CVOICE Quick Reference - 2nd Edition

7/28/2019 CVOICE Quick Reference - 2nd Edition

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CCVP CVOICEQuick ReferenceSecond Edition

Chapter 1:Understanding

Chapter 2:Addressing VoI

Chapter 3:Understanding Analog Voice Po

Chapter 4:Understanding Digital Voice Po

Chapter 5:Exploring Gatew

Chapter 6:Working with D

Chapter 7:Implementing ADial-Plan Featu

Chapter 8:

Understanding Configuring Gat

Chapter 9:Interconnectingwith a Cisco Un

Kevin Wallace



CHAPTER 1

Understanding VoIP Basics

Chapter 1Understanding VoIP BasicsModern enterprise network designs need to support the transmission of voice traffic. analog phones (much like the phones typically found in homes) or IP phones, whichvoice IP packets. Because the analog phones cannot generate IP packets, they connecrouters) that convert the spoken-voice IP packets.

The term Voice over IP, or VoIP, is used to describe the transmission of voice over aThe term IP telephony refers to the use of IP phones and a call processing server (foCommunications Manager [UCM]). However, because many voice-enabled networkscomponents, these terms are often used interchangeably.

This chapter introduces you to the basics of VoIP networks. Specifically, you will becomponents and protocols, review a collection of Cisco VoIP router platforms that cagate approaches for deploying call routing intelligence across multiple sites.

VoIP Components and ProtocolsA VoIP network relies on a collection of specialized hardware and protocols. This se

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CHAPTER 1


VoIP ComponentsFigure 1-1 shows the basic components of an IP telephony network.

n IP phone: Provides IP voice to the desktop.

n Gatekeeper: Provides call admission control (CAC), bandwidth control and ma

n G t P id t l ti b t V IP d V IP t k h th


FIGURE 1-1

IP Telephony Network

V

P

AnPh

Gateway

PSTN

IP WAN

Gateway/Gatekeeper

VideoconferenceStation

MCU UnifiedMessaging

Server

Call Agent

IP Phone

EthernetSwitch

V V

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n Call agent: Provides call control for IP phones, CAC, bandwidth control and mA Cisco UCM server often serves as a call agent in a Cisco IP telephony deploy

n Application server: Provides services such as voice mail, unified messaging, an

n Videoconference station: Provides access for end-user participation in videocostation contains a video capture device for video input and a microphone for austreams and hear the audio that originates at a remote user station. Cisco targets

at desktop videoconferencing applications.

Other components, such as software voice applications, interactive voice response (Iadditional services to meet the needs of enterprise sites.

For VoIP networks to replace traditional telephony networks (for example, PBX telepa collection of services and features that help make the migration to a VoIP environm

These features and services include the following:

n Signaling: Signaling information can help set up, maintain, and tear down a votransmitted using a series of signaling bits (commonly referred to as A, B, C, anbe used. Signaling System 7 (SS7) is a signaling protocol commonly used betwHowever, within a VoIP network, the signaling protocol is probably H.323, Med(MGCP), Session Initiation Protocol (SIP), or Skinny Client Control Protocol (S

n Database services: Database services can add flexibility to a VoIP network. Forto help automate wake up calls in a hotel environment or school closure announ


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n Codecs: A coder/decoder (that is, a codec) can convert between an analog waveby human speech, and a digital representation of that waveform. Some codecs (sent an analog waveform without performing any compression. Although such aquality, it requires more bandwidth than a codec (for example, G.729), which dBecause of the bandwidth savings they offer, codecs that perform compression for traffic traversing an IP WAN.

VoIP Signaling ProtocolsTo expand your understanding of signaling protocols, realize that VoIP gateways usegateway control protocols) to set up, maintain, and tear down a call between themselCisco UCM server. Consider some common VoIP signaling protocols: H.323, MGCP

H.323H.323 is the most mature of the VoIP signaling protocols, and it is considered to be aendpoints in an H.323 network (for example, an H.323 gateway) can have their own The H.323 standard is not a single protocol, but rather a suite of protocols and specifan H.323 network.

Among the H.323 protocols used for call setup are the following:n H 225 0: The H 225 0 protocol (often written as H 225) has a couple of function


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CHAPTER 1


In the topology shown in Figure 1-2, an H.323 gateway interconnects the PSTN withment. Notice an Integrated Services Digital Network (ISDN) circuit is coming into thprotocol for Layer 2 signaling and the Q.931 protocol for Layer 3 signaling. Notice tsignaling streams are terminated at the H.323 gateway and converted into correspond

Session Initiation ProtocolLike H.323, SIP is a peer-to-peer gateway control protocol that is popular in many myou are adding Cisco IP telephony components to an existing third-party IP telephonappropriate gateway control protocol. As discussed later, SIP uses a series of plain-te

Also like H.323, a SIP gateway terminates Q.921 and Q.931 signaling streams comingateway then converts those ISDN signaling messages into corresponding SIP messa

cluster, as shown in Figure 1-3.


FIGURE 1-2

H.323 Gateway

Public SwitchedTelephone

Network (PSTN)

UCM ClusterSwitch

H.323 Voice GWISDN

Q.921

Q.931

H.323

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Media Gateway Control ProtocolMGCP is a client/server gateway control protocol. In a Cisco IP telephony environmcall agent , and a port on a router (for example, an analog Foreign Exchange Station router containing the endpoint is the MGCP gateway, and this gateway registers withinto the gateway via the endpoint, the gateway notifies the call agent, and the call ag

Unlike H.323 and SIP, MGCP has the unique capability to terminate Q.921 signalinginside of IP packets. Those encapsulated messages can then be routed to a Cisco UCand respond to those native Q.931 messages. This feature, illustrated in Figure 1-4, iwhere PRI stands for Primary Rate Interface, or BRI backhauling in a Basic Rate Int

Skinny Client Control Protocol


NOTE

Cisco UCM Version 5.x

or later is required for

communication with a

SIP gateway. Although

Cisco UCM Version 4.x

supports SIP trunks (thatis, connectivity between

a UCM cluster and a SIP

proxy server), it does not

act as a SIP user agent.

FIGURE 1-4

MGCP Gateway


Network (PSTN)

UCM ClusterSwitch

MGCP Voice GWISDN

Q.921

Q.931

MGCP

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In addition to communication between IP phones and UCM servers, SCCP can be us(for example, Cisco VG224 and VG248 analog voice gateways) and a UCM server. Aexample, a collection of digital signal processors (DSPs) residing in an IOS router ca farm. This collection of resources can be used by a Cisco UCM server for a variety ocalling resource, and SCCP is used by the UCM server to communicate with the DSuses of SCCP.


FIGURE 1-5SCCP Signaling

Applications

UCM Cluster

IP Phone

VG224

Voice GWwith DSPs

SCCP

SCCP

SCCP

Switch

V

V

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VoIP Media ProtocolsWhen a VoIP call is established, using the previously discussed signaling protocols, tbe transmitted. These voice samples are often called the voice media. Following are that might be found in a VoIP environment:

n Real-Time Transport Protocol (RTP): RTP is a Layer 4 protocol that is encap

the protocol that carries the actual digitized voice samples in a call.n Real-Time Control Protocol (RTCP): RTCP is a companion protocol to RTP.

Layer 4 and are encapsulated in UDP. UDP ports 16,384 to 32,767 are used by the even port numbers in that range, whereas RTCP uses the odd port numbers.

n Compressed RTP (cRTP): One of the challenges with RTP is its overhead. SpeRTP headers are approximately 40 bytes in size, whereas a common voice paylo

bytes, which includes 20 ms of voice by default. In that case, the header is twicCisco supports cRTP, which is commonly referred to as RTP header compressio

header to 2 or 4 bytes in size (depending on whether UDP checksums are in use


FIGURE 1-6

Compressed RTP

Headers

UDPIP

CompressedHeader

Voice Payload Voice Payload

2 – 4 Bytes

RTP cRTP

V

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VoIP GatewaysEarlier you learned that a VoIP gateway interconnects dissimilar telephony networks network with the PSTN). Cisco offers multiple platforms that can serve as VoIP gateing platforms can act as VoIP gateways:

n Cisco 2800 series router

n Cisco VG248 gateway

n Cisco Catalyst 6500 Communications Media Module (CMM)

n Cisco Analog Telephone Adapter (ADA)

n Cisco 7200 series router

VoIP gateways can loosely be categorized as either analog gateways or digital gatew

or more analog voice ports (which are discussed later) that connect the gateway to ananalog phone or an analog port on a PBX). Similarly, a digital gateway contains oneexample, T1 or E1 ports) for connectivity to a digital circuit. For multiple connectioneconomies of scale, because a single digital voice port typically supports multiple sim

Cisco recommends that when you are selecting a gateway, the gateway meet the follo

n Support for the appropriate signaling protocol (for example, H.323, MGCP, or S

n Th bilit t i t d l t ltif (DTMF) t


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n Support for the Q Signaling (QSIG) protocol, which enhances interoperability bfacturers

n The capability to communicate fax and modem tones across an IP network, withby the codec in use

VoIP Deployment ModelsBecause many organizations have multiple locations, their IP telephony networks mimining how IP telephony components should be deployed, consider the following dement, multisite WAN with centralized call processing deployment, multisite WAN wdeployment, and a clustering over the WAN deployment.

Single-Site DeploymentIf an IP telephony network is contained within a single location, as illustrated in Figuphones, a single-site deployment model is often appropriate.


FIGURE 1-7

Single-Site

Deployment

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Multisite WAN with Centralized Call Processing DeSome organizations might have smaller remote sites that do not contain enough IP phonesfor those locations. In those instances, the UCM servers could be located at the headquartecould then register with the centralized UCM servers over the IP WAN. In the event of an ter with the local Survivable Remote Site Telephony (SRST) routers located at each remottionality. Figure 1-8 shows an example of this multisite WAN with centralized call process


FIGURE 1-8Multisite WAN with

Centralized Call

Processing

Deployment

IP WAN

CCM Cluster

SRST

Remote Office A

Headquarters V

Public Switched

Telephone

Network (PSTN)

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Multisite WAN with Distributed Call Processing DeWhen designing a large IP telephony network with multiple locations, the expense otion might be justified. As an example, Figure 1-9 provides a sample IP telephony todistributed call processing deployment model.


FIGURE 1-9

Multisite WAN with

Distributed Call

Processing

Deployment

UCM Cluster

IP WAN


Network (PSTN)

V

V

V

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Clustering over the WAN DeploymentIn an IP telephony environment that spans multiple locations, having all UCM serversimplify administration. However, having a single cluster at a single site leads to redWAN outage. However, a fourth deployment approach, clustering over the WAN, offservers belong to a single server while having the servers physically dispersed acrossSpecifically, each site has one or more UCM servers on site. These dispersed UCM s

together into a single cluster, as illustrated in Figure 1-10. This type of design does hments to support the intracluster communication among the UCM servers. For examp40-ms round-trip delay between any two UCM servers in the cluster.


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FIGURE 1-10Clustering over the

WAN Deployment

UCM Cluster

V

V

V


Network (PSTN)

IP WAN

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CHAPTER 2

Addressing VoIP Design Considerations

Chapter 2Addressing VoIP Design ConsiderationsDesigning a VoIP network requires the designer to consider the special requirementslenges that can arise from just superimposing voice on top of an existing data networ

quality, in addition to the integrity of modem, fax, and dual tone multifrequency (DTin a VoIP network, and what solutions are available. As you will see, bandwidth avaition, and this chapter examines various factors impacting the bandwidth demand of a

Also, many VoIP features require considerable processing power (for example, mixina conference call). Digital signal processors (DSPs) can be used to provide the proceand this chapter introduces you to the theory and configuration of DSPs.

Voice-Quality ConsiderationsIn a VoIP network, voice and data coexist on the same links. Therefore, when congesbandwidth, voice and data traffic simultaneously contend for access to the network. Tpoor voice quality.

A primary design goal of a VoIP network, however, is to have a level of voice quality


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CHAPTER 2


The Challenge of Compressing NonvAlthough codecs such as G.729 do a great job of compressing voice, they are not desuch as fax, modem, or DTMF tones. Although fax, modem, and DTMF informationwithout a problem, the G.729 codec corrupts these signals to a point where a device cannot interpret them, as illustrated in Figure 2-1.

Fax TransmissionCisco supports several solutions for transmitting fax tones across an IP network. Con

n Fax pass-through: Fax pass-through detects a fax call is in progress and switchexample, G.729) to G.711, which does not do compression and therefore does ntones are then transmitted using G.711. Although this approach does preserve th

might not be the most desirable approach because it uses extra bandwidth (due tof the fax transmission.


FIGURE 2-1Fax Transmission with

G.729

?

FaxFax

G.729V V

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n T.38: Similar to Cisco Fax Relay, T.38 transmits fax information via specially fInternet Fax Protocol (IFP) packets, which are sent using UDP. However, unlikestandard and might therefore be more appropriate in a mixed-vendor environme

n T.37: T.37 is referred to as store-and-forward fax . With T.37, a fax transmissiongateway (called an on-ramp gateway). The on-ramp gateway converts the fax trtation of the fax. Specifically, the fax is converted into a TIFF graphic format, a

the TIFF file, as an email attachment, to an Simple Mail Transfer Protocol (SMbe a user’s email account, which would enable her to read the fax transmission ent could be another voice gateway (known as an off-ramp gateway). The off-rawith the TIFF attachment and convert the graphic back into fax tones, which coto a destination fax machine.

Modem TransmissionSimilar to the challenges of fax transmission, modem tones can also be corrupted as compression. Although Cisco voice gateways support modem pass-through, similar ttones to be transmitted using G.711, a more bandwidth-friendly approach is modem

Specifically, to transmit modem tones across a WAN when you have specified the Guse modem relay, which sends modem information through the Simple Packet Relayhop router then remodulates the data and sends it to the destination router.


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CHAPTER 2


information. However, like fax tones and modem tones, DTMF tones may be degradpoint where the receiving equipment cannot correctly interpret the DTMF values.

Therefore, some sort of DTMF relay technology is required to successfully transmit using a codec that performs compression. The DTMF relay approach largely depend

Options for H.323 DTMF Relay

n Cisco DTMF Relay: DTMF tones are sent via RTP, using a different RTP payl

n H.245 alphanumeric: DTMF tones are sent as text rather than tones.

n H.245 signal: Similar to H.245 alphanumeric, H.245 signal sends tones as text indicate how long individual tones were generated.

n NTEs: Named Telephony Events (NTEs) are used to transmit DTMF tones insi

RFC 2833 Section 3, with a negotiated RTP payload type.

Options for MGCP DTMF Relay

n Cisco DTMF Relay: DTMF tones are sent via RTP, using a different RTP payl

n NSEs: Named Signal Events (NSEs) are used to transmit DTMF tones inside ofRTP payload type.

n NTEs: Media Gateway Control Protocol (MGCP) can use NTEs in one of two m


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SIP DTMF RelayIf you are in a Session Initiation Protocol (SIP) environment, Cisco voice gateways sencoding DTMF tones in SIP NOTIFY messages, resulting in NOTIFY-based out-of

Factors Impacting Bandwidth RequirA variety of factors need to be analyzed when determining how much bandwidth is rfollowing:

n Codec selection: One approach is to use a codec requiring less bandwidth per cdoes not perform any compression, and it requires 64 kb/s of bandwidth (not incall. However, over an IP WAN, where bandwidth is at a premium, Cisco netwowhich requires only 8 kb/s of bandwidth (not including overhead). Because G.7

G.711 does not, voice quality is somewhat compromised when using G.729a.

n Compressed RTP (cRTP): When using G.729a, voice packets contain 20 bytescontains 40 bytes of header information. However, because most information inexample, the same source/destination IP address / UDP port numbers and the sanot send this redundant header information in each frame. Therefore, cRTP redu2 or 4 bytes, allowing more calls to be placed over the same link speed.

n Voice activity detection (VAD): Statistics show that approximately 35 percent


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n Data-link overhead: Layer 2 of the OSI reference model (that is, the data link packet. For example, when you are sending voice packets over an Ethernet netwin size. For Multilink PPP (MLP) and Frame Relay Forum Standard 12 (FRF.12tion is added.

The following formula shows how to calculate a network’s required voice bandwidth

Bandwidth = ((Layer 2 header) + (IP/UDP/RTP header)) * (Codec bit rate) / (Voice paylo

When working with this formula, make the following assumptions:

n IP/UDP/RTP header = 40 bytes

n With cRTP, the header = 2 or 4 bytes

n A WAN’s Layer 2 header = 6 bytes

An easier, and more detailed, bandwidth calculation can be performed using the CiscCalculator, available at http://tools.cisco.com/Support/VBC/do/CodecCalc1.do.

Digital Signal Processors

DSPs are microprocessors residing in Cisco IOS routers or Cisco Catalyst switches (series switch with a CMM module). These DSPs can be leveraged in an IP telephony


NOTE

Your Cisco.com account

must have appropriate

access permissions to

access the Voice Codec

Bandwidth Calculator

URL.

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DSP ApplicationsConsider some of the applications of DSPs:

n Transcoding: Translating between a high-bandwidth codec (for example, G.711example, G.729)

n Voice termination: Interconnecting a time-division multiplexing (TDM) link (f

from the public switched telephone network [PSTN]) with a VoIP network n Media termination point (MTP): Interconnecting two VoIP call legs, both of w

n Conferencing: Mixing multiple audio streams into a single stream

Codec Complexity

Because codec algorithms vary in their arithmetic complexity, the number of calls suthe codec in use. For example, G.711, G.729a, and G.729ab are considered medium-

and G.729b are considered high-complexity codecs.

Another consideration is the chipset used by a router. Newer Integrated Services Rouwhereas older routers often use the C549 chipset.

With a medium-complexity codec, the C549 DSPs can support four calls per DSP, weight calls per DSP. With a high-complexity codec, the C549 DSPs can support two


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Notice the C5510 chipset support for the flex and secure parameters. The flex parammatically adjust to either medium or high complexity depending on the needs of a caG.711 codec, the C5510 chipset automatically adjusts to the medium-complexity mothe C5510 chipset uses the high-complexity mode. Thus, the chipset can accommodausing the most efficient complexity mode. This feature is not supported in the C549 c

Another parameter supported on the C5510 chipset that is not supported on the C549

This parameter supports Secure RTP (sRTP), which adds authentication and encrypti

You can view codec complexity for a DSP by using the show voice dsp command. Scommand shows the chipset installed in the router and the chipset’s configured comp

DSP Configuration

Cisco offers the DSP Calculator, as illustrated in Figure 2-2, which can provide a netDSPs needed for a particular design and a template of the required command-line int

With the appropriate login credentials, you can access the DSP Calculator at http://wbin/Support/DSP/dsp-calc.pl.

To make DSP resources available to the Cisco UCM-based IP telephony environmen

form containing the DSPs and the Cisco UCM server. Figure 2-3 shows a sample con


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CHAPTER 2



FIGURE 2-2

DSP Calculator

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Cisco IOS DSP ConfigurationFirst, consider the configuration of the router KY_Router. As a first step, you enter vassign the DSPs on that card to a DSP farm, as shown in Example 2-1.

Example 2-1 DSP Farm Configuration

KY_Router(config)#voice-card 0

KY_Router(config-voicecard)#dsp service dspfarm

N DSP f fil h KY R Thi fil d fi h


FIGURE 2-3DSP Farm Sample

Topology

IP WAN

192.168.0.10

Fa1/1

Kentucky Arizona

Conference Bridge

KY_Router AZ_Router

V V

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Example 2-2 DSP Farm Profile ConfigurationKY_Router(config)#dspfarm profile 1 conference

KY_Router(config-dspfarm-profile)#codec g711ulaw

KY_Router(config-dspfarm-profile)#codec g729abr8

KY_Router(config-dspfarm-profile)#codec g729ar8

KY_Router(config-dspfarm-profile)#codec g729br8

KY_Router(config-dspfarm-profile)#maximum sessions 2

KY_Router(config-dspfarm-profile)#associate application SCCP

KY_Router(config-dspfarm-profile)#no shutdown

Because SCCP will be used for communication between the DSP farm and the Ciscoconfigured to run SCCP. Example 2-3 shows the syntax for specifying the router intespecifying the IP address of a Cisco UCM server (which is a version 4.1 server in this used to start SCCP. An SCCP group is then configured. The group is associated w(although a keyword of ccm is used rather than ucm, because Cisco Unified Commuknow as Cisco CallManager [CCM]). Also, the SCCP group is associated with a DSDSP resource is specified using the register command. This same name will need toUCM server configuration.

Example 2-3 SCCP Configuration

KY_Router(config)#sccp local fastethernet 1/1

KY Router(config)#sccp ccm 192 168 0 10 identifier 1 priority 1 version 4 1


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Cisco UCM Server Configuration to Reference a DSP FarmNow that the router has been configured to offer a DSP conferencing resource, the Cured to accept a registration from the DSP farm, which will allow the DSP resourcesserver when it is in need of a conferencing resource. From the Cisco UCM Administ> Media Resource > Conference Bridge menu option, as shown in Figure 2-4.


FIGURE 2-4

Accessing theConference Bridge

Configuration Screen

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Click the Add a New Conference Bridge link that appears onscreen. Assume that inthat uses the C5510 chipset. You will therefore specify Cisco IOS Enhanced Confer

Bridge Type field, as shown in Figure 2-5. If your router uses the C549 chipset, you Conference Bridge.


FIGURE 2-5

Specifying the

Conference Bridge

Type

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Enter the name you assigned in the IOS configuration (using the register command)field, as shown in Figure 2-6. Select an appropriate device pool, and click Insert.


FIGURE 2-6

Specifying the Name

of the DSP Resource

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CHAPTER 3

Understanding and Configuring Analog Voice Ports

Chapter 3Understanding and Configuring Analog VNow that you understand the fundamental components of a VoIP network, and you hdesign considerations, in this chapter you learn how to start configuring voice interfa

this chapter addresses the configuration of analog voice ports; Chapter 4, “UnderstanPorts,” addresses digital voice ports.

Call TypesFirst, consider the following categories of voice calls that will be using these voice p

n Local calls: A local call occurs when the calling and called phones are both phy

n On-net calls: On-net calls span more than one router. Specifically, the calling pthe called phone attaches to a different router. In this case, the routers are both p

n Off-net calls: Off-net calls originate on a router but terminate on the public swi

n Private line, automatic ringdown (PLAR) calls: A PLAR call occurs when a phone automatically dials a preconfigured number.


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n On-net to off-net calls: An on-net to an off-net call is a call that was intended tof conditions such as WAN oversubscription or a WAN outage, the call is divertPSTN).

Analog Voice PortsThe three primary types of analog voice ports available for Cisco voice-enabled gateFXS, FXO, and E&M ports.

n FXS: A Foreign Exchange Station (FXS) port enables you to connect plain old a router. For example, you can attach a traditional analog phone, speakerphone,Cisco router, and that FXS port can act like a PBX or central office (CO) switch

provide a dial tone when the phone goes off-hook, interpret dialed digits, and seh


FIGURE 3-1

Analog Voice Ports

PBX

IP WAN

PSTN

AnalogPhone

FXS

FXO

E&M

V

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Analog SignalingConsider the following types of signaling present in the analog telephony world:

n Supervisory signaling: Supervisory signaling indicates the on-hook or off-hookwhether loop current is flowing. In addition, ringing is considered to be supervisent from the phone switch to alert the destination phone that it is receiving an i

the default pattern of ringing (that is,ring cadence

) is 2 seconds on and 4 seconn Address signaling: Address signaling allows a phone to dial (that is, specify th

The older method of dialing digits was with a rotary phone, which used pulse d

and closes the tip-and-ring circuit. This series of open and closed circuit condititers indicates a dialed digit.

A more efficient approach to address signaling is dual tone multifrequency (DT

taneous frequencies are generated, and this combination of frequencies is interpdigit. For example, the combination of a 697-Hz tone and a 1209-Hz tone indic

n Information signaling: Like DTMF, information signaling uses combinations ocate the status of a call (that is, to provide information to the caller). For exampa 480-Hz tone and a 620-Hz tone, with on/off times of 0.5/0.5 seconds.

Configuring Analog Voice Ports


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FXS Port ConfigurationA phone connects to an FXS port, just as a phone would connect to a PBX or the PSparameters such as signal type (that is, loop start or ground start), ring pattern, impedconnecting device), and call progress tones (for example, what a busy signal sounds configuration scenario, as illustrated in Figure 3-2 and Example 3-1.

Example 3-1 FXS Port Configuration

Router(config)#voice-port 1/1/1

Router(config-voiceport)#signal loopstart

Router(config-voiceport)#impedance 600r

Router(config-voiceport)#ring cadence pattern02

Router(config-voiceport)#output attenuation -2

Router(config-voiceport)#input gain 3

Router(config-voiceport)#echo-cancel coverage 32

In this example, voice port 1/1/1 is an FXS port, and you are specifying that it shoult d t t FXS ( d FXO) l t ith l t t d t t


FIGURE 3-2

FXS Port Sample

Topology

FXS1/1/1

AnalogPhone

V

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CHAPTER 3


Also, the impedance is set to 600 ohms resistive (that is, no capacitive component). Tpredefined pattern02, which specifies a cadence of 1 second on and 4 seconds off. Bthe amplitude) of VoIP calls to be approximately the same as the volume of the calls VoIP network as transparent to the users as possible), you can make gain and attenuayou are attenuating (that is, reducing the volume) of calls that are being sent out of tdecibels (dB), with the output attenuation command. The input gain command, in volume of the waveforms that are coming from the phone into the router by 3 dB.

To combat echo, you are increasing the coverage (that is, how long a router memorizfrom that waveform) to 32 ms, with the echo-cancel coverage command. Finally, toenabled the nonlinear feature, which suppresses all incoming waveforms from the phto be interpreted as speech. Note that this nonlinear feature can lead to clipping whencan enter the nonlinear feature with the non-linear voice-port configuration mode co

FXO Port ConfigurationAn FXO port connects to a phone switch (for example, a CO switch or PBX). ThereTypical parameters that you can configure on an FXO port are signaling (which mustphone switch to which you are connecting), dial type (that is, DTMF or pulse), and tthe FXO port answers (that is, goes off-hook). Consider the following FXO configur

and Example 3-2.


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Example 3-2 FXO Port ConfigurationRouter(config)#voice-port 1/2/1

Router(config-voiceport)#signal loopstart

Router(config-voiceport)#ring number 3

Router(config-voiceport)#dial-type pulse

In this example, voice port 1/2/1 is an FXO port, and you are specifying that it shoul

because the FXO port acts like a phone, you can specify how many rings it receives FXO port will answer after three rings. Also, when the FXO port dials, it is configur

E&M Port ConfigurationAn E&M port in a router typically connects to an existing E&M port on a PBX. Para

E&M port include the signaling type (for example, wink start), the E&M type (that iwires that are used for the voice path (that is, the operation). Figure 3-4 and Exampluration.


FIGURE 3-4

E&M Port Sample

Topology PBX PBX

E&M E&M

2/1/1 V V

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CHAPTER 3


In this example, an E&M port is configured as E&M Type I. Type I is most commonmost common outside North America. Another E&M type you might encounter is Tycalling and called equipment (for example, a router or PBX) to be located in the samis not required).

In addition, the voice path, which does not use the E&M leads, uses four wires, meahave their own return path. The default signaling method of wink start is also specifi

calling equipment goes off-hook on its E-lead and waits for the called equipment to wink) on its M-lead. This wink indicates to the calling equipment that the called equ

Timing OptionsNumerous timing options can optionally be configured for Cisco voice ports. To illus

Example 3-4 Timing Parameter Configuration

Router(config)#voice-port 1/1/1

Router(config-voiceport)#timeouts interdigit 20

Router(config-voiceport)#timeouts initial 20

Router(config-voiceport)#timeouts call-disconnect 20

In this example, voice port 1/1/1 is an FXS port. The interdigit parameter determinethat are allowed between dialed digits The initial parameter specifies how long the c


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CHAPTER 3


Dial PeersAt this point, you have seen how to configure voice ports on Cisco voice-enabled routrained the routers to reach specific destinations. That is the focus of this section. Spdial peers that inform the routers how to reach specific phone numbers. Consider the

Routers R1 and R2 each have a POTS dial peer that points to their locally attached phto the IP address of the remote router.

Therefore, when extension 1111 dials extension 2222, router R1 searches for a dial p

of 2222. In this case, R1 has a VoIP dial peer that points to R2’s IP address of 10.1.1Th R2 i th i i ll th t i d ti d f t i 2222 R2 h


FIGURE 3-5

Dial Peers

and Call Legs

x1111 x2222

FXS1/1/1

FXS1/1/1R1 R2

Call Leg 1 Call Leg 2 Call Leg 3 Call Leg 4

x2222 => 10.1.1.2

VoIP Dial Peer

x1111 => 10.1.1.1

VoIP Dial Peer

x1111 => 1/1/1

POTS Dial Peer

x2222 => 1/1/1

POTS Dial Peer

10.1.1.1

10.1.1.2V V

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CHAPTER 3


Notice that you have a total of four dial peers that allow a call in the opposite directithe call (that is, call legs) are defined, two call legs from the perspective of each rout

n Call Leg 1: The call comes into R1 on FXS port 1/1/1.

n Call Leg 2: The call is sent from R1 to IP address 10.1.1.2.

n Call Leg 3: R2 receives an incoming call that is destined for extension 2222.

n Call Leg 4: R2 forwards the call out FXS port 1/1/1.

POTS Dial PeersWhen configuring a POTS dial peer, specify the following two parameters:

n The destination-pattern (that is, the phone numbers)

n The physical port address

Consider the POTS dial-peer configuration presented in Figure 3-6 and Example 3-5


FIGURE 3-6

POTS Dial-Peer

Topology

x1111

FXS1/1/1 R1

x1111 => 1/1/1

POTS Dial Peer

V

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CHAPTER 3


In this example, an analog phone is attached to FXS port 1/1/1. You entered dial-peepeer with the dial-peer voice 1111 pots command. Notice that the 1111 in the dial-p

match the phone number. The number is merely a locally significant tag. However, toitive to interpret, you might want to adopt a practice of using the extension number aextension number of 1111 is specified with the destination-pattern 1111 command.specified with the port 1/1/1 command.

VoIP Dial PeersWhen configuring a VoIP dial peer, you specify a remote phone number with the samthat was used in a POTS dial peer. However, instead of identifying a local port, a VoIpackets’ destination IP address. Consider the VoIP dial-peer configuration provided i

Example 3-6 VoIP Dial-Peer Configuration

R1(config)#dial-peer voice 2222 voip


FIGURE 3-7

VoIP Dial-Peer

Topology

x2222

FXS1/1/1R1 R2

x2222 => 10.1.1.2

VoIP Dial Peer

10.1.1.1

10.1.1.2V V

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CHAPTER 3


Matching Multiple Phone Numbers Using WildcardsAt this point, you understand how to establish a dial plan for two phones that are sepwhat if you have 1000 extensions on the other side of the WAN? You certainly wouldFortunately, the Cisco IOS enables you to use wildcards to specify a range of addresused wildcards are as follows:

n T: A T represents a dial string of any length. For example, you could specify a

any number that begins with a 9, and then you could point any calls matching ththe T can be any number of digits, the router forwards the call, by default, afterseconds or after the caller presses the # key.

n .: A period indicates any single digit. For example, you could specify extension with the 7… destination pattern.

n

[]: Brackets can be used to specify a range of numbers. For example, a 123[4-6digit number of 1234, 1235, or 1236.

Figure 3-8 demonstrates how wildcard characters used in the destination-pattern conation phone numbers, instead of just a single phone number. Note that the 9T patterwith a 9. So, the users could be trained to dial a 9 to get an outside line, after which number. Also note how the 7… destination pattern is used to point to a remote officerange 7000 to 7999.


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CHAPTER 3


Inbound Versus Outbound Dial-Peer MatchingBe aware that not only does a router need to match an outbound dial peer, it also neeThe router attempts to match an inbound dial peer in the following order:

n incoming called-number: This command matches the Dialed Number Identific(that is, the dialed number).

n

answer-address: This command matches the Automatic Number Identification ID number).


FIGURE 3-8Wildcard Pattern

Examples Destination-Pattern 9T

Destination-Pattern 7...

V

PSTN

IP WAN

x7000 – 7999

PBX

V

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CHAPTER 3


In the event of a tie (that is, if multiple dial peers match the same port parameter), thselected. However, if no match is found, the default dial peer (that is, dial peer 0) is u

The default dial peer can be used in the absence of an appropriately configured inboudial peer cannot take the place of an outbound dial peer.

Multiple Dial Peers with Matching Destination PattYou can have a configuration in which multiple destination patterns match a dialed npattern is used? The most specific destination pattern is used when multiple destinatiConsider the configuration presented in Example 3-7.

Example 3-7 Multiple Dial-Peer Matches

Router(config)#dial-peer voice 1 voipRouter(config-dial-peer)#destination-pattern 2468

Router(config-dial-peer)#session target ipv4:192.168.1.1

Router(config-dial-peer)#dial-peer voice 2 voip

Router(config-dial-peer)#destination-pattern 2...

Router(config-dial-peer)#session target ipv4:192.168.2.2

Router(config)#dial-peer voice 3 voip

Router(config-dial-peer)#destination-pattern 2TRouter(config-dial-peer)#session target ipv4:192.168.3.3






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CHAPTER 3


Voice-Port and Dial-Peer VerificationVarious test commands are also available for troubleshooting purposes. For exampleringing voltage to a voice port with the following command:

Router#test voice port 1/1 relay ring on

You can also play a tone out to an attached phone during a phone call with the follow

Router#test voice port 1/1 inject-tone local 500hz

Note that other frequencies can be specified. You can even cause a gateway to dial a command. For example, you can enter the csim start 5551212 command to cause thtion phone number of 555-1212.


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CHAPTER 4

Understanding and Configuring Digital Voice Ports

Chapter 4Understanding and Configuring Digital VDigital voice ports, unlike analog voice ports, have the unique capability to carry muthe same circuit. Therefore, in installations with multiple circuits (about seven or mo

by digital voice ports start to make digital voce ports more economically attractive th

Digital Voice-Port TheoryAlthough analog ports send and receive analog waveforms that continually vary, digibinary 1s/0s), which are represented on the wire as the presence of voltage or the abs

circuits include T1, E1, and Integrated Services Digital Network (ISDN) circuits, as


FIGURE 4-1

Digital Port Examples

T1 T1E1 E1

IP WAN

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CHAPTER 4


Multiplexing, Framing, and Line CodingA common technology used to send multiple conversations over a single connection Consider a T1 circuit, which has 24 separate channels. With TDM, a T1 circuit can schannel, followed by an 8-bit sample from its second channel, followed by an 8-bit son. Each channel, which means each conversation, gets its own time slice in which iillustrated in Figure 4-2.

A T1 digital circuit has 1.544 Mb/s of bandwidth. Consider the calculation of that ba

n Each frame is 193 bits in size: 24 channels x 8 bits per channel + 1 framing bit


FIGURE 4-2Time-Division

MultiplexingV V

T1 Circuit

Channel 1

Channel 2

Channel 3

Channel 24

1232412324

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CHAPTER 4


However, a T1 port does not send a single T1 frame at a time. Rather, it groups multall simultaneously. Two approaches to grouping these frames together are as follows

n Super frame (SF): Combines 12 standard 193-bit frames into a super frame

n Extended super frame (ESF): Combines 24 standard 193-bit frames into an ex

When you are configuring a T1 port (also known as a T1 controller on a Cisco routeframing type. However, for real-world installations, ESF is almost always used.

Another configuration parameter for a T1 controller is the line coding. A T1 circuit’sdictates how binary 1s and 0s are represented over the wire. Binary 1s are often thouvoltage, and binary 0s being the absence of voltage. Although that is true, the goal ovoltage on the line 0 volts, which means when you send a binary 1 using a positive vtive voltage. Therefore, on average, the voltage on the wire is approximately 0.

If two consecutive voltages have the same polarity, an error, called a bipolar violatio

ing binary 1s as alternating voltages is called alternate mark inversion (AMI), and is


FIGURE 4-3

Alternate Mark

Inversion

Bit Pattern

Voltage

Voltage

0

0 1 1 0 1 0 1 0

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CHAPTER 4


Because of AMI’s limitation, another type of line coding was developed. Bipolar 8-za byte containing all 0s by creating a couple of bipolar violations. If a T1 circuit usinbipolar violations at very specific bit positions, as seen in Figure 4-5, the equipment voice-enabled router) knows that a byte containing eight 0s is being transmitted.

Approaches to Digital Signaling

Just as an FXS port needs some type of signaling (for example, loop start or ground on hook or off hook a T1 circuit needs a signaling mechanism Two approaches to s


FIGURE 4-4AMI’s Issue with Eight

Consecutive 0s

Bit Pattern

Voltage

Voltage

0 ERROR

0 0 0 0 0 0 0 0

FIGURE 4-5

B8ZSBit Pattern

Voltage

Voltage

Byte Containing All Zeros

Bipolar Violations

0

01 0 0 0 0 0 0 0 1

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CHAPTER 4


n

Channel associated signaling (CAS): With CAS, specific low-order bits are “rspecific channels. The SF or ESF takes these low-order bits from every sixth chSF and channels 6, 12, 18, and 24 for ESF). This approach is sometimes referrenone of the 24 channels are dedicated to just sending signaling information, unlused.

Common Channel Signaling

Consider CCS in more detail. As the name suggests, all the channels used for sendinchannel (that is, a “common channel”) to send signaling information. A signaling prochannel.

A popular technology that leverages CCS is ISDN. As previously mentioned, an ISDand a D channel. A B-channel is a bearer channel, which carries the voice, data, or vcarry information at a rate of 64 kb/s. The D channel acts as the signaling channel, mdata necessary to set up and tear down calls on the B channels. Depending on your beither the BRI or the PRI types of ISDN:

n Basic Rate Interface (BRI): BRI ISDN connections contain two 64 kb/s B chaa total usable bandwidth of 128 kb/s.

n Primary Rate Interface (PRI): A PRI ISDN connection can use the channels o

PRI is based on a T1 circuit, 23 of the T1’s 24 channels are used as B channels,th D h l H if th PRI i b d E1 i it 30 f th E1’ 32


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CHAPTER 4


Channel Associated SignalingWith CAS on a T1 circuit, the low-order (that is, the rightmost) bit in every sixth frasignaling, rather than for voice sampling, as shown in Figure 4-6. The impact on voiof these bits is generally considered to be negligible.

Just as T1 circuits are popular in North America, E1 circuits are common in Europe.opposed to the 24 channels available in a T1. The first of those 32 channels is dedicaand the seventeenth channel is dedicated to signaling. Based on the discussion of howchannel using CAS, it might be tempting to think you could do the same with an E1nels to send your voice, video, and data. However, an E1 circuit approaches CAS ver

On a standard E1 circuit, the seventeenth channel is always used for signaling, regar

being used. If CCS is being used, a signaling protocol (for example, Q.931) is sent obeing used the 8 bits in the seventeenth channel are used for signaling


FIGURE 4-6

Robbed-Bit Signaling

Robbed Bits

Super Frame(24 193-Bit Frames)

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CHAPTER 4


and 4 bits of signaling information for channel number 18 are being carried in the seframe in an E1 multiframe. Similarly, the seventeenth channel of the third frame in aing information for channel 3 and 4 bits of signaling information for channel 19, as scontinues for each of the remaining frames in the multiframe, such that the multiframchannels, which is exactly the number of channels you use in an E1 to send voice, vi


FIGURE 4-7

E1 Multiframe 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16

1

2

3

4

5

6

7

8

9

10

11

12

13

14

Time Slot

17 18 19 20 21 22 23 24 25 26Frame

NOTE

An E1 CAS trunk is

commonly referred to as

an E1 R2 trunk .

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CHAPTER 4


CAS ConfigurationDigital voice interfaces, such as T1, E1, and ISDN interfaces, have unique interface-These parameters can include the type of line coding (for example, B8ZS) and framithe digital circuit. Consider the following T1 CAS configuration example, as illustratNote that a T1 is configured from controller configuration mode, as opposed to intermode:

Example 4-1 T1 Configuration Example

Router1(config)#controller 2/0

Router1(config-controller)#clock source line

Router1(config-controller)#framing esfRouter1(config-controller)#linecode b8zs


FIGURE 4-8

T1 CAS Sample

Topology

Router1

T12/0

Router2

V V

PSTN

IP WAN

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CHAPTER 4


In this example, T1 controller 2/0 gets its clocking from the network, as indicated byaddition, the T1 is configured for extended superframing (ESF) and B8ZS line codinare grouped together in DS0 group 0, and they support FXO loop-start signaling.

Next, a plain old telephone service (POTS) dial peer is configured to match four-digdial peer forwards the call out of port 2/0:0, where 2/0 identifies the T1 controller (threferences the number of the previously configured DS0 trunk group. In this example

issued to make this POTS dial peer less preferable than a VoIP dial peer (not shown)an IP WAN. Also, because a POTS dial peer only forwards digits matching the wildcpattern command, the forward-digits all command is issued to prevent the stripping

Also, notice the fxo-loop-start parameter. This parameter specifies the signaling typother options are supported for this type parameter. For example, E&M offers differecapabilities, as follows:

n E&M FG-B: E&M Feature Group B can send and receive Dialed Number Infoand it can receive Automatic Number Identification (ANI) information. Howeveonly on the Cisco AS5x00 series platforms.

n E&M FG-D: E&M Feature Group D supports incoming and outgoing DNIS, an

n E&M FG-D EANA: E&M Feature Group D Exchange Access North American

DNIS, and it can transmit ANI information.C id l h l h E&M f S h


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CHAPTER 4


Example 4-2 Using Multiple Trunk Groups to Send and Receive ANIRouter(config)#controller 2/0

Router(config-controller)#clock source line

Router(config-controller)#framing esf

Router(config-controller)#linecode b8zs

Router(config-controller)#ds0-group 0 timeslots 1-12 type e&m-fgd

Router(config-controller)#ds0-group 1 timeslots 12-24 type fgd-eana

Router(config-controller)#exitRouter(config)#dial-peer voice 100 pots

Router(config-dial-peer)#incoming called-number .

Router(config-dial-peer)#port 2/0:0

Router(config-dial-peer)#exit

Router(config)#dial-peer voice 200 pots

Router(config-dial-peer)#destination-pattern 9T

Router(config-dial-peer)#port 2/0:1

CCS ConfigurationTo illustrate the configuration of a CCS circuit, consider the configuration of an ISDExample 4-3. Router1 is connected to a PBX via an ISDN PRI circuit.


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CHAPTER 4


Example 4-3 ISDN PRI Configuration Example

Router1(config)#isdn switch-type primary-qsig

Router1(config)#controller t1 2/0

Router1(config-controller)#pri-group timeslots 1-24

Router1(config-controller)#exit

Router1(config)#interface serial 2/0:23

Router1(config)#isdn incoming-voice voice

In this example, the type of ISDN switch is globally defined as primary-qsig. Althoconfigured, the primary-qsig option specifies that the attached ISDN switch uses Q

used to support multiple ISDN features between ISDN equipment for multiple vendothe 24 channels of the ISDN circuit are grouped together into a PRI group.

To configure the D channel (that is, the signaling channel) of the T1 PRI circuit, youmode. Notice that this example goes into interface configuration mode for interface sT1 controller, and the 23 indicates the D channel on that T1 controller. The number 2though the D channel is the twenty-fourth channel on the T1) because the channel num

ISDN i i l f d i F l h i l


FIGURE 4-9

ISDN PRI Sample

TopologyPBX

ISDN PRIRouter1

T12/0

V

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CHAPTER 4


Q.931 functions are communicated from one PBX, or other QSIG-speaking node, to

Consider a few examples of QSIG features:

n Callback: If you call another party, and you receive a busy signal, because the back feature can be used to alert you after their phone goes on-hook.

n Call transfer: QSIG can support the transfer of a call, without hairpinning (tha

tandem hop as part of the call path).n Message waiting: QSIG supports signaling that can activate or deactivate mess

Non-Facility Associated SignalingIf you have multiple PRIs, instead of using up a D channel on each of those PRIs, yo

NFAS. Specifically, NFAS enables you to control multiple PRI interfaces with a singbackup D channel if the primary D channel is lost.

Verification of Voice PortsFor verification and troubleshooting purposes, consider the following commands:

n show voice port: Displays detailed settings for voice ports


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CHAPTER 5

Exploring Gateway Control Protocols

Chapter 5Exploring Gateway Control ProtocolsIn this chapter, you learn about the theory and configuration of the following gatewaInitiation Protocol (SIP), and Media Gateway Control Protocol (MGCP). The discuss

these protocols, H.323.

H.323 Theory and ConfigurationH.323 is not a single protocol. Instead, it is a suite of protocols. This suite specifies scall signaling.

You should be aware of the following two protocols used to establish an H.323 call:

n H.225: Performs call setup and registration, admission, and status (RAS) functi

n H.245: Performs call control, including a capabilities exchange (for example, co

In addition to various protocols, H.323 identifies the physical components of an IP te

n Terminals: An H.323 terminal is an end-user device that communicates with anH 323 i l h G 711 d


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CHAPTER 5


n Gatekeepers: To prevent voice calls from oversubscribing the WAN bandwidthbe identified. Then, a gatekeeper (GK) can keep track of the number of calls mamade within a zone. Based on the available bandwidth within or between zonesattempt. The GK can also perform centralized E.164 number resolution, in addidiscussed later in these reference sheets.

n Multipoint control units (MCUs): An MCU supports conference calling by ada call and by mixing voice streams together.

GKs are optional components because you can have an H.323 GW directly communiHowever, this approach has scalability limitations. If you introduce GKs into the netusing the RAS channel. In larger topologies, you can have multiple GKs, and those Gusing the RAS channel.

Consider how a call is completed in the following H.323 call scenarios:

n GW-to-GW calls: This topology does not require a GK. Specifically, both GWother. First, H.225 performs the call setup, followed by H.245 performing a capFigure 5-1. However, this negotiation requires numerous packet exchanges betwuse H.323 Fast Connect, which performs call setup and does a capabilities exchmessages between the two GWs.

n GW-to-GK-to-GW calls: With a GK in your topology, the originating GW reqplace a call using an admission request (ARQ) message, after which the GK can


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CHAPTER 5



FIGURE 5-1

H.323 Call Setup

Without a GK

H.225 Call Setup

H.245 Capabilities Exchange

RTP Streams

RTCP Stream

H.323 GW H.323 GWx1111 x2222

V V

FIGURE 5-2

H.323 Call Setup

with a GK

H.225 Call Setup(2)

(4)

H.323 GW

H.323 GK

(1) Admission Request andConfirmation (H.225 RAS)

(3) Admission Request andConfirmation (H.225 RAS)

H.323 GWx1111 x2222

V V

V

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CHAPTER 5


For even larger environments, you can have multiple GKs involved in the call setup. configuration is that when the first GK gets an admission request, it sends a locationlocation confirm (LCF) from the remote GK before sending an ACF to the originatin

To increase the availability of H.323 networks, you can configure multiple GKs/GWHigh-availability technologies such as Hot Standby Router Protocol (HSRP) can alsoH.323 network.

These reference sheets cover H.323 GW and GK configuration in Chapter 8, “UnderGatekeepers.” However, for now, you can verify and troubleshoot an H.323 configuras follows:

n show gatekeeper calls: Displays current phone calls that the GK participated in

n show call active voice [brief ]: Displays details for current voice calls

n show call history voice [last n | record | brief ]: Displays call record logs

SIP Theory and ConfigurationSIP uses the concept of inviting participants into sessions, and those sessions can be Announcement Protocol (SAP). Like H.323, SIP is a peer-to-peer protocol. These pe

two types of UAs are as follows:


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CHAPTER 5


n Redirect server: Informs the UA of the next server to contact

n Location server: Performs address resolution for SIP proxy and redirect server

SIP uses clear text for sending messages. The two types of SIP messages are as follo

n Request: A message from a client to a server

n Response: A message from a server to a client

A request includes messages such as an INVITE (which requests a participant to joindisconnects the current call). Conversely, a response message uses HTML status mesbly attempted to connect to a website and received a “404 error” or a “500 error.” Thin the SIP environment.

For a SIP client to get the IP address of a SIP server, it has to resolve the SIP server’ally URLs that begin with sip: rather than http:, which is commonly used in web bro

variety of information, such as username, password, hostname, IP address, and phona SIP address:

sip:[email protected];user=phone

In this example, the user=phone argument specifies that the user portion of the URLnumber and not a user ID.

SIP devices can dynamically make their addresses known by registering with a SIP r


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CHAPTER 5


The basic call setup begins when a SIP client sends an INVITE to a SIP server (notin

either a client or server, depending on whether it is originating or terminating the calUAS) responds if it is willing to join the session to which it has been invited. The oran acknowledgment (that is, an ACK message) to the destination server, and at this pdirectly between the SIP GWs (or SIP IP phones).

If you introduce a SIP proxy server into your topology, the call setup procedure is simthe INVITE is sent to the proxy server rather than to the destination UAS. The proxy

to learn the IP address of the final endpoint. The destination exchanges call paramete


FIGURE 5-3

SIP Call Setup

INVITE

TRYING

RINGING

OK

ACK

RTP Streams

RTCP Stream

SIP GW SIP GWSIP UAC SIP UAS

V V

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CHAPTER 5


you receive a message saying that the page you are looking for has moved to a new Uredirected to the new URL. When the UAC learns the location of the destination UAbetween the UAC and UAS. Therefore, the main purpose of a redirect server is to offfrom the UAC.

If one of your SIP servers goes down, the voice network could be rendered unavailabis to have multiple instances of proxy and redirect servers. Therefore, the UAs can h

server fails, the second server takes over.Use the following two basic steps to configure SIP on a Cisco router:

Step 1. Enable the UA.

Step 2. Configure dial peers.

Consider Example 5-1.

Example 5-1 Basic SIP Configuration Example

Router(config)#sip-ua

Router(config-sip-ua)#sip-server dns:SERVER1

Router(config-sip-ua)#dial-peer voice 1 voip

Router(config-dial-peer)#destination-pattern 5...

Router(config-dial-peer)#session protocol sipv2

Router(config-dial-peer)#session target sip-server


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CHAPTER 5


In addition to the show call commands that you used for H.323 verification, you cancommand to display three different sets of statistics (that is, SIP response statistics, Sstatistics).

MGCP Theory and ConfigurationAlthough H.323 and SIP are peer-to-peer protocols, MGCP is more of a client/serverSpecifically, MGCP allows GWs to point to a centralized call agent for processing. Iized call agent is the Cisco Unified Communications Manager (UCM) server, as illus


FIGURE 5-4

MGCP Call Setup

RTP Streams

MGCP GW

Call Agent(Cisco UCM Server)

CreateConnection

CreateConnection

MGCP GWx1111 x2222

V V

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CHAPTER 5


The physical pieces that make up an MGCP network, such as call agents, GWs, and However, the logical pieces of an MGCP network, such as calls and connections, arefollowing MGCP components:

n Endpoints: An endpoint is where you interface between the VoIP network and texample, an FXS port that connects to a telephone is considered an endpoint. Eemail address (for example, [email protected]). These names

locally significant name of the endpoint (before the @ sign) and the DNS namesign).

n Gateways: GWs are in charge of converting audio between a VoIP network andexample a residential GW supports devices that you typically find in residentialtelephone service [POTS] telephones).

n Call agents: A call agent is the intelligence of an MGCP network and controls

MGCP GW can report events to the call agent, and the call agent can, for examsignaling to send to the phone.

Recall that an MGCP concept is a logical piece of an MGCP network. Consider the f

n Call: A call is formed when two or more endpoints are interconnected.

n Event: An event is what an endpoint has been instructed (by the call agent) to w

might notice the event of an attached POTS device going off-hook.





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CHAPTER 6

Working with Dial Plans

Chapter 6Working with Dial PlansA dial plan determines how calls are routed through a VoIP network. In this chapter,contains and how to create a dial plan that points outward to the public switched tele

multiple sites can add complexity to a dial plan (for example, because of overlappingcovers potential solutions for such design challenges.

Dial-Plan CharacteristicsDial plans typically organize a group of phone numbers in a hierarchical fashion. Co

plan, which consists of 10 digits, an example of which follows:

859-555-1212

The first three digits (that is, 859) indicate an area code, which is typically associatedNorth America. The following three digits (that is, 555) are the central office (CO) cidentifies a central office location within the area that is specified by the area code. T

point the local CO to a specific local loop that goes out to a subscriber’s physical loc


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CHAPTER 6


Dial-Plan ElementsA dial plan contains elements that perform the following functions:

n Assigning endpoint addresses: Endpoint addressing determines the format of p

n Selecting a path: You might, for example, configure a dial plan to place calls obut calls might be routed over the PSTN as a backup.

n Manipulating digits: Dialed digits and the digits making up a caller ID string mcalling between phones. For example, digits such as area code and office code dcalling out to the PSTN, or one dial string (for example, a 0 to reach a companywith another dial string (for example, the actual directory number of the operato

n Applying call restrictions: Call restrictions can be configured to control whichcall. For example, you might not want a lobby phone calling an international nu

n Supporting call coverage: The call coverage feature allows a group of phones,handle incoming calls (for example, calls coming into a call center).

Assigning Endpoint Addresses

Endpoint addressing assigns directory numbers (DNs) to devices, such as phones. Alnal extensions to incoming direct inward dial (DID) numbers. However, if you do no

internal DN, an auto attendant can be used to take an incoming call and route that ca


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CHAPTER 6


To allow callers in Kentucky and Arizona to have their calls extended to the approprirequire the use of site codes, where the dialed number is prepended with a site code example, if a caller at DN 1501 in Kentucky wants to call DN 1502 in Arizona, the culation must then be performed to strip the 820 site code from the dialed number. Althe caller ID number to prepend the Kentucky site code of 810 to 1501. As a result, t

in Arizona, would look at his phone and see 8101501 displayed as the caller ID. Thist b h h ld di l t ll b k th lli t


FIGURE 6-1

Overlapping DNs

KY_Router

V

AZ_Router

V

IP WAN

PSTN

1500

Ariz

U

Site C

1500 1501

Kentucky

UCM

Site Code: 810

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An IOS router acting as an H.323 gateway makes these call routing decisions based larger deployments, the number of dial peers can be large. Recall that multiple dial psame destination, and the preference command can be issued in dial-peer configuratpeer is used. The preferred dial peer could therefore point across an IP WAN, while point outward to the PSTN.

Manipulating Digits

Dial plans also need to accommodate for digit manipulation. For example, when a cato match a destination pattern that the router knows how to reach. Also, you might wnumber to appear as the number that the called party would have to dial to call back

For outbound calls, you need to present a valid dial string to, perhaps, the PSTN. In 911 calls, which might require Centralized Automatic Message Accounting (CAMA)preserve caller ID information being sent out on an analog trunk. For example, suppomiles and has several buildings. If a caller in Building Z calls 911, but the analog trulocated in Building A, the location information sent out to the 911 public-safety answlocation of Building A rather than Building Z. CAMA can help solve this issue by traan analog trunk to the PSAP.

Applying Call Restrictions

You probably do not want your users calling any destination they choose, such as int


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CHAPTER 6


Supporting Call CoverageThe call coverage component of a dial plan helps minimize the number of dropped indesk, for example, you might have your phone forwarded to another phone.

In a call center environment, the goal is to intelligently distribute incoming calls acroThe phones of these customer service agents can belong to a hunt group. However, ccustomer service agents. Instead, they dial a hunt pilot number , which distributes inc

members.

PSTN Dial PlansConfiguring a dial plan to point outward to the PSTN can be a complex task. Often,by Cisco UCM, Cisco UCM Express, or an IOS-based voice gateway. This section d

considerations, and it examines syntax used to configure PSTN dial plans.

PSTN Dial-Plan Design ConsiderationsCall routing and path selection should be set up in both the incoming and outgoing dibetween the PSTN and the internal VoIP network transparent to the end users, and to

mation to both the called and calling parties, digit manipulation might be required.


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CHAPTER 6


1. DN 1500 in the Kentucky office dials 94805551345 to reach a phone on the PSTUCM server that this call should be routed out the PSTN gateway. At this pointIdentification (ANI; that is, caller ID information) is 1500, and the Dialed Num

is, the dialed number) is 94805551345.


FIGURE 6-2

Outbound PSTN Call

Example

KY_Router 480-555-1345

V

1500 1501

DN 1500 Dials94805551345.

Kentucky

UCM

DID Range: 8595551XXX

1

Destinationphone rings.4

UCM matches the dialstring and strips off the 9.ANI: 1500DNIS: 4805551345

2

KY_Router prependsthe local area code andoffice code to the ANI.

ANI: 8595551500DNIS: 4805551345

3

PSTN

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4. The destination phone (that is, 4805551345) rings, and the caller ID that appear8595551500.

ISDN Dial-Plan ConsiderationsWhen using Integrated Services Digital Network (ISDN) trunks, you need to be awaconsiderations:

n An ISDN network might represent the ANI number as the shortest dialable num(TON) information. As a result, if you just add on the PSTN access code to thisvalid number that can be called back. You can, however, use digit manipulation

n An ISDN network, or PBX, might require that when a call setup message is sention and TON information needs to be included. Digit-manipulation commands type of information to the ISDN network or PBX.

Configuring PSTN Dial PlansFigure 6-3 shows syntax that might be used when configuring a PSTN dial plan.

This PSTN dial plan has two primary requirements:

n Prepend outgoing ANI (that is, caller ID) information with the area code and of


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CHAPTER 6


Prepending Digits to the Outgoing ANI

Example 6-1 shows the syntax that satisfies the first requirement, prepending outgoincode and office code of the Kentucky office (that is, 859555).

Example 6-1 Prepending Area Code and Office Code Information to Outgo

KY Router(config)#voice translation rule 1


FIGURE 6-3

PSTN Dial-Plan

Configuration

Example

KY_Router

T1 PRI1/0/0

Translate DNIS to4 Digits

480-555-1345

V

1500 1501

Kentucky

UCM

DID Range: 8595551XXX

PSTN

Prepend ANI with859555

NOTE

The next chapter digs

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In Example 6-1, voice translation rule 1 matches a dial string beginning with a 1 andthis rule would replace the number of 1500 with 8595551500.

The voice translation rule is then applied to a voice translation profile named ANI-Othe translate calling 1 command. The calling parameter means that the translation wnumber, which is the ANI number.

Finally, the voice translation profile of ANI-OUT is applied in the outgoing direction

of an ISDN PRI circuit built on T1 voice port 1/0/0 is referenced as voice port 1/0/0:

Stripping Area Code and Site Code Information from the Inco

Example 6-2 illustrates the commands used to remove the area code and site code incoming in from the PSTN.

Example 6-2 Digit Stripping from the Incoming DNIS

KY_Router(config)#voice translation-rule 2

KY_Router(cfg-translation-rule)#rule 1 /^8595551/ /1/

KY_Router(cfg-translation-rule)#exit

KY_Router(config)#voice translation-profile DNIS-IN

KY_Router(cfg-translation-profile)#translate called 2

KY_Router(cfg-translation-profile)#exit

KY Router(cfg-translation-rule)#voice-port 1/0/0:23


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Voice translation rule 2 is then applied to the voice translation profile DNIS-IN. Notelated called 2 command, implying the translation rule is applied to dialed digits (thavoice translation profile is applied in the incoming direction to the D channel of the T

Monitoring and Troubleshooting PSTN Dial Plans

Cisco offers a collection of show and debug commands for monitoring and troubleshshown in Table 6-1.

Table 6-1 PSTN Dial-Plan Monitoring and Troubleshooting Commands

Command Description

show dial-peer voice tag Displays detailed configuration information about the sp

show dial-peer voice summary Displays summary information (for example, type of diasession target) for all dial peers configured on a router

show dialplan number dial-string Shows which dial peer would be used by the router if a dial string

debug isdn q931 Displays real-time ISDN Layer 3 signaling messages, w

debug voip dialpeer Displays real-time dial-peer matching

debug voice translation Displays real-time operation of voice translation rules


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Introduction to Numbering PlansJust as an IP network benefits from a hierarchical IP addressing scheme, a VoIP netwnumbering plan. In additional to being hierarchical, a good numbering plan should b

Types of Numbering Plans

Numbering plans can be categorized into one of two broad numbering-plan types: prnumbering plans.

Private Numbering Plans

A private numbering plan is used within an organization, and so these plans do not hdards. Following are a few design considerations for private numbering plans:

n Number of endpoints: The numbering plan should be able to address all existingrowth, while using the fewest number of digits as possible.

n Number of sites: The length of site codes should be minimized, while still accosites, plus any anticipated growth.

n Direct inward dial: If a block of numbers is purchased from the local telephonblock of numbers might be able to map directly to internal directory numbers. Tdirect inward dial (DID) If the block of numbers purchased is not large enough


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PSTN Numbering PlansPSTN numbering plans can vary widely based on the country being supported. Hownumbering plan, the international E.164 numbering plan must be considered in additThe following contrasts these two different categories of PSTN numbering plans:

n E.164: E.164 is an ITU standard for an international numbering plan. In this placountry code (CC), which can be either one, two, or three digits in length. The c

national destination code (NDC), which is then followed by the subscriber numdigits in an E.164 number is 15.

n National numbering plan: A country can define its own national numbering puses a 10-digit numbering plan, where the first three digits represent the area conumbering plan area (NPA) code. The next three digits are the office code, also last four numbers are the subscriber numbers.

Following are common elements found in national numbering plans:

n Numbers for emergency services

n Directory assistance services

n Free calls within a geographic region

n Billing calls to mobile phones (Note that the United States is an exception, and

S f l di ll i h i h


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CHAPTER 6


Integrating Private Numbering Plans with PSTN NuFollowing are design considerations for integrating a company’s private numbering p

n Variable-length dial strings: When calling PSTN numbers from within an orgastrings may vary based on the destination. For example, a local call might not rdistance call does. Therefore, digit manipulation might be required to support Pnumbering plan complexity as transparent as possible to the end users.

n Centrex support: For some smaller office environments, the local telephony cofeatures for the office location, without requiring the office to have a PBX or ketypically have DNs that are four or five digits long. This can add significant numcomes into a VoIP network from the PSTN and that call is routed out to the Cen

n Support for voice mail: Some voice mail systems use a different numbering plsupport. As a result, digit manipulation might be required to support transfers to

n PSTN fallback: If a call cannot be placed over the IP WAN, perhaps because othe IP WAN being unavailable, that call might be able to use the PSTN as a bacdialed digits (for example, a DN or a DN prepended by a site code) are typicallthrough the PSTN. Therefore, digit manipulation needs to be performed to preparea code, office code, or an access code.

n International calls: Because country codes vary in length, numbering plans muinternational destinations that need to be called


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CHAPTER 6


n Number normalization: In a large VoIP network, such as a VoIP network suppsites in the overall VoIP network might use different DN lengths. To support calover the VoIP network, number normalization can be used. The idea of number between sites, the calling and called numbers are modified to a standardized forEquipment at the individual sites can then perform digit manipulation to create strings for use within those individual sites.

Understanding 911 ServicesAlthough understanding the implementation of 911 emergency calling is beyond the should understand some of the basic components that allow callers to have their locacated to a 911 operator. Following is a listing of some of these basic components:

n Automatic Number Identification (ANI): The ANI is the phone number of the

n Automatic Location Identification (ALI): The ALI is a database that associatelocation. Updates to this database can take about 48 hours. Therefore, updating ronment is not an effective solution.

n Public-safety answering point (PSAP): The PSAP is where a 911 call is termi

n Emergency-response location (ERL): The ERL is used in mobile environment

tion from which an emergency call was placed (for example, a specific floor in


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CHAPTER 6


n Selective router: A selective router is a telephone switch that routes 911 calls t

call’s ANI information.

n Centralized Automated Message Accounting (CAMA): A CAMA trunk is ancustomer’s phone switch directly to a selective router. A CAMA trunk carries onanalog trunk, if CAMA were not used, location information visible to the PSAP


NOTE

Network designersshould be aware of and

follow local, municipal,

state, and federal laws

regarding 911 service.

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CHAPTER 7

Implementing Advanced Dial-Plan Features

Chapter 7Implementing Advanced Dial-Plan FeatuA Cisco Unified Communications Manager (UCM) server offers several features to mpathing, and limit which destinations various phones can call. However, IOS voice g

these types of features. This chapter, therefore, explores the theory and configuration

Digit ManipulationAs evidenced in the preceding chapter, digit manipulation is often required in voice gID and dialed number information. Digit manipulation encompasses several features

number, removing digits from a number, and translating one number to another numthings as caller ID or dialed number information.

Using the digit-strip CommandWhen using a plain old telephone service (POTS) dial peer, by default only digits madestination-pattern command are forwarded out of the POTS port. For example, cois calling a number that that router should forward to an attached PBX. The phone se


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CHAPTER 7


FIGURE 7-1

The digit-strip Command

The digit-strip command can be used to override default digit stripping behavior. Instrip could have been issued in dial-peer configuration mode to prevent any digit stri

Using the forward-digits CommandAnother approach to overriding the previously described digit-stripping behavior is tSpecifically, the forward-digits command can be issued in dial-peer configuration mnumber indicates the number of digits, beginning at the right of the dial string, that sport. In this example, the forward-digits 4 command could have been used to cause four of the dialed digits.

U i h fi C d


Router PBX

4123

Destination-Pattern 4…

123

V

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CHAPTER 7


However, the prefix command could also be used in the previous example. Because t

the dial peer stripped off the digit 4, the prefix 4 command could be used to replace

Using the num-exp CommandThe number expansion command (num-exp) can be used to replace one number withconsider a telecommuter working from home, as depicted in Figure 7-2. Even though

via the PSNT by dialing 555-1345, you might want this telecommuter’s phone to be like the other internal numbers. You could therefore use the num-exp command to re(DN) with the public switched telephone network (PSTN) number. For example, if thof 2020, the global configuration mode command of num-exp 2020 5551345 could of 2020 and replace it with a dialed number of 5551345. Be aware, however, you stilnewly created dial string of 5551345.


FIGURE 7-2

The num-exp

Command

555-1345

Telecommuter’sHome

PSTNV

CCVP CVOICE Q i k R f

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CHAPTER 7


The clid CommandIn Integrated Services Digital Network (ISDN) circuits, there is a calling party numbsent via the Q.931 protocol, that is used to send caller ID (or CLID) information. Thtwo different calling numbers. One is a user-provided, or unscreened , number. The onumber. If you want to manipulate this caller ID information, you can use the clid comode.

For example, the clid network-number number command enables you to set the netQ.931 information elements. You can use the clid second-number strip to remove thinformation element. Also, the clid restrict command sets the presentation bit in the display of CLID information.

You can also remove the numbers completely by using the clid strip command twicenumber but also the calling name. Specifically, you enter clid strip to remove the cal

strip name to remove the name.

Using Voice Translation Rules and Voice TranslatioThe most advanced of the IOS-based digit-manipulation approaches involves voice trprofiles. Specifically, a voice translation rule can define a set of rules (as many as 15

type, and number plan. These voice translation rules are associated with a voice trant th i t l ti l


CHAPTER 7 CCVP CVOICE Q i k R f

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CHAPTER 7


These profiles can be applied to, for example, voice ports or dial peers. Each of these

tion profiles applied, one for the inbound direction and one for the outbound directio

A translation rule uses regular expressions to perform digit matching. Table 7-1 provmore commonly used regular expressions.

Table 7-1 Commonly Used Regular Expressions

Regular Expression Description

^ Matches a number at the beginning of a string of numbers

/ Denotes the beginning and ending of matching and replacemen

\ Negates the special meaning of the next character, such that ththe literal character

. Matches one digit

* Matches the previous character 0 or more times

+ Matches the previous character 1 or more times

\(\) Groups elements of regular expressions into sets

To better understand the use of these regular expressions, consider the voice translati

Example 7-1 Voice Translation Rule Example


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In the matching string, the ˆ555 means that for the voice translation rule will match a555. Next, notice the \(…\), which matches any four digits. Also notice that the four

theses, and the parentheses are prepended with backslashes. The backslashes are useciated with the parentheses. The parentheses themselves are used to identify a set, wreplacement string. Because this is the first and only set in the matching pattern, the string to match this voice translation rule, it must begin with a 555 and have at least Also, those four extra digits can be referenced by the replacement pattern as set one.

The replacement string begins with 111, meaning that the first three digits of the tran

The replacement string ends with a \1, which refers to set one from the matching patstring were 5551345, the replacement string would be 1111345.

When the voice translation rule is created, it can be referenced by a voice translationing example, examine the syntax in Example 7-2.

Example 7-2 Voice Translation Profile Example

Router(config)#voice translation-profile OFFICE-CODE


FIGURE 7-3

Voice Translation RuleStructure: Matching

and Replacement

Strings

rule 1 /^555\(….\)/ /111\1/ Matching

StringReplacement

String

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The voice translation profile then needs to be applied to an entity such as a voice por

Example 7-3 shows the voice translation profile being applied to a voice port.

Example 7-3 Applying a Voice Translation Profile

Router(config-voiceport)#translation-profile incoming OFFICE-CODE

The result of the command shown in Example 7-3 is that dialed numbers coming int

matching string (that is, a string of numbers beginning with a 555 and followed by abe replaced with a seven-digit number beginning with 111, with the fourth through tthe matching string to the replacement string.

Influencing Path Selection

Various IOS configurations can influence the route selected for placing a phone call.discussed preference dial-peer configuration mode command can be used. In additioused to leverage the IP WAN to more cost-effectively place calls to the PSTN.

Influencing Path Selection with Dial-Peer ConfigurCommandsE li d b t i di l fi ti d d th t



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CHAPTER 7


Table 7-2 Dial-Peer Configuration Mode Commands for Call Routing

Regular Expression Description

destination-pattern number Used by outgoing dial-peer matching to match a dia

incoming called-number number Matches Dialed Number Identification Service (DNmatching

answer-address number Matches Automatic Number Identification (ANI) inf

direct-inward-dial Allows a router to take the DNIS digits coming in, fforward a call based on those digits, without presen

preference [0 – 10] Breaks a tie between equally matched dial peers, whpreferable

In addition, on some ISDN links, you might be able to benefit from the no dial-peer

command. This command disables the checking of the status of an outbound POTS d

process, which might otherwise disallow a dial peer whose status was down.

Influencing Path Selection with Tail-End Hop OffImagine a scenario where a company has two offices, one in New York and one in Doffice and want to call one of your company’s suppliers in Dallas. You will pay toll c

However, because your office in Dallas has a PSTN gateway that can place a local c



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CHAPTER 7


To configure TEHO, you must

Step 1. Create outbound VoIP dial peers to point across the IP WAN.

Step 2. Perform digit manipulation (for example, to prepend the digits necessary to

Step 3. Create a POTS dial peer at the tail-end site, which points out to the PSTN.

Restricting Calls with Class of RestrTo restrict calls on an IOS router, instead of using partitions and calling search spaceCommunications Manager [UCM]), you can use an approach called class of restricti

using CoR might necessitate the creation of additional dial peers, as compared to wh

Specifically, instead of having a single dial peer to point out to the PSTN, you mightout to the PSTN for emergency calls, another dial peer to point out to the PSTN for distance calls, and perhaps another one for international calls. By being this granularcreate different rules for different categories of PSTN destinations.

CoR TheoryCoR is configured by creating a series of CoRs. These CoRs are then assigned to a C

assigned to incoming/outgoing dial peers (or an ephone-dn in a Cisco UCM environm



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CHAPTER 7


If both the incoming and outgoing dial peers for a call have a CoR list assigned, for

CoR list must have a CoR (that is, a key) that matches each of the CoRs (that is, the either the incoming dial peer or the outgoing dial peer does not have a CoR list assig

Figure 7-4 shows an example.

1. A caller in a VoIP network picks up a phone and dials 5551345


FIGURE 7-4

CoR Sample

Topology: CallPermitted

V

PSTN

FXS1/0/0

Caller dials alocal number of5551345.

1Gateway matches anincoming POTS dialpeer that has theLOCAL-IN COR listassigned.

2Gateway matches anoutgoing POTS dialpeer that has theLOCAL-OUT COR listassigned.

3 4 The call ispermitted.

Incoming COR List – LOCAL-IN

- 911- INTERNAL

- LOCAL

Incoming COR List – LOCAL-OUT

- LOCAL

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4. The call is permitted.

Recall that for the call to be permitted, the incoming CoR list must contain CoRs for each of the CoRs in the outgoing

CoR list. In this example, the incoming CoR list contains the following CoRs: 911, INTERNAL, and LOCAL.

Metaphorically, recall that you can think of CoRs in an incoming CoR list as keys. Also in this example, the outgoingCoR list contains the following CoR: LOCAL. Therefore, in this example, because the incoming CoR list contains the

CoR of LOCAL (that is, the key of LOCAL), it matches the one and only CoR contained in the outgoing CoR list, which

is LOCAL (that is, the lock of LOCAL). Therefore, the call is permitted.

However, consider an example where a call would be denied (see Figure 7-5).

1. In this example, the caller dials a long-distance number.

2. The calls is placed from an analog phone. So, when the call comes into the router, the router matches an incoming

dial peer that points to the local FXS port of 1/0/0. That incoming dial peer has a CoR list of LOCAL-IN associated

with it.

© 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 130 for more details.

FIGURE 7-5

CoR Sample

Topology: Call Denied

V

PSTN

FXS1/0/0

Caller dials along distance

number of8595551345.

1Gateway matches anincoming POTS dial

peer that has theLOCAL-IN COR listassigned.

2Gateway matches anoutgoing POTS dial

peer that has the LD-OUT COR listassigned.

3 4The call isrejected.

Incoming COR List – LOCAL-IN

- 911

- INTERNAL

- LOCAL

Incoming COR List – LD-OUT

- LD

w w w . C a r e e r C e r t .i n f o


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CHAPTER 7


3. The dial string of 8595551345 matched an outbound POTS dial peer pointing o

the LD-OUT CoR list applied.

4. The call is denied.

In this example, the LD-OUT CoR list contains the CoR of LD. However, because thdoes not contain the CoR of LD, the call is rejected.

CoR ConfigurationThe first step when configuring CoR is to configure CoR lists, which is performed indemonstrated in Example 7-4. If you are familiar with creating partitions and callingment, this step is somewhat analogous to the creation of partitions. In this example, fNAL, LOCAL, and LD.

Example 7-4 Creating CoRs

Router(config)#dial-peer cor custom

Router(config-dp-cor)#name 911

Router(config-dp-cor)#name INTERNAL

Router(config-dp-cor)#name LOCAL

Router(config-dp-cor)#name LD

Next CoR lists are created A CoR list is a collection of CoRs and these CoR lists c



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CHAPTER 7


Example 7-5 Creating Incoming CoR Lists

Router(config)#dial-peer cor custom 911-IN

Router(config-dp-corlist)#member 911

Router(config-dp-corlist)#exit

Router(config)#dial-peer cor custom INTERNAL-IN


Router(config-dp-corlist)#member INTERNAL


Router(config)#dial-peer cor custom LOCAL-IN



Router(config-dp-corlist)#member LOCAL


Router(config)#dial-peer cor custom LD-IN

Router(config-dp-corlist)#member 911Router(config-dp-corlist)#member INTERNAL


Router(config-dp-corlist)#member LD

Example 7-6 shows the creation of the outgoing CoR lists used in this example. Memthat need to be unlocked by the keys in the incoming CoR list.



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CHAPTER 7


Example 7-6 Creating Outgoing CoR Lists

Router(config)#dial-peer cor custom 911-OUT



Router(config)#dial-peer cor custom INTERNAL-OUT



Router(config)#dial-peer cor custom LOCAL-OUT



Router(config)#dial-peer cor custom LD-OUT

Router(config-dp-corlist)#member LD

After the creation of the incoming and outgoing CoR lists, these lists need to be appl

tion of an incoming CoR list to an existing POTS dial peer.

Example 7-7 Applying an Incoming CoR List

Router(config)#dial-peer voice 100 pots

Router(config-dial-peer)#corlist incoming LOCAL-IN

Finally, outgoing CoR lists are applied to outgoing dial peers. To illustrate, Example

outgoing CoR list to an outgoing dial peer



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CHAPTER 7


Using CoR can benefit not only POTS and VoIP dial peers but also UCM Express (U

Both UCME and SRST configurations allow Cisco IP phones to register with a routefeature. CoR can therefore be used to enforce call restrictions on those Cisco IP phonoutgoing} list-name command can be used in ephone-dn configuration mode on a Uoutgoing} list-name tag starting number – ending-number command can be used in mode to apply a CoR list to a range of SRST directory numbers.



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CHAPTER 8

Understanding and Configuring Gatekeepers

Chapter 8Understanding and Configuring GatekeeAn H.323 gatekeeper offers multiple benefits for larger enterprise networks. For exambandwidth in the IP WAN does not become saturated with too many simultaneous vo

this and many other features of the gatekeeper, in addition to exploring the theory anconfiguration. You will also see how to scale your network with multiple gatekeepers

Gatekeeper Features and FunctionsAmong a gatekeeper’s many features, some are mandatory, and some are optional. T

two. In addition, gatekeepers use a series of registration, admission, and status (RAScommonly used RAS messages will be described. You will also see how directory gaand the use of technology prefixes. Finally, you will learn that you can offload someto an external server using the Gatekeeper Transaction Message Protocol (GKTMP)

Gatekeeper Functions

Even though a gatekeeper is not a required component in an H 323 network a gateke



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CHAPTER 8


Table 8-1 Gatekeeper Features

Feature Mandatory or Optional Description

Address translation Mandatory Address translation can translate a ding IP address to which a setup pa

Admission control Mandatory Admission control uses a series of to be placed.

Bandwidth control Mandatory Bandwidth control supports mid-ca

Zone management Mandatory Zone management can provide featadmission control, and bandwidth czone with which a terminal or gate

Call authorization Optional Call authorization can permit or depolicies (for example, time-of-day

Call management Optional Call management keeps status info

routing of additional calls based onBandwidth management Optional Bandwidth management is a subset

can permit or reject a call based on

Call control signaling Optional Even though H.225 and H.245 callexchanged directly between the H.3gatekeeper supports a configurationthe gatekeeper.

Q


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CHAPTER 8


Notice that each H.323 GW belongs to a zone, but the gatekeeper itself does not belogateways registers with the gatekeeper, they register as members of their zone. BecauZoneB register with the gatekeeper, both zones are considered to be local zones, from

If the topology contained an additional gatekeeper, as illustrated in Figure 8-2, and ZGK2, ZoneB would be considered to be a remote zone from the perspective of gatekbe considered to be a local zone from the perspective of gatekeeper GK2.

Q

FIGURE 8-1

GatekeeperLocal Zones GK1

V

GW1 GW2606-555-1111

ZoneA ZoneB

859-555-2222

V V

FIGURE 8-2

Gatekeeper

Remote Zones GK1

V

GK2

V

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Definition of a Zone Prefix

When a gatekeeper learns about the H.323 terminals and gateway that have registered as members of various zones, the

gatekeeper can direct calls destined for particular phone numbers to a terminal or gateway in a zone that can appropri-

ately route a call destined for that phone number (see Figure 8-3).

In Figure 8-3, GW1 registered with GK1, as a member of ZoneA. GW2 registered with GK1, as a member of ZoneB. The

gatekeeper GK1 might be configured with zone prefixes that say calls destined for area code 606 should be forwarded to

ZoneA, and calls destined for area code 859 should be forwarded to ZoneB. As a result, if 606-555-1111 calls 859-555-

2222, gateway GW1 will ask gatekeeper GK1 how to route the call. Gatekeeper GK1 will examine the dial string and

notice the destination area code of 859. The gatekeeper’s zone prefix configuration states that if a call is destined for area

code 859, it should be routed to ZoneB. Since gateway GW2 has registered with gatekeeper GK1 as a member of ZoneB,gatekeeper GK1 will tell GW1 that it should send a call setup message directly to GW2.

Definition of a Technology Prefix

You might have a family physician who handles most of your medical needs and regular checkups. On occasion, however,

you might need to see a specialist. For example, you might go to a dermatologist for your skin, a podiatrist for your feet,

© 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 130 for more details.

FIGURE 8-3

Gatekeeper

Zones Prefixes

Zone Prefix 606 Zone Prefix 859

GW1 GW2606-555-1111

ZoneA ZoneB

859-555-2222

GK1

V

V V

w w w . C a r e e r C e r t .i n f o


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CHAPTER 8


or an ophthalmologist for your eyes. Similarly, when a gateway registers with a gatek

specialist for a particular type of call.

For example, a gateway might register as a specialist for video calls, for modem callcall type. Alternatively, much like a family physician, a gateway can register as a def

technology gateway can handle calls that did not request a specialist.

The way a call requests a specialist is by specifying a technology prefix as part of th

receives a dial string that is prefixed with a technology prefix, the gatekeeper can fortered with that specific technology prefix.

Registration, Admission, and StatusGatekeepers communicate with H.323 endpoints and gateways, in addition to other g

Although several RAS messages exist, these reference sheets describe some of the m

Discovery RAS Messages

Before an H.323 terminal or gateway can register with a gatekeeper, it needs to discois preconfigured with the IP address of a gatekeeper. In that case, the gateway sends message to make sure the gatekeeper is up and responsive. This GRQ is sent using a

preconfigured IP address


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CHAPTER 8


Registration RAS Messages

When an endpoint has discovered a gatekeeper, the next step is for that endpoint to rSpecifically, the endpoint sends a registration request (RRQ) RAS message to the garegister. As part of this registration message, the endpoint tells the gatekeeper about message might include information such as the name of the endpoint, the IP addressnumbers reachable by that endpoint.

If the gatekeeper permits registration from that endpoint, the gatekeeper responds wimessage. If the gatekeeper rejects the registration request, however, the gatekeeper c(RRJ) RAS message.

After the endpoint registers with the gatekeeper, the endpoint continues to periodicalkeeper, and the endpoint expects an RCF in response. Although this RAS exchange cfunction between the endpoint and gatekeeper, repeatedly sending all the information

message is not very bandwidth efficient. Fortunately, as of H.323 Version 2, these enmessages, which are considerably smaller than the initial RRQ message.

Admission RAS Messages

After an endpoint discovers a gatekeeper and registers with that gatekeeper, when thacross the IP WAN, it can send an admission request (ARQ) RAS message to the gat

two things First the ARQ asks permission to place the call Second the ARQ asks h


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an ACF message back to the endpoint, not only does the ACF message grant permiss

the ACF message contains the IP address with which the endpoint should establish th

To review how an H.323 call is established using a gatekeeper, take a look at Figure Chapter 5, “Exploring Gateway Control Protocols.”

FIGURE 8-4

ARQ RAS Message

H.225 Call Setup(2)

(4)

H.245 Capabilities Exchange

RTP Streams

RTCP Stream

H.323 GW

H.323 GK

(1) Admission Request and

Confirmation (H.225 RAS)

(3) Admission Request and

Confirmation (H.225 RAS)

H.323 GWx1111 x2222

V V

V


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At this point, the remainder of the call setup is identical to a call setup that does not

tion of the H.245 exchange (and after phone number 2222 goes off-hook), the two gaReal-Time Transport Protocol (RTP) stream between themselves.

Location RAS Messages

If a gatekeeper receives an ARQ and the phone number in the ARQ is in a remote zowith another gatekeeper), the gatekeeper can send a location request (LRQ) RAS me

LRQ asks the other gatekeeper for the IP address of the H.323 endpoint to which calother gatekeeper responds with the requested information, it does so with a location other gatekeeper cannot provide the requested information, however, it might responmessage.

Directory GatekeepersLarge VoIP networks might contain several gatekeepers. Unfortunately, a large numbadd to the required configuration, because every gatekeeper needs to have knowledgeresults in a logical full mesh of connectivity between all the VoIP network’s gatekeep

Fortunately, Cisco offers a proprietary solution called a directory gatekeeper , to whicdirectory gatekeeper contains configuration information about all the zones in the Vo

keepers responsible for those zones This allows the directory gatekeeper to intelligen


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1. The gateway GW1 sends an ARQ message, on behalf of its attached phone, req2222 and requesting the IP address to which an H.225 call setup message shoul

2. Gatekeeper GK1 receives the ARQ message, but it does not have a zone prefix does, however, have a zone prefix configured for all unknown phone numbers. TGK1 to send and LRQ message to the directory gatekeeper DGK, in the ZoneD

3 The directory gatekeeper examines the LRQ message and determines a caller is

FIGURE 8-5

Directory Gatekeeper

GW1 GW2606-555-1111

ZoneA ZoneB

859-555-2222

V V

ZoneDGK

GK1V V

GK2

1

2 3

A R Q

L R Q

L R Q DGK

V

V V


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Advanced Call Routing with GKTMPIf you need more advanced call routing decisions, beyond the capability of basic gateto the Gatekeeper Transaction Message Protocol (GKTMP). Specifically, GKTMP almessage into text messages and send those text messages to an external server. The edecisions based on a variety of criteria, such as the time of day.

The GKTMP server to which you can offload call routing decisions is platform indep

run on Solaris, Linux, or Microsoft platforms.

Basic Gatekeeper ConfigurationTo better understand the syntax used to configure basic gateway and gatekeeper oper

FIGURE 8-6

Gateway andGatekeeper

Configuration

Sample Topology

GW1 GW2606-555-1111 859-555-2222

GK110.7.7.1

V

V V

S0/0 S0/0


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Gatekeeper GK1 ConfigurationExample 8-1 shows the configuration for gatekeeper GK1.

Example 8-1 Gatekeeper GK1 Configuration

GK1(config)#gatekeeper

GK1(config-gk)#zone local ZoneA ciscopress.com 10.7.7.1

GK1(config-gk)#zone local ZoneB ciscopress.com

GK1(config-gk)#zone prefix ZoneA 606.......

GK1(config-gk)#zone prefix ZoneB 859.......

GK1(config-gk)#gw-type-prefix 1#* default-technology

GK1(config-gk)#no shutdown

To configure the gatekeeper, you first enter gatekeeper configuration mode, with thespecify the zones known to this gatekeeper. In this simple topology, both zones regis

are considered to be local zones.

The zone local gatekeeper-name domain-name [ RAS-IP-Address] command is used tthough the syntax specifies a gatekeeper-name parameter, that parameter is actually endpoints from that zone register with the gatekeeper, they attempt to register with azone. This often leads to some confusion. Therefore, it is often easier to think of thezone local command as a zone-name parameter. The domain-name parameter must b

Domain Name System (DNS) services The RAS-IP-Address parameter refers to the

NOTE

If a zone has more than

one endpoint that regis-

ters with the gatekeeper,the gatekeeper can

specify the priority of

these endpoints using the

zone prefix zone-name

dial-string gw-priority

priority endpoint-name

command The validf i i l


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The gw-type-prefix 1#* default-technology command enables you to specify the te

route calls that do not specify a specific technology prefix. Finally, like an interface,istratively enabled or disabled. The no shutdown command enables you to administrservice.

Gateway GW1 Configuration

Example 8-2 Gateway GW1 Configuration

GW1(config)#interface serial 0/0

GW1(config-if)#h323-gateway voip interface

GW1(config-if)#h323-gateway voip id ZoneA ipaddr 10.7.7.1

GW1(config-if)#h323-gateway voip h323-id GW1

GW1(config-if)#h323-gaeway voip tech-prefix 1#

GW1(config-if)#exitGW1(config)#dial-peer voice 100 voip

GW1(config-dial-peer)#session target ras

GW1(config-dial-peer)#destination-pattern 859.......

GW1(config-dial-peer)#exit

GW1(config)#gateway

In this example GW1’s serial 0/0 interface is used to communicate with the gatekeei t f fi ti d th h323 t i i t f d i i d t


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issued to cause this gateway to register with a technology prefix of 1#. This particula

Example 8-1, will be considered by the gatekeeper as a default technology prefix.

For a dial peer to send an ARQ message to a gatekeeper, the dial peer’s session targerather than point to a specific IP address. This is accomplished with the session targ

ration mode.

Finally, for an IOS router to function as an H.323 gateway, the gateway process mus

process is started by issuing the gateway command in global configuration mode.

Gateway GW2 ConfigurationEven through the configuration of GW2 is very similar to that of GW1, Example 8-3the gateway-specific configuration commands for GW2.

Example 8-3 Gateway GW2 Configuration

GW2(config)#interface serial 0/0

GW2(config-if)#h323-gateway voip interface

GW2(config-if)#h323-gateway voip id ZoneB ipaddr 10.7.7.1

GW2(config-if)#h323-gateway voip h323-id GW2

GW2(config-if)#h323-gaeway voip tech-prefix 1#

GW2(config-if)#exit


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Directory Gatekeeper ConfigurationA directory gatekeeper can act as a repository for call routing information in a netwominimized configuration on nondirectory gatekeepers. Specifically, a gatekeeper servconfigured with just two zone prefix commands, one pointing to a local zone for locthe directory gatekeeper’s zone for all other numbers.

Directory Gatekeeper ConfigurationTo better understand the syntax used to configure a directory gatekeeper, take a look

FIGURE 8-7

Directory Gatekeeper

Configuration Sample

Topology

V V

ZoneDGK

GK110.7.7.1 V V

GK210.5.5.1

DGK10.3.3.1

V

V V


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Example 8-4 illustrates the configuration of the DGK directory gatekeeper.

Example 8-4 Directory Gatekeeper Configuration

DGK(config)#gatekeeper

DGK(config-gk)#zone local ZoneDGK ciscopress.com 10.3.3.1

DGK(config-gk)#zone remote ZoneA ciscopress.com 10.7.7.1

DGK(config-gk)#zone remote ZoneB ciscopress.com 10.5.5.1

DGK(config-gk)#zone prefix ZoneA 606.......

DGK(config-gk)#zone prefix ZoneB 859.......

DGK(config-gk)#lrq forward-queries

DGK(config-gk)#no shutdown

Notice the DGK directory gatekeeper is configured for three zones in this example. Itwo remote zones of ZoneA and ZoneB. Because no phone numbers are reachable in

tion has no zone prefix command referencing that zone. The most significant distincconfiguration and a nondirectory gatekeeper configuration, however, is the lrq forwa

instructs the directory gatekeeper to intelligently forward LRQ RAS messages to an aOnly directory gatekeepers use this command.

Configuring Gatekeeper GK1 to Point to the DirectoExample 8 5 shows the configuration of the GK1 gatekeeper which is acting as a re


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Example 8-5 Configuring GK1 as a Regional Gatekeeper



GK1(config-gk)#zone remote ZoneDGK ciscopress.com 10.3.3.1


GK1(config-gk)#zone prefix ZoneDGK *



The regional gatekeeper GK1 needs knowledge of only two zones, one local zone (thzone. Specifically, if GK1 receives an ARQ specifying a phone number in the 606 arZoneA zone. ARQ messages specifying any other phone number (that is, a phone nuwill cause GK1 to forward an LRQ message to the directory gatekeeper, which will appropriate gatekeeper.

Configuring Gatekeeper GK2 to Point to the DirectFor completion, Example 8-6 shows the configuration of the GK2 gatekeeper. This ction, sends an LRQ to a directory gatekeeper for all nonlocal phone calls.

Example 8-6 Configuring GK2 as a Regional Gatekeeper


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Using a Gatekeeper for Call AdmissiAn H.323 gatekeeper, in addition to its number-resolution function, can be used as aanism. A gatekeeper can set up bandwidth limits for calls between defined zones. Thcontrolling bandwidth between zones and the syntax used to add CAC functionality t

Limiting Bandwidth Between ZonesAn IOS gatekeeper can be used to perform CAC between a Unified CommunicationsExpress (UCME) router, or an H.323 gateway. The amount of bandwidth allowed becally configured, rather than dynamically learned.

A gatekeeper can keep track of bandwidth going into and coming out of a zone. Howally measure the bandwidth. Instead, the gatekeeper keeps track of bandwidth based

Because bandwidth is not actually being measured, if you are using a bandwidth-comcompressed RTP), the gatekeeper does not change its bandwidth calculation.

A gatekeeper makes its bandwidth calculation by taking the payload-only bandwidthcodec to be used) and doubles that number. Therefore, the gatekeeper’s bandwidth cacodecs are as follows:

Gatekeeper bandwidth = Payload only bandwidth requirement * 2


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Configuring Zone Bandwidth LimitationsThe command to limit bandwidth between zones is issued in gatekeeper configuratio

Router(config-gk)#bandwidth {interzone | total | session | {default | zo

Table 8-1 describes the options available in the bandwidth command.

Table 8-1 bandwidth Command Parameters

Parameter Description

interzone Refers to the amount of bandwidth from a zone to another zone

total Refers to the cumulative amount of bandwidth allowed in a zone

session Refers to the maximum amount of bandwidth that can be consumed by

default Refers to a default value used for all zoneszone Refers to a specific zone

zone-name The name of a zone for which you are assigning a bandwidth amount

bandwidth An amount of bandwidth, in kilobits per second (kb/s)

To illustrate the use of the bandwidth command, consider Figure 8-8 and Example 8

topology and the corresponding gatekeeper configuration


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Example 8-7 Configuring Bandwidth Limitations



GK1(config-gk)#zone local ZoneB ciscopress.com


GK1(config-gk)#zone prefix ZoneB 859.......


GK1(config-gk)#interzone default 64

GK1(config-gk)#interzone session default 16


In Example 8 7 the interzone default 64 command is used to set the maximum ban

FIGURE 8-8

BandwidthConfiguration

Sample Topology

GW1 GW2606-555-1111

ZoneA ZoneB

859-555-2222

GK110.7.7.1

V

V V

S0/0 S0/0


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Interconnecting VoIP Networks with

a Cisco Unified Border Element

Chapter 9Interconnecting VoIP Networks with a Ci

Border ElementTwo companies might use their own VoIP networks internally, and they might have fanother. Instead of sending these intercompany calls over the public switched telephoremain a VoIP call end to end. To join together two VoIP call legs, however, you needto-IP gateway is known as the Cisco Unified Border Element (UBE). This chapter diof Cisco UBE.

Cisco UBE TheoryA Cisco UBE is physically the same as a Cisco voice-enabled router. However, Ciscothe unique capability to interconnect two VoIP call legs, as opposed to the traditionalwith a plain old telephone service (POTS) call leg.

Cisco UBE Feat res


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Figure 9-1 depicts a Cisco UBE interconnecting two VoIP call legs.

Cisco UBE requires special IOS feature sets. Specifically, you need to use one of the

n INT VOICE/VIDEO, IPIPGW, TDMIP GW AES

n INT VOICE/VIDEO, IPIPGW, TDMIP GW

Cisco UBE in the EnterpriseCisco UBE can be used to interconnect different enterprises, but it can also be used wfor example, VoIP and video over IP users to increase interoperability. Cisco UBE ofbenefit an enterprise network, such as the capability to have an H.323 VoIP call leg cleg. In addition, because Cisco UBE terminates, and then reoriginates, a call, it provipurposes. Also, for security purposes, the Cisco UBE can hide the IP addresses of th

those addresses with the IP address of the Cisco UBE

FIGURE 9-1

Interconnecting Two

VoIP Call Legs with

a Cisco UBE VVoIP VoIP

VoIP Call Leg

Cisco UBE

VoIP Call Leg

V


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As an example of this interworking feature, consider SIP and H.323. H.323 offers a connection, called H.323 Fast Start . With H.323 Fast Start, instead of an exchange oexchange of H.245 messages, H.323 can completely set up a call in a single round tr Early Offer . Without a Cisco UBE gateway, one of the Fast Start mechanisms from oable to set up a call with one of the Slow Start mechanisms. However, with a Cisco Ubecome interoperable.

Media Flow in a Cisco UBE EnvironmentBecause a Cisco UBE gateway forwards signaling protocols, it can influence the setuuse one of two approaches:

n Media flow-through: Media flow-through is the default mode, where the voice

For security purposes, this approach might be desirable when connecting to an of this approach is that it can conceal the IP address of an endpoint, because thewith the IP address of a Cisco UBE gateway.

n Media flow-around: By having media flow through a gateway, the Cisco UBE head, which might limit scalability and might not be desirable. Therefore, for happroach might be the use of media flow-around, which leaves the IP addresses

Although the Cisco UBE is responsible for handling the signaling to set up a ca


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Having a Cisco UBE involved in codec negotiation is not a requirement. A transpareallow the H.245 codec negotiation to be conducted without any intervention by a Cis

RSVP-Based CAC in a Cisco UBE EnvironmentIf a network has two Cisco UBEs, rather than just one, you can use Resource Reserv

call admission control (CAC) mechanism between those two Cisco UBEs, thus preveHowever, be aware that if you are using RSVP, the Cisco UBE must be configured foMedia flow-around is not supported with RSVP-based CAC, because if the media is the treatment of the media cannot be guaranteed.

Figure 9-2 illustrates the call setup between two H.323 gateways when RSVP-based with two Cisco UBEs.

FIGURE 9-2RSVP-Based CAC

in a Cisco UBE

Environment

Cisco_UBE1H.323_GW1

H.225RSVP

RSVP

H 225

H.323_GW2Cisco_UBE2

V V

12

4

3

V V


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1. The H.323_GW1 gateway initiates a call by sending an H.225 message to Cisco

2. Before a call is set up, an RSVP reservation should be established. Therefore, Ction request to Cisco_UBE2.

3. Cisco_UBE2 responds to the RSVP reservation request, and the RSVP reservati

4. Cisco_UBE1 then sends an H.225 call setup message to Cisco_UBE2.

5. Cisco_UBE2 sends an H.225 call setup message to H.323_GW2.

6. H.323_GW2 and Cisco_UBE2 exchange H.245 messages.

7. Cisco_UBE2 and Cisco_UBE1 exchange H.245 messages.

8. Cisco_UBE1 and H.323_GW1 exchange H.245 messages.

9. With the call now set up end-to-end, the H.323 gateways can exchange RTP pac

Using a Gatekeeper in a Cisco UBE EnvironmentJust as a regular H.323 gateway can register with a gatekeeper, a Cisco UBE can regcan then use a Cisco UBE along with via-zones. Specifically, when a call is being rokeeper can be configured to route the call via the zone containing the Cisco UBE. Th

centralized Cisco UBE without having a Cisco UBE at every site


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Cisco UBE in the call setup path between H.323_GW1 and H.323_GW2.

Cisco UBE ConfigurationA Cisco UBE can be configured to support protocol interworking. Among its configuflow through or flow around the Cisco UBE, in addition to whether the Cisco UBE btion. This section shows the syntax used to configure these features and parameters.

In addition a Cisco UBE can be used in conjunction with a via zone gatekeeper The

FIGURE 9-3

Via-Zone Gatekeeper

Cisco_UBEH.323_GW1 H.323_GW2

GK1 GK2VIA-Zone_GK

Call 1 Call 2

Zone_606 Zone_859Zone_VIA

V V V

VV V


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Protocol InterworkingProtocol interworking supports the interconnection of two call legs. For example, an with another H.323 call leg, or with a SIP call leg. The voice service configuration m from-protocol to to-protocol can be used to configure protocol interworking.

Example 9-1 illustrates the configuration of H.323-to-H.323 interworking.

Example 9-1 Configuring H.323 Interworking

Cisco_UBE(config)#voice-service voip

Cisco_UBE(config-voice-service)#allow-connections h323 to h323

Example 9-2 shows how protocol interworking is configured for connecting H.323 a

Example 9-2 Configuring H.323 and SIP Interworking

Cisco_UBE(config)#voice-service voip

Cisco_UBE(config-voice-service)#allow-connections h323 to sip

Cisco_UBE(config-voice-service)#allow-connections sip to h323

Media Flow and Codec Transparency

NOTEThe allow-connections

command is unidirec-

tional. Therefore, to

allow protocol interwork-

ing between H.323 and

SIP, the allow-connec-

tions command must be

i d i f


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If you prefer to have the media not flow through the Cisco UBE (which can ease the UBE), however, you can issue the media flow-around command in dial-peer configu

Also, when the H.323 endpoints are performing their codec negotiation, if you do noinvolved in that negotiation, you can enter the codec transparent command in dial-pthe Cisco UBE will also have to agree to support the selected codec.

Configuring a Via-Zone Gatekeeper for Use with a Figure 9-4 illustrates the configuration of a Cisco UBE and a via-zone gatekeeper wi

NOTEMedia flow-through is

the default setting for a

dial peer. Therefore, the

media flow-through

command does not have

to be entered.

FIGURE 9-4

Via-Zone

Configuration:

Sample Topology

10.10.1.1VIA-Zone_GK

Zone_VIA

GK1 GK2

Call 1 Call 2

Zone_606 Zone_859

V V V

VV V


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Example 9-3 shows the configuration of the via-zone gatekeeper (named VIA-Zone_

Example 9-3 Configuring a Via-Zone Gatekeeper

VIA-Zone_GK(config)#gatekeeper

VIA-Zone_GK(config-gk)#zone local Zone_606 ciscopress.com 10.10.1.1 invia Z

VIA-Zone_GK(config-gk)#zone local Zone_859 ciscopress.com invia Zone_VIA ou

VIA-Zone_GK(config-gk)#zone local Zone_VIA ciscopress.com

VIA-Zone_GK(config-gk)#zone prefix Zone_606 606.......

VIA-Zone_GK(config-gk)#zone prefix Zone_859 859.......

VIA-Zone_GK(config-gk)#gw-type-prefix 1#* default-technology

VIA-Zone_GK(config-gk)#no shutdown

The primary difference between the configuration of a via-zone gatekeeper and a tradthat the zone local commands (for zones other than the via-zone) specify an intermeshould pass before reaching the destination zone. For example, the zone local Zone_

Zone_VIA outvia Zone_VIA command says that call setup traffic going into or out the Zone_VIA zone.


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Configuring a Cisco UBE for Use with a Via-Zone GExample 9-4 provides the gateway-specific syntax for the Cisco UBE gateway in thi

Example 9-4 Configuring a Cisco UBE Gateway

Cisco_UBE(config)#voice service voip

Cisco_UBE(config-voice-service)#allow-connections h323 to h323

Cisco_UBE(config-voice-service)#exit

Cisco_UBE(config)#interface fa 0/1

Cisco_UBE(config-if)#ip address 10.10.1.2

Cisco_UBE(config-if)#h323-gateway voip interface

Cisco_UBE(config-if)#h323-gateway voip id Zone_VIA ipaddr 10.10.1.1

Cisco_UBE(config-if)#h323-gateway voip h323-id Cisco_UBE

Cisco_UBE(config-if)#h323-gateway voip tech-prefix 1#

Cisco_UBE(config-if)#exit

Cisco_UBE(config)#dial-peer voice 606 voip

Cisco_UBE(config-dial-peer)#destination-pattern 606.......

Cisco_UBE(config-dial-peer)#session target ras

Cisco_UBE(config-dial-peer)#exit

Cisco_UBE(config)#dial-peer voice 859 voip

Cisco_UBE(config-dial-peer)#destination-pattern 859.......

Cisco UBE(config dial peer)#session target ras

CHAPTER 9

I i V IP N k i h


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The primary distinctions between a traditional gateway configuration and the Cisco UExample 9-4 are the following:

n The allow-connections command is issued in voice service configuration mode

n The Cisco UBE registers with the via-zone gatekeeper as a member of the via-z

n Phone numbers available in other zones are configured using VoIP dial peers, w

Verifying Via-Zone GatekeepersBecause a via-zone gatekeeper is an H.323 gatekeeper, traditional gatekeeper verificafor verifying a via-zone gatekeeper. For example, consider the following commands configuration and operation:

n show gatekeeper endpoints: Shows an H.323 endpoint’s IP address, H.323 IDcapacity, and the gatekeeper’s total number of endpoint registrations

n show gatekeeper calls: Provides information about the call legs currently joine

CCVP CVOICE Quick Reference,Second Edition

Wallace, Kevin

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