VoIP Fundamentals

Building Scalable Dial Plans

To integrate VoIP networks into existing voice networks, network administrators must have the skills and knowledge to implement a scalable numbering plan and a comprehensive, scalable, and logical dial plan. This section describes the attributes of numbering plans and scalable dial plans for voice networks, addresses the challenges of designing these plans, and identifies the methods of implementing dial plans.


Hierarchical Numbering Plans

The previous section discussed the Public International Telecommunications Numbering Plan (E.164), national numbering plans, and private numbering plans. Each of the numbering plans can benefit, in terms of scalability, from a hierarchical design. A hierarchical design has the following advantages:

• Simplified provisioning Refers to the ability to easily add new groups and modify existing groups

• Simplified routing Keeps local calls local and uses a specialized number, such as an area code, for long-distance calls

• Summarization Establishes groups of numbers in a specific geographical area or functional group

• Scalability Provides additional high-level number groups

• Management Controls number groups from a single point in the overall network

The North American Numbering Plan (NANP) serves as a good role model for a scalable numbering plan. Consider how the NANP might be adapted to your environment. To illustrate the operation of the NANP, examine Figure 5-11.


In Figure 5-11, the calling party dials 1-703-555-0123. The calling party's local central office (CO) forwards the call to a long-distance carrier because the first digit (that is, the 1) indicated the call was a long-distance call. The long-distance carrier then forwards the call, based on the dialed area code, to a Virginia long-distance office. The Virginia long-distance office forwards the call, based on the CO code (that is, the NXX code), to an Alexandria CO. Finally, the Alexandria CO, based on the last four digits, forwards the call out to the called party.

While the NANP acts as a good starting point in designing a numbering plan, it is not always easy to design a hierarchical numbering plan. Existing numbering plans in the network might include proprietary PBXs, key systems, and telephony services such as a Centrex service. The necessity to conform to the PSTN at the gateways also contributes to the complexity of the design. Translation between these systems is a difficult task. If possible, avoid retraining system users. The goal is to design a numbering plan that has the following attributes:

• Minimal impact on existing systems
• Minimal impact on users of the system
• Minimal translation configuration
• Consideration of anticipated growth

Scalable Dial Plans

The North American telephone network is designed around a ten-digit numbering plan that consists of three-digit area codes and seven-digit telephone numbers, as shown in Figure 5-13. For telephone numbers that are located within an area code, the PSTN often uses a seven-digit dial plan. Features within a CO-based PBX, such as Centrex, allow the use of a custom five-digit dial plan for customers who subscribe to that service. PBXs are more flexible and allow for variable-length dial plans containing 3 to 11 digits.


Dial plans contain specific dialing patterns for a user who wants to reach a particular telephone number. Dial plans also contain access codes, area codes, specialized codes, and combinations of the numbers of digits dialed.

Dial plans require knowledge of the customer network topology, current telephone number dialing patterns, proposed router and gateway locations, and traffic-routing requirements. If the dial plans are for a private internal voice network that is not accessed by the outside voice network, the telephone numbers can be any number of digits.

Typically, companies that implement VoIP networks carry voice traffic within the least expensive systems and paths. Implementing this type of system involves routing calls through IP networks, private trunks, PBXs, key systems, and the PSTN. The numbering plan to support the system is scalable, easily understood by the user, and transportable between all of the system components. The use of alternate path components reduces instances of call failure. Finally, the numbering plan conforms to all applicable standards and formats for all of the systems involved.


Scalable Dial Plan Attributes

When designing a large-scale dial plan, Cisco recommends you adhere to the following attributes:

• Logic distribution Good dial plan architecture relies on the effective distribution of the dial plan logic among the various components. Devices that are isolated to a specific portion of the dial plan reduce the complexity of the configuration. Each component focuses on a specific task accomplishment. Generally, the local switch or gateway handles details that are specific to the local point of presence (POP). Higher-level routing decisions are passed along to the gatekeepers and PBXs. A well-designed network places the majority of the dial plan logic at the gatekeeper devices.

• Hierarchical design (scalability) You should attempt to keep the majority of the dial plan logic (routing decisions and failover) at the highest-component level. Maintaining a hierarchical design makes the addition and deletion of number groups more manageable. Scaling the overall network is much easier when configuration changes are made to a single component.

• Simplicity in provisioning Keep the dial plan simple and symmetrical when designing a network. Try to keep consistent dial plans on the network by using translation rules to manipulate the local digit dialing patterns. These number patterns are normalized into a standard format or pattern before the digits enter the VoIP core. Putting digits into a standard format simplifies provisioning and dial-peer management.

• Reduction in postdial delay Consider the effects of postdial delay in the network when you design a large-scale dial plan. Postdial delay is the time between the last digit dialed and the moment the phone rings at the receiving location. In the PSTN, people expect a short postdial delay and to hear ringback within seconds. The more translations and lookups that take place, the longer the postdial delay becomes. Overall network design, translation rules, and alternate pathing affect postdial delay. Therefore, you should efficiently use these tools to reduce postdial delay.

• Availability and fault tolerance Consider overall network availability and call success rates when you design a dial plan. Fault tolerance and redundancy within VoIP networks are most important at the gatekeeper level. By using an alternate path you help provide redundancy and fault tolerance in the network.

• Conformance to public standards Different geographical locations might impose restrictions to your dial plan. Therefore, familiarize yourself with any such limitations prior to designing your dial plan.

VoIP Network Architectures

One benefit of VoIP technology is that it allows networks to be built using either a centralized or a distributed architecture. Corporate business requirements dictate the architecture and functionality that is required. This section discusses centralized and distributed architectures and the gateway requirements to support these architectures in enterprise and service provider environments.

Support for protocols, signaling capabilities, voice features, and voice applications is changing and growing quickly. You must have a good understanding of voice network architectures to know which business requirements each architecture addresses. Also, gateways play an important role in providing access to the right mix of functionality. You must understand the main features and functions required in enterprise and service provider environments to choose the appropriate gateway.


Centralized Network Architectures

One benefit of VoIP technology is that it works with centralized and distributed architectures. This flexibility allows companies to build networks characterized by both simplified management and endpoint innovation. It is important to understand the protocols that are used to achieve this type of VoIP network agility.

The multisite WAN model with centralized call processing, as illustrated in Figure 5-2, consists of the following components:

• Central gateway controller (call agent) The call agent handles switching logic and call control for all sites under the central controller. A central gateway controller includes both centralized configuration and maintenance of call-control functionality. When new functionality needs to be added, only the controller needs to be updated.
  

• Media gateways Media gateways provide physical interconnection between the telephone network, individual endpoints, and the IP network. Media gateways communicate with the call agent to notify it of an event. An example is a telephone going off hook. The gateway also expects direction from the call agent on what action to take as a result of the event. For example, the call agent tells the gateway to provide a dial tone to the port that sees the off-hook condition. After the call-control exchange is completed, the gateways route and transmit the audio or media portion of the calls. This is the actual voice information.

• IP WAN The IP WAN carries both call control signaling and voice payload between the central site and the remote sites. QoS configuration is highly recommended when voice packets are transported across a WAN to ensure that the voice packets get priority over data packets in the same network. To minimize bandwidth use for voice streams that are crossing the WAN, the G.729 coder-decoder (CODEC) is used to compress the voice payload. G.729 compresses voice to 8 kbps per call as opposed to the 64 kbps traditionally used in LAN and PSTN environments.

Centralized call-processing deployments typically offer the following characteristics:

• MGCP or Megaco/H.248 protocol for call control

• Cisco CallManager at central site for managing call control

• Centralized applications pointed to by remote sites

• Up to 30,000 IP phones per cluster

• Call Admission Control (CAC) to limit number of calls per site

• Survivable Remote Site Telephony (SRST) for remote branches

A typical use for centralized architecture is a main site with many smaller remote sites. The remote sites are connected via a QoS-enabled WAN but do not require full features and functionality during a WAN outage. MGCP and Megaco/H.248 are the prevalent signaling protocols used in centralized architectures to control gateways and endpoints.

Applications such as voice mail and IVR systems are typically centralized to reduce the overall cost of administration and maintenance.

Cisco CallManager clusters can support up to 30,000 IP phones per cluster, providing for a scalable solution in enterprise environments. For even more scalability, clusters can be interconnected via intercluster trunks.

CAC is administered by the Cisco CallManager cluster. CAC is critical in enterprise implementations that include WAN connections because these connections typically have limited bandwidth that is shared between voice and data users. Control must be established over the number of calls that can flow concurrently across the WAN at any given time so that as the call volume grows, overall call quality does not diminish.

One disadvantage of implementing a centralized architecture is that if the WAN connection fails between the remote site and the central site that houses Cisco CallManager, no further voice calls can be processed by the remote site. Additional steps need to be taken to ensure that data and voice services at the remote sites remain available. One option is to implement redundant WAN links between the remote sites and the central site. In many cases, this solution is not financially feasible. Alternatively, Survivable Remote Site Telephony (SRST) provides high availability for voice services. SRST provides a subset of the call-processing capabilities within the remote-office gateway. It also enhances the IP phones with the ability to "re-home" to the call-processing functions in the local gateway if a WAN failure is detected. This feature allows the remote site to continue to provide voice connectivity in the absence of the WAN link.

Most centralized VoIP architectures use MGCP or Megaco/H.248 protocols. You can also build session initiation protocol (SIP) or H.323 networks in a centralized fashion. This is done using back-to-back user agents (B2BUAs) or gatekeeper-routed call signaling (GKRCS), respectively.

Figure 5-2 shows a typical centralized call-processing deployment, with a Cisco CallManager cluster acting as the call agent at the central site and an IP WAN with QoS enabled to connect all the sites. The remote sites rely on the centralized Cisco CallManager cluster to handle their call processing but have local voice-enabled routers to perform the voice-gateway translations for media streams. Each remote site connects locally to the PSTN. Long-distance (LD) service might be provided from the head office or through each local PSTN connection.


H.323 Distributed Network Architectures

The multisite WAN architecture with distributed call processing consists of multiple independent sites. Each site has its own call-processing agent, which is connected to an IP WAN that carries voice traffic between the distributed sites.

Each site in the distributed call-processing architecture using H.323 can be comprised of one of the following:

• A single site with its own call-processing agent
• A centralized call-processing site and all its associated remote sites
• A legacy PBX with a VoIP gateway

Figure 5-3 provides an example of a distributed call-processing architecture. Notice, in the figure, that each site contains its own call-processing agents (in the form of CallManager clusters). This type of call-processing approach has the following characteristics:

• No call control signaling for intrasite and off-net calls through the IP WAN
• Transparent use of the PSTN if the IP WAN is unavailable
• Logical hub-and-spoke topology for the directory gatekeeper
• Only one type of CODEC configured for the IP WAN

Multisite distributed call processing allows each site to be completely self-contained. The IP WAN in this model does not carry call-control signaling for intranet and off-net calls because each site has its own Cisco CallManager cluster. Typically, the PSTN serves as a backup connection between the sites in case the IP WAN connection fails or does not have any more bandwidth available.

Distributed architectures are associated with H.323 and SIP protocols. These protocols allow network intelligence to be distributed between endpoints and call control devices. Intelligence in this instance refers to any aspect of call handling including the following:

• Call state
• Calling features
• Call routing
• Provisioning
• Billing

The endpoints can be VoIP gateways, IP phones, media servers, or any device that can initiate and terminate an H.323 VoIP call. The call control devices are called gatekeepers (GKs) in an H.323 network. In an enterprise environment where many gatekeepers are required, a second level of hierarchy is achieved through the use of directory gatekeepers (DGKs). Directory gatekeepers provide summarization capabilities for multiple configured gatekeepers.

The multisite WAN architecture with distributed call processing consists of the following components:

• Media gateways Media gateways provide physical interconnection between the telephone network, individual endpoints, and the IP network. The media gateway translates call signaling between the PSTN or local endpoints and the IP network. The media gateway must contain the call-processing intelligence to perform all call-handling functions related to H.323. Media gateways communicate with gatekeepers for call address resolution and CAC.
  

• Gatekeeper A gatekeeper is an H.323 device that provides CAC and E.164 number resolution. Gatekeepers are among the key elements in the multisite WAN model that have distributed call processing. Gatekeepers provide dial plan resolution, which improves scalability in an H.323 network. Without gatekeepers, each gateway would need to be configured to know where all other reachable telephone numbers were located. The gatekeeper provides a central repository of telephone numbers and the gateways associated with those numbers. When the network is configured for gatekeepers, the learning process is dynamic, because all participating gateways register with the gatekeeper and notify it of available telephone numbers. The gatekeeper also provides CAC to ensure that voice quality is not diminished when a large number of calls enter the network.

• IP WAN The IP WAN carries call control signaling and voice payload for intersite voice communication only. Call signaling and voice transmission for all intrasite calls and off-net calls that are going to the local PSTN remain local to the site. QoS configuration is highly recommended when voice packets are transported across a WAN, to ensure the voice packets get priority over data packets in the same network. As in the centralized system, to minimize bandwidth use for voice streams that are crossing the WAN, the G.729 CODEC is typically used to compress the voice payload. G.729 compresses voice to 8 kbps per call as opposed to the 64 kbps that is traditionally used in LAN and PSTN environments.

VoIP Fundamentals

Voice over IP (VoIP) enables a voice-enabled router to carry voice traffic, such as telephone calls and faxes, over an IP network. This chapter introduces the fundamentals of VoIP, architecture types, and available voice-signaling protocols. Numbering plans, dial plans, and VoIP access to 911 emergency services are explained. The role of gateways and their use in integrating VoIP with traditional voice technologies is described. Traffic engineering and bandwidth calculations are discussed. Finally, the impact of security threats and the components required for a secure voice network are also explained.


Understanding VoIP Requirements

The increased efficiency of IP networks and the ability to statistically multiplex voice traffic with data packets allows companies to maximize their return on investment (ROI) in data network infrastructures. Decreased cost and an increase in the availability of differentiated services are two major reasons companies are evaluating the implementation of VoIP.

As demand for voice services in the IP network expands, it is important to understand the components and functionality that must be present for a successful implementation. Several protocols and tools are available for carrying voice in a data network. In defining the VoIP protocol stack, you must understand at which layer these tools and protocols reside and how they interact with other layers. When voice is packaged into IP packets, additional headers are created to carry voice-specific information. These headers can create significant additional overhead in the IP network.

Understanding which protocols to use and knowing how to limit overhead is crucial in carrying voice efficiently across an IP network.


VoIP Functional Components

In the traditional PSTN telephony network, all the elements that are required to complete the call are transparent to the end user. Migration to VoIP necessitates an awareness of these required elements and a thorough understanding of the protocols and components that provide the same functionality in an IP network.

Required VoIP functionality includes the following features:

• Signaling
• Database services
• Bearer control
• CODECs

The following sections describe each required functional component.


Signaling

Signaling is the ability to generate and exchange control information to establish, monitor, and release connections between two endpoints. Voice signaling requires the ability to provide supervisory, address, and alerting functionality between nodes. PSTN uses Signaling System 7 (SS7) to transport control messages in an out-of-band signaling network. VoIP presents several options for signaling, including H.323, Session Initiation Protocol (SIP), Megaco/H.248, and Media Gateway Control Protocol (MGCP). Some VoIP gateways are also capable of initiating SS7 signaling directly to the PSTN network.

Signaling protocols are classified either as peer-to-peer or client/server architectures. SIP and H.323 are examples of peer-to-peer signaling protocols where the end devices or gateways contain the intelligence to initiate and terminate calls and interpret call control messages. Megaco/H.248 and MGCP are examples of client/server protocols where the endpoints or gateways do not contain call control intelligence but send or receive event notifications to the server commonly referred to as the call agent. For example, when an MGCP gateway detects that a telephone has gone off hook, the gateway does not know to automatically provide a dial tone. The gateway sends an event notification to the call agent, telling the agent that an off-hook condition has been detected. The call agent then notifies the gateway to provide a dial tone.


Database Services

Access to services such as 1-800 numbers or caller ID requires the ability to query a database to determine whether the call can be placed or the information can be made available. Database services include access to billing information, calling name (CNAM) delivery, toll-free database services (1-8xx), and calling card services. VoIP service providers can differentiate their services by providing access to numerous and unique database services. For example, to simplify fax access to mobile users, a provider might build a service that converts fax to e-mail. Another example might be to provide a call notification service that places outbound calls with prerecorded messages at specific times to notify users of such events as school closures, wake-up calls, or appointment reminders.


Bearer Channel Control

Bearer channels are the channels that carry voice calls. Proper supervision of these channels requires that the appropriate call connect and call disconnect signaling be passed between end devices. Correct signaling ensures that the channel is allocated to the current voice call and that the channel is properly de-allocated when either side terminates the call. These connect and disconnect messages are carried in SS7 within the PSTN network, and in SIP, H.323, Megaco/H.248, or MGCP within an IP network.


CODECs

Coder-decoders (CODECs) provide the coding and decoding translation between analog and digital facilities. Each CODEC type defines the method of voice coding and the compression mechanism that is used to convert the voice stream. The PSTN uses TDM to carry each voice call. Each voice channel reserves 64 kbps of bandwidth and uses the G.711 CODEC to convert the analog voice wave to a TDM voice stream. G.711 creates a 64 kbps digitized voice stream. In VoIP design, CODECs often compress voice beyond the 64 kbps voice stream to allow more efficient use of network resources. The most widely used CODEC in the WAN environment is G.729, which compresses the voice stream (that is, the voice payload only) to 8 kbps.


VoIP Protocols

VoIP employs a variety of protocols to set up a call, tear down a call, and send information (for example, the actual spoken voice) during a call. The following are the major VoIP protocols:

• H.323 An ITU standard protocol for interactive conferencing. H.323 was originally designed for multimedia in a connectionless environment, such as a LAN. H.323 serves as an umbrella of standards that define all aspects of synchronized voice, video, and data transmission. H.323 defines end-to-end call signaling.

• Media Gateway Control Protocol (MGCP) A method for PSTN gateway control or thin device control. Specified in RFC 2705, MGCP defines a protocol to control VoIP gateways that are connected to external call-control devices, referred to as call agents. MGCP provides the signaling capability for less-expensive edge devices, such as gateways, that might not have implemented a full voice-signaling protocol such as H.323. For example, any time an event such as an off-hook condition occurs at the voice port of a gateway, the voice port reports that event to the call agent. The call agent then signals that device to provide a service, such as dial-tone signaling.

• Megaco/H.248 A joint Internet Engineering Task Force (IETF) and ITU standard that is based on the original MGCP standard. Megaco defines a single gateway control approach that works with multiple gateway applications including PSTN gateways, ATM interfaces, analog-like and telephone interfaces, interactive voice response (IVR) servers, and others. Megaco provides full call control intelligence and implements call level features such as transfer, conference, call forward, and hold. The basic operation of Megaco is very similar in nature to MGCP. However, Megaco provides more flexibility by interfacing with a wider variety of applications and gateways.

• Session Initiation Protocol (SIP) A detailed protocol that specifies the commands and responses to set up and tear down calls. SIP also details features such as security, proxy, and transport (TCP or User Datagram Protocol [UDP]) services. SIP and its partner protocols, Session Announcement Protocol (SAP) and Session Description Protocol (SDP), can provide announcements and information about multicast sessions to users on a network. SIP defines end-to-end call signaling between devices. SIP is a text-based protocol that borrows many elements of HTTP, using the same transaction request and response model, and similar header and response codes. It also adopts a modified form of the URL-addressing scheme used within e-mail that is based on Simple Mail Transfer Protocol (SMTP).

• Real-Time Transport Protocol (RTP) An IETF standard media-streaming protocol. RTP carries the voice payload across the network. RTP provides sequence numbers and time stamps for the orderly processing of voice packets. In addition to voice packets, RTP can also carry streaming video packets.

• RTP Control Protocol (RTCP) Provides out-of-band control information for an RTP flow. Every RTP flow has a corresponding RTCP flow that reports statistics on the call. RTCP is used for quality of service (QoS) reporting.

Successfully integrating connection-oriented voice traffic in a connectionless IP network requires enhancements to the signaling stack. In some ways, the user voice protocol must make the connectionless network appear more connection oriented through the use of sequence numbers. Table 5-1 provides examples of how various VoIP components and protocols map to the seven-layer OSI model.



VoIP Service Considerations

In traditional telephony networks, dedicated bandwidth for each voice stream provides voice with a guaranteed delay across the network. Because bandwidth is guaranteed in the TDM environment, there is no variable delay (jitter). Configuring voice in a data network requires network services with low delay, minimal jitter, and minimal packet loss. Bandwidth requirements must be properly calculated based on the CODEC that is used and the number of concurrent connections. QoS must be configured to minimize jitter and loss of voice packets. The PSTN offers uptime of 99.999 percent, also known as the five nines of availability. A system that is up 99.999 percent of the time experiences only five minutes of down time in an entire year. To match the availability of the PSTN, the IP network must be designed with redundancy and failover mechanisms. Additionally, security policies must be established to address both network stability and voice-stream security.

Table 5-2 lists the issues associated with implementing VoIP in a converged network and solutions that address these issues.

Voice Dial Peer Configuration

Configuring dial peers is the key to setting up dial plans and implementing voice in a VoIP network. In some situations, a router might also need to manipulate digits in a dial string before passing the dial string to a telephony device. For example, a 9 might need to be added to a dial string before the dial string passes out the router to a PBX, or perhaps a dialed area code and office code needs to be removed from a dial string. This chapter introduces plain old telephone service (POTS) and Voice over IP (VoIP) dial peers, which make an endtoend VoIP call possible. Additionally, this chapter discusses various approaches to manipulating dialed digits.

Consider a call center environment. Calls coming into a call center need to be distributed among available customer service agents. A hunt group takes calls coming into a single number and logically distributes those calls across hunt group members. This chapter also describes how to configure hunt groups and similarly how to reroute a call across the PSTN during times when an IP WAN connection is unavailable.

Finally, this chapter addresses the configuration of special purpose connections such as private line, automatic ringdown (PLAR) and connections that interconnect existing PBX systems.


Configuring Dial Peers

As a call is set up across the network, the existence of various parameters is checked and negotiated. A mismatch in parameters can cause call failure. Therefore, it is important to understand how routers interpret call legs and how call legs relate to inbound and outbound dial peers. Successful implementation of a VoIP network relies heavily on the proper application of dial peers, the digits they match, and the services they specify. A network designer needs in-depth knowledge of dial peer configuration options and their uses. This section discusses the proper use of digit manipulation and the configuration of dial peers.


Understanding Call Legs

Call legs are logical connections between any two telephony devices, such as gateways, routers, Cisco Unified CallManagers, or telephony endpoint devices. Additionally, call legs are routercentric. When an inbound call arrives, it is processed separately until the destination is determined. Then a second outbound call leg is established, and the inbound call leg is switched to the outbound voice port. The topology shown in Figure 4-1 illustrates the four call legs involved in an endtoend call between two voiceenabled routers.


An endtoend call consists of four call legs: two from the source router's perspective and two from the destination router's perspective. To complete an endtoend call from either side and send voice packets back and forth, you must configure all four dial peers. Dial peers are only used to set up calls. After the call is established, dial peers are no longer employed.

An inbound call leg occurs when an incoming call comes into the router or gateway. An outbound call leg occurs when a call is placed from the router or gateway, as depicted in Figure 4-2.


A call is segmented into call legs, and a dial peer is associated with each call leg. The process for call setup, as diagramed in Figure 4-2, is:


1. The POTS call arrives at R1, and an inbound POTS dial peer is matched.

2. After associating the incoming call to an inbound POTS dial peer, R1 creates an inbound POTS call leg and assigns it a call ID (call leg 1).

3. R1 uses the dialed string to match an outbound VoIP dial peer.

4. After associating the dialed string to an outbound voice network dial peer, R1 creates an outbound voice network call leg and assigns it a call ID (call leg 2).

5. The voice network call request arrives at R2, and an inbound VoIP dial peer is matched.

6. After R2 associates the incoming call to an inbound VoIP dial peer, R2 creates the inbound voice network call leg and assigns it a call ID (call leg 3). At this point, both R1 and R2 negotiate voice network capabilities and applications, if required. The originating router or gateway might request nondefault capabilities or applications. When this is the case, the terminating router or gateway must match an inbound VoIP dial peer that is configured for such capabilities or applications.

7. R2 uses the dialed string to match an outbound POTS dial peer.

8. After associating the incoming call setup with an outbound POTS dial peer, R2 creates an outbound POTS call leg, assigns it a call ID, and completes the call (call leg 4).


Understanding Dial Peers

When a call is placed, an edge device generates dialed digits as a way of signaling where the call should terminate. When these digits enter a router voice port, the router must decide whether the call can be routed and where the call can be sent. The router does this by searching a list of dial peers.

A dial peer is an addressable call endpoint. The address is called a destination pattern and is configured in every dial peer. Destination patterns use both explicit digits and wildcard variables to define one telephone number or range of numbers.

Dial peers define the parameters for the calls that they match. For example, if a call is originating and terminating at the same site and is not crossing through slow-speed WAN links, then the call can cross the local network uncompressed and without special priority. A call that originates locally and crosses the WAN link to a remote site may require compression with a specific coder-decoder (CODEC). In addition, this call may require that voice activity detection (VAD) be turned on and will need to receive preferential treatment by specifying a higher priority level.

• POTS dial peers Connect to a traditional telephony network, such as the public switched telephone network (PSTN) or a PBX, or to a telephony edge device such as a telephone or fax machine.

• VoIP dial peers Connect over an IP network.

In Figure 4-3, the telephony device connects to the Cisco voice-enabled router. The POTS dial peer configuration includes the telephone number of the telephony device and the voice port to which it is attached. The router determines where to forward incoming calls for that telephone number.

The Cisco voiceenabled router VoIP dial peer is connected to the packet network. The VoIP dial peer configuration includes the destination telephone number (or range of numbers) and the next-hop or destination voiceenabled router network address.

Follow these steps to enable a router to complete a VoIP call:

1. Configure a compatible dial peer on the source router that specifies the recipient destination address.

2. Configure a POTS dial peer on the recipient router that specifies which voice port the router uses to forward the voice call.


Configuring POTS Dial Peers

Before the configuration of Cisco IOS dial peers can begin, you must have a good understanding of where the edge devices reside, what type of connections need to be made between these devices, and what telephone numbering scheme is applied to the devices.

Follow these steps to configure POTS dial peers:

1. Configure a POTS dial peer at each router or gateway, where edge telephony devices connect to the network.

2. Use the destination-pattern command in dial peer configuration mode to configure the telephone number.

3. Use the port command in dial peer configuration mode to specify the physical voice port that the POTS telephone is connected to.

Voice Interface Configuration

Voice gateways bridge the gap between the VoIP world and the traditional telephony world (for example, a PBX, the 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.


Configuring Voice Ports

Connecting voice devices to a network infrastructure requires an in-depth understanding of signaling and electrical characteristics that are specific to each type of interface. Improperly matched electrical components can cause echo and make a connection unusable. As another consideration, configuring devices for international implementation requires knowledge of country-specific settings. This section provides voice port configuration parameters for signaling and country-specific settings.

Before delving into the specific syntax of configuring these voice ports, this section begins by considering several examples of voice applications. The applications discussed help illustrate the function of the voice ports, whose configuration is addressed at the end of this section.


Voice Applications

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

• CallManager-to-CallManager calls

• On-net to off-net calls

Local Calls
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.


On-Net Calls
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 direct connections and 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 CallManager. 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.


Off-Net Calls
Figure 3-3 shows an example of an off-net call. To gain access to the public switched telephone network (PSTN), the user dials an access code, such as 9, from a telephone that is 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.


PLAR Calls
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 types of signaling, including receive and transmit (ear and mouth [E&M]), FXO, FXS, or any combination of analog and digital interfaces. As an 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.


PBX-to-PBX Calls
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 the IP network. Modern PBX connections to the network are typically digital T1 or E1 with channel associated signaling (CAS) or PRI signaling, although PBX connections can also be analog.


CallManager-to-CallManager Calls
As part of an overall migration strategy, a business might replace PBXs with a Cisco Unified CallManager infrastructure. This infrastructure includes IP telephones that

plug directly into the IP network. Cisco Unified CallManager performs the same call-routing functions formerly provided by the PBX. When an IP phone uses Cisco Unified CallManager to place a call, Cisco CallManager, based on its configuration, assesses whether the call is destined for another IP phone under its control or whether the call must be routed through a remote Cisco CallManager for call completion. Although the call stays on the IP network, it might be sent between zones. Every Cisco CallManager is part of a zone. A zone is a collection of devices that are under a common administration, usually a Cisco Unified CallManager or gatekeeper. Figure 3-6 provides an example of a CallManager-to-CallManager call.


On-Net to Off-Net Calls
When planning a resilient call-routing strategy, it might be necessary 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.

Common Channel Signaling Systems

Common channel signaling (CCS) differs from CAS in that all channels use a common channel and protocol for call setup. Using E1 as an example, a signaling protocol, such as ISDN Q.931, would be deployed in time slot 17 to exchange call-setup messages with its attached telephony equipment, as seen in Figure 2-36.


Examples of CCS signaling are as follows:

• Proprietary implementations Some PBX vendors choose to use CCS for T1 and E1 and implement a proprietary CCS protocol between their PBXs. In this implementation, Cisco devices are configured for Transparent Common Channel Signaling (T-CCS) because the Cisco devices do not understand proprietary signaling information.

• Integrated Services Digital Network (ISDN) ISDN uses Q.931 in a common channel to signal all other channels.

• Q Signaling (QSIG) Like ISDN, QSIG uses a common channel to signal all other channels.

• Digital Private Network Signaling System (DPNSS) DPNSS is an open standard developed by British Telecom for implementation by any vendor who chooses to use it. DPNSS also uses a common channel to signal all other channels.

• Signaling System 7 (SS7) SS7 is an out-of-band network implemented and maintained by various telephone companies and used for signaling and other supplemental services.

The following discussions elaborate on various CCS implementations. Note that proprietary implementations are not discussed because they vary widely among vendors.


ISDN

ISDN (Integrated Services Digital Network) is an access specification to a network. You may have studied ISDN as an access method for dial-up data systems. Because it is a digital system, ISDN makes connections rapidly.

ISDN can be implemented in two different ways: BRI (Basic Rate Interface) and PRI (Primary Rate Interface). BRI features two bearer (B) channels, while PRI supports 23 (for T1) or 30 (for E1) B channels. Each implementation also supports a data (D) channel, used to carry signaling information (CCS).

The following are benefits of using ISDN to transmit voice:

• Each B channel is 64 kbps, making it perfect for G.711 PCM.

• ISDN has a built-in call control protocol known as ITU-T Q.931.

• ISDN can convey standards-based voice features, such as call forwarding.

• ISDN supports standards-based enhanced dial-up capabilities, such as Group 4 fax and audio channels.

Figure 2-37 shows the architecture of an ISDN network. The B channel carries information, such as voice, data, and video, at 64 kbps. The D channel carries call signaling between customer premises equipment (CPE) and the network, usually as the Q.931 protocol but sometimes as the QSIG protocol.


BRI operates using the average local copper pair. It uses two B channels and one signaling channel, which is written as 2 B + D.

PRI implemented on T1 uses 23 B channels and one signaling channel, which is written as 23 B + D. PRI implemented on E1 uses 30 B channels and one signaling channel, which is represented as 30 B + D.

ISDN's Q.931 protocol, which operates at Layer 3 of the OSI (Open System Interconnection) model, uses a standard set of messages to communicate, as illustrated in Figure 2-38.


These standard messages cover the following areas:

• Call establishment Initially sets up a call. Messages travel between the user and the network. Call establishment events include alerting, call proceeding, connect, connect acknowledgment, progress, setup, and setup acknowledgment.

• Call information phase Data sent between the user and the network after the call is established. This allows the user to, for example, suspend and then resume a call. Events in the call information phase include hold, hold acknowledgment, hold reject, resume, resume acknowledgment, resume reject, retrieve, retrieve acknowledgment, retrieve reject, suspend, suspend acknowledgment, suspend reject, and user information.

• Call clearing Terminates a call. The following events occur in the call-clearing phase: disconnect, release, release complete, restart, and restart acknowledgment.

• Miscellaneous messages Negotiates network features (supplementary services). Miscellaneous services include congestion control, facility, information, notify, register, status, and status inquiry.

QSIG

The QSIG (Q Signaling) protocol is based on the ISDN Q.931 standard and provides signaling for private integrated services network exchange (PINX) devices. Figure 2-39 shows how different QSIG operations map to the OSI model.


DPNSS

British Telecom and selected PBX manufacturers originally developed the Digital Private Network Signaling System (DPNSS) in the early 1980s. It was developed and put into use before the ISDN standards were completed because customers wanted to make use of digital facilities as soon as possible.

DPNSS operates over standard ISDN physical interfaces and is described in four documents:

• BTNR 188: Digital Private Networking Signalling System No 1, Issue 6, January 1995.

• BTNR 188-T: Digital Private Networking Signalling System No 1: Testing Schedule.

• BTNR 189: Interworking between DPNSS1 and other Signalling Systems, Issue 3, March 1988.

• BTNR 189-I: Interworking between DPNSS1 and ISDN Signalling Systems, Issue 1, December 1992.

SIGTRAN

SIGTRAN, as illustrated in Figure 2-40, is a signaling protocol defined in RFC 2719 and RFC 2960. It describes the way the IP protocol carries SS7 messages in a VoIP network. SIGTRAN relies on the Stream Control Transport Protocol at Layer 4 of the TCP/IP protocol stack.


Using SIGTRAN, a service provider may interconnect a private VoIP network to the public switched telephone network (PSTN) and ensure that SS7 signals are conveyed end to end.

Signaling Systems

Configuring Cisco Systems voice equipment to interface with other equipment requires an understanding of the signaling that conveys supervision between the systems. Proper troubleshooting also requires an understanding of these signaling systems.

This section describes the various signaling systems used between telephony systems, such as common channel signaling and channel associated signaling. It also explores signaling between PBXs, signaling between PBXs and COs, and specialized signaling, such as ISDN.


Channel Associated Signaling

Channel associated signaling (CAS) is a signaling method commonly used between PBXs. Although this can manifest itself in many forms, some methods are more common than others. Signaling systems can also be implemented between a PBX and a Cisco voice device.


T1 Channel Associated Signaling

PBXs and Cisco devices use T1 and E1 interfaces to convey voice. Originally, this was the main purpose of T1, which carries signaling information using two methodologies: CAS and common channel signaling (CCS). Figure 2-31 illustrates the format of the T1 digital signal.


The characteristics of the T1 digital signal format are as follows:

• A T1 frame is 193 bits long, 8 bits from each of the 24 time slots (digital service zeros [DS0s]), plus 1 bit for framing. A T1 repeats every 125 microseconds, resulting in 8000 samples per second (8 bits * 24 time slots + 1 framing bit * 8000 samples per second = 1.544 Mbps).
  

• T1 has two major framing and format standards:

- Super Frame (SF), or D4, specifies 12 frames in sequence. The D4 framing pattern used in the F position in Figure 2-31 is 100011011100 (a 1 goes with the first frame, a 0 goes with the second frame, a 0 goes with the third frame, and so on, all the way through 12 frames). This unique framing pattern allows the receiving T1 equipment to synchronize within four frames, since any four consecutive frame bits are unique within the 12-bit pattern. Because there are 8000 T1 frames transmitted per second, 8000 F bits are produced and used for framing.

- Extended Superframe (ESF) format was developed as an upgrade to SF and is now dominant in public and private networks. Both types of formatting retain the basic frame structure of one framing bit followed by 192 data bits. However, ESF repurposes the use of the F bit. In ESF, of the total 8000 F bits used in T1, 2000 are used for framing, 2000 are used for cyclic redundancy check (CRC) (for error checking only), and 4000 are used as an intelligent supervisory channel to control functions end to end (such as loopback and error reporting).

Because each DS0 channel carries 64 kbps, and G.711 is 64 kbps, there is no room to carry signaling. Implemented for voice, the T1 uses every sixth frame to convey signaling information. In every sixth frame, the least significant bit (LSB) for each of the voice channels is used to convey the signaling, as shown in Figure 2-32. Although this implementation detracts from the overall voice quality (because only 7 bits represent a sample for that frame), the impact is not significant. This method is called robbed-bit signaling (RBS). When SF employs this method, the signaling bits are conveyed in both the 6th (called the "A" bit) and 12th (called the "B" bit) frames. For control signaling, A and B bits provide both near- and far-end off-hook indication.


The A and B bits can represent different signaling states or control features (on hook or off hook, idle, busy, ringing, and addressing). The robbed bit is the least significant bit from an 8-bit word.

ESF also uses RBS in frames 6, 12, 18, and 24 to yield four signaling bits, providing additional control and signaling information. These four bits are known as the A, B, C, and D bits.

Because the signaling occurs within each DS0, it is referred to as in band. Also, because the use of these bits is exclusively reserved for signaling each respective voice channel, it is referred to as CAS.

The robbed bits, depicted in Figure 2-33, are used to convey E&M status or FXS/FXO status and provide call supervision for both on hook and off hook.


T1 CAS has the following characteristics:

• SF has a 12-frame structure and provides AB bits for signaling.

• ESF has a 24-frame structure and provides ABCD bits for signaling.

• DTMF, or tone, can be carried in band in the audio path. However, other supervisory signals must still be carried via CAS.

E1 Channel Associated Signaling

In E1 framing and signaling, 30 of the 32 available channels, or time slots, are used for voice and data. Framing information uses time slot 1, while time slot 17 (E0 16) is used for signaling by all the other time slots. This signaling format, illustrated in Figure 2-34, is also known as CAS because the use of the bits in the 17th time slot is exclusively reserved for the purpose of signaling each respective channel. However, this implementation of CAS is considered out of band, because the signaling bits are not carried within the context of each respective voice channel, as is the case with T1. E1 CAS is directly compatible with T1 CAS, because both methods use AB or ABCD bit signaling. Although the signaling for E1 CAS is carried in a single common time slot, it is still referred to as CAS because each individual signaling time slot represents a specific pair of voice channels.


In the E1 frame format, 32 time slots make up a frame. A multiframe consists of 16 E1 frames, as depicted in Figure 2-35.

Voice Compression Standards

To conserve valuable WAN bandwidth, you can compress the quantized voice waveforms. Two categories of waveform encoding include:

• Waveform algorithms (coders) Waveform algorithms have the following functions and characteristics:

- Sample analog signals at 8000 times per second

- Use predictive differential methods to reduce bandwidth

- Highly impact voice quality because of reduced bandwidth

- Do not take advantage of speech characteristics

- Examples include: G.711 and G.726

• Source algorithms (coders) Source algorithms have the following functions and characteristics:

- Source algorithm coders are called vocoders, or voice coders. A vocoder is a device that converts analog speech into digital speech, using a specific compression scheme that is optimized for coding human speech.

- Vocoders take advantage of speech characteristics.

- Bandwidth reduction occurs by sending linear-filter settings.

- Codebooks store specific predictive waveshapes of human speech. They match the speech, encode the phrases, decode the waveshapes at the receiver by looking up the coded phrase, and match it to the stored waveshape in the receiver codebook.

- Examples include: G.728 and G.729


The following three common voice compression techniques are standardized by the ITU-T:

• PCM Amplitude of voice signal is sampled and quantized at 8000 times per second. Each sample is then represented by one octet (8 bits) and transmitted. For sampling, you must use either a-law or µ-law to reduce the signal-to-noise ratio.

• ADPCM The difference between the current sample and its predicted value (based on past samples). ADPCM is represented by 2, 3, 4, or 5 bits. This method reduces the bandwidth requirement at the expense of signal quality.

• CELP Excitation value and a set of linear-predictive filters (settings) are transmitted. The filter setting transmissions are less frequent than excitation values and are sent on an as-needed basis.

Table 2-4 describes the CODECs and compression standards.

A common type of waveform encoding is pulse code modulation (PCM). Standard PCM is known as ITU standard G.711, which requires 64,000 bits per second of bandwidth to transport the voice payload (that is, not including any overhead), as shown in Figure 2-30.


Figure 2-30 shows that PCM requires 1 polarity bit, 3 segment bits, and 4 step bits, which equals 8 bits per sample. The Nyquist Theorem requires 8000 samples per second; therefore, you can figure the required bandwidth as follows:

8 bits * 8000 samples per second = 64,000 bits per second

Adaptive differential pulse code modulation (ADPCM) coders, like other waveform coders, encode analog voice signals into digital signals to adaptively predict future encodings by looking at the immediate past. The adaptive feature of ADPCM reduces the number of bits per second that the PCM method requires to encode voice signals. ADPCM does this by taking 8000 samples per second of the analog voice and turning them into linear PCM samples. ADPCM then calculates the predicted value of the next sample, based on the immediate past sample, and encodes the difference. The ADPCM process generates 4-bit words, thereby generating 16 specific bit patterns.

The ADPCM algorithm from the Consultative Committee for International Telegraph and Telephone (CCITT) transmits all 16 possible bit patterns. The ADPCM algorithm from the American National Standards Institute (ANSI) uses 15 of the 16 possible bit patterns. The ANSI ADPCM algorithm does not generate a 0000 pattern.

The ITU standards for compression are as follows:

• G.711 rate: 64 kbps = (2 * 4 kHz) * 8 bits/sample

• G.726 rate: 32 kbps = (2 * 4 kHz) * 4 bits/sample

• G.726 rate: 24 kbps = (2 * 4 kHz) * 3 bits/sample

• G.726 rate: 16 kbps = (2 * 4 kHz) * 2 bits/sample

Code excited linear prediction (CELP) compression transforms analog voice as follows:

1. The input to the coder is converted from an 8-bit to a 16-bit linear PCM sample.

2. A codebook uses feedback to continuously learn and predict the voice waveform.

3. The coder is excited (that is, begins its lookup process) by a white noise generator.

4. The mathematical result is sent to the far-end decoder for synthesis and generation of the voice waveform.

Two forms of CELP include Low-Delay CELP (LDCELP) and Conjugate Structure Algebraic CELP (CS-ACELP). LDCELP is similar to CS-ACELP, except for the following:

• LDCELP uses a smaller codebook and operates at 16 kbps to minimize delay, or look-ahead, from 2 to 5 ms, while CS-ACELP minimizes bandwidth requirements (8 kbps) at the expense of increasing delay (10 ms).

• The 10-bit code word is produced from every five speech samples from the 8 kHz input with no look-ahead.

• Four of these 10-bit code words are called a subframe. They take approximately 2.5 ms to encode. CS-ACELP uses eight 10-bit code words.

Two of these subframes are combined into a 5-ms block for transmission. CS-ACELP is a variation of CELP that performs these functions:

• Codes 80-byte frames, which take approximately 10 ms to buffer and process.

• Adds a look-ahead of 5 ms. A look-ahead is a coding mechanism that continuously analyzes, learns, and predicts the next waveshape.

• Adds noise reduction and pitch-synthesis filtering to processing requirements.

Cisco VoIP environments typically leverage the benefits of G.729 when transmitting voice traffic over the IP WAN. These benefits include the ability to minimize bandwidth demands, while maintaining an acceptable level of voice quality. Several variants of G.729 exist.

Analog-to-Digital and Digital-to-Analog Voice Encoding

This section covers the fundamentals of digitally encoding voice, specifically, the basics of voice digitization and the various compression schemes that are used to transport voice while using less bandwidth.

Digitizing speech was a project first undertaken by the Bell System in the 1950s. The original purpose of digitizing speech was to deploy more voice circuits with a smaller number of wires. This evolved into the T1 and E1 transmission methods of today. Examples of analog and digital waveforms are presented in Figure 2-24.


Table 2-3 details the steps to convert an analog signal to a digital signal.


The three mandatory components in the analog-to-digital conversion process are further described as follows:

• Sampling Sample the analog signal at periodic intervals. The output of sampling is a pulse amplitude modulation (PAM) signal.

• Quantization Match the PAM signal to a segmented scale. This scale measures the amplitude (height) of the PAM signal and assigns an integer number to define that amplitude.

• Encoding Convert the integer base-10 number to a binary number. The output of encoding is a binary expression in which each bit is either a 1 (pulse) or a 0 (no pulse).

This three-step process is repeated 8000 times per second for telephone voice-channel service. Use the fourth optional step, compression, to save bandwidth. This optional step allows a single channel to carry more voice calls.

After the receiving terminal at the far end receives the digital PCM signal, it must convert the PCM signal back into an analog signal. The process of converting digital signals back into analog signals includes the following two processes:

• Decoding The received 8-bit word is decoded to recover the number that defines the amplitude of that sample. This information is used to rebuild a PAM signal of the original amplitude. This process is simply the reverse of the analog-to-digital conversion.

• Filtering The PAM signal is passed through a filter to reconstruct the original analog wave form from its digitally coded counterpart.

With this basic understanding of analog to digital conversion, this chapter considers the sampling, quantization, and encoding processes more thoroughly, beginning with sampling.


Sampling and the Nyquist Theorem

One of the major issues with sampling is determining how often to take those samples (that is, "snapshots") of the analog wave. You do not want to take too few samples per second because when the equipment at the other end of the phone call attempts to reassemble and make sense of those samples, a different sound (that is, a lower frequency sound) signal might also match those samples, and the incorrect sound would be heard by the listener. This phenomenon is called aliasing, as shown in Figure 2-25.


With the obvious detrimental effect of undersampling, you might be tempted to take many more samples per second. While that approach, sometimes called oversampling, does indeed eliminate the issue of aliasing, it also suffers from a major drawback. If you take far more samples per second than actually needed to accurately recreate the original signal, you consume more bandwidth than is absolutely necessary. Because bandwidth is a scarce commodity (especially on a wide-area network), you do not want to perform the oversampling shown in Figure 2-26.


Digital signal technology is based on the premise stated in the Nyquist Theorem: When a signal is instantaneously sampled at the transmitter in regular intervals and has a rate of at least twice the highest channel frequency, then the samples will contain sufficient information to allow an accurate reconstruction of the signal at the receiver. Figure 2-27 illustrates sampling, as prescribed by the Nyquist Theorem.


While the human ear can sense sounds from 20 to 20,000 Hz, and speech encompasses sounds from about 200 to 9000 Hz, the telephone channel was designed to operate at about 300 to 3400 Hz. This economical range carries enough fidelity to allow callers to identify the party at the far end and sense their mood. Nyquist decided to extend the digitization to 4000 Hz, to capture higher-frequency sounds that the telephone channel may deliver. Therefore, the highest frequency for voice is 4000 Hz, or 8000 samples per second; that is, one sample every 125 microseconds.


Quantization

Quantization involves dividing the range of amplitude values that are present in an analog signal sample into a set of discrete steps that are closest in value to the original analog signal, as illustrated in Figure 2-28. Each step is assigned a unique digital code word.


In Figure 2-28, the x-axis is time and the y-axis is the voltage value (PAM). The voltage range is divided into 16 segments (0 to 7 positive, and 0 to 7 negative). Starting with segment 0, each segment has fewer steps than the previous segment, which reduces the signal-to-noise ratio (SNR) and makes the segment uniform. This segmentation also corresponds closely to the logarithmic behavior of the human ear. If there is an SNR problem, it is resolved by using a logarithmic scale to convert PAM to PCM.

Linear sampling of analog signals causes small-amplitude signals to have a lower SNR, and therefore poorer quality, than larger amplitude signals. The Bell System developed the µ-law method of quantization, which is widely used in North America. The International Telecommunication Union (ITU) modified the original m-law method and created a-law, which is used in countries outside of North America.

By allowing smaller step functions at lower amplitudes, rather than higher amplitudes, µ-law and a-law provide a method of reducing this problem. Both µ-law and a-law "compand" the signal; that is, they both compress the signal for transmission and then expand the signal back to its original form at the other end.

Informational Signaling

DTMF tones are used not just for address signaling but also for informational signaling. Specifically, call-progress indicators in the form of tone combinations are used to notify subscribers of call status. Each combination of tones represents a different event in the call process, as follows:

• Dial tone Indicates that the telephone company is ready to receive digits from the user telephone. Cisco routers provide dial tone as a method of showing that the hardware is installed. In a PBX or key telephone system, the dial tone indicates the system is ready to receive digits.

• Busy Indicates that a call cannot be completed because the telephone at the remote end is already in use.

• Ringback (CO or PBX) Indicates that the telephone switch is attempting to complete a call on behalf of a subscriber.

• Congestion Indicates that congestion in the long-distance telephone network is preventing a telephone call from being processed. The congestion tone is sometimes known as the all-circuits-busy tone.

• Reorder Indicates that all of the local telephone circuits are busy, thus preventing a telephone call from being processed. The reorder tone is known to the user as fast-busy and is familiar to anyone who operates a telephone from a PBX.

• Receiver off hook Indicates that the receiver has been off hook for an extended period without placing a call.

• No such number Indicates that a subscriber placed a call to a nonexistent number.

Trunk Connections

Before a telephone call terminates at its final destination, the call is routed through multiple switches. When a switch receives a call, it determines whether the destination telephone number is within a local switch or if the call needs to go through another switch to a remote destination. Trunks interconnect the telephone company and PBX switches, as shown in Figure 2-9.

The primary function of the trunk is to provide the path between switches. The switch must route the call to the correct trunk or telephone line. Although many different subscribers share a trunk, only one subscriber uses it at any given time. As telephone calls end, they release trunks and make them available to the switch for subsequent calls. There can be several trunks between two switches.

The following are examples of the more common trunk types:

• Private trunk lines (tie-lines) Companies with multiple PBXs often connect them with tie trunk lines. Generally, tie trunk lines serve as dedicated circuits that connect PBXs. On a monthly basis, subscribers lease trunks from the telephone company to avoid the expense of using telephone lines on a per-extension basis. These types of connections, known as tie-lines, typically use special interfaces called recEive and transMit, or E&M interfaces.

• CO trunks A CO trunk serves as a direct connection between a PBX and the local CO that routes calls; for example, the connection from a private office network to the public switched telephone network (PSTN). When users dial 9, they are connecting through their PBX to the CO trunk to access the PSTN. CO trunks typically use Foreign Exchange Office interfaces. Certain specialized CO trunks are frequently used on the telephony network. A direct inward dial trunk, for example, allows outside callers to reach specific internal destinations without having to be connected via an operator.

• Interoffice trunks An interoffice trunk is a circuit that connects two local telephone company COs.

• Foreign exchange (FX) trunks FX trunks are interfaces that are connected to switches supporting connections to either office equipment or station equipment. Office equipment includes other switches (to extend the connection) and Cisco devices. Station equipment includes telephones, fax machines, and modems.

Trunk Signaling

Lines and trunks must adhere to signaling standards just as telephony networks and telephone companies do. Trunk signaling serves to initiate the connection between the switch and the network. There are five different types of trunk signaling, and each applies to different kinds of interfaces, such as FXS, FXO, and E&M:

• Loop-start signaling

• Ground-start signaling

• E&M wink-start signaling

• E&M immediate-start signaling

• E&M delay-start signaling

The following sections explain these signaling types.


Loop-Start Signaling

Loop-start signaling allows a user or the telephone company to seize a line or trunk when a subscriber is initiating a call. It is primarily used on local loops connecting to residences rather than on trunks interconnecting telephone switches.

A telephone connection exists in one of the following states, as illustrated in Figure 2-10:

• Idle (on hook)
• Telephone seizure (off hook)
• CO seizure (ringing)

A summary of the loop-start signaling process is as follows:

1. When the line is in the idle state, or on hook, the telephone or PBX opens the two-wire loop. The CO or FXS has battery on ring and ground on tip.

2. If a user lifts the handset off the cradle to place a call, the switch hook goes off hook and closes the loop (line seizure). The current can now flow through the telephone circuit. The CO or FXS module detects the current and returns a dial tone.

3. When the CO or FXS module detects an incoming call, it applies AC ring voltage superimposed over the 48 VDC battery, causing the ring generator to notify the recipient of a telephone call. When the telephone or PBX answers the call, thus closing the loop, the CO or FXS module removes the ring voltage.

Loop-start signaling is a poor solution for high-volume trunks because it leads to glare, which is the simultaneous seizure of the trunk from both ends. Glare occurs, for example, when you pick up your home telephone and find that someone is already at the other end.

Glare is not a significant problem at home. It is, however, a major problem when it occurs between switches at high-volume switching centers, such as long-distance carriers or large PBX systems.


Ground-Start Signaling

Ground-start signaling, illustrated in Figure 2-11, is a modification of loop-start signaling that corrects for the probability of glare. It solves the problem by providing current detection at both ends.

Although loop-start signaling works when you use your telephone at home, ground-start signaling is preferable when there are high-volume trunks involved at telephone switching centers. Because ground-start signaling uses a request or confirm switch at both ends of the interface, it is preferable over other signaling methods on high-usage trunks, such as FXOs. FXOs require implementation of answer supervision (reversal or absence of current) on the interface for the confirmation of on hook or off hook.


E&M Signaling

E&M signaling supports tie-line type facilities or signals between voice switches. Instead of superimposing both voice and signaling on the same wire, E&M uses separate paths, or leads, for each.

To call a remote office, your PBX must route a request for use of the trunk over its signal leads between the two sites. Your PBX makes the request by activating its M-lead. The other PBX detects the request when it detects current flowing on its E-lead. It then attaches a dial register to the trunk and your PBX, which sends the dialed digits. The remote PBX activates its M-lead to notify the local PBX that the call has been answered.

There are five types of E&M signaling: Type I, Type II, Type III, Type IV, and Type V. The E&M leads operate differently with each wiring scheme, as shown in Table 2-1 and Table 2-2. Keep in mind that any of the E&M supervisory signaling types (that is, wink-start, immediate-start, and delay-start) can operate over any of the following wiring schemes.