The Role of Voice Gateways

The basic function of a gateway is to translate between different types of networks. In the data environment, a gateway might translate between a Frame Relay network and an Ethernet network, for example. In a VoIP environment, voice gateways are the interface between a VoIP network and the public switched telephone network (PSTN), a private branch exchange (PBX), or analog devices such as fax machines. In its simplest form, a voice gateway has an IP interface and a legacy telephone interface, and it handles the many tasks involved in translating between transmission formats and protocols. At least one gateway is an essential part of any IP telephony network that interacts with the PSTN or with analog devices. In addition, when gateways are properly configured, many can take over for a Cisco CallManager when it is unreachable.

Figure 1-1 shows a simple VoIP network with a voice gateway connected to both the VoIP network and the PSTN.

Figure 1-1. Voice Gateway Example

The gateway allows communication between the two networks by performing tasks such as these:

• Interfacing with the IP network and the PSTN or PBX.
• Supporting IP call control protocols, in addition to time-division multiplexing (TDM) call control protocols.
• Performing call setup and teardown for calls between the VoIP and PSTN networks by terminating and reoriginating the call media and signaling.
• Providing supplementary services, such as call hold and transfer.
• Relaying dual tone multifrequency (DTMF) tones.
• Supporting analog fax and modems over the IP network.
• In a Cisco CallManager network, a gateway also needs to do the following:

  - Support CallManager redundancy by rehoming to alternate CallManagers.

  - Support call survivability when no CallManager is available.

Gateways communicate with other gateways, gatekeepers, their endpoints, or their call control agents, such as Cisco CallManager or a PBX. The following are the protocols that Cisco gateways use for voice signaling and media:

• Media Gateway Control Protocol (MGCP)
• H.323
• Session Initiation Protocol (SIP)
• Skinny Client Control Protocol (SCCP)
• Real-Time Transport Protocol (RTP)

This chapter introduces these protocols. MGCP, H.323, and SIP are described in depth in other chapters of this book.

Types of Voice Gateways

This book deals primarily with using Cisco routers as voice gateways, but an assortment of different types and models of Cisco voice gateways is available. Gateways contain analog or digital telephony ports, in addition to IP interfaces. Analog options include Foreign Exchange Office (FXO) ports, Foreign Exchange Station (FXS) ports, and Ear and Mouth (E&M) ports. Digital options include T1/E1 PRI ports, ISDN BRI ports, and T1 channel associated signaling (CAS) ports. You can divide Cisco gateways into several categories, each with its own capabilities. For more information on the current models, visit www.Cisco.com.

Routers

You can use many Cisco routers, such as 1700, 2600, 2800, 3600, 3700, 3800, 7200, and 7500 models, as voice gateways. Voice gateway routers can contain analog ports such as FXO and FXS, and digital ports such as E1 and T1. Most can use the MGCP, SCCP, SIP, and H.323 protocols. The remainder of this book focuses mainly on router-based voice gateways.

Standalone Voice Gateways

Unlike routers, standalone voice gateways are used only as voice gateways. They come in two types: digital and analog.

Digital trunk gateways, such as the AS5000 series interface, connect an IP telephony network and the PSTN or a PBX via their trunk ports.

Analog trunk gateways, such as AT-2 or AT-4, connect to the PSTN or a PBX via analog trunks. Analog station gateways, such as an ATA186, VG224, or VG248, connect to analog devices such as telephones, fax machines, or voice-mail systems. The signaling protocols that are available vary by gateway model. The ATA186, an analog-to-IP adapter, can support two analog devices each with its own telephone number. It has one Ethernet port that connects to the VoIP network, and two voice ports for connecting analog devices. It can be controlled by Cisco CallManager and CallManager Express, and it supports SIP and SCCP protocols. However, CallManager 5.0 does not have native SIP support for the ATA186.

VG224 and VG248 are Cisco voice gateways with 24 and 48 analog FXS ports, respectively. These are line-side gatewaysthey do not interface with the PSTN, but with end-user analog devices. Both models allow a CallManager, CallManager Express (CME), or an SRST router multiple to control analog ports. They connect to a LAN via an Ethernet port, and they communicate with CallManager or CME using the SCCP telephony control (STC) application. The VG224 can also act as an MGCP, H.323, or SIP gateway.

The analog telephones, faxes, or modems that are attached to the gateway appear as individual endpoints to the CallManager. You must configure the gateway on the CallManager, and you must configure each port on CallManager with a directory number and any call features it needs. In CME, you must set up an e-phone for each port in use.

The VG248 gateway uses only SCCP. The VG224 gateway can use SCCP, H.323, MGCP, and SIP. CME interaction is supported only with the VG224 gateway, not VG248. The VG224 gateway is Cisco-IOS based, whereas configuration of the VG248 gateway is menu driven.

Switch Modules

Modules containing analog and digital ports are available for Cisco 6500 switches, allowing them to act as voice gateways. For example, the Communication Media Module (which you can also use in the 7600 series router) can contain a combination of T1, E1, and FXS ports. It uses MGCP, H.323, and SIP protocols, and it can provide Survivable Remote Site Telephony (SRST) functionality. The Voice T1/E1 and Services Module, or 6608 blade, can contain T1 or E1 interfaces, and it can perform as an MGCP gateway.