Cisco Voice over IP: Routing calls over analog voice ports

This chapter introduces basic configuration of analog and digital voice ports and demonstrates how to fine-tune voice ports with port-specific configurations. Upon completing this chapter, you will be able to configure voice interfaces on Cisco voiceenabled equipment for connection to traditional, nonpacketized telephony equipment.

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Cisco Voice over IP (CVOICE) (Authorized Self-Study Guide, 3rd edition)
Chapter 3, Routing Calls over Analog Voice Ports

Cisco Voice over IPVoice gateways bridge the gap between the VoIP world and the traditional telephony world (for example, a private branch exchange [PBX], the public switched telephone network {PSTN], or an analog phone). Cisco voice gateways connect to traditional telephony devices via voice ports. Routing Calls Over Analog Voice Ports introduces basic configuration of analog and digital voice ports and demonstrates how to fine-tune voice ports with port-specific configurations. In this chapter, you'll learn how to configure voice interfaces on Cisco voice-enabled equipment for connection to traditional, non-packetized telephony equipment.

Voice Ports

Voice ports on routers and access servers emulate physical telephony switch connections so that voice calls and their associated signaling can be transferred intact between a packet network and a circuit-switched network or device. For a voice call to occur, certain information must be passed between the telephony devices at either end of the call, such as the on-hook status of the devices, the availability of the line, and whether an incoming call is trying to reach a device. This information is referred to as signaling, and to process it properly, the devices at both ends of the call segment, which are directly connected to each other, must use the same type of signaling.

The devices in the packet network must be configured to convey signaling information in a way that a circuit-switched network can understand. They must also be able to understand signaling information that is received from the circuit-switched network. This is accomplished by installing appropriate voice hardware in a router or access server and by configuring the voice ports that connect to telephony devices or the circuit-switched network.

Signaling Interfaces

Voice ports on routers and access servers physically connect the router, access server, or call control device to telephony devices such as telephones, fax machines, PBXs, and PSTN central office (CO) switches through signaling interfaces.

These signaling interfaces generate information about things such as:

  • On-hook status
  • Ringing
  • Line seizure

The voice port hardware and software of the router need to be configured to transmit and receive the same type of signaling being used by the device they are interfacing with so calls can be exchanged smoothly between a packet network and a circuit-switched network.

The signaling interfaces discussed in the next sections include FXO, FXS, and E&M, which are types of analog interfaces. Digital signaling interfaces include T1, E1, and ISDN. Some digital connections emulate FXO, FXS, and E&M interfaces. It is important to know which signaling method the telephony side of the connection is using and to match the router configuration and voice interface hardware to that signaling method.

Analog Voice Ports

Analog voice port interfaces connect routers in packet-based networks to analog twowire or four-wire circuits in telephony networks. Two-wire circuits connect to analog telephone or fax devices, and four-wire circuits connect to PBXs. Connections to the PSTN CO are typically made with digital interfaces. Three types of analog voice interfaces are supported by Cisco gateways.

The following is a detailed explanation of each of the three types of analog voice interfaces:

  • FXS: An FXS interface connects the router or access server to end-user equipment such as telephones, fax machines, or modems. The FXS interface supplies ring, voltage, and dial tone to the station and includes an RJ-11 connector for basic telephone equipment, key sets, and PBXs.
  • FXO: An FXO interface is used for trunk, or tie-line, connections to a PSTN CO or to a PBX that does not support E&M signaling (when the local telecommunications authority permits). This interface is of value for off-premises station applications. A standard RJ-11 modular telephone cable connects the FXO voice interface card to the PSTN or PBX through a telephone wall outlet.
  • E&M: Trunk circuits connect telephone switches to one another. They do not connect end-user equipment to the network. The most common form of analog trunk circuit is the E&M interface, which uses special signaling paths that are separate from the trunk audio path to convey information about the calls. The signaling paths are known as the E-lead and the M-lead. E&M connections from routers to telephone switches or to PBXs are preferable to FXS and FXO connections because E&M provides better answer and disconnect supervision.

    The name E&M is thought to derive from the phrase Ear and Mouth or rEceive and transMit, although it could also come from Earth and Magneto. The history of these names dates back to the early days of telephony, when the CO side had a key that grounded the E circuit, and the other side had a sounder with an electromagnet attached to a battery. Descriptions such as Ear and Mouth were adopted to help field personnel understanding and determine the direction of a signal in a wire.

    Like a serial port, an E&M interface has a DTE/DCE type of reference. In the telecommunications world, the trunking side is similar to the DCE and is usually associated with CO functionality. The router acts as this side of the interface. The other side is referred to as the signaling side, like a DTE, and is usually a device such as a PBX.

Analog Signaling

The human voice generates sound waves, and the telephone converts the sound waves into electrical signals, analogous to sound. Analog signaling is not robust because of line noise. Analog transmissions are boosted by amplifiers because the signal diminishes the farther it travels from the CO. As the signal is boosted, the noise is also boosted, which often causes an unusable connection.

In digital networks, signals are transmitted over great distances and coded, regenerated, and decoded without degradation of quality. Repeaters amplify the signal and clean it to its original condition. Repeaters then determine the original sequence of the signal levels and send the clean signal to the next network destination.

Voice ports on routers and access servers physically connect the router or access server to telephony devices such as telephones, fax machines, PBXs, and PSTN CO switches. These devices might use any of several types of signaling interfaces to generate information about on-hook status, ringing, and line seizure.

Signaling techniques can be placed into one of three categories:

  • Supervisory: Involves the detection of changes to the status of a loop or trunk. When these changes are detected, the supervisory circuit generates a predetermined response. A circuit (loop) can close to connect a call, for example.

  • Addressing: Involves passing dialed digits (pulsed or tone) to a PBX or CO. These dialed digits provide the switch with a connection path to another phone or customer premises equipment (CPE).

  • Informational: Provides audible tones to the user, which indicates certain conditions such as an incoming call or a busy phone.

Download and read Routing Calls over Analog Voice Ports in its entirety.

Excerpted with permission from Cisco Voice over IP (CVOICE) (Authorized Self-Study Guide, 3rd edition), by Kevin Wallace, Copyright 2008. Published by Cisco Press, June 2008. For more information about this book and other titles, please visit Cisco Press.

This was first published in September 2008

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