Introduction to IP Telephony

IP telephony refers to cost-effective communication services, including voice, fax, and voice-messaging applications, transported via the packet-switched IP network rather than the circuit-switched PSTN.

KEY POINT

VoIP uses voice-enabled routers to convert voice into IP packets and route those packets between corresponding locations. Users do not often notice the implementation of VoIP in the network; they use their traditional phones, connected to a PBX. However, the PBX is not connected to the PSTN or to another PBX, but to a voice-enabled router that is an entry point to VoIP.

IP telephony replaces traditional phones with IP phones and uses the Cisco Unified Communications Manager, a server for call control and signaling, in place of PBXs. The IP phone itself performs voice-to-IP conversion, and voice-enabled routers are not required within the enterprise network. If connection to the PSTN is required, a voice-enabled router or other gateway must be added where calls are forwarded to the PSTN.

The basic steps for placing an IP telephone call include converting the analog voice signal into a digital format, and compressing and translating the digital signal into IP packets for transmission across the IP network. The process is reversed at the receiving end.

The IP telephony architecture, illustrated in Figure 8-18, includes four distinct components: infrastructure, call processing, applications, and client devices. These components are described as follows:

■ Infrastructure: The infrastructure is based on data link layer and multilayer switches and voice-enabled routers that interconnect endpoints with the IP and PSTN network. Endpoints attach to the network using switched 10/100 Ethernet ports. Switches may include Power over Ethernet (PoE) ports that sense the presence of IP devices that require inline power, such as Cisco IP phones and wireless access points, and provide that power. Voice-enabled routers perform conversions between the circuit-switched PSTN and IP networks.

■ Call processing: Cisco Unified Communications Manager is the software-based call-processing component of the Cisco enterprise IP telephony solution. Cisco Unified Communications Manager provides a scalable, distributable, and highly available enterprise IP telephony call processing solution and performs much like the PBX in a traditional telephone network, including providing call setup and processing functions.

The Cisco Unified Communications Manager can be installed on Cisco MCS 7800 Series server platforms and selected third-party servers.

Applications: Applications provide additional features to the IP telephony infrastructure. Cisco Unity unified messaging (integrating e-mail and voice mail), Cisco Unified MeetingPlace (multimedia conferencing), Cisco Unified IP IVR, and Cisco Unified Contact Center products (including intelligent contact routing, call treatment, network-to-desktop computer telephony integration, and multichannel automatic call distribution) are among the Cisco applications available for IP telephony. The open-source application layer allows third-party companies to develop software that interoperates with Cisco Unified Communications Manager.

Client devices: Client devices are IP telephones and software applications that allow communication across the IP network. Cisco Unified Communications Manager centrally manages the IP telephones through Ethernet connections in the Building Access Layer switches.

Figure 8-18 IP Telephony Components

Voice Messaging and Applications

Call-Processing Engine

Cisco Unifi Communications Manager

Figure 8-18 IP Telephony Components

Voice Messaging and Applications

Call-Processing Engine

Cisco Unifi Communications Manager

PSTN Gateway or Router

DSP Resources for Conferencing

DSP Resources for Conferencing

— QoS-Enabled WAN Infrastructure

PSTN Gateway or Router

— QoS-Enabled WAN Infrastructure

IP Telephony Design Goals

Typical design goals of an IP telephony network are as follows:

End-to-end IP telephony: Using end-to-end IP telephony between sites where IP connectivity is already established. IP telephony can be deployed as an overlaid service that runs on the existing infrastructure.

■ Widely usable IP telephony: To make IP telephony widely usable, voice quality should be at the same level as in traditional telephony; this is known as toll quality voice.

■ Reduced long-distance costs: Long-distance costs should be lower than with traditional telephony. This can be accomplished by using private IP networks, or possibly the public Internet, to route telephone calls.

■ Cost-effective: Making IP telephony cost effective depends on using the existing WAN capacity more efficiently and the cost-of upgrading the existing IP network infrastructure to support IP telephony. In some cases, this goal can be accomplished by using the public Internet or private IP networks to route telephone calls.

■ High availability: To provide high availability, redundant network components can be used and backup power can be provided to all network infrastructure components, including routers, switches, and IP phones.

■ Lower total cost of ownership: IP telephony should offer lower total cost of ownership and greater flexibility than traditional telephony. Installation costs and operational costs for unified systems are lower than the costs to implement and operate two infrastructures.

■ Enable new applications on top of IP telephony via third-party software: For example, an intelligent phone used for database information access as an alternative to a PC is likely to be easier to use and less costly to own, operate, and maintain.

■ Improved productivity: IP telephony should improve the productivity of remote workers, agents, and stay-at-home staff by extending the productivity-enhancing enterprise telephony features such as voice mail and voice conferencing to the remote teleworker.

■ Facilitate data and telephony network consolidation: Such consolidation can contribute to operational and equipment savings.

The following sections illustrate some sample IP telephony designs.

Single-Site IP Telephony Design

Figure 8-19 illustrates a design model for an IP telephony network within a single campus or site.

Figure 8-19 Single-Site IP Telephony Design

LAN Switch with Voice-Enabled Inline Power Router

Cisco Unified

Communications Voice Mail Manager

Cisco Unified

Communications Voice Mail Manager

Voice Trunk

A single-site IP telephony design consists of Cisco Unified Communications Manager, IP telephones, LAN switches with inline power (PoE), applications such as voice mail, and a voice-enabled router, all at the same physical location. The IP telephones are powered through their Ethernet interface via the LAN switch. Gateway trunks are connected to the PSTN so that users can make external calls.

Single-site deployment allows each site to be completely self-contained. All calls to the outside world and remote locations are placed across the PSTN. If an IP WAN is incorporated into the single-site model, it is for data traffic only; no telephony services are provided over the WAN. Therefore, there is no loss of the call processing service or functionality if an IP WAN failure occurs or if the WAN has insufficient bandwidth. The only external requirements are a PSTN carrier and route diversity within the PSTN network. As a recommended practice, use this model for a single campus or a site with fewer than 30,000 lines.

Multisite WAN with Centralized Call Processing Design

Figure 8-20 presents a multisite WAN design model with centralized call processing; Cisco Unified Communications Manager at the central site connects to remote locations through the IP WAN. Remote IP telephones rely on the centralized Cisco Unified Communications Manager to handle their call processing. The IP WAN transports voice traffic between sites and carries call control signaling between the central site and the remote sites. Applications such as voice mail and IVR systems are also centralized, therefore reducing the overall cost of ownership and centralized administration and maintenance.

Figure 8-20 Multisite WAN with Centralized Call Processing Design

Voice Mail

Cisco Unified

Communications

Manager

Voice-Enabled Router

Figure 8-20 Multisite WAN with Centralized Call Processing Design

Voice Mail

Cisco Unified

Communications

Manager

Voice-Enabled Router

Remote Location

IP Phones Managed with the Central Cisco Unified

Communications Manager

Remote Location

IP Phones Managed with the Central Cisco Unified

Communications Manager

The remote locations require IP connectivity with the Enterprise Campus. IP telephones, powered by a local LAN switch, convert voice into IP packets and send them to the local LAN. The local router forwards the packets to the appropriate destination based on its routing table. In the event of a WAN failure, the voice-enabled routers at the remote sites can provide backup call processing functionality with Cisco Unified Survivable Remote Site Telephony (SRST) services. Cisco Unified SRST extends high-availability IP telephony to branch offices by providing backup call processing functionality on voice-enabled routers.

If an enterprise requires high-quality voice communication over the WAN, the service provider must implement QoS mechanisms. Enterprises and service providers usually sign a service level agreement (SLA) that guarantees bandwidth and latency levels suitable for voice transport.

NOTE The routers are voice-capable to enable voice communication with the outside world through the PSTN.

As a recommended practice, use this model for a main site with many smaller remote sites that connect via a QoS-enabled WAN but that do not require full features and functionality during a WAN outage.

Multisite WAN with Distributed Call Processing Design

Figure 8-21 illustrates a multisite WAN design model with distributed call processing. This model consists of multiple independent sites, each with its own call processing agent and connected to an IP WAN that carries voice traffic between the distributed sites. The IP WAN in this model does not carry call control signaling between the sites because each site has its own call processing agent. Typically, the PSTN serves as a backup connection between the sites in case the IP WAN connection fails or has insufficient available bandwidth for incremental calls.

Figure 8-21 Multisite WAN with Centralized Call Processing Design

Voice Mail

Cisco Unified

Communications

Manager

Voice-Enabled Router

Figure 8-21 Multisite WAN with Centralized Call Processing Design

Voice Mail

Cisco Unified

Communications

Manager

Voice-Enabled Router

Remote Location

Cisco Unified

Communications

Manager

Remote Location

Cisco Unified

Communications

Manager

NOTE A site connected only through the PSTN is a standalone site and is not covered by the distributed call processing model.

As a recommended practice, use this model for a large central site with more than 30,000 lines or for a deployment with more than six large sites (more than 30,000 lines total) interconnected via a QoS-enabled WAN.

NOTE IP telephony functionality can be scaled to a small site or branch office with Cisco Unified Communications Manager Express, which is embedded in Cisco IOS software and provides call processing for up to 240 Cisco Unified IP phones. This product offers customers a low-cost, reliable, and feature-rich solution for deployment.

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  • Daniela
    What is inline telephony?
    1 year ago
  • semrawit asfaha
    What are the IP telephony design goals?
    12 months ago

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