In this paper we examine a case study based on Intel's experience of deploying Voice over Internet Protocol (VoIP) with quality voice within a campus where voice and data were converged on the existing LAN using Session Initiation Protocol (SIP). We demonstrate the voice quality methodology used and the benefit it provided. First, we familiarize the reader with the terminology used throughout the paper and provide an overview of the Intel case study. Next, we discuss the network infrastructure elements, voice quality plan specifics, and trial results. Finally, we explore the challenges and solutions and summarize our key learnings.

Background and Terminology

SIP is an Internet Engineering Task Force (IETF) protocol that is used to initiate interactive user sessions with multimedia elements. SIP is specified in IETF Request for Comments [RFC] 2543 [1]. In the Open Systems Interconnection (OSI) communications model, SIP takes place in the Application Layer (Layer 7) and is responsible for establishing, modifying, and terminating the user sessions. In this case study, the user sessions are Internet telephony phone calls.

In a traditional telephone system for a campus, a PBX or Private Branch Exchange is used to provide call switching that is circuit-switched. Functionality typically provided by traditional PBX systems includes local and least-cost call routing, call forwarding, low-density call conferencing, and call detail recording. Most traditional PBX systems use a proprietary digital protocol over a private local area telecommunications network. Cables are run from the PBX to telephone stations in offices and cubicles, often in parallel to data network cables. Voice quality in a traditional voice network is usually a non issue because the end-to-end path is ensured for the duration of the phone call. The dedicated network provides low latency. Jitter is very low since the paths of the signals take the same route throughout the system. Physical faults such as loose connections or cable malfunctions can degrade the quality of the call. However, these impairments are detected by monitoring systems that measure electrical characteristics of the system.

In a VoIP solution for a campus, an IP PBX is used to provide the local telephone system. Because of the greater access to data and the incorporation of open standards, IP PBX systems generally provide the same features as traditional systems with more intelligence and there is greater opportunity to integrate with standard business applications enabling a higher level of automation. Sometimes media processing applications such as voicemail and automated attendant and Interactive Voice Response (IVR) are converged on the IP PBX, making adjunct devices unnecessary. Physically, the IP PBX may be connected to the data LAN or to a private voice LAN. Monitoring call quality for the IP PBX system becomes more important than in a traditional PBX, due to the nature of packetized voice and variability introduced by traditional networking.

MOS is used to measure the quality of a telephone call, whether it is a traditional circuit-switched or IP telephony call. MOS is determined by a panel of human listeners in a controlled environment who rate the audio from 1 to 5, with 5 indicating best quality. A MOS of 4 is considered acceptable quality, where 90% of the users are satisfied with the quality of the call.

QoS refers to how IP packets are dealt with through network devices. QoS on the network helps facilitate a better MOS for voice on an IP network. QoS is not a standard or protocol, but simply a generic industry term for outlining technologies, standards, and strategies to provide for network quality. In general, QoS, to facilitate good voice quality and high MOS, requires that packets carrying real-time voice traffic cannot be delayed and must be prioritized over data traffic, which can better tolerate being slightly delayed. Most jitter in the network is caused by queuing delays associated with momentary or chronic congestion. QoS for voice can help make this queuing delay transparent to the voice packets.

Introducing the Intel^® VoIP Program

The Intel^® VoIP program took place at one Intel site over a six-month period, with over 50 participants who represented a variety of job functions and telephone usage models. The VoIP system was deployed within the enterprise with standard business voice applications, including automated attendant, voice mail, unified messaging, follow-me (call forwarding) features, fax, and remote access.

The IP PBX was built on Intel architecture with additional communication building blocks from Intel. The range of telephone devices used in this trial enabled the demonstration of SIP interoperability of IP endpoints. Additionally, the IP PBX was integrated with Intel's standard Instant Messaging (IM) solution to share basic user presence, along with the ability to launch a phone call.

The project met each of several high-level business goals that were set. Those goals included using and validating some of Intel's VoIP products, giving Intel's IT department the opportunity to introduce a converged communication solution into the enterprise, and demonstrating a phased migration path from a traditional telephone system to a next-generation IP PBX. How the goals were accomplished is outlined below.

• Validate Intel products.

  Through the success of the project, Intel was able to validate that an open standards VoIP solution using SIP can be deployed with Intel products. Intel is maintaining a showcase platform and demo room at the pilot site where interested parties from inside and outside Intel can experience the converged communication solution firsthand.

  The Intel components used in the solution are as follows:

  • The Intel^® Xeon^® processor is incorporated in the industry-standard server running the IP PBX
  • Intel NetStructure^® Host Media Processing (HMP) Software within the IP PBX performs media processing tasks on Intel servers without the use of specialized hardware.
  • Intel NetStructure PBX-IP Media Gateway (PIMG) is an Internet appliance that integrates the proprietary digital telephones with the IP PBX through telephone set emulation.
  • Intel^® Centrino^® mobile technology on laptops runs the IP PBX client software and softphones.

• Introduce converged communications.

  Intel introduced Intel technology into the enterprise for converged communications at three levels. First, applications were converged at the server including base telephone system and media processing applications. Second, voice and data were converged onto the same LAN. Finally, telephone applications were converged at the client including call control and softphone.

• Demonstrate phased migration.

  The production showcase environment demonstrates to customers and others outside Intel how a phased migration from a legacy PBX to IP PBX might proceed. In this case, the IP PBX was deployed behind the traditional PBX. Voice was layered on top of the existing data LAN.

Planning for and monitoring voice quality were critical elements of the trial. Because users are accustomed to the high-quality performance of a traditional telephone system, they have no tolerance for echo, jitter, or delay in VoIP communication. As a real-time application, VoIP requires the same continuous uptime as the system it replaces. High quality is possible, provided careful planning goes into the network design and implementation details.