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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
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-inclass voice solutions for network operators wishing to maximise the benefits of NextGeneration Network technology.
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.
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.
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.
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 industry-standard Session Initiation Protocol (SIP) v2.0 signalling. Built-in echo cancellation provides optimum audio quality, whilst jitter buffer sizes and silence suppression can be configured for individual requirements.
This picture shows an Ocean fastSSP with 48 E1 capacity and 480 channels of VoIP (the two pairs of cards on the right hand side)
The other well-proven component parts of the fastSSP remain largely unaffected (including NODAL applications, audio 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.
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-atthe-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 easyto-use management tools – Platform Manager and Route.
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 IP Network Providers
Enhance the connectivity of your network by enabling end-user equipment to connect over Session Initiation Protocol (SIP):
Provide TDM connectivity to an NGN:
ISDN DASS2 PBX 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:
Ocean fastIP PSTN Ocean fastSSP Ocean fastIP
Resilient Data Network
*split-site configuration possible
Telsis has been producing world-class high-quality voice solutions since 1987. Our reputation for flexibility, reliability and value-for-money has led to Telsis being chosen by many of the world’s most well-known network operators – mobile and fixed – to provide critical components for their networks. “We use Telsis platforms extensively in our own network and our experience of their competitive cost, robustness and easy programmability gives us confidence in being able to offer our customers a very effective solution.” Thorsten Thews 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 wellestablished and in-depth understanding of the key requirements of network operators: • High availability • Low operational costs • High-quality audio • Flexibility • Resilience • 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.
The Ocean fastSSP programmable switch is available in Compact or Extended versions, both of which are fully non-blocking. The Compact Switch is designed for points of presence and provides 32 E1 trunks (960 ports) in a standalone unit, suitable for desktop or rack-mounted operation. The Extended Switch offers up to 7680 TDM ports or VoIP channels in a single cabinet. The Ocean iPC is a highly-reliable, fully non-blocking Intelligent Protocol Converter, which supports interworking between a wide range of signalling protocols. It combines impressive packing density, small footprint and competitive price with a rich set of protocol interworking standards. The Ocean fastSCP is a flexible, high-performance system which controls call-handling units, such as the Ocean fastSSP family of switches or the Ocean fastIP Intelligent Peripheral. The fastSCP makes service-flow decisions using built-in data storage, with 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 Ocean fastTC INAP interface. The Ocean fastIP high performance intelligent peripheral supports up to 120 simultaneous calls. It offers an exceptional set of telephone call answering, processing and routing capabilities, including interaction. Calls are connected via E1 trunks supporting a number of signalling schemes including SS7 and Euro-ISDN. The Ocean fastSCE is an easy-to-use set of visual service creation and management tools, providing everything needed for a customer to quickly take a service from conception to deployment. Ocean Control Protocol (OCP) is a custom-designed resilient TCP/IP protocol that enables Ocean platforms to talk to one another. It may also be used to communicate with third-party platforms, since OCP is an open, published standard.
For further details on our range of Ocean products, visit our website www.telsis.com
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
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: email@example.com Australia • Germany • Italy • The Netherlands • Singapore • Spain • UK
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