suitable for media
streaming.
6 The ISP connection speed has been tested.
7 The ISP connection is not shared with PCs or other
devices.
8 UDP is being used as the audio transport protocol.
9 Only 50% of ISP connection uplink bandwidth is being
used.
10 There are no wireless connections being used.
Wireless Network Reliability

It is very difficult to guarantee connection quality when there is no way
of knowing how many people are sharing the same wireless connection
at any point in time. For example, wireless 3G broadband IP
connections can easily become congested and result in packet loss
and audio drop-outs, particularly when using cell-phone connections at
special events where thousands of people have mobile phones. This can
result in poor quality connections and audio drop-outs if cell-phone base
stations are overloaded.
Wireless network reliability can be improved through the use of

dedicated portable WiMAX wireless links. Audio codecs should also be
capable of using automated reconnection features to redial and IP
connection immediately if an IP connection is lost.

8.7 Testing a Network

Finally, there are a few very simple tools that you can use to test whether a
codec 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 codec
or 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.



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 on
your 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 available
on 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 is
open.
The process for testing is similar to the ping test.

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) and
press 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 the
port was not available.
Trace the Route of Packets

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



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 your
keyboard.
8.8 Assessing Hardware Requirements

Ultimately the hardware you require will be determined by the broadcast you
want to perform. DSP-based codecs are generally the most reliable over all
IP connections and have greater stability compared to PC systems.
There is a large range of codecs available that are suitable for different
broadcast situations. A sample of these products follows and they can all
connect over 3G wireless broadband networks, wired and wireless LANs,
WANs, the internet, satellite IP, WiMAX and Wi-Fi..
Tielines Bridge-IT is a low-cost, high-

performance, point-to-point or multi-point
stereo IP audio codec solution for broadcast
and professional applications.
2 input analog or AES/EBU with

simultaneous analog and digital outputs
Ideal for STL and audio distribution
applications
IP Multicasting and multiple unicasting
Simple remote broadcast links
Multiple codec installations (2 codecs fit
in 1 x 19 rack unit)
The i-Mix G3 is an advanced IP codec for

radio and television, combining six essential
live remote broadcast products into one
lightweight 16" x 9" box, replacing tens of
thousands of dollars of expensive
equipment.
A wireless-capable 6 input digital mixer

with a cross point digital matrix router
Bi-directional audio & simultaneous
communications circuits with 4

headphone controls/outputs
On-board PA output control with a built-in
telephone hybrid
Wired and wireless IP and POTS codecs
with wireless 3G/3.5GIP, ISDN, X.21,
GSM and Satellite Codec capability
On-board relays and RS-232 with full
studio remote control
The wireless-capable Commander G3 is a

powerful and reliable remote broadcast IP
codec.
3 input stereo mixer with 2 headphone

controls/outputs
Connect over wired IP or use two
interchangeable module slots to connect
over wireless 3G/3.5G, POTS/PSTN,
ISDN, X.21, GSM and B-GAN satellite
networks.
On-board relays and RS-232 with full
studio remote control
The 2RU Commander G3 is the ideal STL

and audio distribution codec, or can be used
to receive IP audio from i-Mix, Commander
field or Bridge-IT.
2 balanced XLR inputs with front and rear

panel headphone outputs and mic inputs.
4 front panel PPM meters displaying your
choice of send, return or channel audio
levels
Insert an analog XLR 2 input/output card
or an AES/EBU XLR 2 input/output card
Internal or external AES/EBU clock
Automatic failover to any compatible audio
transport
The TLG3 GUI software controller emulates

the hardware front panel of the 2RU

Commander G3. It can be used to control

1RU or 2RU codecs using USB or RS-232
serial or LAN connections. This advanced
software GUI can control all normal codec
functions such as dialing, menu navigation,
audio monitoring and level controls
9 Glossary of Terms
AES/EBU Digital audio standard used to carry digital audio signals
between devices.
AES3 Official term for the audio standard referred to often as AES/
EBU.
DNS The Domain Name System (DNS) is used to assign domain
names to IP addresses over the World-Wide Web.
Failover Method of switching to an alternative audio stream if the
primary connection is lost.
GUI Acronym for Graphic User Interface
ISP Internet Service Providers (ISPs) are companies that offer
customers access to the internet
IP Internet Protocol; used for sending data across packet-
switched networks.
Latency Delay associated with IP networks and caused by
algorithmic, transport and buffering delays.
Multicast Efficient one to many streaming of IP audio using multicast
IP addressing.
Narrowcast Transmitting a signal or data to a specific recipient or
recipients.
Network A system for forwarding data packets to different private IP
Address network addresses that reside behind a single public IP
Translation address.
(NAT)
Packet A formatted unit of data carried over packet-switched
networks.
Port Address Related to NAT; a feature of a network device that allows IP
Translation packets to be routed to specific ports of devices
(PAT) communicating between public and private IP networks.

QoS (Quality Priority given to different users or data flows across

of Service) managed IP networks. This generally requires a Service
Level Agreement (SLA) with a Telco or ISP.
Redundancy Choosing an alternative audio stream to use if a primary
audio connection is lost.
RTP A standardized packet format for sending audio and video
data streams and ensures consistency in the delivery order
of voice data packets.
SDP SDP (Session Description Protocol) defines the type of
audio coding used within an RTP media stream. It works
with a number of other protocols to establishes a devices
location, determines its availability, negotiates call features
and participants and adjusts session management features.
SIP SIP (Session Initiation Protocol) works with a myriad of other
protocols to establish connections with other devices. It is
used to find call participants and devices and is the method
used by most broadcast codecs to connect to competing
brands of codec for interoperability.
SLA Service Level Agreements (SLAs) a contractual agreement
between an ISP and a customer defining expected
performance 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 a
receiver. Its two functions include controlling the
transmission rate of data and ensuring reliable transmission
occurs. Generally not well-suited to streaming live audio
because buffering (latency) is employed to ensure data
packets are received in order
UDP UDP (User Datagram Protocol) most commonly used for
sending internet audio and video streams. UDP packets
include information which allows them to travel
independently of previous or future packets in a data stream.
In general, UDP is a much faster and more efficient method
of sending audio over IP.
Unicast Broadcasting of a single stream of data between two points.

Trademarks and Credit Notices 51
10 Trademarks and Credit Notices

1. 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 accuracy
of the information contained in this guide, Tieline is not responsible for
any errors or omissions within it, and the guide should not be relied
upon solely when designing, purchasing and installing broadcast IP
networks. Always consult a qualified and experienced IP broadcast
network professional for advice or to undertake appropriate training prior
to purchasing and installing equipment for use over IP networks.
11 Appendix 1: IP Protocols
Additional IP transport protocols that can affect sending audio over IP
include the following:
RTP (Real-time Transport Protocol)

RTP has been designed to transport real-time multimedia streams over
IP networks. It is a standardized packet format for sending audio and
video data streams and ensures consistency in the delivery order of
voice data packets.
RTCP (RTP Control Protocol)

RTCP is a sister protocol of RTP and it gathers statistics and provides
feedback on the quality of a streaming media connection. The type of
information distributed includes packet counts, lost packet counts, jitter
and round-trip delay times.
SDP (Session Description Protocol)

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

Establishes a codecs location.

Determines the availability of a codec.
Negotiate the features to be used during a call, i.e. the algorithm
and 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 to
establish and control streaming media servers and is typically used in
conjunction with RTP, which controls the transport of streaming data
itself.
SAP (Session Announcement Protocol)

SAP is an announcement protocol used to advertise multicast sessions
and communicate setup information to prospective broadcast
participants.
SNMP (Simple Network Management Protocol)

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

Index 53
10 reasons to broadcast over IP 8

Index Addresses 30
Background on IP networks 5
-3- Connection types 11
3G 16 Interoperability 41
-A- IP versus POTS, ISDN and X.21

IP versus synchronous data 5
5
Algorithms 23 Jitter 28
Audio distribution 9 Jitter Buffering 28
-C- Latency 28
MANs/WANs/LANs 16
Credit notices 51 NAT 32
-D- Network Address Translation 32
Network considerations 21
Data Costs 38 Network types 16
Data Plans 38 Planning Installation 34
Data Requirements 38 Port Forwarding 32
Disclaimer 51 Private and public networks 30
-E- Redundancy 41
Regional factors 34
EBU N/ACIP Tech 3329 41 Testing connections 45
Error Concealment 25 Transport protocols 21
-F- What is IP 5
Wireless 3G and WiMAX 16
FEC 25
IP Addresses 30
Forward Error Correction
About 25 IP Codecs 47
Conserving bandwidth 25 IP Hardware 47
IP LANs 16
-G- IP MANs 16
Glossary 49 IP Protocols 21
-I- Appendix
IP WANs 16
51
Internet Broadcasting 35
Interoperability 41
-J-
Introduction 4 Jitter 28
IP Jitter Buffering 28

-L- -S-
Latency 28 SAP 51
-M- Satellite IP
SDP 51
16
Multicasting SIP 21
About 11
SIP, how it works 41
Applications 11
SNMP 51
Multiple Unicasts
STLs 9
About 11
Studio to transmitter links 9
Applications 11
-N- -T-
TCP and UDP 21
NAT 32
Testing IP Networks 45
Network Address Translation 32
Trademarks 51
Network Types 35
Networks -U-
Considerations 21
Unicasting
-P- About 11
Applications 11
Packet Loss 25
Planning Installation 34 -W-
Port Forwarding 32 WiMAX 16
Private IP Networks 30 Wireless
Public IP Networks 30 3G 16
-Q- EV-DO
Satellite
16
16
Quality of Service (QoS) 35 UMTS/HSDPA/HSPA+ 16
-R- WiMAX 16
Redundancy 41
Reliability Checks 43
Remote broadcasts 9
RTCP 51
RTP 51
RTSP 51

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Part XI Appendix 1: IP Protocols 51

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Broadcasters now clearly recognise that IP networks are more flexible,

How to use this Guide

1. IP is not reliable enough to broadcast over.

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In addition to this guide, you can become a part of a community of

1.1 What exactly is IP?

Some networks like wireless WiMAX networks are called Metropolitan

Tieline Pty. Ltd. 2010

computer networks and telecommunications devices (e.g. IP audio

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

Example of IP Codecs using Wired and Wireless IP Network

Wireless IP connections can be made over 3G and 4G wireless cell

What are the Differences between IP and other

Tieline Pty. Ltd. 2010

Packet-switched networks are more efficient and optimize the use of

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

Despite the potential pitfalls of IP, broadcasters are moving inexorably

Tieline Pty. Ltd. 2010

2 10 Great Reasons to Broadcast

2. The hardware required for IP broadcasting is cheaper

3. Broadcasting over IP is more flexible

4. IP networks can be scaled to suit individual installations

5. Wireless IP networks deliver flexible broadcast connections from

6. IP Networks are Widely Available

Tieline Pty. Ltd. 2010

Wired broadband connections are widely available in most regions of

7. Setting up remote IP broadcasts is extremely simple

8. Audio over IP integrates seamlessly with new broadcast

9. Integration of audio over IP into large radio networks creates

10. Audio over IP is the future of broadcasting

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Studio-to-Transmitter Links (STLs)

Public Internet Connections (WANs).

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Distributing programming between network affiliates or studios.

IP Information: An IP address is a unique number that allows

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Unicast Applications for Broadcasters

STLs between a studio and a single transmitter site.

Example of a Unicast IP Connection

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transmit the original program audio.

Multicast routers over the IP network replicate packets on demand as

Multicast Applications for Broadcasters

Distributing high quality audio over broadcast LANs.

Multicasting can also be a good way to set up permanent STL

Multicast Broadcast Example

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Key Multicasting Concepts

Multicast transmissions are sent using a dedicated IP multicast

Some ISPs provide quality of service (QoS) priority to multicast

The important multicast concepts to remember are:

Multicast streams are not automatically allowed over WANs

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What are Multiple Unicasts?

The total bandwidth of each connection is the bandwidth required to

Multiple Unicast Applications for Broadcasters

Distributing multiple streams of program audio to radio