Weed Eater 2600 User Manual 2

Cisco High-Density Analog Voice and Fax Network Module

Feature History

Release Modification

12.2(2)XT These features were implemented on the Cisco 2600 series, Cisco 3640 and the Cisco 3660 router.

12.2(8)T This feature was integrated into Cisco IOS Release 12.2(8)T.

This document describes the Cisco High-Density Analog Voice and Fax Network Module (NM-HDA) in Cisco IOS Release 12.2(8)T. This document includes the following sections:

• Feature Overview, page 2

• Supported Platforms, page 9

• Supported Standards, MIBs, and RFCs, page 10

• Prerequisites, page 10

• Configuration Tasks, page 11

• Configuration Examples, page 29

• Command Reference, page 31

• Glossary, page 37

Cisco IOS Release 12.2(2)XT and 12.2(8)T

Feature Overview

The Cisco High-Density Analog Voice and Fax Network Module provides DTMF detection, voice compression and decompression, call progress tone generation, voice activity detection (VAD), echo cancellation, and adaptive jitter buffering for up to 16 ports.

The following port combinations are supported:

• Twelve Foreign Exchange Station (FXS) ports

• Eight Foreign Exchange Office (FXO) ports and four FXS ports

• Twelve FXS and four FXO ports

• Four Foreign Exchange Station (FXS) ports

The base card supports four FXS ports. The addition of an eight-port FXS expansion module can increase the capacity to twelve FXS ports. The addition of two four-port FXO expansion modules can increase the capacity to eight- FXO ports and four-FXS ports. The addition of one each of the FXS and FXO expansion modules can increase the capacity to twelve FXS ports and four- FXO ports. The FXO expansion module supports a power failure port which connects directly to the central office (CO) in case of failure.

The digital signal processors (DSPs) on the network module support up to eight-ports of high-complexity codecs or up to sixteen ports of medium-complexity and low-complexity codecs. The number of DSPs must be increased if more than eight ports of high-complexity codecs are needed. In this case, a DSP expansion module must be installed.

Table 1 shows analog voice port numbering, which differs for each of the voice-enabled routers. More

current information may be available in the release notes that accompany the Cisco IOS software you are using.

Cisco High-Density Analog Voice and Fax Network Module

Table 1 High-Density Analog Voice Port Numbering

Base Module Telephony Signaling

Voice Port Numbers

Interface

NM-HDA-4FXS FXS 0 to 3

EM0 FXO 4 to 7

EM0 FXS 4 to 11

EM1 FXO 14 to 17

EM1 FXS 14 to 21

Caution No more than a total of 12 FXS ports can be configured at one time.

Cisco IOS Release 12.2(2)XT and 12.2(8)T

Cisco High-Density Analog Voice and Fax Network Module

Telephony Signaling Interfaces

Voice ports on routers and access servers physically connect the router or access server to telephony devices such as telephones, fax machines, PBXs, and (PSTN) central office (CO) switches. These devices may use any of several types of signaling interfaces to generate information about on-hook status, ringing, and line seizure.

The router’s voice-port hardware and software must be configured to transmit and receive the same type of signaling being used by the device with which they are interfacing so that calls can be exchanged smoothly between the packet network and the circuit-switched network.

The signaling interfaces discussed in this document include foreign exchange office (FXO), and foreign exchange station (FXS), which are types of analog interfaces. It is important to know which signaling method the telephony side of the connection is using and to match the router configuration and voice interface hardware to that signaling method.

The next three illustrations show how the different signaling interfaces are associated with different uses of voice ports. In Figure 1, FXS signaling is used for end-user telephony equipment, such as a telephone or fax machine. Figure 2 shows an FXS connection to a telephone and an FXO connection to the PSTN at the far side of a WAN; this might be a telephone at a local office going over a WAN to a router at headquarters that connects to the PSTN.

Feature Overview

Figure 1 FXS Signaling Interfaces

Figure 2 FXS and FXO Signaling Interfaces

FXS and FXO Interfaces

An FXS interface connects the router or access server to end-user equipment such as telephones, fax machines, or modems. The FXS interface supplies ring, voltage, and dial tone to the station and includes an RJ-11 connector for basic telephone equipment, keysets, and PBXs.

An FXO interface is used for trunk, or tie line, connections to a PSTN CO or to a PBX. This interface is of value for off-premise station applications. A standard RJ-11 modular telephone cable connects the FXO voice interface card to the PSTN or PBX through a telephone wall outlet.

Voice port

1/0/0

FXS FXS

Voice port

1/0/0

FXS FXO

Serial or

Ethernet port

Serial or

Ethernet port

WAN

Serial or

Ethernet port

Serial or

Ethernet port

Voice port

1/0/0

37757

PSTN

37758

Cisco IOS Release 12.2(2)XT and 12.2(8)T

Feature Overview

Cisco High-Density Analog Voice and Fax Network Module

FXO and FXS interfaces indicate on-hook or off-hook status and the seizure of telephone lines by one of two access signaling methods: loop start or ground start. The type of access signaling is determined by the type of service from the CO; standard home telephone lines use loop start, but business telephones can order ground start lines instead.

Loop-start is the more common of the access signaling techniques. When a handset is picked up (the telephone goes off-hook), this action closes the circuit that draws current from the telephone company CO and indicates a change in status, which signals the CO to provide dial tone. An incoming call is signaled from the CO to the handset by sending a signal in a standard on/off pattern, which causes the telephone to ring.

Loop-start has two disadvantages, however, that usually are not a problem on residential telephones but that become significant with the higher call volume experienced on business telephones. Loop-start signaling has no means of preventing two sides from seizing the same line simultaneously, a condition known as glare. Also, loop-start signaling does not provide switch-side disconnect supervision for FXO calls. The telephony switch (the connection in the PSTN, another PBX, or key system) expects the router’s FXO interface, which looks like a telephone to the switch, to hang up the calls it receives through its FXO port. However, this function is not built into the router for received calls; it only operates for calls originating from the FXO port.

Another access signaling method used by FXO and FXS interfaces to indicate on-hook or off-hook status to the CO is ground start signaling. It works by using ground and current detectors that allow the network to indicate off-hook or seizure of an incoming call independent of the ringing signal and allow for positive recognition of connects and disconnects. For this reason, ground-start signaling is typically used on trunk lines between PBXs and in businesses where call volume on loop start lines can result in glare. See the “Configuring Disconnect Supervision Commands” section on page 14 and “Configuring FXO

Supervisory Disconnect Tone Commands” section on page 16 for voice port commands that configure

additional recognition of disconnect signaling.

In most cases, the default voice port command values are sufficient to configure FXO and FXS voice ports.

Disconnect Supervision Commands

PBX and PSTN switches use several different methods to indicate that a call should be disconnected because one or both parties have hung up. The commands in this section are used to configure the router to recognize the type of signaling in use by the PBX or PSTN switch connected to the voice port. These methods include the following:

• Battery reversal disconnect

• Battery denial disconnect

• Supervisory tone disconnect (STD)

Battery reversal occurs when the connected switch changes the polarity of the line in indicate changes in call state (such as off-hook or, in this case, call disconnect). This is the signaling looked for when the battery reversal command is enabled on the voice port, which is the default configuration.

Battery denial (sometimes called power denial) occurs when the connected switch provides a short (approximately 600 ms) interruption of line power to indicate a change in call state. This is the signaling looked for when the supervisory disconnect command is enabled on the voice port, which is the default configuration.

Cisco IOS Release 12.2(2)XT and 12.2(8)T

Cisco High-Density Analog Voice and Fax Network Module

Supervisory tone disconnect occurs when the connected switch provides a special tone to indicate a change in call state. Some PBXs and PSTN CO switches provide a 600-millisecond interruption of line power as a supervisory disconnect, and others provide supervisory tone disconnect (STD). This is the signal that the router is looking for when the no supervisory disconnect command is configured on the voice port.

Note In some circumstances, you can use the FXO Disconnect Supervision feature to enable analog FXO

ports to monitor call progress tones for disconnect supervision that are returned from a PBX or from the PSTN. For more information, see the “Configuring FXO Supervisory Disconnect Tone

Commands” section on page 16.

FXO Supervisory Disconnect Tone Commands

If the FXO supervisory disconnect tone is configured and a detectable tone from the PSTN or PBX is detected by the Digital Signal Processor (DSP), the analog FXO port goes on-hook. This feature prevents an analog FXO port from remaining in an off-hook state after an incoming call is ended. FXO supervisory disconnect tone enables interoperability with PSTN and PBX systems whether or not they transmit supervisory tones.

Feature Overview

Note This feature applies only to analog FXO ports with loop-start signaling on the Cisco 2600 series, the

Cisco 3640, and the Cisco 3660 routers.

To configure a voice port to detect incoming tones, you must know the parameters of the tones expected from the PBX or PSTN. Then create a voice class that defines the tone- detection parameters, and, finally, apply the voice class to the applicable analog FXO voice ports. This procedure configures the voice port to go on-hook when it detects the specified tones. The parameters of the tones need to be precisely specified to prevent unwanted disconnects due to detection of nonsupervisory tones or noise.

A supervisory disconnect tone is normally a dual tone consisting of two frequencies; however, tones of only one frequency can also be detected. Use caution if you configure voice ports to detect nondual tones, because unwanted disconnects can result from detection of random tone frequencies. You can configure a voice port to detect a tone with one on/off time cycle, or you can configure it to detect tones in a cadence pattern with up to four on/off time cycles.

Delay in Voice Networks

Delay is the time it takes for voice packets to travel between two endpoints. Excessive delay can cause quality problems with real-time traffic such as voice. However, because of the speed of network links and the processing power of intermediate devices, some delay is expected.

When listening to speech, the human ear normally accepts up to about 150 ms of delay without noticing delays. The ITU G.114 standard recommends no more than 150 ms of one-way delay for a normal voice conversation. Once the delay exceeds 150 ms, a conversation is more like a “walkie-talkie” conversation in which one person must wait for the other to stop speaking before beginning to talk.

You can measure delay fairly easily by using ping tests at various times of the day with different network traffic loads. If network delay is excessive, it must be reduced for adequate voice quality.

Cisco IOS Release 12.2(2)XT and 12.2(8)T

Feature Overview

Several different types of delay combine to make up the total end-to-end delay associated with voice calls:

• Propagation delay—Amount of time it takes the data to physically travel over the media.

• Handling delay—Amount of time it takes to process data by adding headers, taking samples,

forming packets, and so on.

• Queuing delay—Amount of time lost due to congestion.

• Variable delay or jitter—Amount of time that causes the conversation to break and become

unintelligible. Jitter is described in detail below.

Note Propagation, handling, and queuing delay are not addressed by voice-port commands and fall outside

the scope of this document.

Voice Level Adjustment

As much as possible, it is desirable to achieve a uniform input decibel level to the packet voice network to limit or eliminate any voice distortion due to incorrect input and output decibel levels. Adjustments to levels may be required by the type of equipment connected to the network or by local country-specific conditions.

Cisco High-Density Analog Voice and Fax Network Module

Incorrect input or output levels can cause echo, as can an impedance mismatch. Too much input gain can cause clipped or fuzzy voice quality. If the output level is too high at the remote router’s voice port, the local caller hears echo. If the local router’s voice port input decibel level is too high, the remote side hears clipping. If the local router’s voice port input decibel level is too low, or the remote router’s output level is too low, the remote side voice can be distorted at a very low volume and Dual Tone Multi-Frequency (DTMF) may be missed.

Echo Cancellation

Echo is the sound of your own voice reverberating in the telephone receiver while you are talking. When timed properly, echo is not a problem in the conversation; however, if the echo interval exceeds approximately 25 milliseconds, it is distracting. Echo is controlled by echo cancellers.

In the traditional telephony network, echo is generally caused by an impedance mismatch when the 4-wire network is converted to the 2-wire local loop. In voice packet-based networks, echo cancellers are built into the low-bit rate codecs and are operated on each DSP.

Adaptive Jitter Buffering

Delay can cause unnatural starting and stopping of conversations, but variable-length delays (also known as jitter) can cause a conversation to break and become unintelligible. Jitter is not usually a problem with PSTN calls because the bandwidth of calls is fixed and each call has a dedicated circuit for the duration of the call. However, in Voice over IP (VoIP) networks, data traffic might be bursty, and jitter from the packet network can become an issue. Especially during times of network congestion, packets from the same conversation can arrive at different interpacket intervals, disrupting the steady, even delivery needed for voice calls. Cisco voice gateways have built-in jitter buffering to compensate for a certain amount of jitter; the playout-delay command can be used to adjust the jitter buffer.

Cisco IOS Release 12.2(2)XT and 12.2(8)T

Cisco High-Density Analog Voice and Fax Network Module

Normally, the defaults in effect are sufficient for most networks. However, a small playout delay from the jitter buffer can cause lost packets and choppy audio, and a large playout delay can cause unacceptably high overall end-to-end delay.

Note Prior to Cisco IOS Release 12.1(5)T, playout delay was configured in voice-port configuration mode.

For Cisco IOS Release 12.1(5)T and later releases, in most cases, playout delay should be configured in dial-peer configuration mode on the VoIP dial peer that is on the receiving end of the voice traffic that is to be buffered. This dial peer senses network conditions and relays them to the DSPs, which adjust the jitter buffer as necessary. When multiple applications are configured on the gateway, playout delay should be configured in dial-peer configuration mode. When there are numerous dial peers to configure, it might be simpler to configure playout delay on a voice port. If there are conflicting playout delay configurations on a voice port and also on a dial peer, the dial peer configuration takes precedence.

Voice Activity Detection Commands

In normal voice conversations, only one person speaks at a time. Today’s circuit-switched telephone networks dedicate a bidirectional, 64-kbps channel for the duration of each conversation, regardless of whether anyone is speaking at the moment. This means that, in a normal voice conversation, at least 50 percent of the bandwidth is wasted when one or both parties are silent. This figure can actually be much higher when normal pauses and breaks in conversation are taken into account.

Packet-switched voice networks, on the other hand, can use this “wasted” bandwidth for other purposes when voice activity detection (VAD) is configured. VAD works by detecting the magnitude of speech in decibels and deciding when to cut off the voice from being framed. VAD has some technological problems, however, which include the following:

Feature Overview

Benefits

• General difficulties in determining when speech ends

• Clipped speech when VAD is slow to detect that speech is beginning again

• Automatic disabling of VAD when conversations take place in noisy surroundings

Cost Effective

• Higher density of analog interfaces provides voice and data over a single network

–

16 analog voice ports in a single network module (NM)

• Connects existing analog voice telephone sets, fax machines, and key systems using Cisco gateways

Scalability-Modularity and Flexibility

• Eliminates the need for a channel bank

• The network module may be configured to be an FXO, an FXS or a combination module

Serviceability

• Provides remote monitoring, testing, and diagnoses

Cisco IOS Release 12.2(2)XT and 12.2(8)T

Feature Overview

Restrictions

Cisco High-Density Analog Voice and Fax Network Module

The following features not supported at this time:

• ATM S V C

• SAS signaling

Supported High-Complexity Codecs

• G.711 a-law

• G.711 u-law

• G.723.1 ANNEX-A 5300 bps

• G.723.1 ANNEX-A 6300 bps

• G.723.1 5300 bps

• G.723.1 6300 bps

• G.726 16000 bps

• G.726 24000 bps

• G.726 32000 bps

• G.728 16000 bps

• G.729 ANNEX-B 8000 bps

• G.729 8000 bps

• GSMFR 13200 bps

Supported Medium-Complexity Codecs:

• G.726 16000 bps

• G.726 24000 bps

• G.726 32000 bps

• G.729 ANNEX-B 8000 bps (Annex A version)

• G.729 8000 bps (Annex A version)

Supported low-Complexity Codecs:

• G.711 a-law 64000 bps

• G.711 u-law 64000 bps

Note Three-way calling is not supported with low-complexity codecs.

The ring capacity is 10 Ringer Equivalency Number (REN). The total of all RENs of the telephones connected to the one line must not exceed the value 5, or some or all of the ringers may not operate.

Cisco IOS Release 12.2(2)XT and 12.2(8)T

Cisco High-Density Analog Voice and Fax Network Module

Supported Platforms

• Cisco 2600 series

• Cisco 3640

• Cisco 3660

Determining Platform Support Through Cisco Feature Navigator

Cisco IOS software is packaged in feature sets that support specific platforms. To get updated information regarding platform support for this feature, access Cisco Feature Navigator. Cisco Feature Navigator dynamically updates the list of supported platforms as new platform support is added for the feature.

Cisco Feature Navigator is a web-based tool that enables you to quickly determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common.

To access Cisco Feature Navigator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions at http://www.cisco.com/register.

Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL:

http://www.cisco.com/go/fn

Cisco IOS Release 12.2(2)XT and 12.2(8)T

Supported Standards, MIBs, and RFCs

Table 2 lists the hardware platforms that support this feature, and the releases in which the feature was

first supported. If the First T train Release column is blank, the feature is not yet available in a Cisco IOS T release on that platform.

Table 2 Cisco IOS Release and Platform Support for this Feature

Platform 12.2(2)XT 12.2(8)T

Cisco 2600 series X X

Cisco 3640 X X

Cisco 3660 X X

Supported Standards, MIBs, and RFCs

Standards

• No new or modified Standards are supported by this feature.

Cisco High-Density Analog Voice and Fax Network Module

Prerequisites

MIBs

• No new or modified MIBs are supported by this feature.

To obtain lists of supported MIBs by platform and Cisco IOS release, and to download MIB modules, go to the Cisco MIB website on Cisco.com at the following URL:

http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml

RFCs

• No new or modified RFCs are supported by this feature.

• Obtain analog line or analog trunk service from your PBX, Key System, or PSTN.

• Complete your company’s dial plan.

• Establish a working telephony network based on your company’s dial plan.

• Install at least one other network module or WAN interface card to provide the connection to the

network LAN or WAN.

• Establish a working IP and Frame Relay or ATM network. For more information about configuring

IP, refer to the Cisco IOS IP Configuration Guide, Release 12.2.

• Install appropriate voice processing and voice interface hardware on the router. See the

“Configuring Analog FXO and FXS Voice Ports” section on page 11.

Cisco IOS Release 12.2(2)XT and 12.2(8)T

Cisco High-Density Analog Voice and Fax Network Module

Configuration Tasks

See the following sections for configuration tasks for this feature.

• Configuring Analog FXO and FXS Voice Ports (required)

• Configuring Voice Activity Detection (VAD) (optional)

• Fine-Tuning Analog Voice Ports (optional)

• Verifying Analog Voice-Port Configurations (optional)

Configuring Analog FXO and FXS Voice Ports

To configure basic analog voice port parameters on Cisco 2600 series, Cisco 3640 and Cisco 3660 routers, use the following commands beginning in global configuration mode:

Command Purpose

Step 1

Router(config)# voice-port slot/subunit/port

Configuration Tasks

Enters voice-port configuration mode.

The arguments are as follows:

Step 2

Step 3

FXO or FXS

Router(config-voiceport)# signal {loop-start | ground-start}

Router(config-voiceport)# cptone locale

• slot—Specifies the number of the router slot

where the voice network module is installed.

• port—Indicates the voice port. See Tab l e 1 on

page 2 for valid entries.

• subunit—Specifies the location of the

Cisco High-Density Analog Voice/Fax Netwo rk Module (NM-HDA). The valid entry is 0.

Note A slash must be entered between

arguments.

Valid entries vary by router platform; enter the show voice port summary command for available values.

Selects the access signaling type to match that of the telephony connection you are making. The keywords are as follows:

• loop-start—(default) Uses a closed circuit to

indicate off-hook status; used for residential loops.

• ground-start—Uses ground and current

detectors; preferred for PBXs and trunks.

Selects the two-letter locale for the voice call progress tones and other locale-specific parameters to be used on this voice port.

Cisco routers comply with the ISO 3166 locale name standards. To see valid choices, enter a question mark (?) following the cptone command.

The default is us.

Cisco IOS Release 12.2(2)XT and 12.2(8)T

Configuration Tasks

Command Purpose

Step 4

Step 5

Step 6

Step 7

Step 8

Router(config-voiceport)# ring frequency {25 | 50}

Router(config-voiceport)# ring number number

Router(config-voiceport)# ring cadence {[pattern01 |

interval]}

Router(config-voiceport)# description string

Router(config-voiceport)# no shutdown

Cisco High-Density Analog Voice and Fax Network Module

(FXS only) Selects the ring frequency, in hertz (HZ), used on the FXS interface. This number must match the connected telephony equipment and may be country-dependent. If not set properly, the attached telephony device may not ring or it may buzz.

The keyword default is 25 on Cisco 2600 series, Cisco 3640 and Cisco 3660 routers.

(FXO only) Specifies the maximum number of rings to be detected before an incoming call is answered by the router.

The default is 1.

(FXS only) Specifies an existing pattern for ring, or defines a new one. Each pattern specifies a ring-pulse time and a ring-interval time. The keywords and arguments are as follows:

• pattern01 through pattern12 name pre-set

ring cadence patterns. Enter ring cadence ? to see ring pattern explanations.

• define pulse interval specifies a user-defined

pattern: pulse is a number (one or two digits, from 1 to 50) specifying ring pulse (on) time in hundreds of milliseconds, and interval is a number (one or two digits from 1 to 50) specifying ring interval (off) time in hundreds of milliseconds.

The default is the pattern specified by the cptone locale that has been configured.

Attaches a text string to the configuration that describes the connection for this voice port. This description appears in various displays and is useful for tracking the purpose or use of the voice port. The string argument is a character string from 1 to 255 characters in length.

The default is that there is no text string (describing the voice port) attached to the configuration.

Activates the voice port. If a voice port is not being used, shut the voice port down with the shutdown command.

Cisco IOS Release 12.2(2)XT and 12.2(8)T

+ 26 hidden pages

Weed Eater 2600 User Manual 2

Feature Overview

Telephony Signaling Interfaces

FXS and FXO Interfaces

Disconnect Supervision Commands

FXO Supervisory Disconnect Tone Commands

Delay in Voice Networks

Voice Level Adjustment

Echo Cancellation

Adaptive Jitter Buffering

Voice Activity Detection Commands

Benefits

Restrictions

Related Documents

Supported Platforms

Supported Standards, MIBs, and RFCs

Prerequisites

Configuration Tasks

Configuring Analog FXO and FXS Voice Ports