INTRODUCING PRACTICAL IP TELEPHONY INTO CORPORATE INTRANETS

Transcription

1 DATA COMMUNICATIONS MANAGEMENT INTRODUCING PRACTICAL IP TELEPHONY INTO CORPORATE INTRANETS John Fiske INSIDE Internet Voice Technology Basics, Gateways, Open versus Proprietary Systems, Architecture, Call Control Protocol, Voice Coders, DTMF Detection and Notch-Out, Bandwidth Utilization, Enterprise Deployment INTERNET VOICE TECHNOLOGY BASICS With the Internet promising to offer a new way to do business, the emerging field of Internet protocol (IP) telephony is poised to bring major advantages to enterprise communications. Although the telephone network and data networks have existed side by side for decades, IP telephony is driving them to converge into a single communication network that offers powerful, economical new communications and business opportunities. Currently, real-time voice and fax traffic is primarily transmitted via public switched telephone networks (PSTNs) and customer-provided private networks. Although these transmission media offer a very reliable solution, they may be more expensive than packet networks, such as an Internet Service Provider s (ISP) Internet service, a company s intranet, or Internet pipes. This would be especially true for multinational and global concerns, where the difference between PSTN and Internet pricing is the greatest. Many companies today support the infrastructure and cost of multiple networks including PSTN, private, wireless, intranet, and Internet facilities. The need to optimize the usage of all media components and to take advantage of pricing alternatives among the networks is becoming even more important. Corporate telecom managers, data managers, ISPs, PAYOFF IDEA IP telephony using the Internet for real-time voice and fax traffic is quickly becoming a viable alternative to public switched telephone networks and private networks. Knowing the direction this technology is heading will facilitate planning for later deployments, which may come sooner than expected. 06/98 Auerbach Publications 1999 CRC Press LLC

2 cable operators, and telco managers seek tools to allow transmission of all media types across their network media in order to maximize the use of their infrastructure. These tools are the software that will route voice calls the most important kind of communication in commerce transparently across the most cost- and performance-effective network, without risking the mission-critical nature of the real-time call. GATEWAYS The gateway is the new, and vital link in IP telephony. In Exhibit 1, the Phone System or PBX can be either an internal switch at the company or the CO (central office). In the case of an internal PBX, a strap cable connects the T1 port on the gateway and the T1 port of the switch. In the case of a connection to the CO, a local loop T1 connection (or E1, ISDN, Euro ISDN, or analog) is needed. When the caller picks up the phone and dials a number (long-distance area code or country code), the corresponding IP address in the software preprogrammed dialing profile is looked up in a routing table that is part of the gateway software. The routing table sees the phone number dialed and points the gateway (compressed and digitized call) to the destination IP address where there is a receiving gateway. The result is that the medium at which the call travels is transparent to the originating caller as well as the terminating caller (person receiving the call). The call initially is analog. The gateway takes the standard telephone signal, digitizes it (if it is not already digital), significantly compresses it, packetizes it for the Internet using IP, and routes it to a destination over the Internet. The routing table sends the data to the destination server, where it is changed back from TCP/IP data to analog. The destination gateway server finds a local dial tone and completes the call as a local analog call, thus bypassing the PSTN. The transport of the call can be over any data network, including 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 gateway is a PC, usually modestly powered with a 200 MHz Pentium, 32 MB RAM, CD-ROM, and 2-Gbyte hard drive. Inside the gateway there are specialized boards for connection to telephone PBX (any of the flavors), T1 or analog domestically, or E1 in Europe. With a full T1 connection, a strap cable from the T1 card on the gateway to the T1 card on the PBX would be run, which allows least-cost routing (LCR) on the PBX. The same is done at all points on the wide area network. With LCR, users from their desks experience basically what they experience now. They pick up the phone and can either dial an extension or dial a long-distance number.

3 EXHIBIT 1 Gateway Diagram

4 Within the gateway there is a dialing profile that takes the dialing information and routes it through the LCR mechanism on the PBX, and points to the IP address at the other server. The gateway compresses the analog voice to 13K, adds the IP header information, which is the terminating telephone number, and designates the server on the called end where the server must map to the end system phone. When a call goes intracompany (to a PBX extension), the dialing profile would be sent to an IP address of the extension and terminate right at the switch, rather than going out to a local dial tone. If one wanted to terminate a call somewhere that the network did not go, then one could point to a network s IP address that has a gateway in that area code (or geographical region). A global community of IP networks has formed an IP telephony consortium for this very reason. The terminating gateway levies a charge (fee), which is typically dramatically lower than long-distance/international charges. Gateways also provide another significant IP telephony service: IP addressing. To address a remote user on a multimedia PC, the user s IP address must be known. With a gateway, only the user s phone number needs to be known to address a remote user. If a number that has no IP address is dialed, the gateway recognizes that and sends the call to the outside PSTN. The gateway s telephone connection must exhibit two critical features: 1) there must be approved versions in all major countries, because the largest cost savings for IP telephony is on international calls, and 2) it must be scalable, from two lines for small enterprises to several thousand for service bureaus. A number of configurations can be built from this basic design. 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, an organization or service provider can host one gateway in each country. By bypassing the international connect charges, the configuration costs significantly less than traditional circuit-switched service. QUALITY There are two main factors contributing to a satisfactory IP phone call: voice quality and turnaround time, or latency, which is dependent on network speed. 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. To reach acceptable levels of latency, one must be able to ping between two servers at 100 ms

5 or less for domestic calls, or 200 ms to 250 ms or less on international calls. If the ping times are not within these limits, the latency and call quality will not be acceptable. In general, 210 ms latency is unnoticeable, but above 350 ms the latency becomes irritating. A regular switched phone call latency is 150 ms. What is heard over satellite or microwave calls has traditionally traveled 350 ms roundtrip. A low orbiting satellite can be under 100 ms (as low as 25 ms) and a high orbiting satellite can be anywhere from 100 ms to 1000 ms for a single hop. New digital signal processors (DSPs) are becoming faster and faster. As DSPs get better, millisecond delays are dropping. 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 let bandwidth be reserved. It will take some time for the world routers to be upgraded and for operational aspects (such as how to bill for service) to be resolved. INTEROPERABILITY The way to maximize the efficiency of an Internet gateway network structure is to establish standards that tie together different manufacturers products and promote interoperability. The International Telecommunications Union (ITU) has recognized a set of standards for Internet telephony. This standard, known as H.323, provided the needed common language to Internet telephony, the PSTN, and the PC, which had been compartmentalized by proprietary software and hardware. With the ITU standards, PC desktop software, Internet gateways, modems, videoconferencing, and the existing public telephone system, all will be able to interoperate, expanding the capabilities of each. In addition, the Internet gateway servers will allow the public telephone network to function directly with other networks and products. This is the hidden value of Internet telephony. With set standards coming from IETF, IMTC, ITU, and ETSI, Internet telephony supplies the framework for the integration of computer and voice networks to enable a range of new services, including integrated voice and data, customer-selected audio quality, and internetworked voice mail. An Internet gateway network could be flexible enough to provide new services, limited only by the capacity of computers. OPEN VERSUS PROPRIETARY SYSTEMS One of the key factors for IP 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, some component vendors have been able to respond to the demands of IP telephony quickly,

6 modifying existing products to address the needs of the IP telephony gateway systems. These vendors are also continuing to pour researchand-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. Because system integrators need to excel in fewer aspects of system design, costs fall even more. The advantages of open systems are particularly compelling for IP telephony. The impact of the Internet on telephony is not as a stand-alone 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. CALL CONTROL PROTOCOL The first IP telephony products used proprietary call control protocols. H.323, however, has been defined as the standard call control protocol. This specification defines packet standards for terminal, equipment, and services for multimedia communications over large area networks (LANs) communicating to systems connected to telephony networks such as Integrated Services Digital Network (ISDN). It will be supported by successful IP telephony products. VOICE CODERS A vocoder is the piece of programming that receives the voice and compresses it and packetizes it for TCP/IP. Different vocoders are good for different purposes. They may have differing compression ratios with differing sound qualities, with each designed for and used in a particular type of call. For example, vocoders utilizing as little as a 2K to 4K compression might not have the quality and fidelity for a great voice call but could be effectively used for dual-tone multifrequency (DTMF) transmissions such as credit card autorizations, etc. Key technical requirements for vocoders include: Low bandwidth (8 kb/s or less) CELP or VSELP coding (common in voice compression in analog telephone networks) High quality for voice Low latency Ability to reconstruct lost packets In real-time transmission, 1 in 100,000 of the packets in a transaction might be lost or delayed. Successful IP telephony applications, then, need to recover from lost packets by effectively reconstructing the lost data. Reconstruction is necessary, of course, because retransmission of

7 voice packets is not feasible or acceptable from a performance point of view. The complexity of the coding algorithms has an impact as well. High complexity increases the cost of the host platform. G.723.1, an algorithm for compressed digital audio over telephone lines, is emerging as a popular coding choice. The enduring requirement for coders, however, is that IP telephony systems be capable of supporting multiple coders and adding more as technology emerges and popularity changes. ECHO CANCELLATION IP telephony gateways must perform echo cancellation. In a typical configuration, two gateways are each connected to analog phones via a digital, local CO 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; otherwise, by the time the echo propagates through the network back to the speaking part, it would be quite disruptive. IP telephony represents a unique case. Technically, local connections are used. Hence, the phone system itself is not performing echo cancellation. But long-distance calls are being made, and the echo would disrupt conversations if not canceled. The IP telephony gateways, then, must (and do) supply echo cancellation. FULL DUPLEX Phone calls are full duplex, meaning both parties can speak at the same time. Successful IP telephony products are also duplex. Not all voice cards can support full-duplex operation. DTMF DETECTION DTMF digits do not travel well across the Internet. Coding and packetization distorts and segments them, making them unrecognizable on the remote end. IP 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, a standard is expected to emerge soon. BANDWIDTH UTILIZATION Bandwidth utilization companies are moving to IP telephony to realize greater bandwidth savings because the compression ratios are so good. Bandwidth can be utilized dynamically because calls are going over a data network rather than a separate fractionalized voice network. So

8 bandwidth savings are significant. For example, with a 4:1 compression ratio in your gateway, 24 calls would occupy 512K of bandwidth on an IP network as compared with 24 channels on a T1 to handle those 24 calls. The rest of the bandwidth is saved or reutilized for data. To conserve bandwidth, the Inter-Tel vocoder, a Microsoft GSM 6.10, compresses 64K to 13K on a bidirectional call. The call setup information adds about 5K, for a total of 18K. But to conserve bandwidth the Inter- Tel gateway does not send packets while there is silence. This is not the case in traditional CELP or VSELP, and other vocoders. Most treat silence the same way as the spoken word, which uses bandwidth. Technically, bandwidth could be as high as 432K (bandwidth is usually sold in amounts divisible by 64K requiring a 512K pipe for full 24 simultaneous calls). With the Microsoft GSM bandwidth, utilization is about 192K for 24 simultaneous calls, equivilent to an 8K vocoder, with the compensation for the unused bandwidth during the silent periods of the conversation. CAVEATS When considering IP telephony for your company s intranet, consider the following caveats: Know the vendor, and choose it wisely. Make sure the vendor is going to be around. Choose component vendors with products that meet the user company s technical requirements. More important, the component vendor must be committed to this new market and must demonstrate the ability to adapt on the fly to its changing requirements. Because so many IP telephony systems are global, the vendor also needs to provide a worldwide network of service. Most gateway vendors require the user company s PBX to be T1 with E & M signaling. Typically, PBXs over 15 years old will not have E & M signaling. Make sure the user company has T1 and analog ports available on its PBX. Look at available bandwidth on the data network. Do not be fooled by high-quality H.323 integration. Calls from the Internet will not be of as good quality as those in the intranet because the intranet is a managed piece of bandwidth and the Internet is a hostile environment. An H.323 connection from a PC will not have the same kind of quality as a gateway-to-gateway call in an intranet. If the user company is using H.323, go through a managed network, under managed bandwidth. Finally, look at the structure of the user company internal departments. In many companies telecom departments are separate from data depart-

9 ments. IP telephony bridges these two responsibilities, perhaps these departments should be merged to have the two departments work on IP telephony together as a team to achieve and maintain the combined voice/data system. RECOMMENDATIONS To deploy IP telephony over an intranet some measurements need to be made, and the nature of the voice communications in an organization in general must be determined. Only qualified people in an organization can do this. Technically, it is a fairly simple operation. First, if the organization is not a multinational corporation, or if it is without multiple offices, there may not be any reason to put voice on its network. Utilization of the network should be determined whether it is overloaded or not. The 128 kb/s line cannot be utilized all the way up to 128 kb/s. If the network is operating at more than 70% or 80% capacity and voice is added, there will be delays. The bandwidth may have to be increased. If the organization s intranet does not have enough capacity, the 1) (long-distance) price it is paying for its long-distance traffic, and how much it is paying per month, should be evaluated; and 2) how much it would cost to increase its bandwidth should be determined, as well as the cost of the gateways. The gateways are capital costs with no ongoing costs. The next step is to analyze the current telecommunication use. The number of calls the organization makes to which branches, the number of calls going long distance or internationally, etc., all during periods of heaviest use, should be determined. It can then be determined how many gateways are needed, with a correct number of lines. With this information, the time necessary to achieve payback, which is usually between 8 and 10 months, can be calculated. ENTERPRISE DEPLOYMENT Network managers are not typically looking for more high-bandwidth, real-time applications to run over their networks, and many companies are now beginning to feel comfortable bringing IP telephony onto their own networks. However, there is a source of friction involving telecommunication managers who are not eager to rely on their networking counterparts for service. But the technology is becoming more familiar, the cost savings are quantifiable, and standards are decreasing the risk of choosing a vendor. Forward-thinking organizations are beginning to deploy Internet telephone gateways for internal use, particularly for things like internal help desks. Calls are being routed over intranets.

10 The real key for successful enterprise deployment, however, is when the gateways move from adjunct gateways to become an integral part of the enterprise communication infrastructure. When the gateways are able to route between the PSTN and the Internet (or their private counterpart) based on class of service, cost of service, and quality of service, IP telephony use will take a major step forward, providing an enormously valuable capability. The Internet is used only when it provides correct service for the particular call. More than anything else, IP telephony platforms require flexibility and processing power. Flexibility is important because technologies and capabilities in this new and fast-growing market segment change quickly. New voice coders, new means for packet reconstructions, and new silence suppression techniques will all become available over the next few years. COMING DEVELOPMENTS AND CONCLUSION Coming in the next couple of months is a gatekeeper that will bring intelligence to the network caller ID and other features found on the PSTN. The gatekeeper is the most important and the key element coming to the IP telephony network. The gatekeeper standard is not yet fully defined; the market wants a solution, but the standard is not yet ready. The Internet is fundamentally changing both telephony and the way people communicate. IP telephony is poised to become one of the fastest-growing markets of the decade. Businesses around the world are looking seriously at integrating IP telephony into their private networks in order to shave telecom costs and more completely utilize their existing networks. As the technology advances, the possibilities for business will expand. John Fiske is an independent writer specializing in enterprise networking and management technology. He lives in Prides Crossing, MA, and may be reached at fiske@tiac.net.