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After reading this chapter, you should be able to perform the following tasks:
Describe the various call types in a VoIP network.
Configure analog voice interfaces as new devices are introduced into the voice path.
Configure dial peers so you can add call routing intelligence to a router.
Voice 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. 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 voice-enabled equipment for connection to traditional, nonpacketized telephony equipment.
Before delving into the specific syntax of configuring voice ports, this section considers several examples of voice applications. The applications discussed help illustrate the function of the voice ports, whose configuration is addressed in the next section.
Different types of applications require specific types of ports. In many instances, the type of port is dependent on the voice device connected to the network. Different types of voice applications include the following:
Local calls
On-net calls
Off-net calls
Private line, automatic ringdown (PLAR) calls
PBX-to-PBX calls
Intercluster trunk calls
On-net to off-net calls
The following sections discuss each in detail and provide an example.
Local calls, as illustrated in Figure 3-1, occur between two telephones connected to one Cisco voice-enabled router. This type of call is handled entirely by the router and does not travel over an external network. Both telephones are directly connected to Foreign Exchange Station (FXS) ports on the router.
Figure 3-1
Local Calls
An example of a local call is one staff member calling another staff member at the same office. This call is switched between two ports on the same voice-enabled router.
On-net calls occur between two telephones on the same data network, as shown in Figure 3-2. The calls can be routed through one or more Cisco voice-enabled routers, but the calls remain on the same data network. The edge telephones attach to the network through FXS ports or through a PBX, which typically connects to the network via a T1 connection. IP phones that connect to the network via switches place on-net calls through Cisco Unified Communications Manager. The connection across the data network can be a LAN connection, as in a campus environment, or a WAN connection, as in an enterprise environment.
Figure 3-2
On-Net Calls
Note - The act of routing voice data across the WAN instead of the PSTN is known as toll-bypass. Originally, companies saved significant amounts of money using this strategy, which was one of the first major business benefits of a VoIP-enabled network.
An example of an on-net call is one staff member calling another staff member at a remote office. The call is sent from the local voice-enabled router, across the IP network, and terminated on the remote office voice-enabled router.
Figure 3-3 shows an example of an off-net call. To gain access to the PSTN, the user dials an access code, such as 9, from a telephone directly connected to a Cisco voice-enabled router or PBX. The connection to the PSTN is typically a single analog connection via a Foreign Exchange Office (FXO) port or a digital T1 or E1 connection.
Figure 3-3
Off-Net Calls
An example of an off-net call is a staff member calling a client who is located in the same city. The call is sent from the local voice-enabled router that is acting as a gateway to the PSTN. The call is then sent to the PSTN for call termination.
PLAR calls automatically connect a telephone to a second telephone when the first telephone goes off hook, as depicted in Figure 3-4. When this connection occurs, the user does not get a dial tone, because the voice-enabled port that the telephone is connected to is preconfigured with a specific number to dial. A PLAR connection can work between any type of signaling, including E&M, FXO, FXS, or any combination of analog and digital interfaces. For example, you might have encountered a PLAR connection at an airline ticket counter where you pick up a handset and are immediately connected with an airline representative.
Figure 3-4
PLAR Calls
An example of a PLAR call is a client picking up a customer service telephone located in the lobby of the office and being automatically connected to a customer service representative without dialing any digits. The call is automatically dialed based on the PLAR configuration of the voice port. In this case, as soon as the handset goes off hook, the voice-enabled router generates the preconfigured digits to place the call.
PBX-to-PBX calls, as shown in Figure 3-5, originate at a PBX at one site and terminate at a PBX at another site while using the network as the transport between the two locations. Many business environments connect sites with private tie trunks. When migrating to a converged voice and data network, this same tie-trunk connection can be emulated across an IP network. Modern PBX connections to a network are typically digital T1 or E1 with channel associated signaling (CAS) or Primary Rate Interface (PRI) signaling, although PBX connections can also be analog.
Note - PBX-to-PBX calls are another form of toll-bypass.
An example of a PBX-to-PBX call is one staff member calling another staff member at a remote office. The call is sent from the local PBX, through a voice-enabled router, across the IP network, through the remote voice-enabled router, and terminated on the remote office PBX.
Figure 3-5
PBX-to-PBX Calls
As part of an overall migration strategy, a business might replace PBXs with Cisco Unified Communications Managers. This includes IP phones connected to the IP network. Cisco Unified Communications Manager performs the call-routing functions formerly provided by the PBX. When an IP phone call is placed using a configured Cisco Unified Communications Manager, the call is assessed to see if the call is destined for another IP phone under its control or if the call must be routed to a remote Cisco Unified Communications Manager for call completion. Intercluster trunk calls, as depicted in Figure 3-6, are routed between Cisco Unified Communications Manager clusters using a trunk.
Figure 3-6
Intercluster Trunk Calls
An example of an intercluster trunk call is one staff member calling another staff member at a remote office using an IP phone. The call setup is handled by the Cisco Unified Communications Managers at each location. After the call is set up, the IP phones generate Real-time Transport Protocol (RTP) segments that carry voice data between sites.
When planning a resilient call-routing strategy, you might need to reroute calls through a secondary path should the primary path fail. An on-net to off-net call, as illustrated in Figure 3-7, originates on an internal network and is routed to an external network, usually to the PSTN. On-net to off-net call-switching functionality might be necessary when a network link is down or if a network becomes overloaded and unable to handle all calls presented.
Figure 3-7
On-Net to Off-Net Calls
Note - On-net to off-net calls might occur as a result of dial plan configuration, or they might be redirected by Call Admission Control (CAC).
An example of an on-net to off-net call is one staff member calling another staff member at a remote office while the WAN link is congested. When the originating voice-enabled router determines it cannot complete the call across the WAN link, it sends the call to the PSTN with the appropriate dialed digits to terminate the call at the remote office via the PSTN network.
The following steps, numbered in Figure 3-7, summarize the call flow of an on-net to off-net call:
Step 1. A user on the network initiates a call to a remote site.
Step 2. The output of the WAN gateway is either down or congested, so the call is rerouted.
Step 3. The call connects to the PSTN.
Step 4. The PSTN completes the call to the remote site.
Table 3-1 lists application examples for each type of call.
|
Type of Call |
Example |
|
Local call |
One staff member calls another staff member at the same office. The call is switched between two ports on the same voice-enabled router. |
|
On-net call |
One staff member calls another staff member at a remote office. The call is sent from the local voice-enabled router, across the IP network, and is terminated on the remote office voice-enabled router. |
|
Off-net call |
A staff member calls a client who is located in the same city. The call is sent from the local voice-enabled router, which acts as a gateway, to the PSTN. The call is then sent to the PSTN for call termination. |
|
PLAR call |
A client picks up a customer service telephone located in the lobby of an office and is automatically connected to a customer service representative without dialing any digits. The call is automatically dialed based on the PLAR configuration of the voice port. In this case, as soon as the handset goes off hook, the voice-enabled router generates the prespecified digits to place the call. |
|
PBX-to-PBX call |
One staff member calls another staff member at a remote office. The call is sent from the local PBX, through a voice-enabled router, across the IP network, through the remote voice-enabled router, and terminated on the remote office PBX. |
|
Intercluster trunk call |
One staff member calls another staff member at a remote office using IP phones. The call setup is handled by a Cisco Unified Communications Manager server at each location. After the call is set up, the IP phones generate IP packets carrying voice between sites. |
|
On-net to off-net call |
One staff member calls another staff member at a remote office while the IP network is congested. When the originating voice-enabled router determines that it cannot complete the call across the IP network, it sends the call to the PSTN with the appropriate dialed digits to terminate the call at the remote office via the PSTN network. |
Connecting voice devices to a network infrastructure requires an in-depth understanding of the signaling and electrical characteristics specific to each type of interface. Improperly matched electrical components can cause echo and create poor audio quality. Configuring devices for international implementation requires knowledge of country- specific settings. This section examines analog voice ports, analog signaling, and configuration parameters for analog 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. Figure 3-8 shows typical examples of how voice ports are used.
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.
Figure 3-8
Voice Ports
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 port interfaces connect routers in packet-based networks to analog two-wire 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, as illustrated in Figure 3-9.
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.
Figure 3-9
Analog Voice Ports
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.
Note - Depending on how the router is connected to the PSTN, the voice gateway might provide clocking to an attached key system or PBX, because the PSTN has more accurate clocks, and the voice gateway can pass this capability to downstream devices.
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.
FXS and FXO interfaces indicate on-hook or off-hook status and the seizure of telephone lines by one of two access signaling methods: loop-start or ground-start. The type of access signaling is determined by the type of service from the telephone company’s CO. Standard home telephone lines use loop-start, but business telephones can order ground-start lines instead.
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