Introducing VoIP Networks 2.1.3 Legacy Analog Interfaces in VoIP Networks A VoIP network that includes legacy equipment, such as analog telephones, needs gateways to convert analog signals into digital format and encapsulate them into IP packets. A network built with IP-enabled devices needs no conversion. Gateways use different types of interfaces to connect to analog devices, such as telephones, fax machines, or PBX or PSTN switches. Analog interfaces that are used at the gateways include these three types:

• Foreign Exchange Station (FXS): FXS is a telephone interface that provides battery power, sends a dial tone, and generates ringing voltage. A standard telephone plugs into such an interface to receive telephone service. A telephone exchange is an example of an FXS. In VoIP implementations, the FXS interface connects to analog end systems; the analog end system uses the Foreign Exchange Office (FXO) interface on the end system side. The router FXS interface behaves like a PSTN or a PBX by serving telephones, answering machines, or fax machines with line power, ring voltage, and dial tones. If a PBX uses an FXO interface, it can also connect to a router FXS interface. In this case, the PBX acts like a telephone.
• Foreign Exchange Office (FXO): An FXO is a telephone interface that connects to the PSTN. FXO generates the on-hook and off-hook indicators that are used to signal a loop closure at the FXO’s end of the circuit. In VoIP implementations, the FXO interface connects to the PSTN or a PBX; both the PSTN and PBX use the FXS interface on their side. The router FXO interface behaves like a telephone, obtaining line power, ring voltage, and dial tones from the other side of the interface. As mentioned, a PBX can also use an FXO interface toward the router (which will then use an FXS interface) if the PBX takes the role of the telephone.
• Ear and Mouth (E&M): The E&M interface provides signaling for analog trunks. Analog trunks interconnect two PBX-style devices, such as any combination of a gateway (acting as a PBX), a PBX, and a PSTN switch. E&M is often defined as “ear and mouth,” but it derives from the term “earth and magnet.” “Earth” represents the electrical ground, and “magnet” represents the electromagnet used to generate tones.

In Figure , the gateway serves a telephone and a fax machine using two FXS interfaces. For these two devices, the router acts like a PBX or a PSTN switch. The router connects to the PSTN using an FXO interface. For this connection, the router acts like a telephone toward the PSTN. Another FXO interface connects to a PBX (PBX-1). Again, the router acts like an end system toward the PBX, and hence uses the same port type as the telephone and fax that connect to PBX-1 use. A second PBX (PBX-2) connects to the router FXS interface. For this connection, the PBX behaves like a telephone toward the router, and the router acts like a PSTN switch. Finally, the router connects to another PBX (PBX-3), this time using an E&M interface. On this trunk connection, both the router and PBX-3 act as a PBX.

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Content 2.1 Introducing VoIP Networks 2.1.4 Digital Interfaces Gateways can use digital interfaces to connect to voice equipment. From a hardware perspective, there are BRIs and T1 and E1 interfaces available. All of the interfaces use TDM to support multiple logical channels. T1 and E1 interfaces can use either channel associated signaling (CAS) or common channel signaling (CCS), while a BRI always uses CCS. ISDN PRIs use T1 or E1 CCS. Figure illustrates a router that serves an ISDN telephone using a BRI voice interface. In addition, the router has two T1 or E1 lines: one to a PBX and one to the PSTN. Depending on the interface type (T1 or E1) and the signaling method (CAS or CCS), a maximum of 23, 24, or 30 voice channels are available on these trunks. The PSTN and the PBX also serve ISDN telephones through BRI connections. The table in Figure shows the number of voice channels, the signaling bandwidth, total bandwidth, and framing overhead for all available interface and signaling options.

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Content 2.1 Introducing VoIP Networks 2.1.5 Stages for Completing a VoIP Telephone Call Although different protocols deal with call control in different ways, all protocols provide a common set of services. There are three components of basic call control:

• Call setup: Call setup checks the call-routing configuration to determine the destination of a call. The configuration specifies the bandwidth requirements for the call. Knowing the bandwidth requirements, CAC determines whether sufficient bandwidth is available to support the call. If bandwidth is available, call setup generates a setup message and sends the message to the destination. If bandwidth is not available, call setup presents the call initiator with a busy signal. Different call control protocols, such as H.323, MGCP, and Session Initiation Protocol (SIP), define different sets of messages that devices exchange during setup. However, all of the messages consult the same basic information:
  • The IP addresses of the two devices that will exchange VoIP
  • The User Datagram Protocol (UDP) port numbers that will be used for the Real-Time Transport Protocol (RTP) streams which carry the voice traffic
  • The format (for example, the compression algorithm) used for the digitized voice
• Call maintenance: Call maintenance tracks packet count, packet loss, jitter, and delay during the call. Information passes to the voice-enabled devices to determine whether connection quality is good or has deteriorated to the point where the gateway will drop the call.
• Call teardown: Call teardown notifies voice-enabled devices to make resources available for other calls and make them available for other calls when either side terminates a call.

Note
Later lessons in this module explain the protocols mentioned here. Return to this section to place these protocols in perspective.

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Content 2.1 Introducing VoIP Networks 2.1.6 Distributed Call Control There are two types of call control: distributed and centralized. In the past, all voice networks used a centralized architecture in which dumb endpoints (telephones) were controlled by centralized switches. Although this model worked well for basic telephony services, it mandated a trade-off between simplified management and endpoint and service innovation. One of the benefits of VoIP technology is that it allows networks to use either a centralized or a distributed architecture. This flexibility allows companies to build networks characterized by both simplified management and endpoint innovation, depending on the protocol used. Figure illustrates the distributed model in which multiple components in the network handle call control. Configuring the voice-capable device to support call control directly enables distributed call control using protocols such as H.323 or SIP. With distributed call control, the devices perform the call setup, call maintenance, and call teardown without telephone company involvement:

• Process dialed digits and route the call (sequence shown in Figure ):
  1. After detecting a service request (the telephone is taken off the hook), the first gateway (R1) plays a dial tone.
  2. Next, R1 collects the digits that the caller dials.
  3. R1 then looks up the called number in the R1 local call routing table. According to the call routing table, the called number uses the second gateway (R2).
  4. R1 now enters the first stage of a call, call setup, by sending the appropriate message to R2.
  5. R2 receives the call setup message from R1.
  6. R2 then looks up the called number in its local call routing table. According to the call routing table, the called number uses a local voice port.
  7. R2 sends the call to the local voice port by