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VoIP ProtocolsWith VoIP signaling, end stations perform signaling and session establishment.To successfullyemulate voice services across an IP network, enhancements to the signaling stac

Signaling is used for signaling from the PBX to the router, signaling between routers, and naling from the router to the PBX Without these procedures, calls would not be possible.

sig-Signaling Between Routers and PBXs When signaling from PBX to router, lifting the handset produces an off-hook condition.Theconnection appears as a trunk line to the PBX, which signals the router to seize the trunk.ThePBX then forwards the dialed digits to the router in the same manner the digits would be for-warded to a telephone company switch or another PBX.The signaling interface may be any ofthe common signaling methods used to seize a trunk line, such as FXS, FXO, E&M, or T-1/E-1signaling

As you can see in Figure 9.25, the PBX seizes a trunk line to the router and forwards thedialed digits Within the router, the dial plan maps the dialed digits to an IP address and initiates aQ.931 call establishment request to the remote peer router that is indicated by an IP address(Figure 9.26).This control channel is used to set up the Real-time Transport Protocol (RTP)audio streams, and the RSVP protocol may be used to request a guaranteed QoS

When the remote router receives the Q.931 call request, it signals a line seizure to the PBX

After the PBX acknowledges this seizure, the router forwards the dialed digits to the PBX andsignals a call acknowledgment to the originating router Figure 9.27 shows this line seizure

Figure 9.25 PBX-to-Router Signaling

Internet

Trunk Signaling 555-1212

Figure 9.26 Router-to-Router Signaling

Internet

Q.931 H.323 Agent

H.323 Agent

VoIP Protocols

With VoIP signaling, end stations perform signaling and session establishment.To successfullyemulate voice services across an IP network, enhancements to the signaling stacks are required.For example, an H.323 agent is added to the router for standards-based support of the audio andsignaling streams.The Q.931 protocol is used for call establishment and teardown between H.323agents or end stations

RTCP provides reliable information transfer once the audio stream has been established Atransport protocol such as TCP carries the signaling channels between end stations RTP, whichuses UDP, transports the real-time audio stream RTP uses UDP since it has lower delay thanTCP and voice traffic tolerates low levels of loss and cannot effectively exploit retransmission.H.245 control signaling negotiates channel usage and capabilities H.245 provides for capabil-ities’ exchange between endpoints so that CODECs and other parameters related to the call areagreed upon by the endpoints It is within H.245 that the audio channel is negotiated.Table 9.7depicts the relationship between the ISO reference model and the protocols used in IP voice Wewill discuss several of these protocols in detail in the next few sections

Table 9.7 ISO Reference Model and H.323 Standards

Transport RTP,UDP

H.323 Standard and Protocol Stack

The H.320 series of standards was defined for ISDN videophones and videoconferencing tems H.323 is an ITU-T set of standards that defines the components, protocols, and procedures

sys-necessary to provide multimedia (audio, video, and data) communications over IP networks It is

probably the most important standard for packetized voice technology H.323 provides a method

to enable other H.32x-compliant products to communicate In addition to control and call setup

standards, H.323 encompasses protocols for audio, video, and data as follows:

Figure 9.27 Router-to-PBX Signaling

■ Audio The compression algorithms H.323 supports for audio are all proven ITU dards (G.711, G.723, and G.729) All H.323 terminals must have at least one audioCODEC support specified by G.711.

stan-■ Video Optional Any video-enabled H.323 terminal must support the ITU-T H.261

encoding and decoding recommendation (H.263 is optional.)

■ Data H.323 references the T.120 specifications for data conferencing, which addressespoint-to-point and multipoint data conferences It provides interoperability at the appli-cation, network, and transport levels

Figure 9.28 shows the roles and interoperability of the various H.323 protocols H.323 is asuite of protocols that provide end-to-end call functionality in a converged network.The H.323protocol relies heavily on the services provided by other protocols such as TCP, IP, and UDP aswell as RTP.The protocols that make up the H.323 protocol are Registration, Admission, andStatus (RAS), H.245, and H.225

H.225H.225 establishes and controls calls between two H.323 endpoints, functions that the ITUQ.931performs for ISDN Q.931 uses 22 messages, and 29 in case of Q.932 H.225 adopted asubset of Q.931 messages and parameters such as alerting, call processing, connect, setup, releasecomplete, status, status inquiry, and facility (Q.932)

H.245H.245 control signaling negotiates channel usage and capabilities H.245 exchanges end-to-endcontrol messages managing the operation of the H.323 endpoint Control messages carry infor-mation related to:

Figure 9.28 H.323 Protocol Interoperability

H.245 Control H.225 Call Control H.225 RAS Control

Audio Codec G.711 G.722 G.728 G.729 G.732 Real-time Transport Protocol

(RTP)

Video Codec H.261 H.263

User Data Applications T-120

Video I/O Equipment

Audio I/O Equipment

System Control/

User Interface

RTCP RAS Audio/Video Control

TCP/UDP IP LAN

■ Capabilities exchange

■ Opening and closing logical channels used to carry media streams

■ Flow control messages

■ General commandsAfter call setup, all communications are over logical channels H.245 defines procedures formapping logical channels Logical channel 0 is for H.245 control for the duration of the call whilemultiple logical channels of varying types, such as video, data, voice, are allowed for a single call.Real-Time Transport Protocol

RTP provides end-to-end network transport functions suitable for applications transmitting time data such as audio, video, or simulation data, over multicast or unicast network services It is

real-used to transport data via UDP RTP does not address resource reservation and does not guarantee QoS for real-time services It is augmented by a control protocol (RTCP) to allow monitoring of the

data delivery in a manner scalable to large multicast networks and to provide minimal control andidentification functionality RTP and RTCP are designed to be independent of the underlyingtransport and network layers.The protocol supports the use of RTP-level translators and mixers.RTCP provides a control transport for RTP by providing feedback on the quality of data dis-tribution and carries a transport-level identifier for an RTP source used by receivers to synchro-nize audio and video

Registration, Administration, and Status

RAS is a protocol used between endpoints (terminals and gateways) and gatekeepers to performregistration, admission control, bandwidth changes, and status and to disengage endpoints fromgatekeepers RAS uses UDP port 1719

A solid understanding of H.323 components, their functions, and their importance is

paramount All devices that fall within the H.323 protocol stack can be categorized as one of fourtypes of devices.These device types are terminals, gateways, gatekeepers, and multipoint controlunits (MCUs)

H.323 Terminals (Endpoints)

H.323 terminals provide the user interface for real-time, two-way multimedia communications.All endpoints must support voice communications and, optionally, video or data communications.For voice, the H.323 terminal is generally an IP telephone H.323 is deployed as software such asMicrosoft NetMeeting In order to qualify as an H.323 terminal, the device in question musthave the following three items:

■ A network interface

■ H.323 software

H.323 terminals must support audio (G.711 is mandatory, and G.723.1 and G.729 are mended for networks of low bandwidth) Video and data support is optional; H.261 is mandatorywhen video is supported H.245 and H.225 are required for control functions, and RTP isrequired for sequencing media packets.

recom-H.323 GatewaysGateways translate communications between H.323 and non-H.323 entities (for instance,between H.323 terminals and telephones on the circuit-switched network).They provide callcontrol functions such as address translation and bandwidth management H.323 gateways enableH.323 networks to communicate with other networks such as the PSTN or PBXs Gatewaysprovide translation and call control between dissimilar networks Encoding, protocol translation,and call control mappings occur in gateways between two endpoints Gateways provide manyfunctions, including:

■ Translating protocols Allows the PSTN and the H.323 network to set up and teardown calls

■ Converting information formats Enables different networks to freely exchangeinformation such as speech and video

■ Transferring information Transfers information between dissimilar networks

Gateway functionality is generally provided by a router, such as the 2600 or 3600 series, aCatalyst gateway module such as the 6000 T-1 gateway module, or dedicated gateway devicessuch as the VG200 and DT-24+

H.323 GatekeepersGatekeepers perform call control and policy administration for registered H.323 endpoints

Gatekeepers in H.323 networks are optional If present, it is mandatory that endpoints use their vices.The H.323 standards define several mandatory services that the gatekeeper must provide:

ser-■ Address translation Translate an alias address into a transport address, which is aPSTN-based phone calling a phone on an IP network (an E.164 number such as 555-555-2121 will be translated into an IP network address such as 192.168.12.78)

■ Admissions control Defines RAS messages to authorize network access Does notdefine the rules or policies used to authorize access to network resources.To do so, thegatekeeper can interface with an existing authorization mechanism

■ Bandwidth control and management Determines if there is no bandwidth available

or no additional bandwidth available for calls requesting increases Can instruct a call toreduce its bandwidth usage

■ Zone management An H.323 “zone” is the collection of all components—terminals,gateways, and MCUs—managed by a single gatekeeper Gatekeeper must providerequired functions (for example: address translation, admissions control, bandwidth con-trol) to devices within its zone

The gatekeeper can also perform optional functions such as:

■ Call authorization Authorize or reject a given call; the provider of the H.323 servicespecifies the reasons for authorization and rejection

■ Call control signaling Process all call signaling associated with the endpoints

regis-tered with it (gatekeeper routed call signaling) or allow the call signaling messages to pass

directly between the endpoints

■ Call management Provide intelligent call management.The call management may bebased on address translation functions providing call screening, call forwarding/redirec-tion, and call routing based on time of day, network congestion, or least-cost path

As with gateways, routers are typically incorporated to provide gatekeeper functionality.Multipoint Control Units

MCUs provide conference facilities for users who want to conference three or more endpointstogether MCUs do not provide a direct interconnection to the H.323 protocol stack Rather,they provide a method for H.323 to interconnect voice and videoconferencing

All terminals participating in the conference establish a connection with the MCU It ages conference resources and negotiations between endpoints to determine which audio orvideo CODEC to use.The MCU might or might not handle the media stream An MCU hastwo functional components:

man-■ Μultipoint controller (MC) Mandatory Controls where media streams go Has a

reconciliation capability (common mode) and may be located in the terminal, gateway,

or gatekeeper

■ Μultipoint processor (MP) Optional Mixes, switches, and processes media streams,

including some or all of the streams in the conference call (video, data, or audio)

H.323 Call Stages

The process of establishing and maintaining an H.323 call is a very complex one We will breakthis process down into a logical and hierarchical order to show what is occurring at each stageand the requirements and resources used for each stage

H.323 Discovery and Registration

The five stages of an H.323 call and details of each of these connections are listed

1 Discovery and registration

2 Call setup

3 Call-signaling flows

4 Media stream and media control flows

5 Call termination

A lot happens within each of these stages; from the time the call is requested to the time it isterminated.

Device Discovery and RegistrationThe gatekeeper initiates a “discovery” process to determine the gatekeeper with which the end-point must communicate, as shown in Figure 9.29.This discovery can be either a statically con-figured address or through multicast traffic Once this is determined, the endpoint or gatewayregisters with the discovered gatekeeper

Registration is used by the endpoints to identify a zone with which they can be associated (a

zone is a collection of H.323 components managed by a single gatekeeper) H.323 can then

inform the gatekeeper of the zones’ transport address and alias address

Figure 9.29 H.323 Gatekeeper Call Control/Signaling: Discovery and Registration

1 RRQ

2 RCF

My name or E.164 address.

In Figure 9.30:

1 Gateway X sends an ARQ message using H.225 RAS to the gatekeeper

2 Gatekeeper requests direct call signaling by sending an admission confirmation (ACF) toGateway X

3 H.323 call setup is initiated

Inter-zone Call Placement

The process of placing an inter-zone call is somewhat more complicated and resource–intensive,

as the network is larger and divided into multiple zones In Figure 9.31, Gatekeeper A controlsZone A, and Gatekeeper B controls Zone B Gateway X (or Terminal X) is registered withGatekeeper A, and Gateway Y is registered with Gatekeeper B

To place a call to Gateway Y terminal, Gateway X first sends an ARQ message to the keeper requesting permission to make the call Since Gateway Y is not registered with the gate-keeper in Zone A, we assume that the gateways (terminals) are already registered

gate-Figure 9.31 shows five distinct phases in an inter-zone call placement

1 ARQ Gateway X requests a connection to Gateway Y from its local gatekeeper.

2 Location request (LRQ) Local gatekeeper for Gateway X does not know the IP

address of Gateway Y and is requesting the address from Gateway Y’s local gatekeeper

Figure 9.30 H.323 Gatekeeper Call Control/Signaling: Call Placement (Intra-zone)

H.323 Gateway X Gatekeeper

1 ARQ

WAN

H.323 Gateway Y

3 H.323 Call Setup

Figure 9.31 H.323 Gatekeeper Call Control/Signaling: Call Placement (Inter-zone)

Zone A

H.323 Gateway X

Gatekeeper A

1 ARQ

H.323 Gateway Y

5 Call Setup Zone B

Gatekeeper B

2 LRQ

3 LCF

H.323 Gateway Z

3 Location confirm (LCF) Gateway Y’s local gatekeeper responds to Gateway X’s local

gatekeeper with the IP address of Gateway Y

4 ACF The local gatekeeper responds to Gateway X’s request and provides the remote IP

Figure 9.32 helps us conceptualize this process

The call setup is based on the ITU-Q.931 (H.225 is a subset of Q.931), which provides ameans to establish, maintain, and terminate network connections across an ISDN.This processcomprises six distinct phases, as shown in Figure 9.32

1 Gateway X sends an H.225 call-signaling setup message to Gateway Y to request a nection

con-2 Gateway Y sends an H.225 message back to Gateway X, advising that it may proceedwith the call

3 Gateway Y sends an RAS message (ARQ) on the RAS channel to the gatekeeper torequest permission to accept the call

4 Gatekeeper confirms that the call can be accepted by sending a message (ACF) back toGateway Y

5 Gateway Y sends an H.225 message to Gateway X, alerting that the connection has beenestablished

6 Gateway Y sends an H.225 message to Gateway X, confirming call connection to lish the call

estab-Figure 9.32 H.323 Call Setup

6 H.245 Connection Established Q.931

Messages MessagesQ.931 MessagesRAS

Logical Channel Setup

After call setup, all communications travel over logical channels.The H.245 manages these logicalchannels Multiple logical channels of varying types (video, audio, and data) are allowed for asingle call

The H.245 Logical Channel Signaling Entity (LCSE) opens a logical channel for each mediastream Channels may be unidirectional or bi-directional Figure 9.33 helps us visualize how theH.323 utilizes virtual channels

The H.245 control channel is established between Gateway X and Gateway Y Gateway Xuses H.245 to identify its capabilities via a Terminal Capability Set (TCS) message to Gateway Y.The media channel setup flow is as follows:

1 Gateway X exchanges its capabilities with Gateway Y by sending an H.245 TCS sage

mes-2 Gateway Y acknowledges Gateway X’s capabilities by sending an H.245 TCSAcknowledge message

3 Gateway Y exchanges its capabilities with Gateway X by sending an H.245 TCS sage

mes-4 Gateway X acknowledges Gateway Y’s capabilities by sending an H.245 TCSAcknowledge message

5 Gateway X opens a media channel with Gateway Y by sending an H.245 Open LogicalChannel (OLC) message and includes the transport address of the RTCP channel

6 Gateway Y acknowledges the establishment of the logical channel with Gateway X bysending an H.245 OLC Acknowledge message, including:

■ RTP transport addresses (used to send the RTP media stream) allocated by Gateway Y

■ RTCP address previously received from Gateway X

7 Gateway Y opens a media channel with Gateway X by sending an H.245 OLC messageand includes the transport address of the RTCP channel

8 Gateway X acknowledges the establishment of the logical channel with Gateway Y bysending an H.245 OLC Acknowledge message and includes:

Figure 9.33 Media Channel Setup

Data H.245 Audio Video

Logical Channels 0 4 2 8 H.323 Gateway

■ RTP transport addresses (used to send the RTP media stream) allocated by GatewayX

■ RTCP address previously received from Gateway YFigure 9.34 highlights this process:

Media Stream and Media Control Flows

RTP media streams are transported over UDP ports 16384 through 16384 + 4x (where x is the

number of voice ports on the gateways) For example, a Cisco 3620 router with four E&M portswould use UDP ports 16384–16400 for RTP flows

RTCP manages media streams in the H.323 call flow by supporting QoS feedback fromreceivers.The source may use this information to adapt encoding or buffering schemes RTCPuses a dedicated logical channel for each RTP media stream Figure 9.35 illustrates the steps inthis stage of the H.323 call flow

In the example shown in Figure 9.34, four actions are occurring:

1 Gateway X sends the RTP encapsulated media stream to Gateway Y

2 Gateway Y sends the RTP encapsulated media stream back to Gateway X

Figure 9.34 Media Channel Setup Call Flow

H.323 Gateway X

H.323 Gateway Y

1 2 3 4 5

H.245 Messages

H.245 Messages

6 7 8

Figure 9.35 Media Stream Communication

H.323 Gateway X

H.323 Gateway Y

1 2 3 4

3 Gateway X sends the RTCP messages to Gateway Y.

4 Gateway Y sends the RTCP messages back to Gateway X

Endpoints may seek changes in the amount of bandwidth initially requested and confirmed.The gatekeeper must be asked for bandwidth increases or decreases Endpoints must comply withgatekeeper responses and requests.The bandwidth change flow is diagramed in Figure 9.36.Thisprocess consists of six stages:

1 The initiating gateway sends a bandwidth request (BRQ) to the gatekeeper to requestthe desired bandwidth

2 The gatekeeper responds with a bandwidth confirmation (BCF) message for therequested bandwidth

3 A logical channel is established between the two gateways with the specified bandwidth

4 A BRQ is sent from the remote router to the gatekeeper to change the bandwidth ofthe connection

5 The gatekeeper responds to the gateway with a BCF to confirm the new bandwidth

6 The logical channel is re-established with the new bandwidth

Call Termination

Call termination stops the media streams and closes the logical channels, and may be requested by

any endpoint or gatekeeper It ends the H.245 session, releases H.225/Q.931 connections, andprovides disconnect confirmation to the gatekeeper via RAS Figure 9.37 shows call terminationflow and is described as follows:

Figure 9.36 Bandwidth Change Request.

H.323 Gateway X

H.323 Gateway Y

1 Gateway Y initiates call termination by sending an H.245 End Session Command (ESC)message to Gateway X.

2 Gateway X releases the call endpoint and confirms with an H.245 ESC message toGateway Y

3 Gateway Y completes the call release by sending an H.245 Release Complete message

Termination.”

Figure 9.37 Call Termination (H.245/H.225/Q.931/RAS)

H.323 Gateway X

H.323 Gateway Y

Gatekeeper 1 2 3

4 5 4

5

Session Initiation Protocol

SIP (RFC2543) is a simple signaling protocol for Internet conferencing and telephony Based onSimple Mail Transfer Protocol (SMTP) and HyperText Transfer Protocol (HTTP), SIP was devel-oped by the Internet Engineering Task Force’s (IETF’s) Multiparty Multimedia Session Control(MMUSIC) working group SIP specifies procedures for telephony and multimedia conferencingover the Internet SIP is an application-layer protocol independent of lower layer protocols (TCP,UDP, ATM, X.25)

SIP is based on a client/server architecture in which the client initiates the calls By forming to these existing text-based Internet standards (SMTP and HTTP), troubleshooting andnetwork debugging are facilitated.The protocol can be read without decoding the binary ASN.1payload required in non-text-based protocols, such as H.323 SIP is widely supported and is notdependent on a single vendor’s equipment or implementation

con-SIP is a newer protocol than H.323 and does not have maturity and industry support at thistime However, because of its simplicity, scalability, modularity, and ease with which it integrateswith other applications, this protocol is attractive for use in packetized voice architectures Some

of the key features that SIP offers are:

■ Address resolution, name mapping, and call redirection

■ Dynamic discovery of endpoint media capabilities using the Session DescriptionProtocol (SDP)

Figure 9.38 H.323 Endpoint-to-Endpoint Signaling

H.323 Gateway

Gatekeeper Setup Connect

Capabilities Exchange Open Logical Channel Open Logical Channel Acknowledgement

RTP Stream RTP Stream RTCP Stream

Signaling Plane

H.225 (TCP) Q.931 H.245 (TCP) Media (UDP)

■ Dynamic discovery of endpoint availability

■ Session origination and management between host and endpoints Session Initiation Protocol Components

The SIP system contains two components: user agents and network servers A user agent (UA) is

an endpoint, which makes and receives SIP calls

■ User agent client (UAC) Initiates SIP requests

■ User agent server (UAS) Receives the requests from the UAC and returns responsesfor the user

SIP clients can include:

■ IP telephones (UACs or UASs)

■ Gateways to provide conferencing and translationThere are three kinds of SIP servers:

■ Proxy server Determines the server to which the request should be forwarded

Request can actually transit many SIP servers to its destination Responses return inreverse order

■ Redirect server Notifies the calling party of the actual location of destination

■ Registrar server Provides registration services for UACs at their current locations

Often deployed with proxy and redirect servers

Figure 9.39 illustrates the interaction between SIP components

Figure 9.39 SIP Components

IP Network

Session Initiation Protocol Messages

SIP works on a simple premise of client/server operation Clients or endpoints are identified byunique addresses.These addresses come in a format very similar to that of an e-mail address:

user@domain.com.

■ SIP addresses are URLs: user@host

■ User: name, telephone (E-164 address), number

■ Host: domain, numeric network (IP) addressThe users or clients register with SIP servers to provide location contact information

SIP uses messages for call connection and control.There are two types of SIP messages:

requests and responses SIP messages are defined as follows:

■ Invite Used to invite a user to a call Header fields contain:

■ Addresses of the caller and the person being called

■ Subject of the call

■ Call priority

■ Call routing requests

■ Caller preferences for the user location

■ Desired features of the response

■ Bye Used to terminate a connection between two users

■ Register Conveys location information to a SIP server, allowing a user to tell the serverhow to map an incoming address into an outgoing address that will reach the user

■ ACK Confirms reliable message exchanges

■ Cancel Cancels impending requests

■ Options Solicits information about the capabilities of the end being called, such as thedifference between a plain old telephone handset and a fully-featured multimediaphone

Media Gateway Control Protocol

MGCP (RFC 2705) is a relatively new protocol and as such, it is not as widely deployed as itsH.323 and SIP predecessors MGCP offers many key benefits and is growing in popularity, espe-cially in Cisco CallManager deployments

MGCP is a merger of the Simple Gateway Control Protocol (SGCP) and the InternetProtocol Device Control (IPDC) SGCP calls for a simplified design and a centralized intelligentcall control IPDC was designed to provide a medium to bridge VoIP networks and traditionaltelephony networks MGCP is a media control protocol, suited for large-scale IP telephonydeployment, and supports VoIP only

MGCP incorporates media gateway controllers (MGCs) or call agents to perform all call

connec-tion and call control within an MGCP network.These MGCs signal to, and control, media ways (MGs) to connect and control VoIP calls All the information for making and completing aVoIP call is held in the MG

gate-MGs have very little intelligence and receive all their marching orders from the MGC; theycannot function without a controlling MGC In a Cisco CallManager deployment, the MGC isoften a CallManager server and the media gateway is a router used to connect to a dissimilar net-work Examples of gateway applications are:

■ Trunking gateways Interfaces between the telephone network and a VoIP network

Manage a large number of digital circuits

■ Residential gateways Provide a traditional analog (RJ11) interface to a VoIP work Examples of residential gateways include cable modem/cable set-top boxes, andbroadband wireless devices

net-■ Access gateways Provide a traditional analog (RJ11) or digital PBX interface to aVoIP network Examples of access gateways include small-scale VoIP gateways

■ Business gateways Provide a traditional digital PBX interface or an integrated “softPBX” interface to a VoIP network

■ Network access servers Can attach a modem to a telephone circuit and provide dataaccess to the Internet We expect that in the future, the same gateways will combineVoIP services and network access services

When a gateway device detects that an end-user phone connection goes off-hook, it isdirected by the MGC to provide a dial tone to the phone and receives the dialed digits and for-wards them to the MGC for call processing

MGCP Connections

In an MGCP connection, there are two basic types of logical devices: endpoints and connections

Endpoints are the physical, or logical interfaces that either initiate or terminate a VoIP connection.

Endpoints are most often analog or digital ports in routers acting as gateway devices or digitalinterfaces into a PBX system

Connections are temporary logical flows that are created to establish, maintain, and terminate a

VoIP call Once the call is complete, the connection is torn down and the resources that were

allocated for that connection can be reused to support another connection A one-to-one connection

is really a point-to-point connection; a single endpoint signals to another single endpoint for the

purposes of completing a single VoIP connection Multipoint calls are used for conferencing and

broadcast to multiple endpoints simultaneously

MGCs manage connections in an MGCP network using the Session Description Protocol

SDP uses ASCII commands over IP/UDP to perform all call management functions A series ofeight connection messages is used by the MGC in order to control endpoints

Skinny Station Protocol

Skinny Station Protocol SSP is a Cisco proprietary communications protocol based on the

industry standard Simple Gateway Control Protocol Skinny Station Protocol enables cation between first generation IP telephone handsets/Gateways and CallManager servers

communi-Products that support Skinny Station Protocol include the DT-24 and DE-30 gateways, theCatalyst 6000 8-Port T-1/E-1 voice service modules, as well as the Catalyst 6000 24 port FXSmodule Skinny Station Protocol relies on the CallManager server to relay configuration and con-trol information It is built on TCP/IP and utilizes TCP ports 2000-2002

Simplified Messaging Desk Interface

SMDI is a standard voice-mail protocol for integrating voice-mail systems with legacy PBX tems and/or other similar devices

sys-Cisco VoIP Hardware and Software

Up to this point, this chapter has been focused on the underlying technologies and concepts thatare integral to VoIP We will now turn our attention to Cisco-specific information Cisco offers avariety of hardware and software solutions for implementing VoIP Its routers and switches can beadapted to support voice communications, usually with the addition of voice modules and soft-ware in many cases

Voice Modules and Cards

Routers and switches use voice modules to transform and transport voice traffic across the IPnetwork.They use Voice Interface Cards (VICs) to provide connectivity to telephone equipment.Voice Network Modules (VNMs) and VICs are configured using Cisco IOS VoIP commands.Digital signal processors are used in various Cisco voice-enabled routers in order to convertanalog voice signals to digital for transmission across an IP network and to convert back to analogonce the packet has arrived at the destination router DSPs can be found as modules insertedonto the motherboard, as on the 1700 series routers, or as slots built onto a VNM that is placed

in the router For more information on DSPs, refer back to the section entitled “DSP

Provisioning”

Voice Network ModulesVNMs convert analog voice into a digital form for transmission over the IP network At least oneVNM is needed to enable the router to handle voice traffic VNMs come in several differentmodels for the 2600/3600 series routers Figure 9.40 shows several models of VNMs available forthe 26XX and 36XX routers.

Only VICs are supported in the carriers with a V in the name.The NM-1V is a one-slot

VNM.You can install one VIC in the NM-1V to gain up to two voice ports.The NM-1V/2Vdoes not support WAN interface cards (WICs).The NM-2V is a two-slot version of the VNM

You can install up to two VICs in the NM-2V, providing up to four voice ports.The NM-HDVhigh-density VNM.This network module consists of five slots, one for the voice WIC (VWIC)and four for the packet voice DSP modules (PVDM).You can install one VWIC in the NM-HDV, providing up to two voice ports.The VNMs are the housings for the actual voice interfacecards that provide the necessary functionality and connectivity to achieve voice communications

Voice Interface CardsVoice Interface Cards (VICs) are inserted in the VNM to provide the necessary interface andsupport for the desired type of voice configuration (FXS, FXO, or E&M) Figure 9.41 shows

Figure 9.40 Voice Network Modules

VOICE 1V

EN

VOICE 1V

EN

several VICs to give you an idea of what is available; this is not an exhaustive list, as Cisco tinues to expand in this area.

con-One thing we would caution you about is that physically and outwardly, there is no ence between the FXS and FXO connectors; it can be easy to plug a telephone into what youthink is an FXS port, but is actually an FXO port Ensure that you are using the proper port type

differ-by checking the color and labels before attempting to connect

■ VIC-2E/M The two-port E&M module VIC-2E/M connects an IP network directly

to a PBX system It can be configured for special settings associated with tie-line ports

on most PBXs E&M ports are color-coded brown

such as a telephone, keypad, or fax.These ports provide ringing voltage, dial tone, andother endpoint specific functionality FXS ports are color-coded gray

FXO ports are color-coded pink Other types of FXO cards for use outside NorthAmerica are capable of providing switching and signaling techniques used in other geo-graphic regions such as VIC-2FXO-EU for use in Europe

■ VWIC-2MFT-T-1 The two-port VWIC multiflex trunk interface card is a two-portcard that can be used for voice, data, and integrated voice/data applications.The multi-flex VWIC can support data-only applications as a WAN interface on the Cisco 1700,

2600, or 3600 It can also integrate voice and data with the Drop and Insert multiplexer

Figure 9.41 Voice Interface Cards

VIC-2E/M VIC-2FXO VIC-2FXS

VWIC-2MFT-T-1

Four-Port Analog DID/FXS VIC

functionality and/or configured to support packetized voice (VoIP) when in the digitalT-1/E-1 network module.

Cisco 1700, 2600, and Cisco 3600 series routers.These cards are available as ISDN BRIS/T or NT interfaces for terminating to an ISDN network

■ Four-Port Analog DID/FXS VICs Two direct inward dial interface cards are able One card is a two-port RJ-11 that supports DID only.These cards are used forproviding DID service to extensions on a PBX so that users may transparently dialdirectly to extensions

avail-Installing VNMs and VICs

The types of router chassis described in this section demonstrate how VNMs, VICs, and tional voice port adapters are installed on the various platforms We chose a sampling of Ciscorouters to structure our discussion, but the concepts and processes are similar for all Cisco routers,with a few minor adjustments

addi-E-1/T-1 Voice ConnectivityDigital E-1 and T-1 connectivity allows Cisco series routers and switches to provide E-1 or T-1voice connectivity to PBXs or to a CO.T-1 voice connections are available for various routersand switches, including, but not limited to Cisco 1700, 2600, 3600, 3700, MC3810, 7200, 7500,AS5300, AS5800, and Catalyst 4000 and 6000 series equipment

The 1700, 2600, 3600, 7200, and 7500 series routers are capable of VoFR and VoIP.TheMC3810 (now end of sale) supports VoFR, VoATM, and VoIP.The AS5300 is able to performVoIP, VoHDLC, or VoFR functions

The 7200, 7500 series, and AS5300 series are primarily used as tandem switch points from

T-1 tie lines to PBXs and the PSTN to the internal IP network An example of the use of tandemswitch points is receiving a voice call on one VoIP interface and switching it back out anotherVoIP interface to its final destination.The 1700, 2600, and 3600 routers series can perform thisfunction because support for voice T-1/E-1 interfaces with up to two T-1/E-1 circuits per cardhas been added.The T-1/E-1 enhanced voice port adapter is used in the 7200 and 7500 seriesrouters and can support up to two T-1s per card.The AS5300 series access switch uses the T-1carrier card that can support up to four T-1s

The 7200, 7500, and AS5850 can terminate T-1s for voice traffic into the WAN and forwardthe signals and transmissions to the 1700, 2600, 3600, and AS5300 series routers for completeprocessing.The 7200 Series offers a four- or six-slot configuration, with interfaces includingATM, Synchronous Optical Network Technologies (SONET), ISDN BRI, ISDN PRI,T-1, E-1,T3, and E3 Its multi-service interchange (MIX) allows the 7200 to support digital voice as well

as gateway functionality through the use of two different trunk interfaces, the high-capacity andmedium-capacity T-1/E-1 trunk interface cards.The primary difference between the two cards isthat the high-capacity card includes an on-board DSP card for compression.The 7200 Series cansupport up to 120 voice calls, depending on the module configuration used.This router also sup-ports analog voice applications through the use of voice interface cards (VICs)

1700 Series Router Configurations

The 1700 series modular access routers are designed for small- to medium-sized businesses.The

1700 family has several chassis for different applications, but the two that were designed cally for voice applications are the 1751 and the 1760.These two modular chassis use the CiscoIOS along with various VICs to support analog and digital voice traffic over the IP network.Cisco 1751 Modular Access Router

specifi-The Cisco 1751 is a standalone chassis that can support up to three voice interface slots It comes

in two models: a base model suited primarily for data, but with an easy upgrade path to voice,

and a multiservice model (identified with a V) that includes all features for immediate integration

of data and voice Both models include three slots for data/voice interface cards as well as a10/100 Ethernet port, a console port, and an auxiliary port.The 1700 series VICs are inter-changeable with the Cisco 2600 and 3600 series routers

The Cisco 1751 includes one PVDM-256K-4 (one DSP) that supports one analog VIC Iftwo analog VICs or one or more digital ISDN VICs are used, additional DSPs are required.TheCisco 1751 has two DSP slots to support additional voice channels A PVDM is required to sup-port VICs on the Cisco 1750 and 1760 routers.These two chassis require PVDMs to be placed

on the motherboard, unlike the Cisco 2600, 3600, and 3700 routers, which have DSP support onthe VNMs

The Cisco 1760 Modular Access Router

The Cisco 1760 has four slots for VICs, and is available in two models.The base model is suitedfor data networking, but can be upgraded to support voice.The multi-service model (identified

with a V) includes all features for immediate integration of data and voice Both models include

four slots for data/voice interface cards and a 10/100 Ethernet port

The Cisco 1760 includes one PVDM-256K-4 (one DSP) that supports one analog VIC Iftwo analog VICs or one or more digital ISDN VICs are used, additional DSPs are required.TheCisco 1760 has two DSP slots to support additional voice channels

3600 and 3700 Series Router Configurations

Cisco’s 3600 and 3700 series routers come in a variety of base configurations that differ in theamount and/or type of standard network interfaces (RJ-45 ports, serial ports, and ISDN ports)that are available.The Cisco3600 and 3700 are designed primarily for traditional and powerbranch office solutions

The 3600 series router comes in three varieties: the 3620, which has two network moduleslots; the 3640, which has four network module slots; and the 3660, which is equipped with sixnetwork module slots.The 3640 is end of sale as of this writing

The 3700 series router comes in two varieties: the 3725, which has three integrated WICslots and two network module slots, and the 3745, which has three integrated WIC slots and fournetwork module slots Currently, the built-in WICs do not support VICs

Most Cisco 1700 VICs can be used for the 3600 and 3700 series except that a VNM isrequired in these higher-end routers Installation of these VNMs is covered later in this chapter

For 3700 platforms, the minimum IOS release is IOS 12.2(8) T for all network modules and VICs.

7500 Series Router Configurations

The 7500 series high-end routers support voice, video, and data.The Cisco 7500 series includesthe Cisco 7505, the Cisco 7507, and the Cisco 7513 with 5, 7, and 13 slots, respectively Cisco

7500 adapters include the two-port T-1 and E-1 high-capacity enhanced digital voice portadapter, the two-port T-1 and E-1 moderate-capacity enhanced digital voice port adapter, and theone-port T-1 and E-1 enhanced digital voice port adapters

AS5350 and 5850 Universal Gateway Configuration

The Cisco AS5350 and AS5850 universal gateways provide from 2 to 96 T-1s or E-1s to supportdata, voice, wireless, and fax services on any port.The AS5350 is only one rack unit high andsupports 216 voice, dial, or universal ports.The AS5350 is mainly intended for ISPs and enter-prises, whereas the Cisco AS5850 was designed for large service providers.The AS5850 is 14 rackunits high and supports 2688 voice or universal ports Both chassis support hot-swappable cardsand fans to minimize service interruption.The AS5350 supports two-, four-, or eight-T-1/E-1configurations; the AS5850 supports up to four 24-port T-1 cards for a total of 96 T-1s

The Catalyst 4000 series with the Access Gateway Module (WS-X4604-GWY) can support

up to three VICs for T-1/E-1 voice connectivity.The Catalyst 6000 series utilizing the T-1Service Module (WS-X6608-T-1) can support up to eight T-1s for voice connectivity

Catalyst 3500 Series Switches

The 3500 series is a scalable, entry-level solution for small- to mid-sized networks It is a whollyCisco-developed switch, with a router-like IOS.The 3500 Series of switches are fixed configura-tion switches, and all offer 10/100 Ethernet ports and Gigabit Interface Converter (GBIC) ports.The 3550-24PWR is the only switch in the 3500 Series that supports inline power

Catalyst 4x00 Series Switches

The Cisco Catalyst 4000 series switches are modular chassis available in several configurations.They are designed primarily for enterprise offices, branch offices, and multi-site campuses Bothchassis can provide inline power for end-user Cisco IP phones and thus can also centralize powermanagement

The 4x00 Series is a step up from the 3500, offering a modular configuration in several ferent switches: the 4003, 4006, 4503, 4506, and 4507R.The 4x00 Series also offers supervisorengine functionality, similar to that of the 5500 Series Within the 4x00 Series, only the 4003does not offer inline power.The 4x00 Series also offers voice-gateway functionality through theuse of the Series 4x00 WS-X4604-GWY module, which provides support for both H.323 andSSP (in the future it will support MGCP)

dif-Catalyst 4000 Modules

The Catalyst 4000 currently offers a mid-range hardware-based conferencing and transcodingsolution.The Access Gateway Module (AGM) is equipped with slots for DSP modules, high-den-sity analog, Gigabit Ethernet, and three slots for VICs.The AGM provides the following services:

IP WAN routing, VoIP, and IP telephony It supports VICs and WICs from the

1600/1700/2600/3600 Series routers.The AGM supports the following ports and slots:

■ Two VIC/WIC slots (VWICs, VICs, and WICs)

■ One dedicated VIC slot (VWICs and VICs)

■ FlexSlot High Density Analog (8 port RJ-21 FXS module)

■ Four DSP SIMM slots

■ 64 or 128MB memory SIMM slot (Integrated Service Adapter)VoIP gateway mode requires DSP resources to convert voice calls into data packets.TheCisco IOS IP/DSP Plus feature set is another requirement to allow VoIP gateway capabilities.The AGM supports the following interfaces as a VoIP gateway:

■ T-1 and E-1 ISDN PRI

■ T-1 Channel Associated Signaling

■ Foreign Exchange Office

■ Foreign Exchange Station

■ E&MThe AGM can function as a DSP farm for Cisco CallManager, which can be configured toprovide hardware-based conferencing.The Catalyst 4000 gateway module has four DSP SIMMs,with each SIMM having six DSPs for a total of 24 DSP resources.Table 9.8 summarizes theCatalyst 4000 AGM capabilities

Table 9.8 Catalyst 4000 AGM DSP Resources.

PSTN gateway 96 channels of G.711 voice Conferencing 24 channels of G.711 conferencing (4 conferences x 6 through 8 x 3) MTP transcoding 16 channels of LBR to G.711

Figure 9.42 depicts how the DSP resources are provisioning within the Catalyst 4000 after it

is configured for gateway mode

The Catalyst 4000 DSPs can only support G.711 conferencing sessions.This is not to say thatthere can be a conference session on a Catalyst 4000 with only G.711 participants, but thattranscoding DSP resources must be involved to convert those participants to the G.711 CODECfor the conference.The Catalyst 4000 AGM module supports up to 16 transcoding sessions permodule via hardware and would handle up to 104 channels in software

Cisco Catalyst 4200

The Cisco Catalyst 4200 is a small branch office device that can provide voice gateway ties, IP routing, and Ethernet switching in a single chassis It comes equipped with a 24-port10/100 switch that can provide inline power to Cisco IP phones and a built-in eight-port FXSmodule for support of legacy analog telephony equipment It is capable of using the samevoice/WAN interface cards as the 1700, 2600, and 3600 families of routers

capabili-Catalyst 6500 Series SwitchesThe 6500 Series has highly scalable, enterprise-class switches.The 6500 Series offers a completelymodular design, utilizing supervisor modules, with the capability for redundant supervisor mod-ules, if necessary.There are five switches in the 6500 Series family: the 6503, 6506, 6509, 6506,

6509, and 6513; the last two digits indicate the number of slots on each chassis.The 6500 Seriesprovides inline power through its specialized 48-port switching modules Gateway functionality isprovided via the WS-X6658-x1 module, which supports SSP and MGCP.The 6500 Series alsooffers an eight-port voice T-1/E-1 and services module to provide connectivity to legacy PSTN

or PBX systems, as well as a 24-port FXS module for analog telephone connectivity

Figure 9.42 Catalyst 4000 Gateway Mode DSP Resources

SIMM DSP DSP DSP DSP DSP DSP

DSP DSP DSP DSP DSP DSP DSP DSP DSP DSP DSP DSP DSP DSP DSP

G.711 PSTN Gateway MTP Transcoding Conferencing

4 SIMMs

6 DSPs per SIMM. DSP

DSP DSP DSP

Catalyst 6000 Modules

The Cisco WS-6608-T-1/E-1 module for Catalyst 6000 offers similar functionality as the

Catalyst 4000 AGM, only targeted at CO or headquarters environments.This module providesDigital T-1 or E-1 PSTN and PBX gateway services, transcoding, and conference bridging witheight T-1 or E-1 ports to support common channel signaling or ISDN PRI signaling Each portcan be configured as a PSTN/PBX gateway, MTP transcoder, or a conference bridge.Table 9.9summarizes the DSP resource capabilities of the Catalyst 6000 Voice T-1/E-1 and Services

module

Table 9.9 Catalyst 6000 DSP Resources

■ 24 calls per DS1 port

■ 192 calls per module WS-6608-E-1:

■ 30 calls per DS1 port

■ 240 calls per module

■ 32 conferencing participants per physical port

■ Maximum conference size of six participants

■ 256 conference participants per module G.729:

■ 24 conferencing participants per physical port

■ Maximum conference size of six participants

■ 192 conference participants per module

■ 31 MTP transcoding sessions per physical port

■ 248 sessions per module G.729 to G.711

■ 24 MTP transcoding sessions per physical port

■ 192 sessions per moduleThe 8-port T-1/E-1 Voice and Services module performs 24 transcoding sessions per portwhen translating from G.729 to G.711 It does 31 sessions per port for G.723 to G.711.The 8-portmodule can handle 192 or 248 sessions per module, depending on the LBR CODEC utilized.TheCatalyst 6000 can perform a mix of transcoding and conferencing within the same DSP

Each port on WS-6608-T-1/E-1 is configured with an IP address making it a PSTN gateway,conferencing, or transcoding resource in Cisco CallManager A TFTP server must be configured

to download the configuration

Inline Power Options

Second-generation phones are superior to their first-generation counterparts because they offer

support for inline power First-generation telephones require an external power source Inline

power can be offered to second-generation telephones either via a powered patch panel orthrough inline power modules installed in the switch

LAN Queuing for Video/VoiceQueuing has traditionally been a Layer 3 function for WAN connections, but when discussing aconverged network, specifically that dealing with voice or video traffic, attention must also begiven to the LAN Layer 2 traffic can be classified by type of service using the 802.1Q protocol.

You should separate voice and video traffic from data traffic into a high-priority queue 802.1Qspecifies seven classes of service, with 0 being the lowest priority and 7 being the highest priority.CoS 4-7 should be used for voice and video, and that 0-3 for data traffic An important note tomake regarding Layer 2 queuing is that once the packet encounters a router, the Layer 2 informa-tion is lost—in other words, 802.1Q is only a LAN solution For traffic crossing WAN links,Layer 3 queuing must be incorporated

Quality of Service

Voice traffic and data traffic have different characteristics Unlike data traffic, voice traffic occurs

in real time and is delay-sensitive Voice packets tend to be smaller than data packets When voiceand data networks are merged, it is important to deliver an acceptable QoS for the voice traffic

Voice traffic must be prioritized to minimize delay and jitter Delay is the amount of time

between the original transmission of the voice information and the final processing by the

receiving station Jitter is the variation in the delay between successive voice packets Packet loss

due to network errors or congestion will impact jitter QoS depends on the ability to controlthese two factors that impact voice quality

QoS tools can be divided into three categories:

■ Classification Voice packets can be classified or marked with a specific priority toenhance QoS

■ Queuing Use separate queues for voice and date to ensure consistency and QoS forvoice

■ Provisioning Circuits carrying voice traffic should be provisioned with enough width or capacity to minimize delay and jitter

band-The increasing deployment of VoIP can be attributed to the improvements made in QoS

QoS is a set of ideas, procedures, practices, and numerous protocols that provide for reliable andefficient transportation across data networks

What Is Quality of Service?

QoS is simply a set of tools to ensure that a minimum level of service will be provided to certaintraffic Many protocols and applications are not critically sensitive to network congestion FileTransfer Protocol (FTP), for example, has a rather large tolerance for network delay or bandwidthlimitation

Applications such as voice and video are particularly sensitive to network delay If voice packetstake too long to reach their destination, the resulting speech sounds choppy or distorted QoS can

be used to assure services to these applications Critical business applications can also use QoS

Applications for Quality of Service

When would a network engineer consider designing QoS into a network? Here are a few sons to deploy QoS in a network topology:

rea-■ To prioritize certain mission-critical applications in the network

■ To maximize the use of the current network investment in infrastructure

■ To provide better performance for delay-sensitive applications such as voice and video

■ To respond to changes in network traffic flows

When deploying QoS, analyze the traffic flowing through the bottleneck, determine theimportance of each protocol and application, and determine a strategy to prioritize the access tothe bandwidth QoS allows control over bandwidth, latency, and jitter and minimizes packet losswithin the network by prioritizing Bandwidth is the measure of capacity on the network or aspecific link Latency is the delay of a packet traversing the network, and jitter is the change oflatency over a given period of time

Deploying certain types of QoS techniques can control these three parameters QoS is notwidely deployed within many networks With the push for applications such as multicast,

streaming multimedia, and VoIP, the need for QoS is more apparent, especially since these cations are susceptible to jitter and delay Poor performance is immediately noticed by the end-user However, QoS is not the magic solution to every congestion problem; it may very well bethat upgrading the bandwidth of a congested link is the proper solution to the problem

Best-effort service is when the network will make every possible attempt to deliver a packet to itsdestination With best-effort service, there are no guarantees that the packet will ever reach itsintended destination An application can send data in any amount, whenever it needs to, withoutrequesting permission or notifying the network

accomplish this task, the network uses a process called admission control.

Cisco IOS uses RSVP and intelligent queuing RSVP is currently in the process of beingstandardized by the IETF in one of its working groups Intelligent queuing includes technologiessuch as Weighted Fair Queuing and Weighted Random Early Detection (WRED).

RSVP works in conjunction with the routing protocols to determine the best path throughthe network that will provide the QoS required RSVP routers create dynamic access lists to pro-vide the QoS requested to ensure that packets are delivered at the prescribed minimum qualityparameters

Differentiated ServiceDifferentiated service includes a set of classification tools and queuing mechanisms to providecertain protocols or applications with a certain priority over other network traffic Differentiatedservices rely on edge routers to perform the classification of the types of packets traversing a net-work Network traffic can be classified by network address, protocols and ports, ingress interfaces,

or whatever classification that can be accomplished through the use of a standard or extendedaccess list

Why QoS Is Essential in VOIP NetworksThe challenge facing a converged infrastructure is to provide the efficiency of a packet-switchednetwork with the reliability of a legacy network.This is the role that QoS fills

QoS, through a variety of methods, gives reliability and availability to a converged infrastructureand still affords it the same benefits of efficient utilization of resources by providing the following:

■ Managed response times

■ Jitter (variation in delay) control

■ Prioritization of delay-sensitive traffic

■ Congestion management

■ Congestion avoidance

■ Support and enforcement of dedicated bandwidth requirements

■ Management and recovery of packet lossWith QoS, converged infrastructures can provide end users with a convenient, low-cost, scal-able, and above all, reliable solution for the majority of their communications Without QoS, aconverged infrastructure would be comparable to anarchy, with little to no reliability, conve-nience, or scalability—to a level where a single FTP session could shut down your entire VoIPinfrastructure

Configuring Voice Ports

We have now discussed the basic hardware installation for VNMs and VICs.The next step is toconfigure the cards on the Cisco router IOS Voice card configuration is covered in the followingsections Some basic configuration parameters must be set in order for a voice port to operate.Toconfigure a voice port, complete the following steps

1 Enter Privileged Exec mode:

router> enable

2 Check the DSP voice channel activity with the following command:

router# show voice dsp

3 Enter Global Configuration mode:

router# configure terminal

4 Enter Voice Card Configuration mode On the router, the slot must be 0:

router(config)# voice-card slot

5 Enter the CODEC type for the voice card:

router(config-voicecard)# CODEC {med | high}

This series of steps sets the CODEC compression technique, which is either high or mediumcomplexity High complexity can handle fewer calls per DSP.This is due to the higher CPU uti-lization required for high CODEC complexity operation High and medium complexity

CODECs:

■ High complexity Specifies two voice channels encoded in any of the following mats: G.711ulaw, G.711alaw, G.723.1 (r5.3), G.723.1 Annex A (r5.3), G.723.1 (r6.3),G.723.1 Annex A (r6.3), G.726 (r16), G.726 (r24), G.726 (r32), G.728, G.729, G.729Annex B, and fax relay

for-■ Medium (default) complexity Specifies four voice channels encoded in any of the

following formats: G.711ulaw, G.711alaw, G.726 (r16), G.726 (r24), G.726 (r32), G.729Annex A, G.729 Annex B with Annex A, and fax relay

Configuring FXO or FXS Voice Ports

All these parameters have default settings, and FXS and FXO port default configuration valuesare adequate for most situations.Therefore, user intervention is rarely needed.The following set-tings are mandatory to any FXS/FXO port configuration:

■ Signal type

■ Call progress tone

■ Ring frequency

■ Dial type (FXO only)

■ Music threshold

■ Description

■ Voice activity detection (VAD)

■ Comfort noise Follow these steps to complete a basic setup for all FXS/FXO voice ports:

1 Enter Privileged Exec mode:

router> enable

2 Enter Global configuration mode:

router# configure terminal

3 Identify which port to configure on a 2600 and 3600 series router:

router(config)# voice-port nm-module/vic-module/port-number router(config)# voice-port slot/port (Cisco 175x/1760 and MC3810)

4 Select the appropriate signaling for the start of a call:

router(config-voiceport)# signal [loop-start|ground-start]

5 Select the appropriate country codes for call progression signaling.The default is

northamerica:

router(config-voiceport)# cptone country-code

6 Configure the voice port connection mode type If the connection will be to a PBX,

use the tie-line option If the connection will be for private line automatic ringdown (PLAR), use the plar option If the connection will be for PLAR off-premises exten- sion (OPX), use the plar-opx option.

router(config-voiceport)# connection {tie-line | plar | plar-opx} string

7 Assign the appropriate out-dialing dial type (FXO only):

router(config-voiceport# dial-type{dtmf | pulse}

8 Configure the frequency in Hertz of ringing for the system that is attached on a Cisco

1750, 2600, and 3600 series router (FXS only):

router(config-voiceport)# ring frequency [25| 50]

router(config-voiceport)# ring frequency [20| 30]

9 Configure the maximum number of rings allowed before answering a call (FXO only):

router(config-voiceport)# ring number number

10 Specify an existing pattern for ring tone or define a new one (FXS only) Each patternspecifies a ring-pulse time and a ring-interval time:

router(config-voiceport)# ring cadence {[pattern01 | pattern02 … pattern12]

[define pulse interval]}

11 Specify the termination impedance, which needs to match the specifications of the PBX

it is attaching to:

router(config-voiceport)# impedance [600c|600r|900c|complex1|complex2]

12 Configure the threshold in decibels for hold music:

router(config-voiceport)# music-threshold number

13 (Optional) Configure a text string to the configuration that describes the connection forthis voice port:

router(config-voiceport)# description string

14 Configure background noise generation for the comfort of a user when there is nonoise:

router(config-voiceport)# comfort-noise

15 (Optional) Enable voice activity detection:

router(config-voiceport)# vad

Configuring E&M Ports

E&M default settings are usually not sufficient to enable voice transmissions over IP.This is

because E&M ports are designed to connect directly to a PBX and therefore must match the ticular PBX’s specifications.The following settings are mandatory to implement an E&M port:

The following commands complete a basic setup for all E&M voice ports:

1 Enter Privileged Exec mode:

router> enable

2 Enter Global Configuration mode:

router# configure terminal

3 Identify which port to configure on a 2600 and 3600 series router:

router(config)# voice-port nm-module/vic-module/port-number router(config)# voice-port slot/port (Cisco 175x/1760 and MC3810)

4 Select the appropriate signaling for the interface:

router(config-voiceport)# signal [wink-start|immediate|delay-dial]

5 Select the appropriate country codes for call progression signaling.The default is us.The northamerica keyword is for the Cisco MC3810 multiservice concentrator for versions

prior to Cisco IOS Release 12.0(4)T and for ISDN PRI:

router(config-voiceport)# cptone country code

6 Define cabling scheme operation:

router(config-voiceport)# operation [2-wire|4-wire]

7 Select the appropriate E&M interface type:

router(config-voiceport)# type [1|2|3|5]

8 Specify the termination impedance, which needs to match the specifications of the PBXthe port is attaching to:

router(config-voiceport)# impedance [600c|600r|900c|complex1|complex2]

Some optional configurations for the E&M port are not required for operation As with theFXS/FXO ports, the following configurations are used for optimization and usability:

1 Enter Privileged Exec mode:

router> enable

2 Enter Global Configuration mode:

router# configure terminal

3 Identify which port to configure on a 2600 and 3600 series router:

router(config)# voice-port nm-module/vic-module/port-number router(config)# voice-port slot/port (Cisco 175x/1760 and MC3810)

4 Specify that the port configured for PLAR (which we discuss in more detail later in thischapter):

router(config-voiceport)# connection plar string

5 Define the threshold in decibels for hold music:

router(config-voiceport)# music-threshold number

6 Specify a description field for port:

router(config-voiceport)# description string

7 Set comfort noise to generate background noise for the user when there is no sound onthe line:

router(config-voiceport)# comfort-noise

Voice Port-Tuning Commands

Voice port fine-tuning commands adjust timing, delay, impedance parameters, input gain, andoutput attenuation Once these adjustments are made, you can fine-tune volume control, how thenumber pads are dialed, and how long a voice port will wait before hanging up a signal

Concepts of Delay and Echo

The most challenging part of designing a VoIP network is the transmission of real-time traffic.Voice communication is sensitive to delays and echo Speech patterns become awkward andindistinguishable if there is too much delay in the voice traffic Minimize delay as much as youcan to get the voice traffic as close to real time as possible In today’s voice trafficking, two dif-ferent kinds of delay must be handled: fixed delay and variable delay.The various delay points areillustrated in Figure 9.43

Echo is the reflection of voice traffic back to the source of that traffic A certain amount of

echo is acceptable and desirable because it assures the source that voice traffic has been generatedand sent.Too much echo is disruptive because the speaker will not be able to discern between hisvoice and the echo

Figure 9.43 Voice Packet Delay

Si

Fixed Codec Delay

Variable Queuing Delay Fixed Packetization Delay

Fixed Serialization Delay

Fixed Switch Delay

Fixed De-jitter Delay

Fixed delay is the amount of time the signal needs to transverse the medium, such as copper,

fiber, or microwave.This time is fixed because the laws of physics dictate how fast the data signalswill go on particular media Acceptable levels for most users are below 150ms one-way per ITUG.114 Fixed delays are composed of CODEC delays, packetization delays, and serialization

■ CODEC induced delay Compression/decompression of a voice packet from analog

to digital format and vice versa It ranges from 0.75ms to 30ms, depending on theCODEC used

■ Packetization delay Time it takes the equipment to actually produce a data packet

Should be under 30ms

■ Serialization Time it takes to clock a voice or data frame onto a network interface

Affected by the frame size and line speed

Variable delays are synonymous with jitter and are caused by queuing variances during the

transmission of a packet through the network As the packets are transferred out of the queue,there can be a delay between voice packets that sounds like stuttering speech QoS features can

be used to alleviate the effects of jitter by prioritizing the voice traffic over other traffic.You cancurb delay using several methods

■ Queuing Time it takes for a packet to exit the output queue of the device that isrouting the data Measured from the time the data is generated into the input queue towhen it is released by the output queue

■ Network switching Delay across the public network such as a Frame Relay or ATMnetwork

■ De-jitter Voice traffic works best if there is a constant flow of packets Jitter must beminimized to improve the quality of the conversation De-jitter buffers are utilized onthe receiving end to adjust the variable delays into a fixed delay

The command that adjusts the Cisco de-jitter buffering is delay The delay command was configured under the voice-port configuration mode before IOS release

playout-12.1(5)T Release 12.1(5)T and later implement the command under the dial-peer configurationmode.The following steps are used to configure playout delay:

1 Enter Privileged Exec mode:

router> enable

2 Enter Global Configuration mode:

router# configure terminal

3 Identify the port to configure on a 2600 and 3600 series router:

router(config)# voice-port nm-module/vic-module/port-number

router(config)# voice-port slot/port (Cisco 175x/1760 and MC3810)

4 Determine the mode in which the jitter buffer will operate for calls on this voice port.

■ Adaptive Adjusts the jitter buffer size and amount of playout delay based on rent network conditions.This is the default setting

cur-■ Fixed Defines the jitter buffer size as fixed so that the playout delay does notadjust A constant playout delay is added

router(config-voiceport)# playout-delay mode {adaptive| fixed]

5 Tune the playout buffer to accommodate packet jitter caused by switches in the WAN:

router(config-voiceport)# playout-delay {nominal value| maximum value

| minimum {default | low | high}}

Fine-Tuning FXS/FXO Ports

Special parameters can be adjusted to fine-tune the ports, minimizing issues of delay and echo Inmost cases, the default parameters for FXO/FXS ports will be sufficient, but special values can beset for the following parameters:

■ Timing other than timeouts

To change any of these parameters, follow these steps:

1 Enter Privileged Exec mode:

router> enable

2 Enter Global Configuration mode:

router# configure terminal

3 Identify the port to configure:

router# (config)voice-port nm-module/vic-module/port-number

4 Specify the amount of receiver gain on the interface in decibels Value can be (–6) to 14:

router(config-voiceport)# input gain value

5 Specify the amount of transmit attenuation on the interface in decibels Value can be 0

to 14:

router(config-voiceport)# output attenuation value

6 Enable echo-cancellation for voice signals sent out of the interface and received back onthe same interface Excessive echo can cause disruption to normal conversation patterns

router(config-voiceport)# echo-cancel enable

7 Adjust the size of the echo-cancel coverage time in milliseconds Values are 16, 24, or32:

router(config-voiceport)# echo-cancel coverage value

8 Enable “nonlinear” processing, which shuts off any signal if no speech is detected on thenear end.This is used in conjunction with echo cancellation:

router(config-voiceport)# non-linear

9 Configure how long the system will wait for the first digit to be input by the user after

an off-hook state is detected.This value can be anywhere between 0 and 120 seconds:

router(config-voiceport)# timeouts initial seconds

10 Configure how long the system will wait for subsequent digits after the initial digit isreceived.This value can be anywhere between 0 and 120 seconds:

router(config-voiceport)# timeouts interdigit seconds

11 Specify how long the digital signal lasts for DTMF digit signals.The range is from 50 to

100 milliseconds, with a default of 100 milliseconds:

router(config-voiceport)# timing digit milliseconds

12 Specify the delay between digit signals for DTMF digit signals Range is from 50 to 100milliseconds, the default being 100 milliseconds:

router(config-voiceport)# timing inter-digit milliseconds

13 Configure the length of pulse signal.This command is for FXO ports only using pulsesignals.The range is 10 to 20 milliseconds and the default is 20 milliseconds:

router(config-voiceport)# timing pulse-digit milliseconds

14 Configure length of delay between digit signals.This command is for FXO ports onlyusing pulse signals.The range is from 100 to 1000 milliseconds and the default is 500milliseconds:

router(config-voiceport)# timing pulse-inter-digit milliseconds

Fine-Tuning E&M Ports

E&M ports may require fine-tuning.The following steps are used to fine-tune E&M ports:

1 Enter Privileged Exec mode:

router> enable

2 Enter Global Configuration mode:

router# configure terminal

3 Identify the port to configure:

router# (config)voice-port nm-module/vic-module/port-number

4 Specify the amount of receiver gain on the interface in decibels Value can be (–6) to14:

router# (config-voiceport)input gain value

5 Specify the amount of transmit attenuation on the interface in decibels Value can be 0

to 14:

router# (config-voiceport)output attenuation value

6 Enable echo-cancellation for voice signals sent out of the interface and received back onthe same interface

router# (config-voiceport)echo-cancel enable

7 Adjust the size of the echo-cancel coverage in milliseconds Values are 16, 24, or 32:

router# (config-voiceport)echo-cancel coverage value

8 Enable nonlinear processing, which shuts off any signal if no speech is detected on the

near end.This is used in conjunction with echo cancellation:

router# (config-voiceport)non-linear

9 Configure how long the system will wait for the first digit to be input by the user after

an off-hook state is detected.This value can be anywhere between 0 and 120 seconds:

router# (config-voiceport)timeouts initial seconds

10 Configure how long the system will wait for subsequent digits after the initial digit isreceived.This value can be anywhere between 0 and 120 seconds:

router# (config-voiceport)timeouts interdigit seconds

11 Specify how long the digit signal will last for DTMF digit signals.The range is from 50

to 100 milliseconds:

router# (config-voiceport)timing digit milliseconds

12 Specify the delay between digit signals for DTMF digit signals.The range is from 50 to

500 milliseconds:

router#(config-voiceport)timing inter-digit milliseconds

13 Specify the pulse-dialing rate.This is used for pulse dialing only.The range is from 10 to

20 pulses per second:

router# (config-voiceport)timing pulse pulse-per-second

14 Configure the delay between digit signals.This is used for pulse dialing only.The range

is from 100 to 1000 milliseconds:

router# (config-voiceport)timing pulse-inter-digit milliseconds

15 Configure the delay signal time for delay dial signaling.The range is from 100 to 5000milliseconds:

router#(config-voiceport)timing delay-duration milliseconds

16 Configure the minimum time for outgoing seizure to out-dial address.The range isfrom 20 to 2000 milliseconds:

router# (config-voiceport)timing delay-duration milliseconds

17 Specify the time between generations of “wink-like” pulses.The range is from 0 to 5000milliseconds:

router# (config-voiceport)timing delay-pulse min-delay milliseconds

18 Specify the minimum amount of time between the off-hook signal and the call beingcompletely cleared.The range is from 200 to 2000 milliseconds:

router# (config-voiceport)timing clear-wait milliseconds

19 Specify the delay signal time for delay dial signaling.The range is from 100 to 5000 liseconds:

mil-router(config-voiceport)timing delay-duration milliseconds

20 Configure the maximum wink-wait duration.The range is from 100 to 400 onds:

millisec-router(config-voiceport)# timing wink-duration milliseconds

21 Configure the maximum wink-wait duration for wink-start signal.The range is from

100 to 5000 milliseconds:

router(config-voiceport)# timing wink-wait milliseconds

Some added features always need to be adjusted for the DID ports Contrary to the defaultsettings of the FXO/FXS ports, in most cases DID ports require fine-tuning adjustments Followthese steps to fine-tune DID ports:

1 Enter Privileged Exec mode:

router> enable

2 Enter Global Configuration mode:

router# configure terminal

3 Identify the port to configure:

router(config)# voice-port nm-module/vic-module/port-number

4 This command sets the maximum time to wait for wink signaling after an outgoingseizure is sent.This is optional for wink-start ports only:

router(config-voiceport)# timing wait-wink milliseconds

5 This command sets the maximum time to wait before sending wink signals after anincoming seizure is detected.This is optional for wink-start ports only:

router(config-voiceport)# timing wink-wait milliseconds

6 This command sets the duration of a wink-start signal.This is optional for wink-startports only:

router(config-voiceport)# timing wink-duration milliseconds

7 This command sets the duration of the delay signal.This is optional for delay dial portsonly:

router(config-voiceport)# timing delay-duration milliseconds

8 This command sets the delay interval after an incoming seizure is detected.This isoptional for delay dial ports only:

router(config-voiceport)# timing delay-start milliseconds

Configuring Dial Plans and Dial Peers

The dial plan is the architecture for the numbers and patterns that the user dials to reach the receiving party, or the routing plan for dial numbers Dial plans are manually configured on Cisco

routers A collection of dial peers creates a dial plan Devising the most optimal dial plan allowsfor growth and ease of manageability If an existing plan is used, it should be reviewed for scala-bility and to meet VoIP’s unique requirements A number of extra items should be accounted for:

■ Voice-mail extensions

■ Call parking