Designing

Global VoIP/ and FoIP

Solutions Networks with the

Nuvo200Nuvo 200 IP/SSP

Release 1.7

IP telephony is defined here as any telephony application that can be enabled across a packet-switched data network via the Internet Protocol (IP). Packet-switched networks are not optimized for any one type of traffic, allowing intelligent end-user devices to encode and decode speech to make better use of the available bandwidth. The voice compression algorithms now used in IP-based telephony can deliver voice in a fraction of the bandwidth, sometimes with an increase in efficiency of a factor of eight or greater.

In addition, by treating voice as another form of data and sending it over the same network as data, IP telephony is enabling new applications that use the best characteristics of voice communications and data processing. These applications can include PC-to-PC connections, PC-to-phone connections, and phone-to-phone connections. Example applications include voice over private IP backbones, Internet or intranets, fax traffic (both real-time and store-and-forward), unified messaging, and much more.

Many people talking about IP telephony today are only referring to one application of this technology: long distance replacement, or toll bypass using the Internet.

The idea of making "free" telephone calls over the public Internet has created great excitement among technology-savvy consumers. This approach, identified herein as VON (voice on the net) was widely discussed as the replacement to the PSTN. With over 60100,000 active IP Phone users today, VON is an interesting technology but is lacking in many PSTN features.

The real market for IP telephony is VoIP (voice Voice over IP) on managed IP backbones, (e.g. Qwest, Level 3, WorldCom) and not the public Internet. The Today, the public Internet may be usedis only useful for a least-cost voice or non-time-sensitive data, (secondary backbone) but it is lackingsince it lacks in real-time capabilities, and bandwidth management, . voice quality, and scaleable call control.

The overall advantage of IP telephony comes from treating voice as another form of data. While claims that the PSTN is dying are premature and unfounded, the advantages presented by IP telephony are clearly visible today:

More Bandwidth: One advantage of IP telephony is that it dramatically improves efficiency of bandwidth use for real-time voice transmission, in many cases by a factor of 8 or more. This increase in efficiency is a real long-term driver for the evolution from circuit-switched technology to packet-switched.

New Services: Another advantage IP telephony has over the PSTN is that it enables the creation of a new class of service that combines the best characteristics of real-time voice communications and data processing, such as web-enabled call centers, collaborative white-boarding, multimedia, telecommuting, and distance learning. This combination of human interaction and the power and efficiency of computers is opening up an entirely new world of communications.

Progressive Deployment: The final advantage of IP telephony is that it is additive to today's communications networks. IP telephony can be used in conjunction with existing PSTN switches, leased and dial-up lines, PBXs and other customer premise equipment (CPE), enterprise LANs, and Internet connections. IP telephony applications can be implemented through dedicated gateways, which in turn can be based on open standards platforms for reliability and scalability.

One of the greatest challenges for IP telephony is to develop networks which are not only scaleable but seamless to the subscriber and to the service provider. If the service is difficult to access by the subscriber due to complex dialing plans and special PIN numbers, or requires significant time to complete a call, or has constant call drops, then the IP gateway will only be used by a limited client base.

If it’s difficult for the service provider to install, administer, settle, and bill, it will have longer ROI and will be less likely to be deployed. Incumbent carriers have very specialized billing systems already in place and are not likely to create an entirely new billing system just for IP telephony. An SS7-based solution allows legacy OAM&P systems to be utilized saving considerable expense.

The Nuvo 200 is the answer to these challenges because it can be deployed inside the PSTN network with its SS7 call control features. This allows seamless subscriber interaction such as one-stage dialing and instant call setup. It can be deployed by traditional carriers with legacy billing and settlements since it can generate SSP-style CDR records. It can also be used by next generation carriers who may wish to use an IP telephony billing services.

There is aare a wide range of numbers describing both the current size of the IP telephony market and the growth of the market over the next three to five years. While the specific projections vary, even the most conservative analysts are predicting phenomenal growth. The numbers are summarized below.

Probe Research

Potential customers for IP telephony technology and applications include not only those directly involved in the Internet business but also companies in the PSTN carrier space, private networks or intranets, WANs/extranets, and enterprise networks. Examples include:

- IXCs (InterExchange Carriers), both within the U.S. and internationally. Examples include AT&T, MCI, Sprint, GTE, Deutsche Telekom, Telecom Finland, Telecom New Zealand, and Daewoo International. AT&T’s Globalnet division is currently offering IP telephony service, MCI has already begun building PC-based web servers which may support IP telephony, and Sprint has announced its Global One service that offers dedicated bandwidth on demand.

Even telephone carriers not directly involved in the Internet business will most likely offer services that take advantage of IP telephony, such as reduced-rate voice and fax, and store-and-forward fax and voice messaging.

- CAPs (Competitive Access Providers): there are a multitude of small national and international carriers who offer an alternative to the major carriers for long distance minutes. These CAPS now comprise over 30% of the total international long distance market share. The Today, the strongest play for IP telephony in the CAP-side market is in international long distance where arbitrage between the US and nationalized carriers are artificially high. International Callback callback is one example of a high profit CAP market who can improve their ROI using IP telephony. Both voice and fax are major requirements.

- RBOCs (Regional Bell Operating Companies): such as Bell Atlantic, Pacific Bell and US West,. While the recent ruling allowing RBOCs to offer long distance services will be in dispute for quite some time, it is natural to assume that RBOCs will look at packet-based technology as they begin to roll out their inevitable long distance applications. Additionally, through their alliances with ISPs, RBOCs are a natural convergence point for traditional voice communications and data networks.

- CLECs (Competitive Local Exchange Carriers): such as Winstar, Covad, and ICG have a strong reason to adopt VoIP technologies into their networks. Most CLECs are focusing on specific client types such as commercial businesses instead of home subscribers. As such, CLECs are growing rapidly in metropolitan regions while not maintaining any network infrastructure between regions. It is common for a CLEC to have service in Los Angeles and San Francisco but must use AT&T or other IXCs to move voice between the CLEC regions.

With IP telephony, the CLEC can easily connect their regions together over a private IP network allowing low cost intra-Network calling as well as fax services. IP telephony gateways can become the glue to merge their metro PSTN networks together instead of relying on other PSTN service provider who charge an exorbitant rate for access. CLECs are a fast growing market with over 2,500 applications for CLEC status with the FCC.

- ISPs (Internet Service Providers): There are arguably between one and two dozen large ISPs, notably Alternet, BBN Planet, Digital Express, NetCom, PSI, and UUNET/MFS/WorldCom. There are also hundreds to thousands of smaller and single-region ISPs such as PANIX and TIAC, as well as tens of thousands of bulletin board services (BBSs). Many international ISPs are also offering IP telephony services, such as Rimnet of Japan and OzEmail of Australia. Many ISPs are moving towards CLEC status.

- ITSPs (Internet Telephony Service Providers): Representing a new class of service providers, these companies are building global IP networks specifically designed for low-latency traffic, including voice and fax. This emerging market already had established leaders including ITCX, Delta Three, and Concentric Network, and will undoubtedly spawn new competitors in the coming months.

- Call centers Centers and larger service Service bureausBureaus: These organizations will benefit from new business processes that will capture revenue from web-related sources. This is one of the key applications for IP telephony and is discussed in a later section.

As can be seenshown, the IP telephony marketplace opportunities are manyis large and diverse. The variety range of product s and the scope of applications that need to be developed make it virtually impossible for one company product to target the entire market space. Lately, it has been accepted that a general purpose IP telephony gateway is not enough to satisfy the needs of the diverse network requirements.

Mockingbird Networks’ product sole focus are on to provide answers to carriers who need to offer intelligent network (SS7) oriented solutions to IXCs, CAPs, and CLECscarriers . and service providers who This is still a large market opportunity and will require a number of products that are designed for specific market segments.

The Nuvo200Nuvo 200 is designed for for the most important VoIP/FoIP market opportunity today – International Long Distance. a specific market segment which is in international long distance for voice or fax.This market requires a global approach to IP telephony which requires more than the usual IP telephony features. It also requires SS7/C7 signaling, in-band signaling, digital and analog trunking, and a variety of unique network interfaces to third world countries.

Other Nuvo products will emerge later in 1998/99 this year which will address other portions segments of the market. But it is become quite clear to the market, that a general purpose gateway solution is not enough to satisfy the needs of the diverse network requirements that are identified here.

IP telephony is a new technology based upon diverse (PSTN, SS7, IP, etc.) network topologiess, computer telephony technologyintegration (CTI), and new advances developments in call control such as H.323. In order to fully appreciate the value of IP telephony, this section discussion discusses the fundamentals on of basic IP telephony technologywhich should provide a solid basic understanding of the principles behind the Nuvo200Nuvo 200 family.

The architecture of today's PSTN is a direct descendant of the original manned switchboards of Alexander Graham Bell's day. Voice is transmitted in one way: sampled in 8-bit bytes, 8000 times a second, for an aggregate rate of 64 Kbps. The entire telephone network is designed around this rate and for this one type of traffic (voice).

The PSTN uses a circuit-switched architecture in which a direct connection, or circuit, is made between two users. The circuit provides a full-duplex, or bi-directional, connection with extremely low latency, or delay, between the two end points. This connection was once a physical connection, but is now a logical connection through many switches and across a variety of wiring types (twisted-pair, fiber-optic cable, etc.). The users have exclusive and full use of the circuit until the connection is released.

The fact that the entire PSTN was designed for circuit switching of voice calls has made it very difficult to add new services to the network, or to increase the efficiency of traffic handling. For example, in a paper titled, "The Rise of the Stupid Network," David Isenberg, formerly an executive at AT&T Labs, describes an attempt to improve circuit-switched voice quality as much as possible in the context of the PSTN’s current network architecture:

The PSTN has slowly evolved over the last 100 years from a mechanical switching fabric with analog circuits to a complex mixture of analog and digital circuits with a variety of signaling techniques. In the age of Moore's Law, which states that the power of microprocessors is doubling approximately every 18 months, it took AT&T two years24 months to boost just one part of the signal in the circuit-switched PSTN.

IP Gateways with the PTSN: The typical IP Telephony Gateway is designed to sneak their way into the PSTN by emulating a subscriber who happens to make a cheap long distance phone call for you. The connection is essentially free if it is a local call. They also only take into consideration subscriber type circuits such as FX , T1/E1, and ISDN which offers no SS7 signaling. This approach won’t last as the RBOCs will force IP telephony service providers to behave as IXCs (long distance companies) and will move them to trunk-side circuits. The benefits of free local access will disappear and the need of true PSTN signaling will cause a shake out in the IP gateway industry.

The Nuvo 200 is designed specifically to meet the more complex needs of the PSTN from the trunk side which includes SS7 networks, ISDN, DTMF, MF, and analog circuits. For a global long distance solution, a true carriers perspective must be taken.

The SS7 network is a special private network used for basic call setup, management, and tear down of most of the wireline networks in the PSTN. There are some (third-world) countries which don’t use SS7 yet or use only subsets of the protocol.

Most people rely heavily on the SS7 network without being aware of its existence. SS7 is responsible for basic call control of which we use to place phone calls, receive busy signals and ringing signals. The SS7 network also provides enhanced services which are becoming increasingly reliant on such as:

The ITU definition of SS7 allows for national variants such as the American National Standards Institute (ANSI) and Bell Communications Research (Bellcore) standards used in North America and the European Telecommunications Standards Institute (ETSI) standard used in Europe.

Compared to in-band signaling (non-SS7), SS7 signaling provides: faster call setup times (compared to in-band signaling using multi-frequency (MF) signaling tones), more efficient use of voice circuits, support for Intelligent Network (IN) services which require signaling to network elements without voice trunks (e.g., database systems), and improved control over fraudulent network usage.

- SS7 in IP Networks: SS7 was created to control PSTN voice switches (SSPs) and did not take into account the effect of IP with PSTN networks. A key to IP telephony’s success is to interface the IP Telephony Gateway into the SS7 network in a fashion that allows it to be managed much like a standard PSTN switch.

This is the topic of several IETF BOFs from which a standard might eventually emerge. It may take a considerable period of time before any one protocol becomes an industry-wide standard. For example, the RsVP protocol has been in an IETF committee for over three years and is just now starting to become an adopted protocol by more than two vendors.

In the meantime, leadership companies such as Mockingbird Networks are deploying IP telephony solutions which work seamlessly with the SS7 network. Mockingbird’s SS7 call control agent is software upgradeable allowing interoperability with any potential new protocol which might become widely adopted. Even then it would only be necessary for interoperability purposes with other IP telephony access products as our SS7 call control is fully certified to run in ANSI and ITU networks.

Packet switching is a data transmission technology in which data is assembled into distinct digital "packets" with addresses that are read by switches or routers as the packets are received. The switches/routers forward the packets on to the appropriate destination, but there is not a dedicated circuit connection between the two.

In fact, packets from a particular source may take different routes to the same destination, depending on varying network traffic conditions and other factors. This type of transmission is only half-duplex, or unidirectional, which can easily lead to high delays between sending and receiving. This type of network is highly efficient for data, which can be read into memory and reassembled at the destination.

The non-deterministic nature of packet switching, however, means that some packets will arrive out of sequence and that there must be a mechanism to notify the sending device when a packet is "lost" so that it can be re-sent.

The Internet Protocol is a portion of the ubiquitous TCP/IP suite. TCP/IP was originally developed by the US Department of Defense to link dissimilar computers across many kinds of networks, both local area networks (LANs) and wide area networks (WANs).

Its key feature, providing multi-vendor connectivity, has made it popular among network managers and administrators. The Internet Protocol tracks Internet addresses of nodes, routes outgoing messages, and recognizes incoming messages. In other words, IP provides the addressing needed to enable routers to forward packets across multiple networks.

IP provides a connectionless Datagram service, which means that it attempts to deliver every packet, but has no provision for re-transmitting lost or damaged packets. IP leaves such error correction, if required, to higher-level protocols such as TCP.

IP addresses are 32 bits in length and have two parts: the Network Identifier (Net ID) and the Host Identifier (Host ID). The Net ID is assigned by a central authority and specifies the IP address, unique among all entities across the Internet. The Host ID is assigned by the local network administrator and identifies a particular host, station, or node within a given network, which means that it only has to be unique within the local network.

IP's ability to run over any network medium (Ethernet, FDDI, SONET, ATM, Frame Relay, etc.) has led to its widespread adoption around the world, and it's one of the technologies that have enabled the growth of the public Internet. Its popularity, though, goes far beyond the Internet, to encompass the majority of data networks worldwide.

The IP telephony gateway is is the key component in any IP telephony solution, enabling interoperability between the circuit-switched and packet-switched networks. An typical IP telephony gateway, in its most basic form, provides the following five functions:

A gateway may provide other functions, such as an Interactive Voice Response (IVR) interface to initiate calls, billing software, etc., but the functions outlined above must be performed for any application to work.

When implementing an IP telephony gateway, several features need to be addressed — voice quality, standards, and scalability.

The voice quality provided by IP telephony is an often discussed but largely misunderstood subject. Generalizations such as, "it's unacceptable over the public Internet but okay over an Intranet" seem to be typical. The reason for these characterizations is that it is very difficult to talk absolutely or quantitatively about anything related to the Internet, much less something as subjective as a voice conversation transported over it.

In fact, voice quality achieved by IP telephony can vary greatly and depends on a myriad of factors. Gateway equipment, phone systems, client software, telephone carrier(s), ISPs and even time of day impact the performance of an IP telephony call. The issue of voice quality can be broken down into three interrelated parts:

Intelligibility is mainly determined by the choice of the vocoder, the device that converts analog voice signals into digital signals, which are then converted back into speech sounds. The major IP telephony vocoders in the market today include:

· G.711: The G.711 algorithm encodes non-compressed speech streams running at 64 Kbps. This is toll-quality speech, equivalent to voice of the PSTN network, and requires the full-bandwidth of traditional circuit-switched voice channels.

· ITU G.723.1: This International Telecommunications Union (ITU) algorithm runs at 6.4 or 5.3 Kbps and uses linear predictive coding and dictionaries which help provide smoothing. The smoothing process is CPU-intensive, however (30 MIPS on an Intel Pentium), which means that a scalable solution requires substantial computing power.

Among ITU speech encoders, G.723.1 delivers toll-quality performance at the lowest bit rate. It also specifies the generation of comfort noise frames during silence periods, which greatly enhances the perceived quality of speech. G.723.1 has been chosen as the default speech encoder for IP telephony by the International Multimedia Teleconferencing Consortium (IMTC) Voice over IP (VoIP) Forum.

· ITU G.729A: This algorithm runs at 8 Kbps with a 35 ms total system delay. It provides near toll-quality performance and is ideal for applications requiring high-quality speech coding with low delay. G.729A has been deployed for many years as the speech encoder of choice for the Frame Relay market. G.729A is the alternate default encoder for IP telephony chosen by the IMTC - VoIP.

· Voxware MetaVoice RT 24: The MetaVoice RT 24 algorithm is a 2.4 Kbps, real-time voice encoding algorithm for applications requiring minimal processing power and high-quality speech. MetaVoice is designed to ignore ambient noise (such as traffic in the background), deliver cleaner sound, and reduce the amount of data analyzed by the compressor.

Voxware’s MetaVoice coding technology allows excellent speech quality in a fraction of the bandwidth of other vocoders.

· Voxware SC6: Voxware has also developed the new, scalable SC6 vocoder which runs at 6.4 Kbps and is designed to run on high-density systems, including carrier-grade gateways.

· Elemedia SX7300P: The Elemedia SX7300P vocoder offers near toll-quality speech and, with a rate of 7.3 Kbps, enables speech transmission over either V.34 or V.32bis modems. The small frame size of the vocoder provides low end-to-end delay. The SX73000P supports a frame reconstruction algorithm that maintains high performance in the presence of channel errors. The SX7300P offers speech quality comparable with ITU G.723.

· MS-GSM: This algorithm, marketed by Microsoft, runs at 13 Kbps and is a derivative of the ITU (International Telecommunications Union) GSM standard. GSM is used in 85 countries around the world as the standard for digital cellular communications.

Microsoft’s implementation varies from the standard in several ways including how the encoded data is represented and what aspects of the encoder are supported.

The transmission of uncompressed speech data consumes a great deal of network bandwidth. This is why a vocoder is used to compress speech before it is transmitted and decompressed it when it arrives at its destination. To do this, however, the vocoder must buffer the data briefly to evaluate speech segments. A small delay is incurred as the vocoder "looks ahead" during its mathematical computations. Typical ‘low delay’ vocoders look ahead 15 to 45 ms for this purpose. The combination of buffering delay and vocoder delay is often called algorithmic delay.

The vocoder also introduces some additional delay while it conducts the actual computations that compress speech for transmission. Those calculations are conducted on the computer processor, on which the vocoder is running, for example a Sun Solaris UltraSPARC processor or Lucent DSP, and they consume actual time; it does not happen instantaneously. The faster the processor, the lower the delay. The time required to conduct these calculations, and the system delay incurred as a result of it, is known as compression delay.

- Voice Quality: Managing compression delay is a function of a gateway architecture, especially as a gateway scales beyond a few ports of voice to multiple T/E spans in a single gateway. Additionally, because vocoders only approximate the analog waveform of spoken speech, there is some degradation in the quality of the sound when it is reproduced from the compressed format. The quality of the most popular vocoders has been measured by many groups, typically using the Mean Opinion Score (MOS).

On the MOS scale, zero equals the worst quality and five is the highest. The MOS scores, bit rate, and sample size for standard VoIP vocoders and a few of the more popular de facto standard vocoders are shown below.

Algorithms that run at lower bit rates take longer to encode speech than those that run at higher bit rates. In general, a lower bit rate means more potential delay. Typically a balance must be struck among voice quality, latency and bit rate. This balance is a function of the applications and the network conditions.

In many countries, the transit network is built entirely using four wires (any digital link is a virtual four-wire link). In other countries the "last mile" is covered using only two wires. The two-wire to four-wire separation occurs at the local switch where the phone is connected. If the impedance between these two portions of a network does not exactly match, then a portion of the incoming signal is fed back in the outgoing signal. This parasitic signal is the hybrid echo, which can have the following consequences:

• In some configurations, signals will loop between the two amplifiers and create an echo if the one-way delay is about 20 ms. To avoid instability in the network, a loss of 6 dB at least is introduced in the 4-wire path.

• A phone with a mismatched impedance will echo all incoming signals from the network, and the speaker at the other side of the line will hear an echo after a round trip delay.

In most cases, echo cancellation is required for intelligible speech across IP networks. For phone-to-phone communications a special type of hybrid echo cancellation is required — far-end or tail-end echo cancellation. This type of echo cancellation provides a very short but effective removal of return speech echo at the far end of an IP telephony call. Speech cannot be understood unless this feature is supported in the IP telephony gateway.

Delay, or latency, between talkers is one of the most crucial factors in determining the subjective quality of any phone call. In IP telephony, minimization of latency is the primary goal of the gateway design engineer and the network architect. The total end-to-end delay between two telephones connected over IP is equal to the sum of the delays through the gateways and the delay in the IP network between the gateways.

- Gateway Delay:

There is generally very little delay in the PSTN or PBX interface to the gateway. An incoming analog signal is digitized to 64 Kbps Pulse Code Modulation (PCM), or is already a digital signal in the case of T1 or E1, and is passed on to the compression subsystem.

Delay may be introduced in the buffering between the telephony subsystem and the compression subsystem, especially if the compression is not performed on a dedicated DSP processor already servicing the telephony interface. A DSP which is handling the incoming PCM samples can also perform the voice compression task without introducing buffering and data transfer latency.

Beware of compression that is performed on processors located across a bus and running on a different operating system from that of the telephony interface. Potential delays may show up in the data handling between the subsystems. An implementation created out of programmable DSP hardware, TDM buses, and dedicated IP processors will scale without increasing latency as users are added.

Another component impacting speech delay is the choice of vocoder. The G.723.1 algorithm, for example, is based on a 30 ms frame size and contains an additional algorithmic delay of 7.5 ms. This means that it takes a minimum of 37.5 ms for a PCM sample from the telephony interface to be encoded and passed out of the compression engine.

Many gateways also perform echo cancellation, DTMF tone detection and regeneration, and fax detection in the compression subsystem. An optimized DSP software architecture is critical to support the operation of these functions while maintaining low latency.

The IP subsystem is also a contributor of delay in a gateway. The principle job of this subsystem is to packetize the compressed voice data received from the compression engine and send it out of the gateway onto the data network. Historically, such packet protocol engines are optimized for file transfer and work on relatively few large data packets and do not interface directly with real time data streams.

To achieve high quality IP telephony communications, a gateway must send a full-duplex stream of many small data packets in real time to satisfy the time division multiplexed telephony interface. This packet-to-circuit, or asynchronous-to-synchronous, translation is implemented by adding buffers between the subsystems.

Again, minimization of delay in the gateway is driven by the implementation details of these system-buffering issues. Key architectural features that allow high performance in the IP subsystem include use of a deterministic operating system, dedicated processor(s) running protocol stacks, and dedicated hardware interface(s) to the compression engine.

- Network Delay: y

Latencies in the IP network are the largest unpredictable variable in IP telephony. IP has become a nearly universal protocol for sending data among computers, enabling the Internet to spread rapidly throughout the world. This explosive growth has lead to an infinite number of configurations and routing possibilities. Today, it is virtually impossible to predict delays between gateways arbitrarily connected to each other over the worldwide Internet.

When transporting voice over the Internet, there is limited information about, and basically no control over, the path the voice packets might take between any two gateways. There may be long delays, high packet loss and variable packet jitter (out of sequence packets, lack of synchronization); there may be good performance; or the performance may change over time.

In an Intranet configuration, however, there is very good information about, and significant control over, the path the voice packets take between gateways. An Intranet, a well-defined network of routers and switches, might be a corporation's private data network, or it could be a virtual private network (VPN) service purchased from an ISP providing premium service. In either case, the knowledge and control of this IP transport allow the creation of a high-performance IP telephony network.

- Packet Delay:

Packets transported on an IP network are delayed passing through each router they traverse. The delay in a router will depend upon its configuration, performance, capacity and load. A rule of thumb is that each router introduces about 10 ms of latency.

Many factors can negatively impact this number, most notably a high volume of large data packets, which arrive at the router simultaneously with the IP voice traffic. However, in a managed network it is possible to prioritize IP telephony ports over generic data ports, or to reroute this time-sensitive traffic to minimize router latency.

In the same way that packets are delayed through a router, the possibility exists that packets may be lost in a router. Routers are designed to transmit all incoming packets to the correct outgoing port, but an overloaded router can choose to drop entire queues of IP packets. The loss of packets is detrimental to the performance of an IP telephony call. Although most gateways employ vocoding algorithms and techniques to handle this possibility, packet loss of more than 5% has a significant adverse effect on the quality of the conversation. Again, through network planning and management, packet loss can be minimized and often eliminated.

IP network performance over the Internet will always be highly variable and often unacceptable. On the other hand, IP voice quality over a managed network, k or Intranet, or Virtual Private network (VPN) can be very good and relatively quantifiable.

In the past, Intranets have been implemented using expensive leased line networks by large corporations and government agencies. But increasingly, ISPs and data communications providers are offering cost-effective virtual private networking services with guaranteed quality of service and round-the-clock maintenance.

The ability to co-locate a gateway in an ISP's point of presence has also become an inexpensive way to minimize IP network latency, eliminating a couple of routers at each end of the IP telephony network. Given the proliferation of IP networking service providers and product offerings, it is now possible to create an IP telephony backbone with relative ease.

Costs are not prohibitive and packet delay, loss and jitter can be managed to provide excellent quality around the world. In such a managed network, delays can be kept below 100 ms, packet loss under 3%, and jitter below 60 ms, resulting in high-quality conversations.

Recognizing the limitations of available IP telephony approaches and the potential for building a truly scalable, high-performance product, Mockingbird Networks created the Nuvo family of products. Nuvo is a hardware and software platform that can be used to engineer and build scalable, high-quality IP telephony systems within a robust Unix platform, thus greatly simplifying development and deployment.

The Nuvo 200 is the latest member of the Nuvo family and offers greater scalability, higher density, and improved call control features than the original Nuvo 100. Future variants of the Nuvo family will target other market segments. Regardless of the variant, all Nuvo products share very important common technologies.

Nuvo products are very different from the average gateway platform. The most distinguishing advantage is its SS7 call control features. With the NX-series of call control servers, the Nuvo architecture appears to the PSTN’s SS7 network as a Class-4 SSP. But instead of being just a point in a circuit-switched call, the Nuvo architecture functions as an IP telephony gateway, compressing voice, packetizing it, and sending into a packet environment to a peer device which reinserts the call back into the PSTN.

The advantage of the Nuvo approach to Plain Old Switching (POS) is the return on investment time period. By compressing voice, using low cost transmission networks and lower or no arbitrage costs, the Nuvo architecture realizes its ROI in a matter of months.

The Nuvo 200 is a distributed architecture, where the SS7 call control is physically separated from the trunking interfaces. The separation of call control from the media gateways were designed to allow carriers to build a more scaleable and flexible solution.

This also allows the network designer to deploy the Nuvo 200 in either SS7 or non-SS7 environments. For example, its possible to support SS7 in one point of a Nuvo network while offering only in-band signaling in other locations. It’s also possible to offer ANSI-variants in the United States while signaling to a European Network which is an ITU variant. These are important distinctions from the typical IP telephony gateways which are really just subscriber-side solutions.

The NX-series Signaling Gateways offer varying degrees of SS7 scalability, redundancy, and performance. The NX1000 is the most basic server and can either be deployed as an individual or tandem (redundant) server. The NX-2000 offers more than twice the compute power and is always redundant. The NX-4000 is the highest performing SS7 call agent and includes features such as RAID to allow the deployment of SCP-oriented databases.

The TX-series Media Gateways are the physical trunking interfaces between a PSTN switch and an IP network. The TX-200 is the current offering which is an 8U NEBS-compliant platform with up to 4 T1 or E1 interfaces, DSP processors, and IP interfaces.

A new version of the Media gateway will be released later which can offer Analog interfaces for points of presence where T1 or E1 are not available.

The Signaling Gateways interface to the Media Gateways over a redundant IP network which handles the call control traffic.

A specialized SS7 call agent identified as TXG-NX is the heart of the NX-series Signaling Gateways. TXG-NX is a tandem (redundant) switchless SS7 call control engine. It is a pioneering approach to building SS7 IP telephony since it doesn’t require an open programmable switch such as the Summa-Four VCO.

Instead, TXG-NX interoperates with open systems platforms which use open switching technology such as MVIP, SCSA or H.100/H.110 (MVIP is used in the TX 200 Media Gateway). The elimination of the programmable switch reduces the cost of an SS7 solution by a wide margin.

The Call Agent can control a large number of Media Gateways via a redundant Ethernet network. The Media Gateways are usually located in the same location as the SS7 Call Agent but the SS7 Call Agent may be located remotely controlling Media Gateways over several PSTN locations.

Each Call Agent is uniquely identified by a numeric point code in the SS7 network. Point codes are carried in signaling messages exchanged between signaling points to identify the source and destination of each message. Each signaling point uses a routing table to select the appropriate signaling path for each message.

There are three kinds of signaling points in the SS7 network

· SSP (Service Switching Point)

· STP (Signal Transfer Point)

The Signaling Gateway combined with the Media Gateway perform the same function as an SSP in the PSTN. SSPs are switches that originate, terminate, or tandem calls. An SSP sends signaling messages to other SSPs to setup, manage, and release voice circuits required to complete a call. An SSP may also send a query message to a centralized database (an SCP) to determine how to route a call (e.g., a toll-free 1-800/888 call in North America).

An SCP sends a response to the originating SSP containing the routing number(s) associated with the dialed number. An alternate routing number may be used by the SSP if the primary number is busy or the call is unanswered within a specified time. Actual call features vary from network to network and from service to service.

TXG-NX also contains hooks for TCAP, IS-41, and GSM signaling protocols. It is possible for Mockingbird to offer additional enhanced features very quickly with the TXG-NX such as interfaces to wireless networks in TDMA and CDMA networks, and TCAP to IP routing.

Contact Mockingbird for more information regarding enhanced services beyond the ISUP-IP signaling that is standard in the TXG-NX Call Agent.

TXG-NX is supported by a family of UltraSPARC Signaling Gateways all which are NEBS compliant. These platforms are specialized severs which perform the function of earlier fault tolerant technology such as Tandem or Stratus computer..

It is necessary to choose the correct Signaling Gateway when planning a network deployment. The NX-2000 and NX-4000 can process between 64-256 calls per second and are field upgradeable to support more processors to handle the higher call rates. The NX-1000 is better suited for smaller networks where system cost is a major consideration.

The NX platforms are based upon the UltraSPARC II architecture in single, dual, or quad processor configurations, the NX-series also offers a wide variety of network capacity and features. All are based upon proven Sun Microsystems hardware in Mockingbird designed platforms.

High Reliability without Fault Tolerance: All NX Signaling Gateways have built-in redundancy at hardware and software levels. The Signaling Gateways provide redundant SS7 links to the SS7 network and redundant network links to the Media Gateways The products are designed with redundant processor, disk, and application redundancy to perform with the reliability that’s required in a telecom network.

As such, these are heavy duty call processors in the much same manner as a Stratus or Tandem platform. The main difference between a Tandem K1000 NEBS computer and the NX-2000/NX-4000 is one of design approach and cost.

-The market for lock-step fault tolerant computers has dwindled due to the massive proliferation of unproven software and high quality hardware. The original lock-step approach worked well when software was well behaved and mode failures of components were common. Today, failures occur much more often in the software-side of a computer. That’s because software is much more complex than it used to be. In a hardware lock-step machine, many software errors will bring a lock-step computer down on its knees.

The NX-series takes a loosely coupled high availability approach which prevents poor software design from bringing down the network. When hardware failures occur, the SS7 services (in redundant platforms) guarantee no loss of service. When software failures occur, the system is able to make an appropriate response instead of locking up.

This approach is not only superior to the legacy fault tolerant computers, it is significantly less costly.

- NX-1000 Signaling Gateway: The NX-1000 is a low profile (4U) high performance UltraSPARC IIi platform. It can be initially deployed with a lower cost non-redundant SS7 Call Agent and can be upgrade later to full redundancy with an additional NX-1000 interlinked via a private network.

- NX-2000 Signaling Gateway: The NX-2000 our mid-range platform with twice the compute power of the NX-1000 and has built in redundancy of all components. Each major component is field replaceable/upgradeable in a matter of minutes and can be done without bringing the machine offline.

- NX-4000 Signaling Gateway: The NX-4000 offers twice again the performance of the NX-2000 and is designed for the most demanding network configurations. It supports up to four UltraSPARC processors in each of its redundant computing elements. Like the NX-2000 all components are field upgradeable.

The NX-4000 also offers an internal RAID subsystem which allows up to 100 GB of fault tolerant disk space. This is designed to support house billing information, gatekeeper information, and/or user databases for call control.

Mockingbird Media Gateways are typically trunk-side IP telephony gateways which are usually controlled by a Signaling Gateway. They can also be configured to be subscriber-side devices in locations which the Network Designer doesn’t have access to switch trunks.

All Mockingbird Media Gateways are based upon the Sun Microsystems UltraSPARC technology. This is an important differentiation from the typical gateway which is usually based upon Windows NT and sometimes supports an embedded operating system as well.

The Solaris operating system is the primary driver why the media gateways use the Sun UltraSPARC architecture. Solaris for Intel is not nearly as robust in third party applications which may become necessary or useful in Media Gateways.

The typical Mockingbird Media Gateway consists of five logical components which are:

• Host Processor
• PSTN Interface (T1/E1, FX0/FXS)
• Compression Engine
• IP Interface (Ethernet, T1/E1)
• Redundant IP interface to Signaling Gateway

The host processor is a high performance UltraSPARC IIi processor which runs the Solaris operating system. This Unix approach is much more robust than a Windows NT platform requiring a typical reboot of once per year instead of the typical instability of NT which is approximately 10 times less stable.

This is important in a network access device since down time needs to be measured in minutes per year. Unix platforms have been long established as the only choice for telecommunication environments.

Also, the Solaris operating system has superior multi-threading capabilities which allows significantly higher performance than the same processor running an NT operating system. Some gateway vendors who run on the Intel platform have reported a near doubling of performance when running Solaris vs. NT in the same gateway application.

The PSTN Interface is how the Media Gateway interconnects into the PSTN, as well as breaks that connection into more manageable parts called DS0’s.

Depending on each customer’s requirements, the PSTN Interface needs to fulfill various needs. It can be deployed inside the CO on the trunk side, at a local ISP POP on either trunk or subscriber side of the PSTN switch, or at remote POP on the subscriber side of the PSTN.

Each of these locations has different requirements. In order to address as many configurations as possible, flexibility is important.

The Compression Engine is the part of the Media Gateway that takes each DS0 and has a Digital Signal Processor (DSP), reduce its size into a packet that can be transported over an IP network. The current approach is for there to be the same number of DSPs as there are DS0 that need to be processed. In the case of T1 lines, there would be 24 DSPs; an E1 would have 30 DSPs. This data is reduced in size by using standard compression algorithms.

-The IP Interface: The third stage is where the compressed voice channel is packetize in a UDP packet and sent to the IP interface. The IP interface function is to provide the connection into the IP network, once the voice data is received, processed, and the compressed. With various network topologies that are available today, again, flexibility is key.

The TX-200 Media Gateway is designed for global IP telephony applications in either Internet or IPLD (International Private Long Distance) circuits which are leased from carriers. It offers a wider range of PSTN options than the typical IP telephony gateway. It supports the common digital trunking interfaces such as T1, E1 and ISDN but also supports a wide range of Analog interfaces.

In much of the third world, there are few digital circuits available making it difficult for typical IP telephony gateways to be easily deployed. By providing broader support of these interfaces, the Nuvo 200 address long distance circuits and also regional locations where perhaps only analog interfaces are available.

The TX-200 Media Gateway , the primary media gateway product in the Nuvo 200 family, can support up to 96 DS0’s in each 8U rack platform. In other words, 4 T1 lines or 3 E1 lines. These DSPs can use a variety of compression algorithms, including the following:

The compression algorithms are configurable in the Nuvo 200 on a call by call basis. This is allows the network designer with flexibility in providing a range of voice quality to their subscribers as well as allow their circuits to handle higher volume calls when necessary.

In its default configuration, the TX-200 provides a 10/100 Base-T interface. But due to its expansion capabilities, the TX-200 can support a variety of networks, including FDDI, ATM, Gigabit, Frame Relay, or even a T1/E1 connection (data grade, not voice grade).

The TX-200 offers the following features: These features include :

• Call Detail Records (CDR)

• System Status Monitoring
• Call Utilization Monitoring (by hour and by day)
• Bandwidth Utilization
• Per Port Status, including compression code selected, packet size, and type of call
• ANI, DNIS support
• Flexible T1 signaling, including E&M Wink start, FXS, and Feature Group D
• Two Stage Dialing

The US domestic long distance market is saturated with service opportunities which make IP telephony truly useful only when VoIP networks provide more service offerings than circuit-switching. Some companies are offering VoIP services based only on price, but these prices will be quickly met by the traditional carriers in a short period of time.

It’s the opposite for global long distance where monopolistic telecom providers still dominate many countries. In general, consumers in these countries experience high rates, poor quality service, and few features, and even fewer alternative. American alternative service providers can gain a foothold in these new markets by offering lower rates, better service, and more service offerings.

According to the Yankee Group, wholesale alternative service long distance services constitute a market with estimated 1997 revenues of $5.4 billion, growing at a rate approximately between 10% and 15% annually.

In spite of the uncertain economic conditions, telecommunication needs are growing. Innovative products and lower prices will continue to win market share. Growth rates exceeding 8% are common in developing countries and are exceeding that in such countries as China, India, Brazil, Malaysia, and Chile.

International callback is a popular service for this very reason as a CAP can offer real telco-class service by re-originating a call from the US. In 1997 international callback accounted for $1.6 billion in telecom revenue and that growth will increase by 25% in 1999.

For example, one alternative service provider offered callback services to western Europe and South America, their rates were on average four times lower than the comparative rates offered by local PTTs. By offering lower rates without sacrificing service, Justice was able to grow over 30% per month in these countries.

The FCC supports callback and other forms of re-origination in countries where it is legal. It will help support companies who follow US laws. The FCC is vigorously fighting the EU’s decision to apply a 15% Value Added Tax on callback services!

IP telephony is not callback as a call can originate in the country the call is made from. The difference is that the call is bypassing the monopolistic carrier via a local call which reroutes the call over a leased line or an IP backbone.

There are numerous services now being created that are VoIP networks constructed of the traditional IP telephony gateway. These gateways though are hard to use and do not offer the service quality of international callback.

They usually can’t meet the service quality because voice is moved over the internet much like VON technologies. Delays are unacceptable due to the small IP pipes that exist which cause severe packet drop. When more than 15% of packets are dropped the conversation become unintelligible.

IP/SSP Telephony is a new concept which allows an IP telephony device such as the TX-200 appear to the PSTN as a Class-4 SSP. The Nuvo 200 is one of the few IP telephony products that can meet all of these criteria and is available today without reverting to a traditional PSTN switch service.

This is due to the service features of the Nuvo 200 which allows it to interface directly into the PSTN. It is possible for existing and new alternative providers to easily upgrade their networks to support the Nuvo 200 without overhauling their network.

The Nuvo 200 family offers the following features:

A CAP is much more than an ISP with IP telephony gateways running off the Internet. A CAP is a true alternative carrier who offers high quality phone to phone service over managed circuits such as T1 or E1, SS7 interconnectivity options, and a wider choice of PSTN connectivity to suffice the needs of many third world countries.

It’s easy for a CAP to become an IP/CAP since the relationships, circuits, marketing and infrastructure are already there. A Nuvo 200 solution can be easily installed in the CAPs existing network usually in the US with an SS7 connection and over a E1 circuit to an inband connection in foreign markets.

If not already a CAP, this takes an effort that goes beyond the scope of this this White Paper. But a few tips might be useful:

• In countries with telecom monopolies, expect significant resistance from the government. Be creative in overcoming obstacles.

• Provide services licensed or approved by the US government. Do not perform illegal actions under any circumstances.

• One of the most important decisions facing upstart firms is choosing the appropriate underlying carrier.

• With deregulation in the telecom community spreading out to all destinations, long distance has become viewed as a user-friendly commodity. As a result, a fine balance must be struck in order to acquire dependable service at an affordable price.

• Form strategic partnerships with complementary companies and agents Agents based in the country will know the market better than an outsider company. It is advisable to enter into strategic partnerships with companies and agents located in the country. They will know how best to exploit opportunities in these markets. The process is reciprocal. US telecom companies can offer significant resources and attractive compensation plans, while leveraging the sales savvy of market insiders.

• Obtain demographic and psychographic information on consumers in targeted countries. This can be obtained through field research and partnerships with in-country companies and agents.

For companies with established long distance services, the return on investment (ROI) for installing IP telephony gateways at multiple sites is a simple calculation. Some companies are reporting ROI of six months or less.

The Nuvo 200 can offer both VoIP and FoIP or a mixture in the same network. Its universal port architecture will allow each call to be automatically detected for voice or fax and setup the call in an appropriate manner.

Conclusion: IP telephony is now in its infancy but, by all indications, will quickly become a major factor in the telecommunications industry. IP telephony represents a phenomenal business opportunity that can be tapped with open, standards-based products such as Mockingbird Network’s Nuvo Family.