By David Perez
In recent years, the corporate world has acquired significant benefits from the applications and products produced by the convergence of voice and data technologies: Internet protocol (IP) telephony; Internet telephone software; and virtual private networks (VPNs).
One of the newer technologies spawned from the integration of voice and data is the ability to transmit voice over data networks. Referred to as Voice over IP (or VoIP, for short), this technology enables voice to be carried over IP-based, packet-switched local-area networks (LANs) and wide-area networks (WANs).
VoIP holds enormous potential that many companies find compelling. Consider, for example, that in a traditional circuit-switched network, establishing a connection creates a dedicated, end-to-end channel for the duration of the communication. As a result, any unused bandwidth remains just that — unused — until the call is released.
By comparison, packet-switched communications allow bandwidth to be shared among various types of communications, filling the available bandwidth capacity more effectively than in circuit-switched networks. Moreover, the ability to combine all traffic onto a single network represents considerable cost savings in the physical enterprise. Another key VoIP advantage attracting significant attention is the ability to conduct long-distance calls while bypassing the public switch telephone network (PSTN) and its associated toll charges.
Quality Issues
While the current and potential benefits afforded by VoIP are compelling, the feasibility of having voice transported over networks that also carry data and video traffic raise a number of quality issues that must be addressed before VoIP is embraced as a mainstream business tool. One of the key quality issues that have received significant attention centers on voice connections over the Internet.
VoIP technology made its first big splash with the arrival of Internet telephony. Consumers got excited by the prospect of using a PC and an Internet connection to dial friends and family members anywhere in the world and talk for hours, without having to pay long-distance charges.
Users expected such connections to provide the same voice quality as the public switched telephone network (PSTN). Unfortunately, the protocols for defining a data network were designed for non-realtime data traffic, where network congestion results in dropped packets and requests for retransmissions.
While this approach works well with data, it wreaks havoc with voice calls, where dropped and delayed voice packets force callers to experience disorienting repetitions or gaps. These behaviors are not acceptable for users who expect the same voice quality of the PSTN and will not tolerate poor performance from a VoIP network.
The secret to success
For corporations interested in deploying VoIP technology, the success or failure of their venture will depend, in large part, on the performance of the network elements that carry and route the voice packets.
Gateways are required to perform the conversion of voice to IP packets for applications that cross between the PSTN and VoIP networks. In addition to the concern on network element performance, these gateways must process voice reliably under extreme loads.
A standard is needed — and more
Having a reliable standard is vital with a technology as new and formative as VoIP. The motivation for developing such a standard provide the positive attributes of a centralized control architecture (such as scalability, carrier grade reliability, and PSTN regulatory compliance) while, at the same time, encouraging greater multi-vendor operation.
H.323
The first VoIP standard to be developed is the umbrella standard known as ITU-T H323. H.323 is a version of the H.320 Multimedia-over-ISDN standard optimized for packet-based networks such as TCP/IP. H.323 is not specific to IP (it can also be used with AppleTalk and IPX). However, it relies on a number of significant IETF technologies, most notably the realtime protocol (RTP) and realtime control protocol (RTCP).
Figure 1: Typical analog telephone network
H.323's modularity makes it extremely flexible, particularly for joining an existing voice network to VoIP equipment. Figure 1, for example, shows a typical corporate telephone network composed of traditional analog technology. Figure 2 shows how H.323 components can replace some of components in this network, while preserving other portions of the analog network.
Figure 2: H.323 replaces telephone network components
H.323, for the most part, has been met with ambivalence by VoIP solution providers. In fact, of the many VoIP products available today, only a handful supports any standards-based implementation. This ambivalence toward IP has raised some interoperability concerns.
For example, the H.323 call-setup negotiation routines require end-point systems to allocate random port numbers for the RTCP control channel and RTP data channel. While this approach promotes considerable portability across different kinds of packet-based networks, it makes implementing H.323 across an IP firewall challenging if not downright difficult.
Rather than using a well-known port for all voice traffic, every H.323 node on the network must listen for and send on any port number above 1,024. Most businesses find this approach untenable because it forces them to open their entire corporate network to all UDP and TCP traffic, placing their very raison d’être at risk.
The easiest workaround for this dilemma is to contain all H.323 traffic within a specific region of the corporate network. If traffic is filtered between an organization’s corporate backbone and its branch-office networks, the H.323 traffic can be contained within those sites, with voice trunks used for any interconnection services. Alternatively, an H.323-compliant firewall can be used.
Because H.323 has not been more widely adopted by the industry, you may want to consider limiting your VoIP implementations to a few key areas. Given H.323's modularity, you can replace only select components on your network.
For example, you might provide users in a new facility with VoIP equipment at the desktop, yet retain your existing PBX network at your corporate headquarters. Conversely, you might replace an outdated PBX cluster with IP-centric systems, while maintaining existing user-side equipment at the desktop.
Other VoIP standards
The less than total acceptance of H.323 as a VoIP standard has spurred the development of other emerging standards. One of these standards, called session initialization protocol (SIP), is currently under development. This standard is being developed within the IETF's Multimedia Working Group. SIP offers similar architectural features of H.323, with a particular focus on IP-specific technologies such as DNS. In addition, SIP incorporates the concept of fixed port numbers for all devices and supports proxy servers, both of which ease firewall implementations.
Another standard that had been gaining interest at the IETF was the simple gateway control protocol (SGCP). Developed by Bellcore, SGCP introduced a new call-management tier called the call agent. The call agent was designed to off-load much of the signaling intelligence from the end node, making this standard ideal for traditional telephone handsets. SGCP also promised to reduce delays associated with H.323’s use of signaling translators and TCP/IP.
A fourth standard making the rounds was the Internet protocol device control (IPDC). Developed by Level 3 and friends, IPDC was intended for use between centralized switches and IP-based gateways, providing large-scale integration and management.
Recently, the IETF formed the MeGaCo working group, which merged SGCP and IPDC into a unified standard called MGCP. MGCP was submitted to the IETF’s MeGaCo working group. Lucent Technologies submitted a third protocol, called media device control protocol (MDCP), and from these inputs emerged a new and improved protocol named MeGaCo protocol (also known as H.248).
SIP is currently in draft form and far from implementation, while MGCP is undergoing implementation and interoperability testing, with prototype testing and implementations occurring in selected organizations. Therefore, while only a small percentage of vendors support H.323, it's the only standard so far that guarantees interoperability.
Conclusion
There is no question that transmitting voice over IP-based data networks will become an enormously popular application — the cost-savings and rapidly improving technology are too compelling. Nevertheless, there remain significant questions about the viability of having voice carried over data networks that compete with data and video traffic.
There are several methods currently under discussion by VoIP equipment and service providers for improving quality of service (QoS) and providing customers with QoS guarantees. If implemented, these methods will improve the way that conversations in the VoIP environment sound, adding consistency to quality performance. This hurdle must be overcome before general business acceptance of outside enterprise intranets will occur.
For corporations interested in deploying VoIP technology, the success or failure of the venture will depend, in large measure, on the performance of the network elements that carry and route the voice packets. Therefore, organizations should not be hasty when deciding whether or not to implement VoIP. Every area of an enterprise’s network will be governed by individual factors that motivate (or discourage) the adoption of VoIP technologies.
Moreover, each portion of an enterprise network has its own issues that must be considered when planning a VoIP implementation. For instance, the opportunities to reduce costs in remote offices are not the same as they are for local users. Similarly, bandwidth and infrastructure requirements for a telecommuter or small office are radically different form those for a large office or campus. For these reasons, you may want to conduct a trial deployment to measure usage and other issues in your environment.
In the final analysis, the success of the industry hinges on the positive perception of people using telephones.
