Skip to content Skip to navigation

Voice Communications


This document covers the strategic vision for Stanford's voice services infrastructure, which provides telephony services for the campus (both faculty and administration), all residential students, Stanford Hospital and Clinics (SHC), and Lucile Packard Children's Hospital (LPCH). While a successful voice service must take into consideration the business needs of a wide variety of clients, this document focuses on the technical aspects of the voice infrastructure.

The goal of the voice services infrastructure is to provide voice-based services that are robust, reliable, always-on, scalable, and cost-effective for the university, using both Time Division Multiplexing (TDM) and Voice over IP (VoIP) technologies. Wherever possible, the voice infrastructure will use IT Services-adopted system standards such as the Linux operating system and centralized authentication.

Distinctions between voice and data services on the network are evaporating and the technologies are converging into both wired and wireless (802.11) IP networks. Cellular phones have become an integral component of our voice services, and increasingly clients are looking for a service that converges their landline service and cellular phone service in a seamless, cost-effective way.

In a converging environment with wired and wireless phones, enterprise phones and cellular phones, desk phones and soft phones, voicemail and email, we must have the products and services in place so that each of these technologies are simple and seamless to use. One of the keys is to provide single-number "reachability" that is independent of the voice technology or device the client is using at the time. The Stanford phone number, regardless of technology or device, will continue to be the main user identifier for voice services for the foreseeable future; it is a requirement in order to obtain any of the Stanford voice services; and is an important part of the Stanford identity for clients.

Session Initiation Protocol, or SIP, will become the primary protocol for voice and other multimedia communications services. Major investments in voice services will focus primarily on SIP-capable systems and devices. However, SIP is still an emerging technology, and the features and functionalities available on SIP may not be as robust as those currently available on vendor-specific proprietary protocols. Therefore, IT Services may choose to deploy services with products utilizing proprietary protocols in order to meet current business needs, but will insure that any product purchased is capable of being converted to SIP.

Current State

The Voice Core network is physically separate from the main Stanford data network, SUNet. For security purposes, a redundant pair of geographically diverse firewalls separates the two networks. Within the Voice Core, the network is further sub-networked by technology and function. All of the major systems within this Voice Core are built with geographically diverse, redundant systems running in high-availability mode. The redundant pairs are dispersed between the Forsythe Data Center, East ECH (Electronic Communications Hub) and West ECH.

SUNet consists of ten semi-autonomous areas known as Operational Zones. The main academic areas of Stanford are served by six Operational Zones (OZ), with each OZ consisting of geographically diverse, redundant firewalls, routers and distribution switches. Each OZ's distribution switches connect to the Voice Core network over a 1Gbps connection. Departmental networks connect over a 1Gbps connection to both of the redundant OZ distribution switches, with one of the links designated as the primary for data traffic and the other link designated as the primary link for voice traffic. The data primary link is configured as the secondary link for voice traffic, and vice versa. Under normal operating conditions, voice traffic to and from the Voice Core network, and data traffic to and from SUNet does not compete for the same bandwidth until it reaches the departmental network's distribution switch. A high level overview of the topology is shown below.

Stable core technologies:

Time Division Multiplexing (TDM)
TDM services as a technology are well-established and well-understood. They are reliable, secure, and treated as a utility. Stanford's TDM-based services matches or exceeds the grade of service provided by the telephony industry at large. Stanford will use the existing TDM equipment to its full useful life where possible, while we deploy the core VoIP infrastructure and research emerging wireless technologies that could cause a fundamental shift in the direction of communication technologies and services. The TDM equipment currently meets the voice needs of much of the campus.

Voice Over IP (VoIP)
Stanford is positioning itself to take advantage of technology changes by a careful migration of voice services to VoIP technologies. The always-on expectation for voice services itself requires the supporting infrastructure (network, closets, and power) be engineered for nonstop, reliable operations. Redundant equipment and network paths are required to provide the required level of reliable service. The infrastructure upgrade of the approximately 1,470 communications closets on campus is extremely costly, but can be done over time. Since there is no compelling business reason for immediate replacement of all legacy TDM voice services with VoIP services, Stanford can deploy VoIP-based services when and where it makes business sense, such as in new or remodeled buildings, as well as in off-campus locations. A building's infrastructure can be enhanced to support VoIP while other construction is being done. VoIP and TDM voice services can coexist depending on the client's needs. VoIP simplifies new set installations, and also allows the client to relocate their phone. The VoIP telephone has become another network device, and voice another network application. As an application, the potential for expanded voice services using VoIP is enormous. Services such as presence, a concept that allows clients to control their own contact parameters, will provide clients with the ability to decide how, when, and with whom they communicate.

Enhanced Automatic Call Distribution (eACD)
eACD provides even call distribution to all members of any help desk, call center, or hospital clinic. eACDs provide call statistics, such as total number of calls answered, number of calls abandoned, maximum and average call holding times, and call wait times. Metrics such as these are critical to effective management of answering services, and are used by some organizations (SHC) to aid in accreditation. eACD must support:

  • Call routing by calling line ID, called party information, agent skill set, and Integrated Voice Response input.
  • Script-controlled call flow.
  • Traffic/event-based call flow.
  • Detailed management reporting.
  • Real-time monitoring.

Technologies new to IT Services:

Unified Messaging (UM)
Unified Messaging is the next generation of voicemail services. Voice messages can be delivered to the -mail desktop as a .wav file attachment, where the user can listen, delete, or forward the message, as with any other email. UM provides for other options to retrieve voice mail — in the traditional method from the desk phone, or via a web portal.

Find Me / Follow Me (FM-FM)
Find me / follow me allows the Unified Messaging system to route the call through a user-defined list of numbers. The numbers may be called simultaneously or sequentially, in a preferred order or in accordance with the user's scheduled activities and locations. Once the list has been called and no connection made, the system may route the call to voicemail.

Fax Server
Fax messages can be delivered to the email desktop as .tiff file attachments. Once the fax is delivered to email, the user can view, delete, print, or forward the message as with any other email. Users may also choose to log into the Unified Messaging portal to retrieve the fax.

Soft Phone
Just as hardwired computer connections have their place in the network fabric, fixed or wired VoIP desk phones will remain cost-effective in the voice arena. However, the soft phone is poised to supplant many wired VoIP implementations. With the demands of mobile clients and a mobile society growing daily, telephone-set emulation on a laptop will become the preferred method of communication for many, especially for those who telecommute on a regular basis.

VoIP over WiFi
WiFi VoIP phones, which have the same form factor as a cell phone, allow the client to become un-tethered from the desk, while still utilizing all of the features of an enterprise telephone system. The most common application of this technology is in the hospitals and clinics, where physicians and nurses can stay in contact wirelessly, not only through voice communications, but also with other medical specialty systems. WiFi capability is also useful in areas with poor cellular phone coverage.

Emerging technologies:

Presence is the ability for a client to choose how, when, and to whom voice services are delivered. An example of an application that provides presence is Instant Messaging. In IM, it is possible to see if a user is online and available, or off-line; what communications modes they have been enabled, etc. These presence capabilities are increasingly available for voice services.

Session Initiation Protocol (SIP)
SIP is becoming the signaling method of choice in the VoIP world, rapidly replacing the Signaling System No.7 (SS7), which has been in use by the Bell Operating Companies since 1980. Because SIP is IP-based, advanced services that require sophisticated signaling can now be accomplished without massive capital investments.

  • Informational applications on phones (LDAP directory integration, time schedules, streaming media).
  • Interactive applications on phones (Instant Messaging, multimedia conferencing).

Deprecated technologies:

  • TDM-only services
  • Voicemail without unified messaging
  • ACD (Automatic Call Distribution) without skills-based and event-driven routing


Stanford is a complex, diverse, and decentralized environment. The SHC's business needs are quite different from those of the main university's. Within the university, each school and department may have widely differing business needs. As the central provider of voice services, our products and services must be flexible in order to meet the demands of a highly diverse organization, and yet scalable in order to support the needs of a large population. Therefore, our strategic direction incorporates the concept of a mission-specific PBX, rather than deploying a single PBX solution that tries to meet the needs of the entire campus. With a mission-specific PBX strategy, the goal is to define the best systems for specific functions or for special communities of interest, and seamlessly tie them together within the core voice network, utilizing SIP. Avaya is defined as is a mission-specific PBX for providing call center functionality for all organizations across campus, including the SHC, LPCH and North Campus clinic. Nortel is the current mission-specific PBX for providing legacy voice services across campus, with Cisco Unified Communications the current standard for VoIP to the desktop. A Cisco IP-PBX is currently deployed to serve the needs of the SHC, a second Cisco IP-PBX is deployed to serve the need of the main academic campus, and the LPCH and the Stanford Linear Accelerator Center (SLAC) are candidates for having their own Cisco IP-PBX instance.

Despite the widespread deployment of IP-based telephony solutions, a business need remains for legacy voice capabilities. Many devices including fax machines and conference room speakerphones require an analog telephone line, as do elevator emergency phones. Some specialized areas, for example hospital emergency rooms may require both VoIP and legacy voice service in order to maximize availability and maintain technological diversity. Although technologies exist to convert a VoIP line to an analog phone line, in most cases it will be more efficient and operationally stable to continue maintaining the investment the university currently has in copper wiring and legacy voice systems.


  • Understand the impact of wireless VoIP on the existing voice and data infrastructures. Items of concern in this area are security, capacity, and wireless coverage.
  • Determine which SIP engine and services meet the needs of the Stanford community.
  • Determine which presence engine meets the needs of the Stanford community.
  • Provide a method for third-party applications to utilize IP portion of VoIP sets.


  • Deploy Avaya ACD system — completed May 2008.
  • Deploy Cisco VoIP to North Campus Clinic — completed February 2009.
  • Deploy Unified Messaging — May 2009.
  • Deploy Find Me / Follow Me — August 2009.
  • Deploy Cisco VoIP as part of Converged Communications service — ongoing.
  • Second-round pilot of fixed / mobile convergence options on smartphones.
  • Pilot presence technologies.
  • Sunset legacy Nortel VoIP service; replace with Cisco VoIP stand-alone or with Converged Communications service.
  • Unified Communications as a service umbrella which runs on a full IP Multimedia Subsystem (IMS).
  • Deploy SIP and SIP services.