Assessing A Sound Option
While voice over internet ushers in convergence of telecom and the web, network compatibility, acoustic reliability and overall viability are still being evaluated by enterprises.
Every major networking vendor, a few start-ups and smaller vendors are now touting a plethora of Voice-over-IP (VoIP) products targeted
at both enterprise and service provider customers. Analyst reports describe rapidly growing markets and forecast a tremendous take up of this new technology. Yet many customers remain cautious about implementing VoIP in their networks, and new applications based on VoIP technologies are emerging slowly. So are enterprise customers ready for VoIP, as we enter the new millennium?
As some customers say, “Circuit switched telephony works. Why play with it?” There are numerous answers to this question. By all standards, today’s circuit switched telephony environment is expensive to acquire, install, maintain and use. It does not integrate easily with the other desktop business tool, the PC.
Further, there are many more useful capabilities that can be provided with a more advanced system. Additionally there are clearly significant benefits to be gained from network simplification, reduced operating costs and universal service capabilities. But if VoIP offers lower costs and better capabilities, why are some customers still cautious? The answer can be summed up in one phrase–legacy systems or the installed base.
Decibel data dynamics
Enterprise customers have spent millions of dollars to equip their premises with separate voice and data network infrastructures. Voice networks are built
around large circuit switched PBXs and data networks, based on earlier generation, software-based routers. Each of these networks have their own separate management systems, and in many cases, it is managed by different technical staffs. While the data network experts will be concerned about the impact of voice traffic on the network, the voice experts will stress the need for reliability and voice quality in the context of packetized voice. Both are right to be concerned, but the latest VoIP solutions cater to both sets of concerns.
Reliability and availability are major issues for telephony. PBXs are built with redundant logic and battery back up which ensure a very high degree of availability. Data networking devices, on the other hand, are less fault-tolerant and therefore very reliable. However, the philosophy behind data network availability is a little different. One reason the internet itself was created, was to establish a computer network not susceptible to damage in the event of a hostile attack. The distributed nature of data networking implies that while one router may fail, the network can still provide the necessary connectivity. While we can make individual networking devices highly reliable, the need for battery back up, is not normally considered necessary. So can data networks be made highly reliable? Yes. Do most companies build their data networks to such standards today? Not yet, but voice and data convergence will likely drive, higher availability for both sides of the house.
A second factor is that VoIP will change the way telephony services are provided. The PBX and its associated hard-wired, proprietary telephones can be equated to the mainframe with its dumb terminals in the earlier days of computing. When the PC came along, the client-server approach took over and intelligence was passed to the clients–the PCs. Now information is stored in shared servers such as email servers or web hosts. Services like printing are provided by servers too. Servers are typically PCs that come without any special redundancy features. Back up is provided by replicating the server, so users can get the services they want even if a primary server is down. The same thing is about to happen in the telephony world.
Networking sound traffic
In new telephony systems, phone users will be able to use more intelligent, off-the-shelf, standards-based IP phones, or a variety of standards-based analog and black phone gateways or wireless phone technologies. Voice traffic will pass between phones via the network, but the traffic will not pass directly through the telephony server. Only connection establishment and the provisioning of voice services like conferencing, holding and transfer will be handled by the server. Two or more telephony servers can be used to provide the necessary redundancy and services availability. If a server goes down, it will only affect calls that are in the process of being established at that time, requiring users to dial again to obtain the services of an alternate server.
Existing calls will not be affected.
A word of caution is necessary. VoIP gateways interconnect existing PBXs and telephone exchanges and play a very cost-effective roll in converging voice and data traffic onto inter-site, WAN networks. These gateways do carry both voice traffic and signaling information between PBXs. In fact, they act as tandem switches in PBX networks and therefore, need to be as reliable as the PBXs themselves. PC-based products are unsuitable for this role, not only because of a lack of power supply redundancy, but also because of the inability to hot swap interface and processor cards. Purpose built, hardware-based solutions offer the best approach.
VoIP systems rely on a variety of standards-based algorithms to sample the natural voice streams, packetize and compress the traffic. There is clearly a trade-off between the degree of compression and the achieved voice quality.
Yet, network effects aside, the voice qualities achievable are generally considered perfectly adequate for normal business conversations. That is, speech is clear, accents and voice characteristics are also easily recognizable. The problems that do occur are mostly related to network design and to the choice of equipment used to create the network infrastructure.
Two major issues that need to be dealt with are echoes and latency. Echoes are dealt with using standard echo suppression schemes such as G.168.
Except in extreme circumstances, G.168 performs very well. Latency is the delay injected in the voice path by the various devices between the phones at each end. For good quality voice calls, latencies should be kept below about 180 milliseconds. The packetization and depacketization process required for VoIP typically takes around 100msec, give or take 20msec or so, leaving approximately 80msec only for network derived delay. Voice packets routed by traditional software routers can suffer excessive delays because of the time it takes to process each packet. New protocols, such as Multi Protocol Label Switching (MPLS) have been designed to speed packets to their destinations across the network, but these too add processing load to the routers and can reduce their packet throughput.
Switch to converge
The latest routers solve this problem by ‘switching’ packets in hardware. The switching occurs at ‘wire speeds’ or gigabit rates, so the latencies incurred by packets going through individual routers are minimized. In addition, layer 4 switching capabilities provide the ability to recognize voice packets by their User Datagram Protocol
(UDP) socket numbers, for example, enabling the router to prioritize these packets ahead of data packets, to avoid delays in congested networks. Layer 4 ‘flows’ can also be directed along specific paths using policy-based routing schemes. This reduces the jitter effects of connectionless routing and ensures voice packets are directed via routers with adequate bandwidths and performance.
By combining good network design with layer 3 and 4 switches or routers, it is now possible to build converged networks to carry good quality VoIP and data traffic. Enterprises are replacing their older generation routers with these next generation products at an increasing rate and so are installing network infrastructures that can support both voice and video capabilities. The new routers also support much higher bandwidths with gigabit backbone connectivity and 10/100 base-T to the wiring closets and desktops. Extra bandwidth that should eliminate the data experts’ fears regarding network congestion. The layer 4 switching and QoS mechanisms, are also available in the wiring closet together with management tools that can be used to affect flow-based management, policy establishment and service level management for both voice and data applications.
Voice gateway technology has also evolved. It is now possible to provide reliable and resilient links between existing PBXs using VoIP while maintaining good voice quality. Many enterprise customers are evaluating and deploying this technology because of its potential to save significantly on long distance charges. This first step in using VoIP leverages existing investments in PBXs and phone systems. This reduces the risks associated with wholesale upgrades while saving on telephony charges, leased line tariffs and administrative costs. It also benefits users at smaller locations who can now gain access to the corporate PBX network via a cost-effective data network or
The holy grail of desktop IP telephony will be achieved more slowly. There are still problems to be solved such as how best to power the phones and what to do about lifeline services. Many early IP PBX products are only suitable for smaller offices and cannot be grown to support the entire enterprise. Solutions to these issues are now becoming available. For example, battery powered wireless phones solve both the power and the lifeline issues, by operating alternatively as wireless in building phones or as cellular phones outside the office, or when the in-house system fails. The new systems will be used to extend existing PBX and phone networks to new locations rather than replace them, reducing relocation costs and giving users the chance to evaluate new technologies and new features. Enterprises are not ready for wholesale upgrades to their voice systems, but their data networks are becoming capable of a gradual migration, as business needs dictate.
Senior Director, Voice and Converged Solutions Cabletron Systems