This document describes how to set up the analog voice module. It has the following sections:

Overview of Analog Voice Module

• Voice compression and fax processing

• Call state processing of the signalling information

• Tone generation, including call progress, dial tone, busy signal, Dual-Tone Multi-Frequency (DTMF) ringback, and others.

• Ring signal/voltage generation

• Transport of voice band, signalling, and control information.

• Telco power conversion, regulation, and isolation

• FXO, foreign exchange office; FXS, foreign exchange station; or E&M, ear and mouth.

The following sections describe how to create and configure voice channels on the analog voice module.

Switched Frame Transfer Mode

SFTM gives you the benefits of SVC (switched virtual circuit) service using PVC (permanent virtual circuit)-based frame relay networks. Whether the traffic is voice/fax, frame relay, X.25, SNA/SDLC, asynchronous, or nearly any LAN protocol, SFTM automatically determines the best available network path for all traffic. SFTM continually monitors line outages and restorations, adjusting traffic routing accordingly with no loss of network manageability or control.

Before You Begin

Specifically, it covers the following topics.

• Assigning an IP Address to the Analog Voice Module

• Using NAT With Analog Voice

• Using the NetrixView 2000 Software

• Logging In to the NetrixView 2000

• Initializing Your NetrixView PC Database

• Creating and Configuring an SFTM Trunk

• Using the Default Analog Database

1. Connect to the CLI and then display the Voice Config> prompt.

*config

Config>voice 

VOICE Config>

2. Use the set address command to set the IP address.

VOICE Config> set address = 192.168.20.10

3. Restart the router to cause the new address to take affect.

VOICE Config> exit
Config>
*restart
Are you sure you want to restart the gateway? (Yes or [No]): Yes
OpenROUTE(tm) Software
OpenROUTE is a registered trademark of Netrix Corp.
Copyright Notices:
Copyright 1985-2000 by Netrix Corp., All rights reserved
Copyright 1984-1987, 1989 by J. Noel Chiappa

MOS Operator Control

* 

Using NAT With Analog Voice

To do this, you set up a fixed address mapping for the voice module so that NAT does not translate the voice IP address. You need to assign the same address as the public outside address and the private inside address. This address must also be on the same subnet as the Internet connection.

The following example shows how to set up a fixed address mapping, where 128.185.2.2 is the IP address of the voice module. Enter these commands in the OpenROUTE CLI.

*config
Config>PROTOCOL ip
Internet protocol user configuration
IP config>nat
Network Address Translation Configuration
NAT Config>add FIXED-IP-MAPPINGS
Interface number [1]? 3
Public outside address [0.0.0.0]? 128.185.2.2
Mask [255.255.255.255]?
Private inside address [0.0.0.0]? 128.185.2.2

To check the global IP address that NAT is using, enter list nat at the NAT monitoring prompt.

*monitor
+PROTOCOL IP
IP>nat
Network Address Translation Console         
NAT>LIST NAT-INTERFACE
Interface number [1]?
NAT Enabled on interface 1
  Address is:     128.185.2.1       Service Table Used:  Global
  Current # entries: 0
  Maximum # entries: 500            Global ageout:       1800 secs
  TCP ageout (secs): 9000           TCP closed ageout:   30 secs

NAT Config>SET NAT-INTERFACE IP-ADDRESS
Interface number [1]?
NAT IP address (0.0.0.0 = use automatic default) [0.0.0.0]? 128.185.2.1

The bottom of each NetrixView 2000 screen lists the three actions that mouse buttons can perform. These actions change depending on what screen is active and where the cursor is located.

The left mouse button performs the action on the left, the middle mouse button performs the action in the center, and the right mouse button performs the action on the right. If your mouse has two buttons, click both buttons at the same time to simulate the middle button.

In this document, the actions that mouse buttons can take appear in bold. Here are some examples.

This instruction . . .	means to . . .
Get Info on the NX Analog Board icon.	Move your mouse over the icon and click the right mouse button.
Select the NX Analog Board Component.	Move your mouse over the component and click the left mouse button.
View into the facility.	Move your mouse over the facility composite object and click the middle mouse button.

To access online help, place the cursor on the field, button, or other screen item and press the Help mouse button. The Help mouse button is usually the center button.

To close a window, or popdown to the previous level, click the right mouse button.

To edit text or numbers in a field, you must leave the cursor on the entry field. When you move the cursor from the field, the text or numbers in the field are accepted as entered, as if you pressed the Enter key.

Logging In to the NetrixView 2000

The default login entries are

Network: netrix

Name: tech

Password: (Do not type a password, just press Enter.)

Initializing Your NetrixView PC Database

1. Select the Create button located at the top of the screen to put NetrixView in create mode.

2. On the control column, click the Goto button.

3. In the Node No. field, type 1 and Select View.

The following view appears, which shows various components on your PC.

4. Click each component with your left mouse button.

5. Select the Monitor button located at the top of the screen to put NetrixView in monitor mode.

6. Select each component with your left mouse button and Select the Configure button on the screen that appears.

Each component on your PC should turn green, which means that component is responsive.

For detailed information on setting up trunks, refer to Setting Up Trunks, Chapter 6 of Configuring and Monitoring a Network Exchange 2200.

Note: Make sure the NetrixView 2000 is in Create mode.

1. Select the Composite Menu button located on the control column.

The Composite Configuration Management popup appears.

2. In the Composite Objects selection list, Select the IP SFTM Trunk composite. (You may need to scroll down through the list of objects to see it.) In the Composite Name field, type a name for the trunk. Select the Create button, and Place the icon.

The IP SFTM trunk icon is created.

1. In node view, get Info on the IP SFTM trunk icon.

The Top Level IP SFTM Trunk Info popup appears.

2. Select the UDP/IP PFD component.

The IP PFD Configuration popup appears.

3. In the Remote IP Address field, enter the IP address you assigned to the analog voice module.

4. In the Remote Node Number field, enter 3001, which is the default node number for the analog voice module.

Note: In analog voice software releases prior to r01.01.00, the default node number is 4090.

5. To update the database and implement the configuration changes, Popdown to the Top Level IP SFTM Trunk Info popup and Select the Update & Config button.

1. In node view, get Info on the IP SFTM trunk icon.

The Top Level IP SFTM Trunk Info popup appears.

2. Select the Packet Stream component.

The Packet Stream Configuration popup appears.

3. Set the CIR (bps) to 8000.

4. Set the Class of Service to Privileged.

5. To update the database and implement the configuration changes, Popdown to the Top Level IP SFTM Trunk Info popup and Select the Update & Config button.

1. On the control column, click the Goto button.

2. In the Node No. field, type 3001 and Select View. (If you are using a software release prior to r01.01.00, type 4090 as the node number.)

A screen similar to the following appears. This screen has an analog voice module with four channels and a Virtual Network, VNET 127. To configure or monitor any of these objects, move your mouse over an icon, and click the right mouse button.

Setting Up the Analog Voice Channels

• Configuring Standard Voice Channels

• Configuring the Voice Call Handler Component

• Configuring the Channel Driver Component

Configuring Standard Voice Channels

1. Get Info on the channel.

The Top Level XV Voice Channel Info popup appears.

2. To name the channel icon, Select the General Information button, type a name in the Name entry field, then Select the Update General Information button.

The name appears beneath the icon in the view.

3. To set addresses and address translation, Select the Network Interface component.

To assign an address,

a. on the Network Interface Configuration popup, Select the Addresses button.

b. On the Addressing popup that appears, which is the same as for all subscribers, enter the address for this voice channel. This is generally the dial number for this port.

Designate the address as the Main address; typically, you specify that the address can handle calls to and from the address. To do so, click on the Address Usage field with your left mouse button until Route TO and FROM Address displays.

c. Click Add and Popdown to the Network Interface Configuration popup.

To set up address translation,

a. on the Network Interface Configuration popup, Select the Address Translation button.

b. On the Address Translation popup that appears, which is the same as for all subscribers, enter one set of the following address translation rules if they are not already present. To do so, type each string in the Rule Definition field and click Add Rule.

· If this voice channel connects to a telephone, or if this channel does not need to pass the dialed digits through to the attached equipment for any other reason, use the following rules:

O:{+}=
I:{+}={1}

The outbound rule strips off the dialed network address before forwarding the call user data. The inbound rule forwards all received data, for use in directing the call.

· If this voice channel connects to a PBX or to other equipment that needs to receive the dialed digits, use the following rules:

O:{+}={1}
I:{+}={1}

The outbound and inbound rules forward all the dialed or received data to the attached equipment, for use in directing the call.

Note: For complete guidelines on creating and using address translation rules, including modifying address translation rules to fit your site, see the Network Management System User Guide.

4. To update the database and implement the configuration changes, Popdown to the Top Level XV Voice Channel Info popup and Select the Update & Config button.

You can configure the channel to be one of the following:

• Dialed address channel lets you dial up a connection at any time by entering an address. This is the recommended configuration for most voice/fax channels. To set up this channel type, proceed to Configuring a Dialed Address Channel.

• Automatic ring down channel connects to a specified address when the receiver goes off hook. This is the recommended configuration for STDM data channels, to connect to a remote STDM port when a scheduled Connect Request occurs. To set up this channel type, proceed to Configuring an Automatic Ring Down Channel.

• Hoot-n-holler channel keeps up a constant connection to a specified address, or raises a connection to a specified address at preset times of day. To set up this channel type, proceed to Configuring a Hoot-n-Holler Channel.

1. In node view, get Info on the appropriate channel.

2. Select the Voice Call Handler component.

The Voice Call Handler Configuration popup appears.

3. In the Voice Channel Type selection list, Select the Dialed Address option.

The parameters required for this channel type are activated on the popup.

4. Type the following dial plan rule in the Dial Rule Definition entry field (adjust the number of $ placeholders to reflect your dialing scheme). Enter the maximum numbers of digits that telephone numbers in your network may have.

$$$$$$$$$$$=DONE 

This example tells the channel to collect 11 digits dialed and use them as the address to make the call. If you dial less than this number of digits, the channel attempts the call after the amount of time specified on the Interdigit Timeout slide lever.

Note: For security reasons, you may prefer to make more specific dial rules. See the online help on the Dial Rule Definition field for assistance and a discussion of dial rule syntax.

5. Select the Add Rule button.

The rule appears in the Dial Plan Rules selection list. If your rule syntax was incorrect or if you use lower-case letters, an error message appears.

When a user finishes dialing a call, the channel compares all dial rules to the entered digit(s), starting with the top rule on the list and working down to the last rule. The channel uses the first rule that matches the address string for that address.

6. To add other dial rules, enter the rule in the Dial Rule Definition field and then Select the Add Rule button.

7. To test the configured rules, perform the following steps:

a. Select the Test Rules button.

The Dial Plan Rule Test popup appears.

b. Select the numbers of the address on the popup's keypad. After you select each digit, the software tests all dial rules in the list with the digits accumulated thus far. It tests the top rule first, then the second rule, and so on to the bottom rule. As soon as the address matches a configured rule, that rule is highlighted. Select the Clear Test button to clear the Digits Dialed entry field.

You can also enter the digits of a possible address in the Digits Dialed entry field, and the address is tested when you press Return.

c. Popdown the Dial Plan Rule Test popup.

8. To increase or decrease the amount of time the Voice Call Handler waits between dialed digits, adjust the Interdigit Timeout slide lever. If this time period elapses before a caller enters another digit or digits match a dial rule, the call attempt is canceled and the caller hears a rapid busy tone.

9. To update the database and implement the configuration changes, Popdown the Voice Call Handler Configuration popup and Select the Update & Config button.

Configuring an Automatic Ring Down Channel

1. In node view, get Info on the appropriate channel.

The channel's top level info popup appears.

2. Select the Voice Call Handler component.

The Voice Call Handler Configuration popup appears.

3. In the Voice Channel Type selection list, Select the Auto Ring Down option.

The parameters required for this channel type are activated on the popup.

4. In the ARD Address to Call field, type the address to be automatically dialed when the phone goes off hook (voice/fax) or when the channel receives a Connect Request (STDM).

Note: Ensure the address entered here is the address of a compatible channel type—for example, another voice/fax channel.

5. To update the database and implement the configuration changes, Popdown the Voice Call Handler Configuration popup and Select the Update & Config button on the top level info popup.

Configuring a Hoot-n-Holler Channel

• Set up a Time Manager statistic designating the times of day you want the call to go up and go down.

• Define a hoot-n-holler action that launches a call to a specific address, and define an action that ends the call.

• Create a primitive that triggers the correct hoot-n-holler action to occur at the time specified in the Time Manager statistic.

Setting Up a Time Manager Statistic

1. In node view, get Info on the appropriate channel.

The channel's top level info popup appears.

2. On the Top Level Voice/Fax Channel Info popup, Select the Time Manager component.

The Time Manager Info popup appears. (For detailed information on the Time Manager, refer to The Time Manager, Chapter 14 of Configuring and Monitoring a Network Exchange 2200.)

3. Select the Time of Day Statistic 1.

The Time of Day Statistic Schedule Events popup appears.

4. Set the time you want the call to be launched as follows:

• In Statistic Value, select 1.

• In Hour and Minute, designate the exact time you want the call launched.

• In Weekdays, select the days of the week you want the call to be launched (for every day, select all of the days).

• To launch the call on a certain day of the month, designate that day in the Day slide lever (otherwise leave it at 0).

• To launch the call only in a specific month, designate that month in the Month slide lever (otherwise leave it at 0).

• Then select the Add Event button.

This Time of Day statistic has its first event, which gives this statistic a value of 1 at the designated time.

5. Set the time you want the call taken down as follows:

• In Statistic Value, select 0.

  · In Hour and Minute, designate the exact time you want the call to be taken down.

  · In Weekdays, Day and Month, keep the same values you used in the previous step.

  · Select the Add Event button.

  This Time of Day statistic has its second event, which gives this statistic a value of 0 at the designated time.

6. To update the database and implement the configuration changes, Popdown to the Top Level Voice/Fax Channel Info popup and Select the Update & Config button.

1. On the Top Level Voice/Fax Channel Info popup, Select the Voice Call Handler component.

The Voice Call Handler Configuration popup appears.

2. In the Voice Channel Type selection list, Select the Hoot-n-Holler option.

The parameters required for this channel type are activated on the popup.

3. In the HnH Action Definition entry field, create the hoot-n-holler "Action 0" by typing the action using the following format, then selecting the Add Action button:

ACTION0 BEGIN nnnnnnn XXXX

where nnnnnnn is the address/phone number you want this call to connect to.

and where XXXX is the voice call algorithm used for the call. For a description of these voice call algorithms, see Section 9.2.5.1 Voice Call Algorithms in Chapter 9 of Configuring and Monitoring a Network Exchange 2200.

The action appears in the Hoot-n-Holler Actions selection list.

4. In the HnH Action Definition entry field, create the hoot-n-holler "Action 1" by typing in the action using the following format, then selecting the Add Action button:

ACTION1 END

The action appears in the Hoot-n-Holler Actions selection list. Once you set up the required primitive, the call to the address designated in Action 0 ends when the Time Manager statistic you created in the previous section reaches a value of 0.

Note: You can create up to 10 hoot-n-holler actions. You only need to create one "END" action that can be used to terminate any active HnH call.

1. On the Top Level Voice/Fax Channel Info popup, Select the General Information button. On the General Information popup, Select the Primitive Information button.

The Status Configuration Information popup appears.

2. Create the RLX Action0 primitive by entering the following values:

• In the Primitive Name field, type RLX Action0.

• In the Dividend Statistic list, select Time of Day Statistic 1.

• In the Divisor Statistic list, select Global Statistic 100.

• Leave the thresholds and weights at their default values (minor 1 and major 10).

• Under Transitions select Green -> Yellow.

• Under Transition Actions select VCH Action 0.

• Under Transitions select To Green.

• Under Transition Actions select VCH Action 1.

• Select the Add button.

The primitive name appears in the Status Primitives selection list. This primitive activates the hoot-n-holler action that you set up in the previous section.

3. To update the database and implement the configuration changes, Popdown to the General Information popup and Select the Update General Information button.

4. To update the database and implement the configuration changes in the Voice Call Handler, Popdown to the Top Level Voice/Fax Channel Info popup and Select the Update & Config button.

Configuring the Channel Driver Component

Note: Channel Driver Parameters for Analog Voice describes each field on the Channel Driver Configuration popup. In addition, the online help text describes these fields.

The Channel Driver Configuration popup appears.

Notes:

• On some NetrixView 2000 screens you may have difficulty viewing the lowest fields on the Channel Driver Configuration popup. If this occurs, you can resize the screen until it displays the full popup. Simply pull the right side of the window closer in, making the window narrower. Continue until the entire popup is visible.

• Initially, you should leave most of these parameters at their default settings, adjusting them as necessary to improve system performance. However, there are certain parameters that you should set according to your network configuration. These parameters are listed below.

1. Set the Electrical Protocol for the voice channels. Make sure this setting is compatible with the protocol used on the analog equipment to which this channel connects. (The equipment at the remote end of the call, however, is not required to use the same electrical protocol used at this end.)

2. Set the telephone Signaling Protocol to use on this channel.

3. In the Dial Method to the PBX field, specify how to transmit to the PBX, dialing digits received from the switch.

4. In the Dialing Method from the PBX field, specify how to handle dialing digits received from the PBX.

5. Set the Voice Compression Algorithm used for calls originating on this channel.

Note: For calls between two ports configured with different voice call algorithms, the algorithm on the initiating side of the call is used for the call.

You can control the amount of delay through the voice-compression engine. A higher delay causes more data to be packed into each frame, resulting in a lower frame-switching rate, without appreciably noticeable delay to the human ear.

• If you use ACELP as the voice compression algorithm, you can control the delay using ACELP Additional Frame Delay.

• If use Low Bit Rate ACELP, you can control the delay using the Low Bit Rate ACELP Additional Frame Delay.)

6. Use the Transmit and Receive TLP (Transmission Level Points) fields to control the quality and volume of the audio sent to and received from external voice equipment.

7. In the Echo Canceller Enable field, specify whether or not to run echo cancellation software on this channel. If the PBX supplies echo suppression, select disable.

8. In the Echo Suppression field, specify whether Silence or recorded Background Noise is played during echo suppression. On connections with high levels of background noise, you may wish to specify Background Noise. This helps avoid complete dropouts of far-end audio received during near-end speech.

9. In the Maximum Fax Rate field, set the maximum rate at which this channel processes faxes.

• For shorter fax transmission time, select a higher rate.

• To give voice and data traffic preference over faxes, select a lower rate.

• To prohibit faxes on this line, use No FAX Allowed

10. To compensate for delay variances in the network, adjust the Maximum Jitter Compensation Delay.

11. If you plan to allow oversubscription of ACELP calls on SFTM trunks in the network, enable Voice Activity Detection. This feature takes advantage of the naturally-occurring silences in voice calls to enable a single 64 kbps trunk, for example, to carry up to 12 voice calls at one time.

12. To update the database and implement the configuration changes in the Channel Driver, Popdown to the Top Level Voice/Fax Channel Info popup and Select the Update & Config button.

Electrical Protocol

Signaling Protocol

Delay Dial	Timing for this protocol is similar to wink start timing, except that the E lead response to M lead assertion is not a pulse of fixed length. Instead, Delay Dial asserts the E lead immediately after seeing the M lead assertion, and the E lead remains asserted until dialing begins. The voice path is established once the local exchange of delay pulses takes place. (If no delay pulses are exchanged within 14 seconds, the voice path is automatically opened.)
Immediate Start	Provides a complete audio path to the far end during dialing, allowing you to hear call progress tones from the far end. You typically use this protocol on ports connecting to PBXs, keysystems, or central offices (unless the PBX or central office uses Wink Start). The PBX, keysystem, or central office generates the ringback tone to the call's originating end.
Hoot and Holler	Also known as Transparent, this option lets you allocate bandwidth for transparent connections during scheduled time periods. Transparent connections are useful for applications where a permanent connection without call setup time is desired. During scheduled times, the analog voice module maintains a constant voice path and passes E&M leads transparently through the analog voice module.
Wink Start	When the trunk side wishes to seize the line, it asserts the M lead, then waits for a "wink" pulse on the E lead to acknowledge completion of the connection. Wink pulses range from 140 to 200 milliseconds in length. The voice path is established once the local exchange of wink pulses has taken place. (If no wink pulses are exchanged within 14 seconds, the voice path is automatically opened.)
Phone	Generates a ringing signal on the call's originating channel and does not complete the audio path until the far end is off hook. This protocol does not pass digits to the connected telephone equipment. Therefore, we recommend it for use with telephones, but not with other analog equipment, such as a PBX or keysystem. (This is a Nx Networks proprietary protocol.)

Incoming M-lead Treatment

Normal	Does nothing.
Inverted	Changes the polarity
Set	Always shows M-lead present.
Clear	Always shows M-lead absent.

Busy Indicator

Immediate Off Hook

Enable	The analog voice module goes off-hook toward the originating PBX immediately after detecting off hook. Must be enabled for FXS
Disable	The analog voice module does not send an answering off hook signal until it receives a Call Answered signal from the remote end of the call.

Flash Enable

Enable	The analog voice module interprets M-lead disassertion longer than the minimum flash detect time and shorter than the disconnect time as hook flashes and generates a flash message toward the switch when this occurs. It also generates an E-lead disassertion of the flash generate time when it receives a flash message from the switch.
Disable	The analog voice module does not generate or detect flashes.

Flash Generate Time

Minimum Flash Detect time

Call State Answered (CSA) Enable

Call State Answered pulses and generates a CSA message toward the switch when this occurs. It also generates an E-lead disassertion of the CSA generate time when it receives a CSA message from the switch.

If disabled, CSA pulses are not generated or detected.

CSA Generate Time

Min CSA Detect Time

Wink Generate Time

Dial Tone Enable

Dial Method to the PBX

Specifies how to handle dialing digits the analog voice module receives from the switch. Digits arrive from the switch with an indication of whether they were generated as pulse or tone digits. You can override that indication using this parameter. A setting of transparent tells the analog voice module to reproduce the digits as the switch generated them.

Disabled	Disable dialing to the PBX. The analog voice module does not generate or transmit dialed digits.
Pulse Only	Overrides the indication set on incoming digits to pulse digits.
Tone Only	Overrides the indication set on incoming digits to tone digits.
Transparent	Reproduces the digits as they were entered, pulse or tone.

Dialing Method from the PBX

Both	Accept both tone and pulse digits.
Tone Only	Ignore pulse digits.
Disabled	Ignore all dialed digits.
Pulse Only	Ignore tone digits.

Initial Dial Delay

Dial Pulses per Second

Pulse Inter-digit Time

Pulse Dial Make Ratio (%)

Tone Dial Detect Time

Tone Dial Generate Time

Tone Dial Inter-digit Time

Voice Compression Algorithm

You can set up a voice channel to use a voice call algorithm for transmission of calls via trunks connected to other 3000 Series voice modules or to Network Exchange 2200 voice equipment. Uncompressed voice channels use 64 kbps of bandwidth; voice compression allows information to be packed into smaller bandwidth.

When the analog voice module receives a voice call on a channel configured for voice compression, it compresses the voice signals according to the configured algorithm, and switches them through the network. The analog voice module on the other end of the call decompresses the voice signals before passing them to the voice subscriber at its end of the call.

Note: For calls between two ports configured with different voice call algorithms, the algorithm configured on the initiating side of the call is used for the call, and overrides the algorithm configured on the receiving side of the call.

TDHS	Time Domain Harmonic Scaling (TDHS) detects and compresses pitch periods (digitized samples of speech). Also includes linear prediction coding (LPC) to further compress the amount of information needed to transmit and recreate the signal. (Actual throughput bandwidth levels vary between 9 kbps and 12 kbps, depending on break factor configuration and system load.) Calls using the TDHS voice call algorithm are assigned the TDHS class of service.
4.8Kbps CELP OR 7.4Kbps CELP	CELP (Code Excited Linear Prediction) breaks a fixed-size sampled-input signal into blocks of samples that are processed according to standard models. The speech signal's format structure and pitch are modeled and quantified for recreation at the remote end. You can set the bandwidth of the compressed voice stream to 4.8 kbps or 7.4kbps. Calls using either of these settings are assigned the CELP class of service. 4.8 kbps CELP—Allocates a CIR of 6.2 kbps (this includes the overhead required by the 2200 network). 7.4 kbps CELP—Provides a voice quality that is slightly higher than 4.8 kbps. Allocates a CIR of 9 kbps (this includes the overhead required by the 2200 network).
ACELP OR Low Bit Rate ACELP	ACELP (Algebraic Code Excited Linear Prediction) is the highest quality compressed voice call algorithm. ACELP is a modified CELP algorithm that supports silence suppression (also known as voice activity detection, or VAD), so that trunk bandwidth is used only when active speech is present. This permits more efficient use of network bandwidth by allowing voice or data calls to make use of the bandwidth that naturally occurring silences in voice calls provide. You can use either of the following ACELP settings. Low Bit Rate ACELP—the peak bandwidth of the compressed voice stream is 5.5 kbps. Allocates a CIR of 7.4 kbps (this includes the overhead required by the 2200 network). ACELP—the peak bandwidth of the compressed voice stream is 8 kbps. Allocates a CIR of 9.8 kbps (this includes the overhead required by the 2200 network). Calls using the ACELP or Low Bit Rate ACELP voice call algorithm are assigned the CELP class of service.
H.323 (PCM)

Transmit and Receive TLP

Note: The default values (-5 Transmit, 0 Receive) are considered reasonable starting points when configuring TLP levels on voice channels.

Figure 1 shows example Transmit and Receive TLP settings.

Figure 1 Transmit and Receive TLP

In this example,

• the voice signal coming from the voice equipment to 3000 Series A is -12 dBm. The Receive TLP is set to -12 dBm, which causes the analog channel to expect to receive voice signals with an average range of -12 dBm, and correct accordingly to ensure that voice signals travel across the network at 0 dBm. (This means that when you set the Receive TLP to a negative value, the analog voice module applies gain to the signal.)

• the voice equipment attached to 3000 Series B expects a voice signal of -16 dBm. The Transmit TLP is set to -16 dBm. This setting causes the analog voice module, which expects to receive voice signals from the network at 0 dBm, to attentuate the signal 16 dBm to send them to the voice equipment at -16 dBm.

  Note: Because the receive TLP is specified as the measured receive level that requires correction, negative numbers represent the gain that must be applied to bring a low value up to zero, while positive numbers represent the attentuation that must be applied to bring a high value down to zero.

It is important to remember that incorrect settings at one end of the call may result in symptoms that seem to point to incorrect settings at the other end. In some cases, slight (1 to 2 dBm) adjustments are necessary to optimize voice quality or add small amounts of gain or attenuation to suit the users' preferences. In this case, you should first ensure that the Transmit and Receive TLPs on both sides of the call are set correctly to correspond with the requirements of the connected phone equipment. Then you can make adjustments as follows:

If the user hears . . .	then . . .
good quality, but at a low level	increase the Transmit TLP on the local analog voice module.
good quality, but at a high level	reduce the Transmit TLP on the local analog voice module.
poor quality at a low level	increase the Receive TLP on the remote analog voice module.
poor quality at a high level	increase the Receive TLP on the remote analog voice module.
poor quality at an appropriate level	reduce the Receive TLP at the remote analog voice module, and reduce the Transmit TLP at the local analog voice module.

For more detailed information, see section 9.2.4.1.3 Adjusting the Volume on the XV Channel in Chapter 9 of the Configuring and Monitoring a Network Exchange 2200 guide.

Minimum TDHS Brake Factor

Maximum TDHS Brake Factor

• A lower value gives better quality at the expense of using more bandwidth.

• A higher value gives lower quality, but saves bandwidth.

Maximum Fax Rate

• For shorter fax transmission time and better fax quality, select a higher rate.

• To give voice and data traffic preference over faxes, but allow more FAXs on a trunk line at the same time, select a lower rate.

• To prohibit faxes on this line, enter No FAX Allowed.

Hold Time

Maximum Jitter Compensation Delay

A channel handles delay on a line by learning about the line's delay as each frame arrives, and compensating accordingly. If delay is consistent, within a few frames the channel can identify the delay and buffer and deliver the incoming frames as necessary to ensure a smooth voice call. This happens over satellite links, for example. Consistent delay is not a problem for voice calls.

However, when jitter occurs and a burst of frames arrives at the remote end, the receiver has to determine how to handle the frames. It can throw many of the frames away, which keeps perceived delay low, but risks dropping pieces of the call. Or it can retain the frames in a buffer and deliver them in order, which increases the delay but reduces voice distortion.

Adjusting for Delay

• On lines that rarely experience jitter, set this delay to a low value, such as 100 milliseconds. This prevents occasional jitter from adding delay. If jitter does occur, the analog voice module stores 100 ms worth of the bursting frames in a buffer and discards the rest. If the backup is brief and does not involve many frames, the voice call algorithm can compensate and voice quality is not compromised. If a significant number of frames are backed up, however, a larger group of frames are discarded and speech quality may be temporarily degraded.

• On lines that experience jitter more often, such as low-speed frame relay or Internet connections, set this delay to a higher value, such as 500 milliseconds. If jitter occurs, the analog voice module stores more of the bursting frames in a buffer and discards fewer frames. This results in a higher delay, but few or no voice gaps resulting from discarded frames.

If you plan to allow oversubscription of ACELP calls on SFTM trunks in the network, enable Voice Activity Detection. This feature takes advantage of the naturally-occurring silences in voice calls to enable a single 64 kbps trunk, for example, to carry up to 12 voice calls at one time.

If you have conditions of frequent, high jitter, we recommend not using Voice Activity Detection on the line.

For more information, see section 9.2.6 Oversubscribing ACELP Voice Calls on SFTM Trunks in Chapter 9 of the Configuring and Monitoring a Network Exchange 2200 guide.