Plain-old-telephone-service (POTS) networks

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1 INTERNATIONAL JOURNAL OF NETWORK MANAGEMENT Int. J. Network Mgmt., 8, (1998) Transporting Voice Traffic Over Packet Networks POTS networks are being rapidly superceded by newer, packet-based ones, which allows a greater facility for voice traffic. This article explores the practical issues involved in deploying voice networks over ATM, frame relay and IP by John Wiley & Sons, Ltd. By Larry Greenstein* Introduction Plain-old-telephone-service (POTS) networks traditionally have relied on timedivision multiplexed transport systems for interconnection. We now see a rapid migration to newer packet-based networks which combine statistical multiplexing with voice compression, which results in many more voice calls carried over the same transmission systems. When examined closely, one finds there is much more to this issue than simply TDM versus packet transmission. This article addresses the practical considerations of deploying a high-performance, reliable, manageable packet voice network. It examines voice over ATM, frame relay and IP. Packet voice networks have an important Larry Greenstein is the Director of Product Management at Nuera Communications in San Diego, CA, USA ( He is Vice President of Technology on the board of directors of the Frame Relay Forum and Vice Chairman of the Frame Relay Forum Technical Commmittee. His background includes hardware design, switch architecture, and carrier services planning, development and deployment. *Correspondence to: Larry Greenstein, Nuera Communications, Pacific Center Court, San Diego, CA, USA. lgreenst nuera.com appeal they save money, and lots of it. And you don t have to sacrifice quality to save money unless you want to be aggressive and trade off some quality for additional savings. The financial savings are the result of the lower tariffs for connections over packet networks, the bandwidth savings as a result of voice compression, and the additional bandwidth savings due to statistical multiplexing. Voice Packet Networks There are three types of packet networks which have great popularity and are suitable for carrying voice traffic. These are Asynchronous Transfer Mode (ATM) networks (ATM packets are called cells because they have a fixed length), frame relay networks, and internet protocol (IP) networks. ATM The ATM Forum and various standards bodies have defined procedures for carrying traditional TDM voice traffic inside ATM cells as constant bitrate traffic. While this provides a reliable and highperformance solution for migrating to ATM net John Wiley & Sons, Ltd. CCC /98/ $17.50

2 228 LARRY GREENSTEIN works, it does not take advantage of the statistical nature of voice traffic. As a result of the cell overhead bits, considerably more bandwidth is used. Recently, the ATM Forum, in conjunction with standards bodies, has begun work on transporting compressed voice over ATM while taking advantage of ATM s statistical multiplexing architecture. By defining a new ATM adaptation layer, AAL2, multiple voice samples from multiple voice channels can be placed in a single cell, or span multiple cells. This more robust packing of traffic into the cell allows the cell to be used as efficiently as possible by reducing the amount of unused payload in a cell. To take advantage of this new adaptation layer, a variety of application layer procedures are required to be defined in the specification. For example, voice call establishment, compression algorithm negotiation, fax modem detection and fax relay procedures, and interworking with legacy telephone equipment. The ATM Forum is expected to make strides in this area and publish the finished specifications in Interoperable and compliant equipment will likely be available in 1999 and it is quite probable that this equipment will be deployed by the world s largest telcos. (For more information about voice over ATM check the web site Frame Relay The Frame Relay Forum has published an Implementation Agreement (FRF.11) which defines procedures for transporting voice over frame relay. The initial reaction many people have to the idea of transporting voice over frame relay is that it probably does not work well. After all, frame relay does not offer the quality of service guarantees that are built into ATM. So how does one explain the very rapid growth of voice over frame relay (VoFR) equipment sales? The answer is simple; it works quite well. Frame Relay: Who s Using It for Voice Traffic? So are we likely to see the world s largest carriers scrap their other voice network technology and migrate to frame relay as their fundamental transport? Not likely. But we are likely to see these same carriers offer VoFR services in addition to their other voice services. And there are large numbers of smaller carrier networks and large corporate voice networks which can benefit greatly from VoFR technology. In some cases, the corporate network will use it initially only for intra-company communications. After an evaluation period, it is common for these corporate networks to then use it when calling outside the corporate enterprise. These two phases of deployment are illustrated in Figures 1 and 2. In the first phase, the PBX routes some calls to the frame relay network and some to the telco network. It makes this determination based on the dialed digits. Intra-corporate calls are typically either a four-digit extension or they can look like an outside call, which typically begins with a 9 followed by seven or more digits. These calls are routed to the frame relay network by the PBX. The PBX is programmed with a dialing plan which indicates which called numbers are intra-corporate and which ones are not. It matches the called numbers against its list and routes the call accordingly. In the example shown in Figure 1, the caller who is located in a branch office dials extension 1317 to reach another employee of the company who is located in another branch office. If the call were to be carried in the traditional manner over the longdistance telephone network, the caller would incur toll charges. The PBX recognizes the called number as an intra-corporate destination and routes the call to the VoFR equipment. The VoFR equipment, which has also been configured with the network dialing plan, routes the call over the frame relay PVC (permanent virtual connection) to the central site where it is switched within the VoFR equipment back to the frame relay PVC which will carry it to the destination VoFR equipment. As a result, the call is completed without any toll charges. When the same caller places a call to someone outside of the company, the PBX again determines the proper route for the call based on its dialing plan. In this case the called number is The PBX routes the call to the PSTN (public switched telephone network) which routes the call to its destination. Of course, in this case there are long-distance toll charges. Multiple voice calls can share a frame relay PVC in the same way that traditional voice calls share

3 TRANSPORTING VOICE TRAFFIC OVER PACKET NETWORKS 229 Figure 1. Phase 1 deployment. a transmission facility. And voice calls can route through multiple VoFR hops to get to the destination because the VoFR access equipment functions as a digital voice switch. In the second phase of deployment, the voice call routing capabilities of the VoFR equipment can be used to further reduce monthly long-distance toll charges. In this phase, the PBX routes all calls (except for calls from one extension to another within the same office) to the VoFR network. The VoFR network dial plan allows the call to be routed to the destination (if it is an intracorporate destination) or to a PSTN connection closest to the destination. At this point, the VoFR equipment can automatically translate the called number (if necessary) and out-pulse the tones, or pulses to further progress the call over the PSTN. As a result, the frame relay network carries the voice call over the long distance and the PSTN is used only for a local call, which results in a lower toll charge. Why would called number translation be needed? As shown in Figure 2, suppose a call is placed from the USA to the UK. The caller dials At the destination VoFR equipment, which connects to the PSTN, the VoFR equipment out-pulses the local phone number which is and the call is further routed to the final destination. With robust translation capabilities, any called number can be trans- lated by the VoFR equipment at any point in the VoFR network. The more sophisticated VoFR equipment on the market today can also route voice calls around failures, around fully occupied transmission facilities, or to alternative voice ports at another destination if all the ports are busy at the primary destination. This capability permits the network manager to assure that voice calls always get through. When VoFR bandwidth or resources are fully occupied during peak periods, the call can always be routed back to the PSTN for connection to the destination. The ability to route calls in the manner described above is very important. It allows the network manger to deploy the network in stages without the need to train employees in new procedures for dialing the destination phone number. Mixing Voice and Data Frame relay has had phenomenal success in the marketplace. Until now it has been used primarily as a replacement for leased lines which interconnecct IP routers. In many private frame relay data networks there is excess bandwidth available. By adding VoFR equipment, both voice and data traffic can be carried over the frame relay network. In branch offices which need only two or four voice channels, VoFR equipment can usually be

4 230 LARRY GREENSTEIN Figure 2. Phase 2 deployment. Frame relay has had phenomenal success in the marketplace. queuing, rate enforcement and traffic metering in exactly the same way it is performed by frame relay network switches. added without the need to increase the interface speed. But despite what some equipment vendors may advertise, the voice traffic does not get transported for free. If you are using a public frame relay service, you ll probably want to add one PVC for the voice traffic and configure it with some committed information rate (CIR). How much CIR? Typically 8 Kbps CIR is recommended for an 8 Kbps voice channel. If you want to be aggressive, or if there are more than eight voice ports at a particular endpoint, you can go with 4 Kbps CIR per voice port. However, underprovisioning of CIR can lead to congestion and while the data traffic can tolerate that condition, voice traffic cannot. To add voice capability to the frame relay data network, the router is plugged into the VoFR equipment which in turn is plugged into the frame relay network as shown in Figure 3. This alows the VoFR equipment to manage the router traffic and assure that a misbehaving router will not compromise voice quality. This is accomplished by the VoFR equipment through the use of priority Congestion But what happens when the routers start blasting away with file transfers? By plugging the router into the VoFR equipment, the VoFR equipment can prioritize the data traffic one setting below the voice traffic priority. Additionally, long data frames can be fragmented so that they don t delay voice traffic when traversing a low-speed frame relay interface. This combination of prioritization and fragmentation allows the VoFR equipment to assure that the data traffic cannot disrupt the flow of voice traffic. Delay Frame relay networks insert more delay than traditional telco networks typically about 100 milliseconds greater than the electrical or optical transmission time. Also, voice compression/ decompression and jitter buffering (to accommodate delay variations) adds another 65 to 95 milliseconds. This increased end-to-end delay is almost

5 TRANSPORTING VOICE TRAFFIC OVER PACKET NETWORKS 231 Figure 3. Mixed voice and data. never an issue for the user as long as it does not exceed 250 milliseconds. Due to the increased endto-end delay, it is necessary to eliminate echoes reflected back to the speaker. Echo cancellation on the tail circuit is built into all the popular VoFR equipment on the market. Compression By compressing voice traffic down to a lower bit rate (e.g., 8 Kbps), huge bandwidth savings can be achieved. And bandwidth savings translates directly into money savings or increased capacity, or both. There has been tremendous progress in voice processing technology in recent years, not only in the area of much higher-performance digital signal processors (DSPs) but also in the voice-encoding algorithms (vocoders) which run on these processors. The toll quality vocoder against which all other vocoders are compared is pulse code modulation (PCM), which has been used since the 1960s and consumes 64 Kbps. Adaptive differential PCM (ADPCM) at 32 Kbps is commonly used by large carriers for toll quality service on international circuits. Today s high-performance vocoders, which are based on code-excited linear prediction (CELP) and use 8 Kbps (such as the ITU G.729 standard), encode voice with radically different techniques from those used in PCM and ADPCM. The result is a high-quality voice channel which cannot be distinguished from ADPCM by most listeners. By using CELP, eight times as many voice channels can be carried over the same amount of bandwidth required for one PCM channel. Many virtual private networks use PBXs to route the voice call through the network. Since these networks do not use voice compression, there is no problem with the call being tandem switched through a PBX at a central site in the network. However, when voice compression is used, it is important that the voice sample is not decompressed and recompressed at an intermediate switching point in the network. If that were to happen, the quality of the voice traffic would degrade and the end-to-end delay would increase. Therefore, it is important that the VoFR equipment is able to provide frame switching capability. By switching frames coming in on one PVC to go out on another PVC, the voice quality is preserved. Voice Activity Detection When people speak on the phone, one person speaks while the other listens (most of the time). On average, about half the bandwidth is unused in each direction when multiple conversations are carried over one transmission facility. Packet-

6 232 LARRY GREENSTEIN based systems can simply cease packet transmission when a person stops speaking. As a result, the eight-to-one improvement in bandwidth savings due to compression quickly becomes a 16-to- 1 savings when voice activity detection is added to the network. Interoperability The Frame Relay Forum s VoFR implementation agreement provides for interoperability among vendors of VoFR equipment and was published in May 1997; compliant implementations are expected by the end of The implementation agreement focuses on providing VoFR service over PVCs because Switched Virtual Circuits (SVCs) are not yet common in the marketplace. However, the Frame Relay Forum is expected to continue to progress VoFR and include service over SVCs in the next version of the document. (For more information on VoFR technology, check the Frame Relay Forum web site at com.) VoFR Service Providers A couple of US-based frame relay service providers have anounced support for VoFR users. Many other frame relay service providers have tested VoFR equipment in their labs and approved the equipment for connection to their network. It is very reasonable to expect these and other service providers to soon offer managed VoFR service in the same way they now offer turnkey managed data service over frame relay. Voice Over IP Many of the issues described above (e.g., compression, voice activity detection, delay, congestion, etc.) also pertain to voice-over IP (VoIP). However, there are some important differences. Delay Delay over IP networks, especially over the Internet, can be unpredictable. While this does not typically cause a problem for data applications, it can lead to unacceptable voice connectivity. To address this issue, the Internet Engineering Task Force (IETF) has defined a new IP protocol called Real Time Protocol (RTP). RTP itself does not provide any mechanism to ensure timely delivery or other quality-of-service guarantees, but relies on lower-layer services to do so. The services provided by RTP include payload type identification, sequence numbering, timestamping and delivery monitoring. Once the RTP and the supporting infrastructure is deployed, voice over IP will be a viable solution for voice networking. The first RTP networks will be corporate ones. Eventually there will be separate classes of Internet service provided to you by your local Internet service provider (ISP). But don t expect the new high-performance service to be as inexpensive as what you now get from your ISP. We will have to wait and see what happens with pricing, but it is likely that RTP service will be billed by the minute or possibly billed based on how much traffic you send. Resource Reservation Protocol (RSVP) is an Internet control protocol and is used for the purpose of resource reservation setup and is designed for accommodating integrated services over IP networks. The RSVP protocol is used by a host to request specific qualities of service from the network for particular applications. RSVP requests result in resources being reserved in each node along the data path. The combination of RSVP, RTP and the underlying support infrastructure will provide reliable integrated real-time services on demand. Voice Over IP Forum The Voice Over IP Forum is drafting implementation agreements which define procedures and guidelines so that VoIP equipment vendors can develop interoperable equipment. The scope of these implementation agreements is quite large and they provide for sophisticated supplementary services similar to what has been standardized by the ITU for intelligent telco networks. These more complex systems provide for additional features such as allowing the subscriber to specify a custom profile which can cause an incoming call to route to voice mail if it is coming from someone the subscriber does not want to hear from. Additionally,

7 TRANSPORTING VOICE TRAFFIC OVER PACKET NETWORKS 233 calls from important callers can be routed to a cell phone or to a home phone (depending on the time of day) instead of to voice mail. VoIP clearly has a bright future, but it is not quite ready for prime time deployment. Look for the pioneering corporate users to begin deploying networks in 1998 with more general deployment in (For more information on VoIP check Economics The economics of VoATM is not as compelling as the others because constant bit rate ATM service is considerably more expensive (about 80% more) than variable bit rate service (which is priced similar to frame relay). Also, ATM service at interface speeds below 1.5 Mbps is rare, and many applications require only a few voice channels. VoIP over the Internet can result in very low cost and frequently yields performance that is not adequate for the commercial enterprise. Costs of VoIP over the intranet are on a par with those of VoFR and both provide good-quality service. The economic analysis provided below is based on service over frame relay and the costs and savings are also applicable to VoIP. Voice over frame relay is fairly well accepted as being economical when used in international and non-us applications. With international call prices sometimes approaching $1.00 per minute, the payback period for voice over frame, even with the higher international rates for frame relay, are very dramatic. One way of looking at the economic justification for voice over frame relay involves the dollars per month based on the cost of dial voice. For instance, if you assume that you re using only one hour each business day for each of eight lines at $1.00 per minute, the monthly cost is almost $10,000 a month. When you re dealing with this type of cost to start with, voice over frame relay is cost justified very quickly. But the economic justification is much more of a challenge in domestic (US) networks, where competition has already pushed voice down into the 5 to 10 cents per minute range. In order to solidify these economic arguments, quantitative calculations are needed. These calculations are surprisingly straightforward, and the results are most impressive. For instance, if you assume that you re using 3 hours on each of eight lines at 5 cents per minute for 20 days a month, the cost is $ per month for all eight lines significantly less than international phone bills. By comparison, consider buying frame relay service by the month on a CIR (Committed Information Rate) basis. For most of the carriers, this is the only available option. In this case, the fixed monthly costs must be compared with the costs on a per-minute basis. This is usually done by comparing the fixed cost with the cost per minute assuming a certain level of usage. Let s start by assuming a worst case for frame relay a single 64 Kbps PVC between two points with a single 64 Kbps port at each end. Using the assumptions of an eight-to-one compression ratio with voice activity detection enabled, this will readily support eight voice channels (on an unlimited usage basis). This also assumes that this is additional bandwidth for the exclusive use of voice, so it totally disregards the usual economies found in combining the voice application with existing data applications for frame relay. For this situation, the undiscounted cost (excluding access lines) is about $ per month. (Most services charge on the basis of access cost, port charge, and the CIR per PVC. In this analysis, we ignore the access costs since these are roughly the same regardless of the transmission method used.) Compared with the assumption of $ per month for toll calls, this results in savings of about $ per month. And if you assume 10 cents per minute for five hours per line per day, the toll cost skyrockets to $ This would result in a difference of almost $4000 per month. Furthering this analysis, doing a simple comparison of cost per minute to cost of lines, assuming 5 cents per minute, yields a break-even point (where the cost per minute equals the fixed costs) of about 200 hours of usage per month. Since we re assuming, quite conservatively, that we re getting eight-to-one compression, this amounts to the equivalent of an hour and 10 to 20 minutes per line per day. Assuming a (more realistic) rate of 10 cents per minute for voice, the break-even point drops to 35 to 40 minutes per line per day! And this is worst case. If frame relay is also used for data, the payoff is much quicker since the bandwidth ccan be dynamically allocated between voice and data. Further, the general avail-

8 234 LARRY GREENSTEIN ability of PVCs with CIRs in relatively small increments (like 16 Kbps) makes the minimum increment much smaller than the eight lines per location as assumed here. Conclusion While packet voice network technology is rapidly advancing on all fronts, it is clear that not all these technologies are at the same stage of maturity. Efficient voice over ATM will eventually be a core technology in the largest carrier networks, and voice over IP will be a big success in future corporate networks and for personal use over the Internet. However, voice over frame relay is here now and it is already providing moneysaving, practical, and high-quality solutions for business applications. If you wish to order reprints for this or any other articles in the International Journal of Network Management, please see the Special Reprint instructions inside the front cover.