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Table of Contents

MRP Manager

MRP Manager

The Multiservice Route Processor (MRP) is a voice and data-capable router that can carry voice and data traffic over an IP network or can carry voice traffic to public switched telephone network (PSTN). The two slots on the MRP accept WAN interface cards (WICs), voice interface cards (VICs), or voice/WAN interface cards (VWIC) to provide connectivity to the IP network, the PSTN, a Private Branch Exchange (PBX), analog phones, and fax machines.

The MRP card provides both analog and digital voice ports for its implementation of Voice over IP (VoIP). The Cisco WAN Interface Card Hardware Installation Guide and the Cisco ICS 7750 Software Configuration Guide provide additional information regarding the WICs, VICs, and VWICs.

The MRP Manager is a web-based tool in the Cisco ICS 7700 System Manager that is used to configure the MRP. Access to MRP Manager uses the authentication process in the System Manager. You must log in and have configuration privileges to MRP Manager. For more information on configuration privileges, see "Account Management and Security."

No restrictions or special handling is performed by System Manager if more than one user attempts to configure the same MRP. Multiple users can access MRP at any given time and change configurations. The person making the last modification overrides any previous changes.

If more than one person is concurrently making changes, the HTML page might not reflect the latest changes until the page is refreshed or reloaded from your web browser. The same could be true if MRPs are changed or removed.

MRP Manager configures the following features:

Cisco CallManager Integration

The Cisco CallManager views the MRP as an H.323 gateway device. H.323 is the International Telecommunications Union (ITU-U) umbrella standard for the set of standards defining real-time multimedia for packet-based networks. This is also referred to as IP telephony.

When an MRP is installed in the Cisco ICS 7750, MRP Manager automatically adds the H.323 gateway in the Cisco CallManager. You must configure this device using the specific route patterns that the Cisco CallManager will use to forward calls to this MRP. Once you have configured the route patterns in the Cisco CallManager, specify route patterns for the MRP using the MRP Manager Voice Ports feature by selecting the route pattern that corresponds to each port.


Note   Changes made in Cisco CallManager do not update the MRP configuration. If you change or delete route patterns, directory numbers or hostnames in Cisco CallManager, you must update this information in MRP Manager.

If an MRP is physically removed from the system, the H.323 gateway device information remains. You can then move the MRP to a different slot position without losing the specific MRP information. If you permanently remove the MRP, you must manually remove the H.323 gateway device from Cisco CallManager.

MRP Manager and Cisco CallManager Shared Information

Cisco CallManager and MRP Manager share some of the configuration information. However, if the shared data is changed or deleted in Cisco CallManager, the same must be done in MRP Manager. The shared information is described in this section.

Route Patterns

For each route pattern configured for an MRP in Cisco CallManager, the route pattern should also be configured on the MRP voice ports so that the MRP will route calls to the outside world.

Route patterns must be configured in Cisco CallManager before assigning them to a MRP by using MRP Manager. If you change or delete the route patterns in Cisco CallManager, you must reflect those changes in MRP Manager.

The PBX connected to an foreign exchange office (FXO) or recEive and transMit (E&M) port is normally a range of phone numbers handled by that PBX. For example, the route pattern could be 555... to represent all phone numbers in the 555 exchange. The PSTN connected to an FXO port is a number that is used to access the outside world and is typically set to 9.

The foreign exchange station (FXS) uses the phone number for the analog phone or fax connected to this port. This should be a phone number that CallManager is routing to this MRP. That is, it should be a phone number within one of the defined route pattern ranges.

Directory Numbers

The directory numbers in Cisco CallManager are the telephone numbers for the IP Phones. If someone uses an analog phone to call an IP phone managed by Cisco CallManager, all MRPs connected to a PSTN need to route these directory numbers (IP phone numbers) to Cisco CallManager by using VoIP. These routes can be configured by using Voice Port Management (VoIP Dial Peers) in MRP Manager.

Hostname

When an MRP is physically added to the system, MRP Manager automatically creates an H.323 gateway device in Cisco CallManager and uses the Ethernet IP address of the MRP in the Device Name field. You should not change this value in Cisco CallManager, as this would require manually entering the change in MRP Manager, as well.

Physical Data and Voice Connections

A WIC connects the MRP to a data network. A VIC connects the MRP directly to an analog phone, a fax (FXS), PSTN (FXO), or to an analog PBX. A VWIC supports both data and voice connections.

The multiflex trunk T1 (MFT-T1) interface card provides both voice and data connections. There are two versions of the MFT-T1 card: the single-port T1 interface card (1MFT-T1 VWIC) and the dual-port T1 interface card (2MFT-T1 VWIC).

MRP Manager can configure the ports in two modes:

  In the case of data connections, MRP Manager cannot support more than one channelized group. If you specify a range less than 24 channels within the channelized group, the rest of the channels are not used.

Accessing MRP Manager

To access MRP Manager, do the following:


Step 1   Click Configure on the System Manager home page to display the Configure page.

The Configure page is shown in Figure 3-1.


Figure 3-1: Configure Page


Step 2   Select MRP Manager from the secondary tab or the blue-highlighted link. Figure 3-2 shows the MRP Manager main page.


Figure 3-2: MRP Manager Main Page



Feature and Management menus contain related links that can be used to view or configure the features of the selected MRP.

Effectiveness of Changes

When you change or enter a value by using MRP Manager, the change is effective immediately on the selected MRP. Commands are generated only for those fields that you change in the web form, as opposed to regenerating and sending the entire Cisco IOS configuration to the MRP. However, because web pages are static, the information on a page does not reflect entries or changes until the next time you load the web page. For example, if you hot-swap the MRPs, you cannot view any configuration changes until you refresh or reload the page.

Selecting an MRP

When you select an MRP, MRP Manager checks to see if anyone has accessed the device since the system bootup. If you are the first to access the device since bootup, MRP Manager prompts you to read the existing Cisco IOS configuration from memory.

To select an MRP, do the following:


Step 1   Click the down arrow to display the Current MRP drop-down list.

Step 2   Select an MRP from the list. The page refreshes, displaying the selected MRP.


The Basic option in the Features group General list displays the basic information for the MRP, and you can change the name of the device.

Changing the Name of an MRP

By default, the General list in the Features group is open. The Basic option displays by default and shows the device model and hostname for the selected MRP, as shown in Figure 3-3. You can change the device name by using this page.


Figure 3-3:
Basic Information


To change the device name, do the following:


Step 1   Delete the existing entry.

Step 2   Enter the new entry in the Name field. The limitations of the string are described in the "Device Name Rules" section.

Step 3   Click Submit. A page displays, indicating the Cisco IOS configuration has been delivered to the MRP.

Step 4   Click Continue to refresh the page. The basic device information, including the new name, is displayed.


Device Name Rules

The following rules apply to device names:

Enabling DNS and Web Access

Use the Services page to optionally enable Domain Name System (DNS) or Web access for the selected MRP. DNS translates domain names or network node names into IP addresses.

Enabling the MRP as a Web (HTTP) server allows you to point your browser to the IP addresses of a MRP to view the built-in web pages on the device. You do not need to enable this feature if you plan on using only the MRP Manager to configure your MRP.

To enable DNS lookup:


Step 1   Click Services in the Features list to display the Services page, as shown in
Figure 3-4.


Figure 3-4: Services Page


Step 2   Check Enable IP domain name lookup.

Step 3   Enter the IP address of the DNS server in the Domain Name Server IP Address field. This field is required if you enable DNS lookup. Contact your network administrator or service provider to obtain the DNS server IP address.

Step 4   Click Submit if the status of the device as a Web server is acceptable, and a message displays, indicating the configuration is being delivered to the device. Otherwise, continue to the next procedure.

Step 5   Check Enable Web (HTTP) server on device to enable browser access.

Step 6   Click Submit to complete the procedure.

Step 7   Click Continue to refresh the page.


SNMP Device Management

Simple Network Management Protocol (SNMP) is a protocol used to manage networks through a limited set of management commands and responses. The MRP contains managed objects, called MIB variables, arranged in a database called a management information base (MIB). MIB variables define the properties of the managed device.

The management system issues SNMP requests to the managed device (in this case, the MRP) to retrieve object information, such as the status of a port on the device. The agent replies with a response that includes the requested data. The managed device's agent can also send an unrequested event notification, called a trap, to the SNMP management system to identify the occurrence of certain conditions, such as excessive traffic.

Displaying SNMP Information

To view SNMP information, click SNMP in the Features list to display the SNMP page, as shown in Figure 3-5.


Figure 3-5: SNMP Page


Reading the Hardware Configurations

The read-only Hardware Configuration page displays the current hardware configuration of the selected MRP. Information shown includes the WICs, VICs, and VWICs detected by the system.

To display the Hardware Configuration page, shown in Figure 3-6, click Hardware Config. in the Features list.


Figure 3-6: Hardware Configuration Page


Voice Port Management

The Voice Ports page in the Features menu Voice list is used to configure voice ports on a VIC or WVIC. You can add and delete channel groups, enter controller settings for your T1 link, and modify port settings.

Displaying Voice Ports

To display the Voice Ports page, do the following:


Step 1   Click Voice in the Features list to display the voice options.

Step 2   Click Voice Ports to display a list of the ports, as shown in Figure 3-7.


Figure 3-7: Voice Ports Page



Configuring Controller Settings

Controller settings configure the properties of a MFT-T1 controller. The MFT-T1 controller ports are numbered as follows:

0/0—MRP slot 0, MFT-T1 port 0.

0/1—MRP slot 0, MFT-T1 port 1.

1/0—MRP slot 1, MFT-T1 port 0.

1/1—MRP slot 1, MFT-T1 port 1.

To configure the settings for MFT-T1 controller, do the following:


Step 1   In the Features list, click Voice to display the voice options.

Step 2   Click Voice Ports to display a list of the ports, as shown in Figure 3-7.

Step 3   Click Controller Settings to display the Controller Settings page, shown in Figure 3-8.


Figure 3-8: Controller Settings Page


Step 4   From the TDM Clock drop-down list, select the internal time-division multiplexing (TDM) clock. Options for a clock source are as follows:

Export Line—TDM clock for this controller is supplied by the line to which this controller is connected (PBX or PSTN). This controller may also provide clock to other controllers.

Import T1 slot/port Line—TDM clock for this controller is supplied by the line clock of the exporting T1 0/0 controller. The line clock is provided by the network interface (NI).

  Import T1 slot/port Internal—TDM clock for this controller is supplied by the line clock of the exporting T1 0/0 controller. This controller provides clock to the line to which this controller is connected.
  Import onboard Internal—TDM clock for this controller is supplied by the onboard clock (Cisco ICS 7750 chassis). This controller provides clock to the line to which this controller is connected.

Step 5   Select the external clock from the External Clock Source drop-down list. The options are as follows:

  Line—The port uses the clock signal produced by an external device on the line (for example, the PSTN).
  Internal—The port exports the internal clock signal to the line. The source of the internal clock signal is determined by the Internal TDM Clock setting.

Step 6   Select the payload type from the Payload Type drop-down list.

Step 7   Select framing format from the Frame Type drop-down list that matches that of the PBX or the central office that connects to the digital T1 interface. The MRP supports two types of framing for T1: ESF (Extended SuperFrame) and SF (SuperFrame, also called D4 framing), described as follows:

  Extended Superframe (ESF)—24 frames of 192 bits each, with the 193rd bit providing timing and other functions. ESF is an enhanced version of the Superframe (SF) format.
  SuperFrame (SF)—12 frames of 192 bits each, with the 193rd bit providing error-checking and other functions. SF is superceded by ESF, but is still widely used.

Step 8   Select the line code from the Line Code drop-down list. Use the line encoding specified by your service provider. The options are as follows:

  Bipolar-8 zero substitution (B8ZS)—encodes a sequence of eight zeros in a unique binary sequence to detect line coding violations.
  Alternate mark inversion (AMI)—represents zeros using a 01 during each bit cell, and ones are represented by 11 or 00, alternately, during each bit cell. AMI requires that the sending device maintain ones density. Ones density is not maintained independent of the data stream.

Step 9   Select the cable length for your T1 link from the Cable Length drop-down list based on the closest value available. The cable length setting must conform to the actual cable length you are using. If it is greater than 655 feet, you must also set the pulse gain and pulse rate; otherwise, these two fields are not used.

Step 10   Select the pulse gain for your T1 link from the Pulse Gain drop-down list. The options for this field are as follows:

Step 11   Select the pulse rate for your T1 link from the Pulse Rate drop-down list. The options for this field are as follows:

Step 12   Click Submit. A message displays, indicating the configuration has been sent to the router.

Step 13   Click Continue to refresh the page.


Adding Channel Groups

Channel groups combine channels as determined by your telephone service provider or PBX administrator. The timeslots specify which range of the 24 channels you want in this group. A channel can belong to only one channel group. The Type field determines the voice port type (FXO, FXS, or E&M) as well as the signal type (for example, loop start, ground start, and so forth).

To configure the settings for MFT-T1 controller, do the following:


Step 1   In the Features list, click Voice to display the voice options.

Step 2   Click Add Voice Channel Group to display the Add a Channel Group page, shown in Figure 3-9.


Figure 3-9: Add a Channel Group


The following sections describe the Add a Channel Group page options.

Step 3   In the Group Number field, enter the group number.

Step 4   In the Timeslots field, enter the range of timeslots. For T1, allowable values are within the range from 1 to 26. This value must indicate one start and end range of timeslots, for example, 1-15.

Step 5   From the drop-down Type list, select the connection type.

The signaling method selection type depends on the connection you are making, as follows:

The options for type are as follows:

  e&m delay—Specifies that the originating endpoint sends an offhook signal and then waits for an offhook signal followed by an onhook signal from the destination.
  e&m immediate—Specifies no specific offhook and onhook signaling.
  e&m wink—Specifies that the originating endpoint sends an offhook signal and waits for a wink signal from the destination.
  fxo ground-start—Specifies FXO ground-start signaling support.
  fxo loop-start—Specifies FXO loop-start signaling support.
  fxs ground-start—Specifies FXS ground-start signaling support.
  fxs loop-start—Specifies FXS loop-start signaling support.

Step 6   Click Add to configure the port.


Configuring Voice Port Route Patterns

A route pattern is a number or a range of numbers. For example, 555.... represents extensions beginning with 555 followed by any four digits (as denoted by one period for each digit).

The VIC voice ports are listed as FXS and E&M in the Voice Port column and have a corresponding route pattern.

  FXS—Enter the phone number of the phone that is connected to this port or channel. This is the number that CallManager routes out to the MRP.
  E&M—If this port is connected to a PBX, select the range of numbers used to access the PBX. Use x as a wildcard digit. The possible values are the route patterns that should already be configured in CallManager for this MRP. For example: 555-xxxx refers to all numbers in the 555 exchange.

If this port is connected to a PSTN, enter the number used to access the PSTN. This number is typically 9. The possible values are the route patterns that should already be configured in CallManager.


Note   Configure the route patterns in CallManager before assigning them to the corresponding voice ports in MRP Manager.

To configure a voice port route pattern, do the following:


Step 1   In the Features list, click Voice to display the voice options.

Step 2   Click Voice Ports to display the Voice Ports page.

Step 3   In the Actions column, click Edit to display the Voice Port edit page, as shown in Figure 3-10.


Figure 3-10: Voice Port Edit Page


Step 4   From the Route Pattern drop-down list, select the route pattern or enter the value in the field.

Step 5   From the Call Progress Tones drop-down list, select the tone pattern.

The format of the tones are used to indicate the state of the call, such as call ringing, line busy, all circuits busy, and so forth. Set the value based on your country and location.

Step 6   From the Signal Type drop-down list, select the signal type.

Step 7   Click Submit. A message displays, indicating the configuration has been sent to the router.

Step 8   Click Continue to refresh the page.


Configuring Voice Ports

This section describes the configuration of voice ports. FXS and FXO voice ports are similar to E&M voice ports, but they do not have Interface Type and Cabling Scheme fields. MFT voice ports are similar to E&M voice ports, but they do not have Signal Type, Interface Type, and Cabling Scheme fields.

The voice ports are configured as follows:


Step 1   In the Features list, click Voice to display the voice options.

Step 2   Click Voice Ports to display a list of the ports.

Step 3   Click Voice Port Settings to display the Voice Port page, shown in Figure 3-11.


Figure 3-11: Voice Port Configuration


Step 4   Select Enable Comfort Noise to enable background noise to fill silence gaps if voice activity detection (VAD) is enabled. Enabling this parameter usually improves performance, although some users might perceive truncation of consonants. If comfort noise is not enabled and VAD is enabled at the remote end of the connection, the user hears silence when the remote party is not speaking.

The configuration of comfort noise only affects the silence generated at the local interface; it does not affect the use of VAD on either end of the connection or the silence generated at the remote end of the connection. The default value is enabled.

Step 5   Select Enable Nonlinear Processing (also know as residual echo suppression and is associated with the echo canceller operation) to enable non-linear processing (required). Nonlinear processing shuts off a signal if no near-end speech is detected. Enabling this setting normally improves performance, although some users might perceive truncation of consonants at the end of sentences. The default value is enabled.

Step 6   Select Enable Echo Cancel to cancel voice sent out on the interface that is received on the same interface; sound that is perceived by the listener as an echo. Disabling echo cancellation might cause the remote side of the connection to hear an echo. This process minimally degrades voice quality and should be disabled if it is not needed. Echo cancel does not affect the echo heard by the user on the analog side of the connection. The default value is enabled.


Note   There is no echo path for a 4-wire receive-and-transmit (E&M) interface. The echo canceller should be disabled for that interface type.

Step 7   Enter the Echo Cancel Coverage to specify the coverage size of the echo canceller. Configure the maximum amount of time between voice sent out over the interface and is received back on the same interface for voice cancellation to occur. If the local loop (the distance from the analog interface to the connected equipment producing the echo) is longer, the configured value should be extended. If you configure a longer value for echo cancel coverage, it will take the echo canceller longer to converge; in this case, the user might hear slight echo when the connection is initially set up. If the configured value is too short, the user might hear some echo for the duration of the call, because the echo canceller is not cancelling the longer delay echoes.


Note   There is no echo or echo cancellation on the IP side of the connection. Echo cancel coverage is only valid if echo cancel has been enabled.

Step 8   Enter the Music On Hold Threshold to specify the decibel level of music played when calls are put on hold for this interface. The firmware will pass steady data above the level you specify. It only affects the operation of VAD when receiving voice.

If the value is set too high, VAD interprets music-on-hold as silence, and the remote end does not hear the music. If the value is set too low, VAD compresses and passes silence when the background is noisy, creating unnecessary voice traffic. The values range from -70 to -30 db. The default value is -38 db.

Step 9   Set the Input Gain level for the number of decibels that is inserted at the receiver side of the interface. A system-wide loss plan must be implemented using both input gain and output attenuation. The default value of 0 db assumes that a standard transmission loss plan is in effect, meaning that normally there must be -6 db attenuation between phones. Connections are implemented to provide -6 db of attenuation when the input gain and output attenuation are configured with the default value of 0.

You cannot increase the gain of a signal going out into the public switched telephone network (PSTN), but you can decrease it. If the voice level is too high, decrease the volume by either decreasing the input gain value or increasing the output attenuation. If the voice level is too low, increase the gain of a signal coming into the MRP by increasing the input gain. The values range from -6 to 14 db.

Step 10   Set the Output Attenuation in decibels that is inserted at the transmit side of the interface. A system-wide loss plan must be implemented using both input gain and output attenuation. The default value of 0 db assumes that a standard transmission loss plan is in effect, meaning that normally there must be -6 db attenuation between phones. Connections are implemented to provide -6 db of attenuation when the input gain and output attenuation are configured with the default value of 0.

You cannot increase the gain of a signal going out into the public switched telephone network (PSTN), but you can decrease it. If the voice level is too high, decrease the volume by either decreasing the input gain value or increasing the output attenuation. If the voice level is too low, increase the gain of a signal coming into the MRP by increasing the input gain. The values range from 0 to 14 db.

Step 11   Select the Connection Mode of the interface. Available values include Normal, Trunking and Plar (default).

  Normal—the standard session application creates a dial tone when the interface goes off-hook until enough digits are collected to match a dial peer to complete the call.
  Trunking—specifies a straight tie-line connection to a private branch office (PBX). This setting can be used for E&M-to-E&M trunks, FXO-to-FXS trunks, and FXS-to-FXS trunks. Signaling is transported for E&M-to-E&M trunks and FXO-to-FXS trunks; signaling will not be transported for FXS-to-FXS trunks. The telephone number you enter in the Number field is the destination telephone number for this PBX.
  Plar (private line, automatic ringdown)—associates a dial peer directly with the interface; when an interface goes off-hook, the dial peer sets up the second call leg and creates a conference call without the caller having to dial any digits. The telephone number you enter in the Number field is used as the called number for all calls coming in over this voice port. The destination dial peer is determined on the basis of this called number.

Step 12   Enter the Number, the full E.164 telephone number (maximum of 32 characters), that is used to establish connection with trunking or plar modes. If the connection mode is set to Normal, this parameter does not apply and defaults to null.

Step 13   Specify the Call Disconnect Timeout; the amount of time the system waits when the connection changes from off-hook to on-hook before a disconnect signal is sent from one end to indicate to the other that the transmission connection is no longer needed. The timer is activated when the accepted call is terminated at one end, but not the other. If the configured timeout value is exceeded, the caller is notified through the appropriate tone to terminate the call. To disable call disconnect timeout, set the value to 0 seconds.

Step 14   Specify the Initial Digit Timeout, the amount of time the system waits for the caller to enter the first digit of the destination address. The timer is activated when the call is accepted and is deactivated when the caller enters the first digit. If the configured timeout value is exceeded, the caller is notified through the appropriate tone, and the call is terminated. The values range from 0 to 120 seconds. The default value is 10 seconds. To disable initial digit timeout, set the value to 0 seconds.

Step 15   Specify the Inter Digit Timeout, the amount of time the system waits (after the caller has entered the initial digit), for a subsequent input digit from the caller. The timer is activated when the caller enters a digit and restarted each time the caller enters another digit until the destination address is identified. If the configured timeout value is exceeded before the destination address is identified, the caller is notified through the appropriate tone, and the call is terminated. The values range from 0 to 120 seconds. The default value is 10 seconds. To disable inter digit timeout, set the value to 0 seconds.

Step 16   Set the value of the Call Progress Tones, based on your country and location. The format of the tones that indicate the state of the call such as call ringing, line busy, all circuits busy, and so forth.

Step 17   Select the Signal Type, the protocol used when the phone line goes off-hook and on-hook. It is used between a voice interface on a router and a voice device. Values when configuring port settings for your voice interface card (VIC) include wink-start, immediate, delay-dial. Contact your service provider for this information.

Step 18   Select the Interface Type that best describes your E&M voice port:

  Type 1—The tie-line equipment generates an output signal to the PBX by grounding the output lead. If you select 1, a common ground must exist between the line equipment and the PBX.
  Type 2—The interface requires no common ground between the equipment, thereby avoiding ground loop noise problems. Although type 2 interfaces do not require a common ground, they do have the tendency to inject noise into audio paths because they are asymmetrical with respect to the current flow between devices.
  Type 3—The interface operates the same as type 1 interfaces with respect to the output signal. However, if you select 3, a common ground must be shared between equipment.
  Type 5—Type 5 is quasi-symmetrical in that, while the line is up, current flow is more or less equal between the PBX and the line equipment, but noise injection is a problem.

Step 19   Select the Cabling Scheme to indicate whether you will be using a 2-wire or 4-wire scheme for your private branch exchange (PBX). Contact your service provider for this information.

Step 20   Click Submit. A message displays, indicating the configuration has been sent to the router.

Step 21   Click Continue to refresh the page.


Dial Peers

All of the voice technologies use dial peers to define the characteristics associated with a call leg. A call leg is a discrete segment of a call connection that lies between two points in the connection, for instance, between a telephone and a router, a router and a network, a router and a PBX, or a router and the PSTN. Each call leg corresponds to a dial peer. An end-to-end call comprises four call legs, two from the perspective of the source router, and two from the perspective of the destination router. Dial peers apply specific attributes to call legs and to identify call origin and destination. Attributes applied to a call leg include Quality of Service (QoS), coder-decoder (CODEC), voice activity detection (VAD), and fax rate.

To display the Voice over IP (VoIP) dial peers, do the following:


Step 1   In the Features list, click Voice to display the voice options.

Step 2   Click Dial Peers to display the Dial Peers page, shown in Figure 3-12.


Figure 3-12: Voice Over IP Page



Adding a POTS Dial Peer

A POTS (also known as plain old telephone service or basic telephone service) dial peer associates a physical voice port with a local telephone device.

To add a POTS dial peer, do the following:


Step 1   Click Add POTS Dial Peer to display the POTS Dial Peer page, shown in
Figure 3-14.


Figure 3-13: POTS Dial Peer Page


Step 2   (Optional) Enter a tag in the Tag field. The tag is a unique number used to identify a VoIP or POTS dial peer. The related IOS commands are dial-peer voice dialpeer-tag voip and dial-peer voice dialpeer-tag pots. Refer to the Cisco IOS Software Command Summary for more information.

Step 3   In the Route Pattern field, enter the route pattern. The route pattern defines the telephone number associated with the POTS dial peer. See the "Configuring Voice Port Route Patterns" section in this chapter for additional information.

Step 4   In the Prefix Digits field, enter the prefix digits.

Step 5   From the Voice Port drop-down list, select the voice port. The voice port associates the POTS dial peer with a specific logical dial interface, normally the voice port connecting your router to the local POTS network.

Step 6   Click Add to configure the port.


Adding a Voice Over IP Dial Peer

To add a Voice Over IP dial peer, do the following:


Step 1   Click Add VoIP Dial Peer to display the Voice over IP Dial Peer page, shown in Figure 3-14.


Figure 3-14: Voice Over IP Dial Peer Page


Step 2   (Optional) In the Tag field, enter a tag. The tag is a unique number used to identify a VoIP or POTS Dial Peer. The related IOS commands are dial-peer voice dialpeer-tag voip and dial-peer voice dialpeer-tag pots. Refer to the Cisco IOS Software Command Summary for more information.

Step 3   In the Route Pattern field, enter the route pattern. The route pattern defines the telephone number associated with the VoIP dial peer. See the "Configuring Voice Port Route Patterns" section in this chapter for additional information.

Step 4   In the Target IP Address field, enter the target IP address.

Step 5   Select the CODEC from the Codec drop-down list. CODEC transforms analog signals into a digital bit stream and digital signals back into analog signals. CODEC values determine how much bandwidth the voice session uses.

Step 6   From the IP Precedence drop-down list, select the precedence. The IP precedence command gives voice packets a higher priority than other IP data traffic. The default IP precedence is set to 5.

As the precedence value increases, the algorithm allocates more bandwidth to that conversation to make sure that it is served more quickly when congestion occurs. weighted fair queuing (WFQ) assigns a weight to each flow, which determines the transmit order for queued packets. In this scheme, lower weights are served first. IP precedence serves as a divisor to this weighting factor. For instance, traffic with an IP Precedence field value of 5 gets a lower weight than traffic with an IP Precedence field value of 3, and thus has priority in the transmit order.

Step 7   Select the Voice Activator Detection (VAD) check box to enable voice activity detection (VAD). VAD disables the transmission of silence packets over the network, so that only audible speech is transmitted. If you enable VAD, the sound quality is slightly degraded, but the connection monopolizes less bandwidth. If VAD is disabled, voice data is continuously transmitted to the IP backbone.

Step 8   Click Add to configure the port.


Discovering VoIP Dial Peers

The Discover VoIP Dial Peers link displays the CallManager directory numbers (Auto Registration Range) as well as directory numbers for each IP phone configured in CallManager. If a VoIP dial peer needs to be added, you are prompted to add it.

To discover a Voice Over IP dial peer, click Discover VoIP Dial Peers. The VoIP Discovery Page displays the result, as shown in Figure 3-15.


Figure 3-15: VoIP Discovery Page


Configuring Number Expansion

In most corporate environments, the telephone network is configured so that you can reach a destination by dialing only a portion (an extension number) of the full telephone number. VoIP can be configured to recognize extension numbers and expand them into their full dialed number by mapping the extension numbers to full phone numbers. Extensions are mapped to full phone numbers by configuring the Number Expansion table.

To configure your number expansion table, do the following:


Step 1   In the Features list, click Voice to display the voice options.

Step 2   Click Number Expansion to display a list of the ports, as shown in Figure 3-16.


Figure 3-16: Number Expansion Table Page


Step 3   Click Add an entry to display the Pattern page, shown in Figure 3-17.


Figure 3-17: Add Number Pattern Page


Step 4   In the Extension Pattern field, enter the extension pattern, following the limitations described in the "Valid Phone and Extension Patterns" section.

Step 5   In the Phone Pattern field, enter the phone number pattern.

Step 6   Click Add to submit the parameters.

Step 7   Click Continue to refresh the page.


Valid Phone and Extension Patterns

When entering patterns, the character x represents a wildcard. For example, if your PBX has the following blocks of phone numbers:

And they mapped to the following blocks of extensions:

You would enter the following information in the Phone Extension table:

Extension Pattern
Phone Pattern

1xxx

4085551xxx

2xxx

4085552xxx

3xxx

4085553xxx

Routing

Routing is the simple matter of transmitting packets to a destination along a predetermined route. Routing protocols direct data and routed protocols through an internetwork. Examples of these protocols include Enhanced Interior Gateway Routing Protocol (Enhanced IGRP), Open Shortest Path First (OSPF), and Routing Information Protocol (RIP).

To exchange routing updates with a connected device, you must select a routing protocol for your MRP that is common with the device to which it is connected. Although not required, we recommend that all devices use the same routing protocol.

Dynamic and Static Routing

Dynamic routing automatically selects routes and adjusts to changes in the network. Static routing requires manually selecting a route for data traffic, and dynamic routes can be overridden by static routes when the static routes are reachable. Dynamic routing requires more bandwidth than static routing and might reduce your MRP performance.

We recommend you use static routing if the following are true:

Enabling Dynamic Routing

Dynamic routing can be turned on or off. If Dynamic Routing is on and the MRP determines that a static route is not reachable, it attempts to use a dynamic route. When dynamic routes appear unreachable, it sends the packets to the router of last resort.

A router of last resort is the router to which all unroutable packets are sent, ensuring that all messages are handled in some way. The packets might be dropped, or the router of last resort might find a route unknown to the MRP and forward the packets.

If dynamic routing is turned off, static routes must be configured to enable network traffic between devices.

To display the Dynamic Routing page, do the following:


Step 1   In the Features list, click Routing.

Step 2   Click Dynamic Routing to display the Dynamic Routing page, as shown in Figure 3-18.


Figure 3-18: Dynamic Routing Page


Step 3   Select from the following:

  Routing Information Protocol 2 (RIP2) is the most common interior gateway protocol (IGP) in the Internet. It is used to exchange routing information within an autonomous system. The MRP uses RIP2 to build its routing tables. Turn on RIP2 if the switch to which this MRP is connected has RIP2 enabled.
  Enhanced Interior Gateway Routing Protocol (EIGRP) is an advanced protocol developed by Cisco Systems for route information exchange that combines the advantages of link state protocols with those of distance vector protocols. Turn on EIGRP if the switch to which this MRP is connected has RIP2 enabled.
  In the Autonomous System field, enter the EIGRP autonomous systems number.
  Open Shortest Path First (OSPF) is an industry-standard routing protocol for exchanging and updating routing information. Its features include least-cost routing, multipath routing, and load balancing. Enable this option if the switch to which this MRP is connected has OSPF turned on.
  If you enable OSPF and specify the OSPF area for each major network, you must enforce IP class boundaries. Areas must all have the same value for IP addresses in the same class boundary. For example, if an interface has a Class A IP address of 10.1.1.1 and another interface has the IP address 10.1.2.1, both of these interfaces must have the same OSPF area because they are on the same Class A boundary.
  Enter the OSPF area in the OSPF Area field.

Step 4   Click Submit. A message displays, indicating the configuration has been sent to the router.

Step 5   Click Continue to refresh the page.


Enabling Static Routing

Static routes are table mappings established by a network administrator prior to the beginning of routing. These mappings do not change unless the network administrator alters them. Static routes work well in environments where network traffic is predictable and where network design is relatively simple. You must provide the IP address and subnet mask of the route destination and identify the forwarding interface.

To display the Static Routing page, do the following:


Step 1   In the Features list, click Routing.

Step 2   Click Static Routing to display the Static Routing page, as shown in Figure 3-19.


Figure 3-19: Static Routing Page


Table 3-1 describes the Static Routing fields.


Table 3-1: Static Routing Page Options
Option Description

Prefix

Network portion of your IP address.

Prefix Mask

Bits in the host field that specify the subnet or the subnet mask.

Interface

Interface used to route packets.

IP Address

IP address of the interface used to route packets.

Action

Links that modify or delete a static route.

Step 3   Click Add IP Static Route to display the Add IP Static Route page, as shown in Figure 3-20.


Figure 3-20: Add IP Static Route Page


Step 4   In the Prefix field, enter the network prefix.

Step 5   In the Prefix Mask field, enter the subnet mask.

Step 6   Click Interface to choose the forwarding port by the interface or, click IP Address to identify the port by its IP address.

Step 7   From the drop-down menu next to the Interface label, select the interface and highlight the forwarding port to select it, or enter the IP address in the field next to the IP Address label.

Step 8   Click Add to submit the parameters. A message displays, indicating the configuration has been sent to the router.

Step 9   Click Continue to refresh the page.


Network Address Translation

Network Address Translation (NAT) reduces the need for globally unique IP addresses. You can use addresses locally that are not globally unique, yet connect to the global network (Internet) by translating the local addresses into globally routable addresses. This address translation extends addressing capabilities by providing both static address translations and dynamic address translations. It also gives you limited protection by effectively hiding internal addresses and removing all your internal services from the external name space.

The NAT option in the Features menu provides the following links:

Enabling NAT

You cannot configure dynamic or static NAT for your network unless this option is enabled.

To enable NAT, do the following:


Step 1   Click NAT to display the Enable NAT page, as shown in Figure 3-21.


Figure 3-21: Enable NAT Page


Step 2   Click the Enable NAT checkbox. When NAT is enabled, network address translation must occur on your entire network.

Step 3   From the Select outside network interface list, select the outside network interface. The selected interface becomes a part of the outside network. All other interfaces become part of the inside network.

Step 4   Click Submit. A message appears, saying that the configuration has been sent to the router.

Step 5   Click Continue to refresh the page.


Enabling Dynamic NAT

Dynamic NAT uses a single, global IP address to allow hosts with private addresses access to the Internet. There are two methods of using dynamic NAT:

To enable Dynamic NAT, do the following:


Step 1   In the Features menu, click NAT to display the list of options.

Step 2   Click Dynamic NAT to display the Dynamic NAT page, as shown in Figure 3-22.


Figure 3-22: Dynamic NAT Page


Step 3   Accept the No dynamic NAT default, or select one of the following:

Step 4   Click Submit. A message appears, saying that the configuration has been sent to the router.

Step 5   Click Continue to refresh the page.


Static IP Address Translation

Static address translation allows hosts with private addresses global access and to be publicly accessible from the outside. It statically maps a private IP address to a public or global address. There are two types of static address translations:

  For example, packets that contain port number 80 in the header are routed to the device bound to port number 80, the well-known services port number for HTTP services; packets with SMTP port number 25 in the header are routed to the device bound to port number 25, and so forth.

If you use the same public address on multiple devices and mix a simple translation with extended translations, the simple translation includes all of the services not specified by the extended translations. For example, if you translate the public address 200.1.1.1 with FTP to Device 1 and the same public address is simply translated to Device 2, then FTP access is allowed on Device 1, but all services except FTP are allowed for Device 2. Defining services using these address translation methods impacts the availability of services you can select in the MRP Manager.

Adding Static NAT

Static translations are generally used to allow access to a particular device through the NAT. For example, if a network has an internal DNS server that needs to communicate with an external DNS server, you would configure a static translation to enable such connectivity. The NAT thus allows traffic to be passed between these statically known, but translated addresses.

Note that addresses used in static translations must explicitly be omitted from the dynamic translation pool. An IP packet traversing a NAT can have both its source and destination addresses translated by the NAT.

To add Static NAT, do the following:


Step 1   In the Features menu, click NAT to display the list of options.

Step 2   Click Static NAT to display the Static NAT page, as shown in Figure 3-23.


Figure 3-23: Static NAT Page


Step 3   Click Add a Static Address Translation. The Add a Static NAT Entry page shown in Figure 3-24 appears.


Figure 3-24: Add a Static NAT Entry Page


Step 4   In the Local IP Address field, enter the local IP address of the MRP.

Step 5   In the Global IP Address field, enter the global (public) address. The private IP address of a device will be translated to the IP address you enter here.

Step 6   From the drop-down IP Service list, select an IP service. The IP Services are as follows:


Connections

The MRP Manager supports the following WAN connection types if a WIC or multiflex trunk T1 interface (MFT-T1) card is installed:

The Ethernet port is inside the chassis and connects to the internal System Switch Processor (internal Ethernet Switch). These connections provide read-only pages, as you cannot remove this connection and IP addresses cannot be changed by using this page.

ISDN BRI connections connect an MRP with ISDN devices. ISDN connections use one or both data channels for the connection to the ISDN service provider. Normally, the ISDN service provider is your local telephone company.

Adding an ISDN BRI connection to the MRP creates a logical dialer interface. ISDN BRI is a dial-up connection. (ISDN connections differ from their dial-up counterparts because with a dial-up connection, a new connection must be established each time the line is used).

A Point-to-Point Protocol (PPP) Sync Serial connection is used to connect an MRP to the serial port of another MRP or router using PPP. The PPP was designed to provide router-to-router connections over synchronous and asynchronous circuits. PPP has built-in security mechanisms such as Challenge Handshake Authentication Protocol (CHAP) and Password Authentication Protocol (PAP).

High-Level Data Link Control (HDLC) specifies a data encapsulation method on synchronous serial links by using frame characters and checksums. Choose HDLC encapsulation for connections when the MRP on the remote end uses HDLC encapsulation. HDLC is a proprietary protocol implemented differently by each router manufacturer.

Frame Relay offers a relatively high-speed, streamlined service for connecting LANs across a WAN. Frame Relay has a wide range of transmission speeds. Typically, a Frame Relay link transmits data at 56 or 64 kbps, with T1/E1 speeds up to 2 Mbps becoming more common. Frame Relay functions as a best-effort, unreliable service, and its connections use virtual circuits to move data between locations.

MRP Manager configures only point-to-point subinterfaces on the MRP. (Multi point subinterfaces are not supported.) Subinterfaces enable a single serial interface to connect to more than one destination. MRP Manager does not support configuring a serial interface for a multi-point connection.

An interface can have more than one virtual circuit. When you order a Frame Relay line from the service provider, you need to specify the destinations for the data. The service provider configures the virtual circuits for each location and assigns a Data Link Connection Identifier (DLCI) for each end of a virtual connection.

Displaying the Connections

Use the Connections page to configure your WAN connections for an MRP that has a WIC or a multiflex trunk T1 interface card (VWIC) installed.

In the Features menu, click Connections to display the Connections page, as shown in Figure 3-25.


Figure 3-25: Connections Page



Note   Controller settings configure the properties of your MFT-T1 controller. The procedure is described in the "Configuring Controller Settings" sections of this chapter.

Adding a Dialer Interface Connection

When you add an ISDN connection to a BRI interface, a logical dialer interface is created and added to the table.

To add a logical dialer interface, do the following:


Step 1   Click the Add a Dialer Interface Connection to display the Add Dialer Interface Connection page, as shown in Figure 3-26.


Figure 3-26: Add a Dialer Interface Connection Page


Easy IP provides two substantial services. It offers the ability to translate many IP addresses to one address and enables a Cisco router to automatically negotiate its own registered WAN interface IP address. The ability to negotiate a WAN IP address works only for PPP or ISDN connections because the underlying encapsulation method must be PPP. Select this option to enable the MRP to automatically negotiate its own registered WAN interface IP address.


Note   You should use Easy IP for your address translation needs if the MRP will function as the DHCP server for your network.

Step 2   Select IP unnumbered to Fast Ethernet or Specify an IP Address.

If you select IP unnumbered... addressing, you are specifying that the IP address assigned to an Ethernet interface of an MRP can be applied to all additional interfaces on the MRP. Using this option saves an IP address. This feature is available for PPP, HDLC, Frame Relay, and ISDN BRI.

If you select Specify an IP Address, enter the IP address and the subnet mask. Contact your network administrator or service provider for these values.

Step 3   In the Login Name field, enter the login name for CHAP authentication.

CHAP secures the connection between the MRP and the Internet by requiring a password to log onto your network. Your Internet service provider gives you the login name. If this name is not entered correctly, you cannot make outgoing calls.

This security feature, supported on lines using PPP encapsulation, prevents unauthorized access by identifying the remote end of the connection. The MRP then determines if the user is allowed access.

Step 4   In the Enter Password field, enter the password.

We recommend that passwords are any combination of alphanumeric and punctuation characters of mixed case without embedded spaces or / \*?"<>|#()&% characters. The first character must not be a number. The maximum length is 11 characters. The password is case-sensitive.

Step 5   In the Confirm Password field, re-enter the password.

Step 6   Enter the telephone number used to make outgoing calls to the Internet in the phone numbers field. Contact your network administrator or Internet service provider (ISP) for the telephone number.

Step 7   Select the telephone switch speed 56 kbps or 64 kbps radio button. The default speed is 56 kbps and works for all switches. Select 64 kbps only if all the end-to-end service telephone switches are 64 kbps.

Step 8   Select Use multiple B-channels... if you want to set all B channels of an ISDN BRI line to be used for a single call. Leave unchecked if using one B channel for each call. The default setting is to use one B channel for a single call. Note that your ISDN service provider might bill each B channel as a separate call.

Step 9   Select the switch type from the drop-down Switch Type list. The switch type identifies the equipment located at the ISDN service provider. MRP Manager requires that all lines connected to the MRP have the same switch type.

Step 10   Enter the Service Profile Identifier (SPID) number in the SPID Value field. You must provide two SPID values for each port.

When you order an ISDN line, your phone company might give you a SPID for every phone number you have. Each ISDN BRI line typically has two phone numbers and, thus, two SPIDs. The SPID is an eight to fourteen digit number that identifies the services you ordered. SPIDs are used only by the DMS-100 and ISDN-1 switches. They are optionally used by the AT&T 5ESS switch. All other switches do not use SPIDs.

Step 11   (Optional) Enter the local directory number (telephone number) associated with each SPID in the Local Directory No. field.

Step 12   Click Add to submit the values.


Adding a PPP Serial Connection

To add a PPP Serial connection, do the following:


Step 1   In the Features menu to display the Connections page, click Connections.

Step 2   Click Add PPP under Actions.

Step 3   Choose one of the following options for your installation.

  Easy IP can translate many IP addresses to one address and enable a Cisco router to automatically negotiate its own registered WAN IP address. The ability to negotiate a WAN IP address works only for PPP or ISDN connections because the underlying encapsulation method must be PPP.
  When you select IP unnumbered addressing, the IP address assigned to an Ethernet interface of an MRP can be applied to all interfaces on the MRP. Using this option saves one or more IP addresses. This feature is available for PPP, HDLC, Frame Relay, ISDN BRI, and asynchronous modem line connections.

Step 4   Click Add to send the configuration.

Step 5   Click Continue to refresh the page.


Adding an HDLC Connection

To add an HDLC connection, do the following:


Step 1   In the Features menu, click Connections to display the Connections page.

Under Actions, click Add HDLC. The HDLC connection properties page, shown in Figure 3-27, appears.


Figure 3-27: HDLC Connection Properties Page


Step 2   Choose one of the appropriate options for your installation.

  When you select IP unnumbered addressing, the IP address assigned to an Ethernet interface of an MRP can be applied to all interfaces on the MRP. Using this option saves one or more IP addresses. This feature is available for PPP, HDLC, Frame Relay, ISDN BRI, and asynchronous modem line connections.

Step 3   Click Add to send the configuration.

Step 4   Click Continue to refresh the page.


Adding a Frame Relay Connection

Adding a Frame Relay connection on a serial interface creates a logical subinterface that is added to the interface table (similar to adding dialer interfaces).

To add Frame Relay, do the following:


Step 1   In the Features menu, click Connections to display the Connections page.

Step 2   Under Actions, click Add Frame Relay. The Frame Relay connection properties page, shown in Figure 3-28, appears.


Figure 3-28: Frame Relay Connection Properties Page


The following sections describe the fields in this page.

Step 3   Choose one of the appropriate options for your installation.

  When you select IP unnumbered addressing, the IP address assigned to an Ethernet interface of an MRP can be applied to all interfaces on the MRP. Using this option saves one or more IP address. This feature is available for PPP, HDLC, Frame Relay, ISDN BRI, and asynchronous modem line connections.

Step 4   Enter the Local Management Interface (LMI) type for all virtual circuits of your Frame Relay connection.

The LMI (also known as LMT in ANSI terminology) is a specification used in Frame Relay networking that defines a method of exchanging status information between Frame Relay network ports and customer premises devices such as routers and frame-aware DSUs (FRADs).

LMI includes support for the following:

Your three options are Cisco, ANSI, or ITU-T Q.933. Select ANSI if you are not sure of the LMI type.

Step 5   In the DLCI field, enter the Data-link connection identifier (DLCI). The values are typically between 16 and 1007.

The Data-link connection identifier (DLCI) specifies if a permanent virtual circuit (PVC) or switched virtual circuit (SVC) is being used in the Frame Relay connection. Frame Relay connections use virtual circuits to move data between locations. A single physical interface on the router can have more than one virtual circuit. When you order a Frame Relay line from the service provider, you must specify the destinations for the data. Your service provider configures the virtual circuits for each location and assigns a DLCI for each end of the virtual connection. Contact your Frame Relay connection service provider for this value.

Step 6   Check Use IETF Frame Relay Encapsulation if your MRP is connected to a non-Cisco router. The MRP supports both Internet Engineering Task Force (IETF) standard Frame Relay encapsulation and Cisco proprietary encapsulation.

Step 7   Click Add to send the configuration.

Step 8   Click Continue to refresh the page.


Quality of Service for VoIP Traffic

Quality of Service (QoS) for Voice over IP (VoIP) refers to the ability of a network to provide better service to selected network traffic over various underlying technologies, such as PPP and Frame Relay. QoS features provide better and more predictable network service by:

Take advantage of QoS for optimum efficiency by configuring VoIP QoS features for your device connections.

You can reserve bandwidth for voice and modify the packet size for PPP, which affects fragmentation and interleaving. Specifying a smaller packet size reduces delays that might occur if the processing of large data packets is delaying the transmission of voice packets.

For Frame Relay, you can reserve bandwidth for voice and specify the committed information rate (CIR), burst rates, and the maximum transmission unit (MTU).

For all WAN connections, RTP header compression is automatically turned on, and the IP precedence is set to 5.

Configuring VoIP QoS for PPP

Failing to configure QoS can result in unacceptable voice quality.

To configure Voice Over IP QoS for your PPP connection, do the following:


Step 1   In the Features menu, click Quality of Service to display the Quality of Service page, as shown in Figure 3-29.


Figure 3-29: Quality of Service Page


Step 2   Click Edit QoS Settings to display the QoS Edit page, as shown in Figure 3-30.


Figure 3-30: Quality of Service for PPP Connections Page


Step 3   Enter the speed of this connection in the Speed of this... field.

The speed of the connection in entered in kbps. For example, if you are using a 56-kbps line, enter 56. Your service provider can tell you the speed of this connection.

Step 4   Enter the bandwidth reserved for VoIP in the Bandwidth you want... field.

Each VoIP call requires approximately 12 kbps of bandwidth and uses default codec G.729 with RTP header compression. RTP header compression and an IP precedence of 5 are automatically set for all of the WAN connections. If there is no VoIP traffic, the reserved bandwidth is not wasted because it is used by data traffic.

Step 5   Click the Enable Multilink PPP Fragmentation and Interleaving check box to enable fragmentation and interleaving.

Interactive traffic, such as Telnet and VoIP, is susceptible to increased latency and jitter when the network processes large packets, such as LAN-to-LAN FTP or Telnet transfers traversing a WAN link. This susceptibility increases as the traffic is queued on slower links.

To reduce delay and jitter on slower speed links, large datagrams are fragmented and low-delay traffic packets are interleaved with the resulting smaller packets. This reduces the possibility of large non-critical packets blocking time-critical packets (such as voice) from passing through.

Enabling PPP fragmentation and interleaving. PPP fragmentation and interleaving is highly recommended for low-speed links up to 256 to 512 kbps. It will improve the voice quality by reducing delay and jitter.

It is recommended that you disable PPP fragmentation and interleaving on high-speed links above 512 kbps because the delay is minimal. Fragmentation also increases the CPU utilization. This box is checked by default for speeds up to 512 kbps.

Step 6   In the Maximum Delay for Each Fragment field, enter the maximum delay for each fragment.

Delay denotes the time required to move a fragment from source to destination over a given path. Normally, the fragment delay of 30 ms is recommended to attain good quality of service for VoIP traffic. Decrease the fragment delay if you notice delay or jitter in voice call.

Reducing the fragment delay adversely affects data throughput and increases CPU utilization.


Configuring VoIP QoS for Frame Relay

You can configure your Frame Relay connection for the VoIP QoS features. Failing to configure QoS can result in unacceptable voice quality.

To configure VoIP QoS for your Frame Relay connection, do the following:


Step 1   In the Features menu, click Quality of Service to display the QoS page, as shown in Figure 3-31.


Figure 3-31: Quality of Service Page


Step 2   Click Edit QoS Settings for the interface you want to modify. The Quality of Service for Frame Relay connection page, shown in Figure 3-32, appears.


Figure 3-32: Quality of Service for Frame Relay Connection Page


Step 3   In the Committed Information Rate (CIR) field, enter the committed information rate.

The CIR is the rate at which a Frame Relay network agrees to transfer information under normal conditions, averaged over a minimum increment. This rate is negotiated between you and your Frame Relay service provider.

Your Frame Relay service provider can tell you the CIR for each circuit between your devices. If the CIR is different on each side of the circuit, enter the smallest value.

If the fields are left blank, the Burst Size (Bc) and Excess Burst Size (Be) values are automatically calculated based on CIR value. Your Frame Relay service provider can tell you the optional values for Bc and Be.

Step 4   In the Burst Size (Bc) field, enter the burst size (optional).

Frame relay allows a device to transmit data at a higher rate than the CIR for a few seconds at a time. This is called bursting. The burst size (Bc) is the maximum amount of data above CIR, in bps, that the network agrees to transfer under normal conditions during the measurement interval. This data is not eligible for discard and should be transmitted. Once the burst amount of data has been transmitted during the measurement interval, all frames are marked as discard-eligible.

Step 5   In the Excess Burst Size (Be) field, enter the excess burst size (optional).

The excess burst size (Be) is the maximum number of uncommitted data bits above CIR that the network attempts to deliver during the measurement interval. This traffic-shaping parameter indicates the amount of extra data that can be transmitted as discard-eligible. A frame marked as discard-eligible is not guaranteed delivery through the Frame Relay network. Discard-eligible frames are not transmitted when the Frame Relay network becomes congested.

Bc plus Be represents the maximum amount of data that can be transmitted during the measurement interval. All data after that is discarded.

Step 6   In the Bandwidth you want reserved for VoIP field, enter the bandwidth reserved for VoIP.

Each VoIP call requires about 12 kbps of bandwidth and uses default codec G.729 with RTP header compression. RTP header compression and IP precedence of 5 are automatically set for all of the WAN connections. If there is no VoIP traffic, the reserved bandwidth is not wasted and is used by data traffic.

Step 7   In the Maximum Transmission Unit (MTU) field, enter the maximum transmission unit (MTU).

The MTU denotes the maximum packet size (in bytes) that an interface is allowed. Lowering the MTU on high-speed links reduces the data throughput and increases CPU utilization. On low-speed links of 256 to 512 kbps, we recommend that you lower the MTU.

Step 8   Click Submit to send the configuration.

Step 9   Click Continue to refresh the page.


Management Menu Links

The Management menu provides Cisco IOS access. The Issue Command option uses thecommand-line interface (CLI) show commands to display configuration information for the selected MRP. The Telnet feature provides remote access to the system for configuring and monitoring. The View IOS Config options display the startup configuration or the running configuration for the selected MRP.

Displaying Configuration Information

MRP Manager issues only commonly used Cisco IOS show commands. The full suite of commands is not available from this page.

To issue other show commands, use the CLI. For a complete listing of the latest Cisco IOS Commands, refer to the Cisco IOS Software Command Summary at the following URL:

http://www.cisco.com/univercd/cc/td/doc/product/software/ios112/sbook/  
index.htm 
 

located on the Cisco Connection Online web site.

To select and send predefined Cisco IOS show commands, do the following:


Step 1   In the Management menu, click Issue Command to display the Issue Command page, as shown in Figure 3-33.


Figure 3-33: Issue Command Page


Step 2   Select the show command from the drop-down list.

Table 3-2 describes the available show commands.


Table 3-2: MRP Manager Show Commands
Show Command Description

show cdp neighbors

Show other Cisco-manufactured devices that advertise their existence on the LAN or the remote side of the WAN by using the Cisco Discovery Protocol (CDP).

show diag

Show diagnostic information for a WIC.

show flash

Show contents of the system Flash memory.

show interfaces

Show interface status and configuration information.

show protocols

Show the network routing protocols active on the MRP.

show running-config

Show the configuration that is currently operational on the MRP.

show snmp

Show the status of communications between the SNMP agent and the SNMP manager.

show startup-config

Show the contents of the startup configuration. This is the IOS configuration loaded when you bootup the Cisco ICS 7700. The startup configuration is the same as the running configuration when using the MRP Manager.

show tech-support

Show the system information needed when calling the Cisco Technical Assistance Center.

show version

Show hardware and software version information.

Step 3   Click Go to display the results of the command.


Opening a Telnet Session

MRP Manager uses the Telnet application that resides on your workstation to log into the MRP. If you make any changes to the MRP configuration, you must click the Sync button in MRP Manager so that the MRP reads the updated Cisco IOS configuration. To open a Telnet session on a device, the device must already be configured with an IP address.

To open a Telnet session, do the following:


Step 1   To display the Telnet page, in the Management menu click Telnet, as shown in
Figure 3-34.


Figure 3-34: Telnet Page


Step 2   Click the IP address of the device. The MRP Manager launches your default Telnet application.


Viewing a Cisco IOS Configuration

The View IOS Configuration page displays the Cisco IOS configuration file currently running on the MRP or the Cisco IOS startup configuration file created for the selected MRP.

To view a Cisco IOS configuration, do the following:


Step 1   Click View IOS Configuration in the Management menu to display the View IOS Configuration page, as shown in
Figure 3-35.


Figure 3-35: View IOS Configuration Page


Step 2   Click the Current running configuration or the Contents of startup configuration radio button.

Step 3   Click Go to display the results of the command.



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Posted: Mon Oct 2 13:47:02 PDT 2000
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