HERE IT IS!!
Internet Telephony Basics
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The Internet is a worldwide network of computer networks. Computers can send messages to each other via the Internet - email is the most familiar example. However, audio can also be digitized and passed between computers just like any other data type. In fact, many Web sites are now sound-enabled.
Internet telephony uses the Internet to send audio between two or more computer users in real time, so the users can converse. Vocaltec introduced the first Internet telephony software product in early 1995. Running a multimedia PC, the Vocaltec Internet Phone (and the numerous similar products introduced since) lets users speak into their microphone and listen via their speakers.
Within a year of its birth, Internet telephony technology had caught the world's attention. The technology has improved to a point where conversations are easily possible. And it continues to get better. Dozens of companies have introduced products to commercialize the technology, and virtually every major telecommunications company has launched research to better understand this latest threat to their markets.
In March of 1996, Vocaltec announced it was working with Dialogic to produce the first Internet telephony gateway. The original Internet telephone products based on multimedia PCs are tremendous -- offering the ability to combine voice and data on one network. They also offer low-cost long-distance 'telephone' service (assuming the user already has a multimedia PC and a fixed-rate Internet service provider account).
Gateways are the key to bringing Internet telephony into the mainstream. By bridging the traditional circuit-switched telephony world with the Internet, gateways offer the advantages of Internet telephony to the most common, cheapest, most mobile, and easiest-to-use terminal in the world: the standard telephone. Gateways also overcome another significant Internet telephony problem, addressing. To address a remote user on a multimedia PC, you must know the user's Internet Protocol (IP) address. To address a remote user with a gateway product, you only need to know the user's phone number.
How Does It Work?
Conceptually, Internet telephone gateways work like this:
On one side, the gateway connects to the telephone world. It can communicate with any phone in the world. A phone line plugs into the gateway on this end.
On the other side, the gateway connects to the Internet world. It can communicate with any computer in the world. A computer network plugs into the gateway on this end.
The gateway takes the standard telephone signal, digitizes it (if it is not already digital), significantly compresses it, packetizes it for the Internet using Internet Protocol (IP), and routes it to a destination over the Internet.
The gateway reverses the operation for packets coming in from the network and going out the phone.
Both operations (coming from and going to the phone network) take place at the same time, allowing a full duplex (two-way) conversation.
A number of configurations can be built from this basic operation. Phone-to-PC or PC-to-phone operation can take place with one gateway. Phone-to-phone PC operation can occur with two gateways. To offer international long-distance service using gateways, for example, an organization or service provider can host one gateway in each country. By bypassing the international connect charges -- even paying in-country long distance rates -- the configuration costs significantly less than traditional circuit-switched service.
How Well Does It Work?
Nothing replaces trying it for yourself. However, we can make some general observations. There are two main factors contributing to quality: voice quality and turnaround time, or latency.
Voice quality has improved greatly from early versions of the technology, which were characterized by distortions and disruptions in speech. Improved technologies for voice coding and lost packet reconstruction have yielded products where speech is easy to understand.
Latency affects the pace of the conversation. Humans can tolerate about 250msec of latency before it has a noticeable effect. Today's Internet telephony products exceed this latency, so most connections sound like traditional calls routed over a satellite circuit (which are usable, but require some getting used to). Even today, the products are well suited to many applications. Moreover, the latency will continue to improve, driven by three factors:
Improved gateways. Developers are just beginning to squeeze latency out of the first generation of products.
Deployment over private networks. By deploying gateways on private circuits, organizations and service providers can control the bandwidth utilization and, hence, latency.
Internet development. Today's Internet was not designed with real-time communication in mind. The Internet Engineering Task Force (IETF),together with Internet backbone equipment providers, is addressing this with technologies like Reservation Protocol (RSVP), which will allow bandwidth to be reserved. While it will take some time for the world's routers to be upgraded and operational aspects (like how to bill to high quality of service) to be resolved, the Internet word is moving fast -- and in the right direction.
Open Versus Proprietary Systems
One of the key factors for Internet telephony gateway developers to consider is the value of open systems versus proprietary systems. It is tempting to develop proprietary versions of new technology where off-the-shelf components are not readily available. However, component vendors like Dialogic have been able to respond to the demands of Internet telephony quickly, modifying existing products to address the needs of the Internet telephony gateway systems. These vendors are also continuing to pour research and development money into enhancing their components.
The general advantages of open systems design are overwhelming. Competition -- at all levels - leads to lower prices, enhanced features and continual innovation. Since system integrators need to excel in fewer aspects of system design, costs fall even more.
The advantages of open systems are particularly compelling for Internet telephony. The impact of the Internet on telephony is not as a standalone system or feature. It is fundamental and systemic. New generations of telephony systems will evolve to better incorporate Internet capability. These new generations will be built using open systems and standards.
Choosing a Component Supplier
To build open systems, system integrators must chose component vendors with products that meet their technical requirements. Even more importantly, the component vendor must be committed to this new market and must demonstrate the ability to adapt on-the-fly to its changing requirements. And since so many Internet telephony systems are global, the vendor also needs a worldwide network of service.
Architecture
For telco-grade installations, system integrators will consider VersaModule-Europe (VME) or Compact Protocol Control Information (PCI) system designs. Using equipment that meets Bellcore's Network Equipment Building Standards will improve the installation and maintenance in many facilities. Customer premise equipment can be hosted in an Industry Standard Architecture (ISA) or PCI chassis. Windows NTT and UNIX are both suitable operating system choices.
Telephone Interface
The telephone connection of the gateway needs to exhibit two critical features:
There must be approved versions in all major countries, since the largest cost savings for Internet telephony is on international calls.
It must be scalable. Depending on the design goals of the system integrator, systems might range from two lines for small enterprises to several thousand for service bureaus.
Call Control Protocol
The first Internet telephony products used proprietary call control protocols. H.323, however, is clearly emerging as the standard call control protocol. This spec defines packet standards for terminal, equipment and services for multimedia communications over LANs communicating to systems connected to telephony networks such as ISDN. It will be supported by successful Internet telephony products.
Voice Coders
Key technical requirements for coders include:
Low bandwidth (8 kpbs or less)
High quality for voices (3.5 mean opinion score (MOS) rating or better)
Low latency
Ability to reconstruct lost packets
In real-time transmission, up to 30% of the packets in a transaction might be lost or delayed (which is the same as lost in real time). Successful Internet telephony applications, then, need to recover from lost packets by effectively reconstructing the lost data. The complexity of the coding algorithms has an impact too. High complexity increases the cost of the host platform.
G.723.1 is emerging as a popular coding choice. G723 is an algorithm for compressed digital audio over telephone lines. The enduring requirement for coders, however, is that Internet telephony systems be capable of supporting multiple coders and adding more as technology emerges and popularity changes.
Echo Cancellation
The Internet telephony gateways must perform echo cancellation.
In a typical configuration, two gateways are each connected to analog phones via digital, local central office switch. The phone system generally does not perform echo cancellation on local circuits. Echo is present (caused by the four wire to two wire hybrid), but is not a problem on local calls. The latency is not long enough for the echo to come back as a separate transmission. The phone system does perform echo cancellation on long-distance circuits. By the time the echo propagates through the network back to the speaking part, it is quite disruptive.
Internet telephony represents a unique case. Technically, local connections are being used. Hence, the phone system itself is not performing echo cancellation. But long distance calls are being made. Hence, the echo will disrupt conversations if it is not canceled. The Internet telephony gateways, then, must supply the echo cancellation.
Full Duplex
Phone calls are full duplex, meaning both parties can speak at the same time. Successful Internet telephony products are also duplex. Surprisingly, not all voice cards can support full duplex operation.
DTMF Detection and Notch-Out
Dual tone multi-frequency (DTMF) digits do not travel well across the Internet. Coding and packetization distorts and segments them, making them unrecognizable on the remote end. Internet telephony gateways, then, must detect DTMF digits locally, suppress their transmission, then generate them on the remote side.
There is not yet a standard for passing DTMF control information across an Internet telephone call; however, Dialogic is monitoring the efforts of various standards groups and expects such a standard to emerge soon.
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