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VoIP

& NGN

1
Title
Preface
The move towards Next-Generation Networks
(NGNs) based on Voice Over IP (VoIP) represents
the most significant change in core network design
since the transition from analogue to digital. NGNs
provide operators with the capability to carry both
data and voice traffic over a single, converged
network. This can reduce operational costs and
increase flexibility. However, it also presents a
number of new challenges.

The demands of data transmission are much


different from those of voice transmission.
Fundamentally, data is very sensitive to errors but quite tolerant of delay. Conversely,
voice is very sensitive to delay and to variations in delay, but can tolerate a certain
amount of error. VoIP involves the transmission of voice conversations over a
technology originally designed for data transmission.

Like the transition from analogue to digital telephony, networks will normally migrate
gradually towards a full NGN environment and therefore it will often be necessary to
bridge between current circuit-switched networks and NGNs using Media Gateways.
Careful design and configuration is required to ensure that operators can realise the
benefits of NGNs, while their customers continue to enjoy high-quality service.

Since its formation in 1987 Telsis has been providing high-performance voice
solutions to network operators and service providers. Indeed, back in 1988, Telsis
implemented one of the world’s first Voice over LAN systems, providing hundreds of
callers with simultaneous access to multiple live sports commentaries.

The company has always focused on developing technology that enables our
customers to cost-effectively meet the needs of the end-user for a high quality,
predictable and reliable service.

If you wish to learn more about Telsis’ expertise and how it applies to NGN and VoIP
networks, please call +44 1489 76 00 00.

Jeff Wilson
Chairman, Telsis

2
Introduction

The Ocean fastSSP is a carrier-grade switch from Telsis that has been chosen by a large
number of telephone network operators around the world to provide solutions for revenue
generation, cost reduction and service differentiation. These operators have included
incumbent operators (like BT and KPN), other licenced operators (such as the German
regional carrier EWE TEL and UK-based Opera Telecom) and mobile network operators
(including O2, Telefónica Móviles, T-Mobile and Vodafone).

As these customers will all testify, the fastSSP offers many benefits, including:
• Flexibility – programmability, coupled with optional audio playback and DTMF
detection, enables value-added services to be created and adapted to meet the needs
of the ever-changing telecommunications market place.
• Resilience – dedicated hardware, designed for the rigours of the non-stop
telecommunications operating environment, enables operators to concentrate on
developing their business, safe in the knowledge that the fastSSP can continue
handling calls even in the unlikely event of a component failure.
• Performance – the fastSSP’s high Busy Hour Call Attempts (BHCA) enables operators
to use it for a large variety of applications, including short-duration calls such as those
associated with TV-stimulated voting traffic.
• Value for money – the fastSSP is competitively priced and, due to its small footprint,
low power consumption and easy-to-use remote configuration and management
capability, operational expenditure is kept to a minimum.
• Scalability – the fastSSP’s capacity can be easily expanded to accommodate traffic
growth, thereby offering a pay-as-you-grow business model.
• Signalling expertise – Telsis’ in-house signalling expertise means that the fastSSP
can be readily adapted to overcome interworking issues – the Ocean fastSSP has
passed interconnect approvals in a large number of countries, in many cases against
very short timescales.

Thanks to the clear modular design of the Ocean fastSSP, with dedicated switch matrix,
signalling processing and trunk interface cards, Telsis is able to offer the fastSSP with E1
trunk interfaces (for connecting to a conventional telephone network), Ethernet interfaces
(for VoIP connections to a packet-based Next-Generation Network), or a combination of
the two (acting as a Media Gateway, providing a bridge between conventional telephone
networks and Next-Generation Networks).

Nearly twenty years of telephony experience enables Telsis to design and develop best-in-
class voice solutions for network operators wishing to maximise the benefits of Next-
Generation Network technology.

3
VoIP Essentials

Overview
Traditional digital telephone networks convert from the analogue input on a handset to a
digital stream using 8-bit sampling at 125µs intervals (equating to a required bandwidth
of 64kbit/s). This is known as Pulse Code Modulation (PCM) and is often referred to as
G.711. In order to be sent over an IP network, this audio stream needs to be split into
separate packets.

Media
Concentrator 0100011110011100111
Gateway

Packet size involves a trade-off between the delay and bandwidth requirements.

78 bytes of header information are required to enable the IP network to route a packet
successfully to its intended destination. A 1ms packet size would have 8 bytes of data,
plus the 78 bytes header, requiring an IP network bandwidth of 688kbit/s. On the other
hand, a packet size of 100ms requires a bandwidth of only 70kbit/s. Of course, it is
desirable to minimise bandwidth in order to reduce cost of transmission.

However, delay is introduced whenever audio is converted into packets – as the media
gateway needs to wait for the required stream length to arrive before it can send out a
packet. Therefore, the larger the packet size, the greater the delay. It is desirable to
minimise delay because it affects the ability of callers to interact with one another
naturally. If the delay is too long, normal conversations become difficult – by the time the
person at one end has heard what the other person is saying, they may have started
talking at the same time, making the call experience highly unsatisfactory.

Typically, a VoIP packet size of 10ms is used, giving generally tolerable delay with a
bandwidth requirement of approximately 127kbit/s per channel.

It is therefore important to realise that migration to VoIP does not reduce bandwidth –
packetisation increases the overall bandwidth requirement. Instead, the network
operator’s case for using VoIP is usually the reduced Opex of running a single network for
both voice and data, and that IP transmission costs are generally lower than Time Division
Multiplex (TDM).

There are also other factors to consider when designing VoIP networks.

4
Silence Suppression
One way of reducing bandwidth is to employ silence suppression at the media gateway, so
that packets are only sent when speech levels are above a given threshold. As all voice
conversations have pauses in them at various times, silence compression has the advantage
of reducing the required bandwidth. However, care is required when setting the threshold
level – set it too high and the regenerated speech is often ‘clipped’.

To prevent a silence period being mistaken for the end of the call, a technique called
Comfort Noise Generation (CNG) can be used to fill the gap with background noise.

Jitter
When a call is set-up in a conventional TDM network, an end-to-end connection is
established to provide a continuous audio stream. However, in an IP environment, individual
audio packets may take different paths over the network and therefore take different
amounts of time to arrive at the destination, potentially even arriving in a different order. As
a consequence, when the audio is to be regenerated at the far end, arriving packets must
be buffered for a period of time before they are reconstructed as a stream of audio. This
buffer, known as a jitter buffer, introduces yet more delay.

The size of jitter buffer is generally set when a call is established. If a jitter buffer is too
small, the regenerated audio stream may be susceptible to missing packets (because they
have not arrived in time), with a consequent impact on the quality of the audio. On the other
hand, if the jitter buffer is too large, the extra delay impairs the quality of the conversation.

Jitter buffer sizes can be reduced by taking steps to reduce packet routing divergence across
a network. For example, Multi-Protocol Label Switching (MPLS) can be used to ensure that
sequences of packets are routed via the same MPLS routers across a network.

Echo
Echo is primarily caused by imperfections in the hybrid (part of the handset that converts
between 2-wire and 4-wire). As humans, we are able to filter out echoes where the delay is
both constant and very short (typically less than 30 milliseconds). This means that, over
circuit-switched networks, echo is only a problem when the transmission time is long. For
this reason, echo cancellers are provided only at international gateways.

For VoIP networks, echo is an issue for all calls because of the delays introduced by
packetisation and jitter buffers. Therefore, for good audio quality, media gateways need to
employ echo cancellation devices. Echo cancellers analyse audio sent on the transmit side
in order to ‘predict’ the corresponding echo on the receive side, and to counteract it. ITU-T
recommendation G.168 provides a minimum specification for the performance of echo
cancellers.
5
Ocean fastSSP, Ocean iPC and VoIP

The Telsis Ocean fastSSP VoIP option


enables network operators to quickly and
easily add VoIP capability to their network.

By providing both E1 and IP connections,


the fastSSP can act as a media gateway,
bridging conventional TDM and VoIP
networks. Unlike many other media
gateways, the fastSSP with VoIP option is
based on proven carrier-grade telephony
equipment, trusted by network operators
such as BT, Telefónica, T-Mobile and
Vodafone.

The VoIP option is provided by VoIP Card


pairs, with each pair providing up to 480
channels of VoIP (the equivalent of 16 E1
TDM trunks). Call control is provided by This picture shows an Ocean fastSSP with 48
E1 capacity and 480 channels of VoIP (the two
industry-standard Session Initiation Protocol
pairs of cards on the right hand side)
(SIP) v2.0 signalling. Built-in echo
cancellation provides optimum audio quality, The other well-proven component parts of
whilst jitter buffer sizes and silence the fastSSP remain largely unaffected
suppression can be configured for individual (including NODAL applications, audio
requirements. playback and DTMF detection on VoIP
channels), thereby simplifying the addition
of VoIP Cards to an existing TDM-only
fastSSP.

Because the VoIP facility is fully integrated


with other fastSSP functionality, NGN
operators can enjoy the flexible call routing
and number manipulation capabilities
offered by the switch.

Furthermore, for simple protocol conversion


applications, the Ocean iPC Intelligent
Protocol Converter can also be specified
with VoIP capability. A VoIP-enabled
Ocean iPC may be used to connect an
IP PBX to a TDM network, for example.

6
These capabilities of the fastSSP can be extended further via external control – for
example, using an Ocean fastSCP Service Control Point. The fastSCP executes service
logic programmes that offer advanced call routing features such as time-based routing,
proportional routing, origin-based routing, call-barring and special routing for VIP callers.

This flexibility is offered to network operators via an easy-to-use graphical user interface:

Alternatively, customers can use the fully resilient Ocean Control Protocol (OCP) to control
fastSSPs from their own host systems.

Although the fastSSP supports SIP addressing for VoIP calls, standard telephone numbers
will continue to be used – numbers are required for PSTN access. Therefore, the
fastSSP’s comprehensive call routing and number management facilities are just as
applicable for VoIP as they are for TDM networks, offering network operators the potential
to maintain the same services and call routing during the period of transition towards an
NGN architecture.

As with all Ocean fastSSPs, there are options for DTMF or voice detection and start-at-
the-beginning audio playback from a store with up to 4 hours capacity – available on all
ports simultaneously.

Customers can optionally write their own fastSSP applications so that value-added
services can be created quickly, easily and cost-effectively for calls to or from both TDM
and next-generation networks.

The VoIP-enabled Ocean iPC and fastSSP platforms continue to be controlled by the easy-
to-use management tools – Platform Manager and Route.

7
VoIP Applications

Connectivity drives revenues and therefore the Ocean fastSSP’s and Ocean iPC’s support
for both TDM and NGN interfaces enables network operators to grow their customer base
by offering connectivity to a wide range of customer premises equipment (such as PBXs).
Also, the ubiquity of IP routing and switching equipment offers potentially reduced
transmission costs within a network, thereby increasing operating margins.

Save on Opex by using IP connectivity between POPs, instead of dedicated E1 capacity. In


some cases, for equivalent bandwidth, leased capacity cost savings of up to 90% can be
realised:

3rd Party
Ocean Ocean
PSTN IP Network PSTN
fastSSP fastSSP
Providers

Enhance the connectivity of your network by enabling end-user equipment to connect


over Session Initiation Protocol (SIP):

SS7 Ocean SIP


TDM PBX
Network iPC

8
Provide TDM connectivity to an NGN:

ISDN
SIP DASS2
Next-Generation Ocean
PBX
Network iPC
DPNSS
QSIG

Enable an advanced call routing capability with additional value-added services for both
TDM and VoIP networks when connected to a full Ocean Service Node, including optional
Ocean fastIP Intelligent Peripherals:

VoIP Network

Ocean
fastIP

Ocean
PSTN
fastSSP
Ocean
fastIP

Ocean
Resilient Data Network
fastSCP*

*split-site configuration possible

9
Summary

Telsis has been producing world-class “We use Telsis platforms extensively in
high-quality voice solutions since 1987. our own network and our experience
Our reputation for flexibility, reliability and of their competitive cost, robustness
value-for-money has led to Telsis being and easy programmability gives us
chosen by many of the world’s most confidence in being able to offer our
well-known network operators – mobile and customers a very effective solution.”
fixed – to provide critical components for Thorsten Thews
their networks. EWE TEL

Unlike many VoIP equipment suppliers, the roots of Telsis are firmly in the
telecommunications market place rather than IT. This means that we have a well-
established and in-depth understanding of the key requirements of network operators:

• High availability • Flexibility


• Low operational costs • Resilience
• High-quality audio • Predictable performance

Our design philosophies are based upon the needs of the caller – all of our products are
designed according to rigorous processes to ensure that they provide the best user
experience.
Furthermore, Telsis products are designed
“We’ve answered over 200 million calls for rapid installation and configuration, so
and we haven’t had a single major that they can start earning revenue at the
problem. Reliability is everything. Telsis earliest opportunity. Resilience offered by
promises us that the system will work.
dual-redundant components offers remote
We promise our customers that it will
‘lights-out’ operation so that network
work. And it does.”
operators can focus on innovation and
Stewe Wahlström growing their business, rather than
TeliaSonera worrying about the availability of the
network. The ability to plan ahead is vital
in today’s increasingly competitive
telecommunications market place.

10
Ocean Glossary Glossary

rds Next-Generation Networks The Ocean fastSSP programmable switch is available in Compact or Extended versions, BHCA – Busy Hour Cal
n Voice Over IP (VoIP) represents both of which are fully non-blocking. The Compact Switch is designed for points of
cant change in core network design presence and provides 32 E1 trunks (960 ports) in a standalone unit, suitable for desktop CNG – Comfort Noise G
tion from analogue to digital. NGNs or rack-mounted operation. The Extended Switch offers up to 7680 TDM ports or VoIP DASS2 – Digital Acces
rs with the capability to carry both channels in a single cabinet.
traffic over a single, converged DPNSS – Digital Privat
an reduce operational costs and
The Ocean iPC is a highly-reliable, fully non-blocking Intelligent Protocol Converter, which Signalling System
ty. However, it also presents a
supports interworking between a wide range of signalling protocols. It combines impressive
challenges. DTMF – Dual Tone Mul
packing density, small footprint and competitive price with a rich set of protocol
f data transmission are much interworking standards. E1 – European Digital T
hose of voice transmission. based on 32 x 64 kbit/
data is very sensitive to errors but quite tolerant of delay. Conversely, The Ocean fastSCP is a flexible, high-performance system which controls call-handling
nsitive to delay and to variations in delay, but can tolerate a certain units, such as the Ocean fastSSP family of switches or the Ocean fastIP Intelligent IP – Internet Protocol
. VoIP involves the transmission of voice conversations over a Peripheral. The fastSCP makes service-flow decisions using built-in data storage, with
nally designed for data transmission. ISDN – Integrated Serv
rapid, visual application development via the Ocean fastSCE service creation environment.
The SCP can also control third-party switches when used in conjunction with the MPLS – Multi-protocol
on from analogue to digital telephony, networks will normally migrate Ocean fastTC INAP interface.
ds a full NGN environment and therefore it will often be necessary to NGN – Next Generation
current circuit-switched networks and NGNs using Media Gateways.
The Ocean fastIP high performance intelligent peripheral supports up to 120
and configuration is required to ensure that operators can realise the PBX – Private Branch E
simultaneous calls. It offers an exceptional set of telephone call answering, processing
Ns, while their customers continue to enjoy high-quality service.
and routing capabilities, including interaction. Calls are connected via E1 trunks PCM (G.711) – Pulse C
ion in 1987 Telsis has been providing high-performance voice supporting a number of signalling schemes including SS7 and Euro-ISDN.
work operators and service providers. Indeed, back in 1988, Telsis PSTN – Public Switche
ne of the world’s first Voice over LAN systems, providing hundreds of The Ocean fastSCE is an easy-to-use set of visual service creation and management
QSIG – Q Signalling
ultaneous access to multiple live sports commentaries. tools, providing everything needed for a customer to quickly take a service from
conception to deployment. SIP – Session Initiation
as always focused on developing technology that enables our
ost-effectively meet the needs of the end-user for a high quality, Ocean Control Protocol (OCP) is a custom-designed resilient TCP/IP protocol that SS7 – Signalling Syste
reliable service. enables Ocean platforms to talk to one another. It may also be used to communicate with
TDM – Time Division M
third-party platforms, since OCP is an open, published standard.
arn more about Telsis’ expertise and how it applies to NGN and VoIP
VoIP – Voice over Inter
e call +44 1489 76 00 00.

For further details on our range of Ocean products,


visit our website www.telsis.com

s
Glossary

BHCA – Busy Hour Call Attempts

CNG – Comfort Noise Generation

DASS2 – Digital Access Signalling System

DPNSS – Digital Private Networking


Signalling System

DTMF – Dual Tone Multi-frequency

E1 – European Digital Telephony Standard


based on 32 x 64 kbit/s channels

IP – Internet Protocol

ISDN – Integrated Services Digital Network

MPLS – Multi-protocol Label Switching

NGN – Next Generation Network

PBX – Private Branch Exchange

PCM (G.711) – Pulse Code Modulation

PSTN – Public Switched Telephone Network

QSIG – Q Signalling

SIP – Session Initiation Protocol

SS7 – Signalling System No. 7

TDM – Time Division Multiplex

VoIP – Voice over Internet Protocol


www.telsis.com
Telsis Limited, Barnes Wallis Road, Segensworth East, Fareham, Hampshire PO15 5TT, United Kingdom
Tel: +44 1489 76 00 00 Fax: +44 1489 76 00 76 E-mail: info@telsis.com
Australia • Germany • Italy • The Netherlands • Singapore • Spain • UK

1590-1122-01 Copyright © 2006 Telsis Ltd. All rights reserved.

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