|
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This chapter explains how to configure the multiservice route processor (MRP) and contains the following sections:
 |
Note This guide describes the use of IOS commands that have been created or changed for use with the MRP that supports IOS Release 12.1.(4)T. The complete IOS Release 12.1 documentation is available through CCO by selecting Service and Support > Technical Documents > Documentation Home Page > Cisco IOS Software Configuration > Cisco IOS Release 12.1. |
|
Note For default settings on the MRP and more information on the commands described in this chapter, refer to the Cisco ICS 7750 Command Reference. |
The MRP is a voice-and-data-capable router that can carry voice and data traffic over an IP network and can link small-to-medium-size remote Ethernet LANs to central-offices (COs) LANs over different types of WAN links.
The MRP has the following main features:
Table 5-1 lists the WICs, VICs, and VWICs supported by the MRP. A VWIC functions both as a WIC and a VIC. Each VWIC can be used for voice or data, but both cannot be configured on a single T1. However, you can have voice on one T1 and data on another T1.
| Card Description | Voice or Data | Abbreviated Name |
|---|---|---|
1-port serial | Data | 1T WIC |
2-port serial | Data | 2T WIC |
2-port asynchronous/synchronous serial | Data | 2A/S WIC |
Data | 1B-ST WIC | |
1-port ISDN BRI U | Data | 1B-U WIC |
1-port 56/64-kbps DSU/CSU3 | Data | 1DSU-56K4 WIC |
1-port T1/FT1 | Data | 1DSU-T1 WIC |
2-port FXS4 voice/fax interface | Voice | 2FXS VIC |
2-port FXO5 voice/fax interface | Voice | 2FXO VIC |
2-port E&M6 voice/fax interface | Voice | 2E&M VIC |
1-port T1 multiflex trunk interface | Voice and data | 1MFT-T1 VWIC |
2-port T1 multiflex trunk interface | Voice and data | 2MFT-T1 VWIC |
Table 5-2 contains the combination guidelines for installing WIC, VIC, and VWIC cards in the MRP. Cards that can be installed in Slot 0 are listed across the top of the table. Cards that can be installed in Slot 1 are listed in the left column.
| 1B-ST WIC 1B-U WIC | 1T WIC 1DSU-56K4 WIC 1DSU-T1 WIC | 2T WIC 2A/S WIC | 2MFT-T1 VWIC- Two Voice | 2MFT-T1 VWIC- One Voice, One Data | 1MFT-T1 VWIC-Voice | 1MFT-T1 VWIC -Data | FXS, FXO, E&M VICs | |
|---|---|---|---|---|---|---|---|---|
| 1B-ST WIC 1B-U WIC | Yes | Yes | Yes | Yes | Yes | Yes | Yes | Yes |
| 1T WIC 1DSU-56K4 WIC 1DSU-T1 WIC | Yes | Yes | Yes | Yes | Yes | Yes | Yes | Yes |
| 2T WIC 2A/S WIC | Yes | Yes | Yes | Yes | Yes | Yes | Yes | Yes |
| 2MFT-T1 VWIC Two Voice1 | Yes | Yes | Yes | No | No | No | Yes | Yes |
| 2MFT-T1 VWIC One Voice, One Data2 | Yes | Yes | Yes | No | No | Yes | No | Yes |
| 1MFT-T1 VWIC-Voice | Yes | Yes | Yes | No | Yes | Yes | Yes | Yes |
| 1MFT-T1 VWIC-Data | Yes | Yes | Yes | Yes | No | Yes | No | Yes |
| 1Even though the MRP can configure both ports on a 2MFT-T1 for voice, it can only make 24 simultaneous calls. 2The MRP does not support 2MFT-T1 VWIC for two T1 data channels. |
The MRP supports two CODECs, G.729a and G.711, for Voice over IP (VoIP) calls. The default is G.711. Each digital signal processor (DSP) can handle a combination of G.711 and G.729a calls.
The MRP handles the calls based on the grouping of the DSPs. The DSPs are located on the packet voice data module (PVDM). There can be up to five DSPs on a single PVDM. Each PVDM corresponds to one DSP group.The MRP has two PVDM slots and therefore can have a maximum of two DSP groups. Each DSP group serves either an analog port or a T1 port on the VIC. Therefore, one analog VIC and one T1 VIC make up two groups and two T1 VICs with two different clock sources also make up two groups.
Each DSP group serving a T1 port can support as many DSPs as there are in the PVDM.
A DSP has a maximum capacity of 100 MIPS to handle a particular number of simultaneous calls. One G.729a call requires 25 MIPS and one G.711 call requires 12.5 MIPS. The number of calls on a DSP is determined by the total used MIPS reaching 100 on that DSP. The DSP resource manager rejects the call if it cannot find a DSP with required unused MIPS for the selected CODEC.
The following table lists some of the scenarios of the number of calls supported on a single DSP depending on the CODEC used:
| Scenarios | Calls per DSP | CODECs | MIPS per session | MIPS Required | Call Status |
|---|---|---|---|---|---|
1 | 4 | G.729a | 25 | 25 x 4 = 100 | 4 calls accepted |
2 | 8 | G,711 | 12.5 | 12.5 x 8 = 100 | 8 calls accepted |
3 | 4 1 | G.729a G,711 | 25 12.5 | 25 x 4 = 100 12.5 x 1 = 12.5 |
1 call rejected |
Some of the combinations of calls that can be used on a single DSP are as follows:
| G.711 Calls | G.729a Calls |
|---|---|
2 | 3 |
4 | 2 |
6 | 1 |
Each DSP group serving analog ports needs one DSP per VIC. Therefore, two analog VICs need two DSPs. Each DSP in this case can handle two calls that can be either a G.711, G.729a, or a fax relay.
 |
Caution The user is strongly recommended to use only the Cisco ICS 7700 System Manager (also referred to as System Manager) GUI interface to define and manipulate the MRP configurations that are supported by this interface. It is possible for a user with sufficient knowledge to change configurations by direct interaction with the system components through the CLI. However, if there are problems due to CLI configuration, you will not be able to use the System Manager to recover. |
It is also recommended that the following list of tasks should not be done through the CLI, because they are configured through the System Manager and might conflict with its configurations:
To configure the MRP through the CLI, you must access the MRP through the system alarm processor (SAP). For more information on how to get the CLI prompt on the MRP, refer to "Accessing the System."
To prevent the loss of the MRP configuration, save the running-config file to the startup-config file by following these steps:
| Command | Purpose | |
|---|---|---|
Step 1 | | Enter enable mode. Enter the password. You have entered enable mode when the prompt changes to |
Step 2 | Save the configuration changes to the startup-config file so that they are not lost during resets, power cycles, or power outages. |
This section describes basic configuration, including enabling the interface and specifying IP routing. Depending on your own requirements and the protocols you plan to route, you might also need to enter other configuration commands.
Before you begin configuring the interfaces, make sure you do the following:
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Note It is recommended not to change the pre-configured Fast Ethernet interfaces because the change might conflict with the System Manager configurations. |
| Command | Purpose | |
|---|---|---|
Step 1 | | Enter enable mode. Enter the password. You have entered enable mode when the prompt changes to |
Step 2 | | Enter global configuration mode. You have entered global configuration mode when the prompt changes to |
Step 3 | | Enable routing protocols as required for your global configuration. This example uses IP routing. |
Step 4 | | Enter interface configuration mode. You have entered interface configuration mode when the prompt changes to |
Step 5 | | Assign an IP address and subnet mask to the interface. |
Step 6 | | Exit back to global configuration mode. Repeat Step 4 through Step 6 if your MRP has more than one interface that you need to configure. |
Step 7 | | When you finish configuring interfaces, return to enable mode. |
This section contains the following subsections:
You can configure the serial interfaces on your asynchronous/synchronous serial WIC manually by entering IOS commands on the command line in configuration mode.
Before you begin configuring the interfaces, make sure you do the following:
| Command | Purpose | |
|---|---|---|
Step 1 | | Enter enable mode. Enter the password. You have entered enable mode when the prompt changes to |
Step 2 | | Enter global configuration mode. You have entered global configuration mode when the prompt changes to |
Step 3 | | Enable routing protocols as required for your global configuration. This example uses IP routing. |
Step 4 | | Enter the interface configuration mode. You have entered interface configuration mode when the prompt changes to |
Step 5 | | Assign the IP address and subnet mask to the interface. |
Step 6 |
| All serial ports are initially configured as synchronous. Enter this command if you want to configure the port as asynchronous. |
Step 7 | | Configure asynchronous parameters according to your needs. |
Step 8 | Configure the asynchronous line setting. | |
Step 9 | | Specify the time that the interface waits in controlled carrier mode. See Table 5-5 for a list of half-duplex timer commands. |
Step 10 | To use a port in DCE mode, connect a DCE cable and set the internal transmit clock signal (TXC) speed in bits per second. See Table 5-6 through Table 5-8 for a list of clock rate settings for your specific interface. (For ports used in DTE mode, the MRP automatically uses the external timing signal.) | |
Step 11 |
| When a port is operating in DCE mode, the default operation is for the DCE to send serial clock transmit (SCT) and serial clock receive (SCR) clock signals to the DTE, and for the DTE to return a serial clock transmit external (SCTE) signal to the DCE. If the DTE does not return SCTE, enter this command to configure the DCE port to use its own clock signal. |
Step 12 |
| MRPs that use long cables might experience high error rates when operating at higher transmission speeds, because the clock and data signals can shift out of phase. If a DCE port is reporting a high number of error packets, inverting the clock using this command can often correct the shift. |
Step 13 |
| Exit back to global configuration mode. Repeat Step 4 through Step 14 if your MRP has more that one serial interface that you need to configure. |
Step 14 | | When you finish configuring interfaces, return to enable mode. |
Step 15 | MRP#copy running-config startup-config | Save the configuration changes to the startup-config file so that they are not lost during resets, power cycles, or power outages. |
Table 5-5 shows a list of half-duplex timer commands.
| Timer | Syntax | Default Setting (Milliseconds) |
|---|---|---|
CTS delay1 | 100 | |
CTS drop timeout | 5000 | |
DCD drop delay | 100 | |
DCD transmission start delay | 100 | |
RTS2 drop delay | 100 | |
RTS timeout | 2000 | |
Transmit delay | 0 |
| 1CTS = Clear To Send 2RTS = Request To Send |
Table 5-6 through Table 5-8 show lists of clock rate settings for your specific interface.
1200 bps | 38400 bps | 148000 bps |
|---|---|---|
2400 bps | 56000 bps | 500000 bps |
4800 bps | 57600 bps | 800000 bps |
9600 bps | 64000 bps | 1000000 bps |
14400 bps | 72000 bps | 1300000 bps |
19200 bps | 115200 bps | 2000000 bps |
28800 bps | 125000 bps | 4000000 bps |
32000 bps | 128000 bps | 148000 bps |
1200 bps | 28800 bps | 72000 bps |
|---|---|---|
2400 bps | 32000 bps | 115200 bps |
4800 bps | 38400 bps | 125000 bps |
9600 bps | 56000 bps | 128000 bps |
14400 bps | 57600 bps |
|
19200 bps | 64000 bps |
|
1200 bps | 28800 bps | 72000 bps |
|---|---|---|
2400 bps | 32000 bps | 115200 bps |
4800 bps | 38400 bps | 125000 bps |
9600 bps | 56000 bps | 128000 bps |
14400 bps | 57600 bps |
|
19200 bps | 64000 bps |
|
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Note To prevent the loss of MRP configuration, refer to the "Saving Configuration Changes" section in this chapter. |
You can configure the interfaces on your BRI WAN interface card manually by entering IOS commands on the command line in configuration mode.
Before you begin configuring the interfaces, make sure you do the following:
| Command | Purpose | |||
|---|---|---|---|---|
Step 1 | | Enter enable mode. Enter the password. You have entered enable mode when the prompt changes to | ||
Step 2 | | Enter global configuration mode. You have entered global configuration mode when the prompt changes to | ||
Step 3 |
| Enter an ISDN switch type. See Table 5-8 for a list of ISDN switch types.
| ||
Step 4 | | Enable routing protocols as required for your global configuration. This example uses IP routing. | ||
Step 5 | | Enter the interface configuration mode. You have entered interface configuration mode when the prompt changes to | ||
Step 6 | | Assign the IP address and subnet mask to the interface. If you are configuring this interface for voice, enter the switch type instead of an IP address. | ||
Step 7 | | Exit back to global configuration mode. Repeat Step 5 through Step 7 if your MRP has more than one BRI interface that you need to configure. | ||
Step 8 | | By default, the card allocates 25 percent of DRAM to shared memory (used for data transmitted or received by network modules and WAN interface cards). If your card includes 16 or more ISDN BRI interfaces, you must increase the amount of shared memory by entering the memory-size iomem command. This example increases shared memory from 25 percent to | ||
Step 9 | | When you finish configuring interfaces, return to enable mode. | ||
Step 10 | MRP#copy running-config startup-config | Save the configuration changes to the startup-config file so that they are not lost during resets, power cycles, or power outages. |
Table 5-9 shows a list of ISDN switch types for North America.
| ISDN Switch Type | Description |
|---|---|
basic-5ess | AT&T basic rate switches |
basic-dms100 | NT DMS-100 basic rate switches |
basic-nil1 | National ISDN-1 switches |
Before using an MRP with an ISDN BRI interface, you must order a correctly configured ISDN BRI line from your local telecommunications service provider.
The ordering process varies from provider to provider and from country to country; however, here are some general guidelines:
ISDN BRI provisioning refers to the types of services provided by the ISDN BRI line. Although provisioning is performed by your ISDN BRI service provider, you must tell the provider what you want. Table 5-10 lists the provisioning you should order for your MRP.
| Switch Type | Provisioning |
|---|---|
5ESS Custom BRI | For data only: 2 B channels for data Point to point Terminal type = E 1 directory number (DN) assigned by service provider MTERM = 1 Request delivery of calling line ID on Centrex lines Set speed for ISDN calls to 56 kbps outside local exchange |
5ESS National ISDN (NI-1) BRI | Terminal type = A 2 B channels 2 directory numbers assigned by service provider 2 SPIDs required, assigned by service provider Set speed for ISDN calls to 56 kbps outside local exchange Directory number 1 can hunt to directory number 2 |
DMS-100 BRI | 2 B channels 2 directory numbers assigned by service provider 2 SPIDs required, assigned by service provider Functional signaling Dynamic terminal endpoint identifier (TEI) assignment Maximum number of keys = 64 Release key = no, or key number = no Ringing indicator = no EKTS = no PVC = 2 Request delivery of calling line ID on Centrex lines Set speed for ISDN calls to 56 kbps outside local exchange Directory number 1 can hunt to directory number 2 |
Some service providers assign service profile identifiers (SPIDs) to define the services to which an ISDN device subscribes. If your service provider requires SPIDs, your ISDN device cannot place or receive calls until it sends a valid SPID to the service provider when initializing the connection. A SPID is usually a 7-digit telephone number plus some optional numbers, but service providers might use different numbering schemes. SPIDs have significance at the local access ISDN interface only; remote MRPs are never sent the SPID.
Currently, only DMS-100 and NI-1 switch types require SPIDs. Two SPIDs are assigned for the DMS-100 switch type, one for each B channel. The AT&T 5ESS switch type might support SPIDs, but Cisco recommends that you set up that ISDN service without SPIDs.
If your service provider assigns you SPIDs, you must define these SPIDs on the MRP. To define SPIDs and the local directory number (LDN) on the MRP for both ISDN BRI B channels, use the following isdn spid commands:
MRP (config-if)# isdn spid1 spid-number [ldn] MRP (config-if)# isdn spid2 spid-number [ldn]
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Note Although the LDN is an optional parameter in the command, you might need to enter it so that the MRP can answer calls made to the second directory number. |
For further information on configuring ISDN, see the chapters "Configuring ISDN" and "Configuring DDR" in the Wide-Area Networking Configuration Guide for the Cisco IOS Release 12.0.
The 1-port T1 and fractional (FT1) WICs includes an integrated data service unit /channel service unit (DSU/CSU) and can be configured either for full T1 service at 1.544 Mbps or for fractionalized T1 service. You can configure the interfaces on your T1 WICs manually by entering IOS commands on the command line. This method, called configuration mode, provides the greatest power and flexibility.
Before you begin configuring the interfaces, make sure you do the following:
The IOS software provides the following default configuration for DSU/CSU- and T1-specific parameters:
service-module t1 clock source lineservice-module t1 data-coding normalservice-module t1 timeslots all speed 64service-module t1 framing esfservice-module t1 lbo noneservice-module t1 linecode b8zsno service-module t1 remote-alarm-enableservice-module t1 remote-loopbackno service-module t1 fdl
To view the current configuration, enter the show service-module serial slot/port command. For further information about these commands, refer to the Cisco IOS configuration guides and command references.
Use the following procedure to configure a new T1/FT1 interface or to change the configuration of an existing interface:
| Command | Purpose | |
|---|---|---|
Step 1 | | Enter enable mode. Enter the password. You have entered enable mode when the prompt changes to |
Step 2 | | Enter global configuration mode. You have entered global configuration mode when the prompt changes to |
Step 3 | | Enable routing protocols as required for your global configuration. This example uses IP routing. |
Step 4 | | Enter the interface configuration mode. You have entered interface configuration mode when the prompt changes to |
Step 5 | | Assign the IP address and subnet mask to the interface. |
Step 6 | | Enter the framing type and linecode type. In this example, the framing type specified is |
Step 7 | | If you are using fractional T1 service, enter the time slot range and speed. In this example, the time slot range specified is from 1 to 20, and the speed specified is 64 kbps. |
Step 8 | | Exit back to global configuration mode. Repeat Step 4 through Step 8 if your MRP has more than one interface that you need to configure. |
Step 9 | | When you finish configuring interfaces, return to enable mode. |
Step 10 | MRP#copy running-config startup-config | Save the configuration changes to the startup-config file so that they are not lost during resets, power cycles, or power outages. |
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Note To prevent the loss of MRP configuration, refer to the "Saving Configuration Changes" section in this chapter. |
Configure the multiflex trunk interface (MFT-T1) card as a WIC (data). In the WIC mode, MRP treats the T1 as a single serial interface for data up to 1.544 Mbps. You can specify the number of channels (up to 24) for this connection. On a data T1 you can configure only one channelized group. The rest of the channels are simply not used.
The MRP supports the following T1 configurations:
To configure a T1 multiflex trunk interface, use the Cisco ICS 7700 System Manager available on your system or use configuration mode (manual configuration). In this mode, you enter IOS commands at the MRP prompt.
This section describes basic configuration, including enabling the interface and specifying IP routing. Depending on your own requirements and the protocols you plan to route, you might also need to enter other configuration commands.
Before you begin configuring the interfaces, make sure you do the following:
Digital T1 interfaces require not only that you set timing, but that you consider the source of the timers. You must configure the tdm clock to specify the clock source. You can specify up to two external clock sources for each MRP. This means that only two of the T1 ports can use line as the clock source. The clock source is selected via the tdm clock global configuration command.
For detailed commands and tasks to configure the tdm clock, refer to the "Configuring TDM Clock" section. For default tdm clock values, refer to the "Default TDM Clock" section.
Refer to the "TDM Clocking Scenarios" section, which describes the basic timing scenarios that can occur when a digital T1 interface is connected to a PBX, CO, or both.
Use the following procedure to configure a new T1 interface or to change the configuration of an existing interface.
| Command | Purpose | |
|---|---|---|
Step 1 | | Enter enable mode. Enter the password. You have entered enable mode when the prompt changes to |
Step 2 | | Enter global configuration mode. You have entered global configuration mode when the prompt changes to |
Step 3 | | Enable routing protocols as required for your global configuration. This example uses IP routing. |
Step 4 | Select the T1 interface to configure. This example configures a T1 interface in slot 1 and port 0. | |
Step 5 | MRP(config-controller)# framing {esf | sf} | Specify the framing type. The default is esf. |
Step 6 | MRP(config-controller)# linecode {b8zs | ami} | Specify the line code format. The default is b8zs. |
Step 7 | MRP(config-controller)# channel-group X timeslots 1-24 | Specify the channel group and time slots to be mapped. For multiflex trunk interfaces, only one channel group can be configured. |
Step 8 | MRP(config-controller)# interface serial 1/0:0 | Configure each channel group as a virtual serial interface. Specify the T1 interface, port number, and channel group to modify. |
Step 9 | | Assign an IP address and subnet mask to the interface. |
Step 10 |
| Exit back to global configuration mode. Return to Step 4 if your MRP has more than one T1 interface that you need to configure. |
Step 11 | | When you finish configuring interfaces, return to enable mode. |
Step 12 | MRP#copy running-config startup-config | Save the configuration changes to the startup-config file so that they are not lost during resets, power cycles, or power outages. |
|
Note To prevent the loss of MRP configuration, refer to the "Saving Configuration Changes" section in this chapter. |
This section contains the following subsections:
The MRP provides both analog and digital voice ports for its implementation of Voice over IP (VoIP). The type of signaling associated with these analog voice ports depends on the VIC installed in the device.
You can install either a VIC or a VWIC in the MRP to make voice-related calls through the network. A VIC connects the MRP directly to a regular analog phone, fax, or to a PBX. Alternatively, a VWIC enables 24 channels or this number of simultaneous calls incoming or outgoing on the network. This provides a much higher density than with the VIC, which has only two ports per MRP. VWIC also provides the flexibility to combine channels to form channel groups with the same characteristics.
Each VIC is specific to a particular signaling type; therefore, VICs determine the type of signaling for the voice ports. Voice-port commands define the characteristics associated with a particular voice-port signaling type.
The voice ports support four basic voice signaling types:
Figure 5-1 shows how to connect the VICs to the network.
This section contains the following subsections:
This section explains how to configure ports on FXS VICs that connect directly to a standard telephone, fax machine, or similar device.
Figure 5-2 shows a basic voice network. A small business uses a MRP card (named West) to provide telephone and fax connections among employees in its office. Two of these telephones are connected to an FXS VIC port in the West MRP.
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Note You can name your MRP by using the global configuration hostname command. |
Table 5-11 lists telephone numbers and voice ports for the West MRP.
| Telephone Number | Voice Port |
|---|---|
408 555-3737 | 0/0 |
408 555-4141 | 0/1 |
|
Note If your MRP is configured with two VICs, a total of four telephones and fax machines can be connected to it. As the MRP has only two slots, you need to replace one VIC with a WIC to provide an interface for IP connectivity to the WAN and for data traffic. To accommodate more than four voice devices, you need to add more MRPs or use an E&M VIC and a local PBX, rather than connecting every telephone to its own FXS VIC. |
To route a received voice call to the right destination, the MRP needs to know which telephone number belongs to each voice port. For instance, if a call comes in for 408 555-3737, the MRP needs to know that this telephone is connected to voice port 0/0 (as shown in Figure 5-2.) In other words, the MRP needs to know the information in Table 5-11.
To hold this information, IOS software uses objects called dial peers. A telephone number, a voice port, and other call parameters are tied together by associating them all with the same dial peer. Configuring dial peers is similar to configuring static IP routes—you are telling the MRP what path to follow to route the call. All voice technologies use dial peers to define the characteristics associated with a call leg. A call leg is a segment of a call path; for example, segments occur between a telephone and an MRP, an MRP and a network, an MRP and a PBX, or an MRP and the PSTN. Each call leg corresponds to a dial peer.
Dial peers are identified by numbers, but they are usually referred to as tags to avoid confusion with telephone numbers. Dial-peer tags are arbitrary integers that can range from 1 to 231-1(2147483647). Within the allowed range, you can choose any dial-peer tag that is convenient or makes sense to you. Dial peers on the same MRP must have unique tags, but you can reuse the tags on other MRPs.
Table 5-12 assigns a dial-peer tag to each telephone number and its associated voice port on the West MRP. This type of dial peer is called a POTS dial peer or a local dial peer. The term POTS (plain old telephone service) means that the dial peer associates a physical voice port with a local telephone device. (VoIP dial peer is explained in the section "Calling Between MRPs" later in this guide.)
| Telephone Number | Voice Port | Dial-Peer Tag |
|---|---|---|
408 555-3737 | 0/0 | 401 |
408 555-4141 | 0/1 | 402 |
You should construct a table similar to Table 5-12 for your own MRPs, assigning your own telephone numbers and dial-peer tags.
|
Note The telephone numbers used in this guide are only examples and are invalid for public use in the United States. When you configure your network, be sure to substitute your own telephone numbers. |
To configure the MRP with the dial-peer information in Table 5-12, enter the following global configuration commands:
| Command | Purpose | |
|---|---|---|
Step 1 |
| Enter enable mode. Enter the password. You have entered enable mode when the prompt changes to |
Step 2 | | Enter global configuration mode. You have entered global configuration mode when the prompt changes to |
Step 3 | West (config)# dial-peer voice 401 pots | Enter dial-peer configuration mode. |
Step 4 | West (config-dial-peer)# destination-pattern 14085553737 | Define the destination telephone number associated with the POTS dial peer. |
Step 5 | West (config-dial-peer)# port 0/0 | Enter the port and slot number associated with this POTS dial peer. |
Step 6 | West (config) # dial-peer voice 402 pots | Enter dial-peer configuration mode. |
Step 7 | West (config-dial-peer)# destination-pattern 14085554141 | Define the destination telephone number associated with the POTS dial peer. |
Step 8 | West (config-dial-peer)# port 0/1 | Enter the port and slot number associated with this POTS dial peer. |
Step 9 | West (config-dial-peer)# exit | Return to configuration mode. |
These commands are summarized in Figure 5-3.
The dial-peer command always takes the argument voice. The number following it is the dial-peer tag, and pots is the type of dial peer.
Cisco IOS software refers to a telephone number as a destination pattern because it is the destination for an incoming or outgoing call. Enter these numbers with the destination-pattern command. A destination pattern can include asterisks (*) and pound signs (#) from the telephone keypad, and commas (,) and periods (.), which have special meanings. Parentheses ( () ), hyphens (-), slashes (/), and spaces ( ), which are often used to make telephone numbers easier for humans to read are not allowed.
Notice that the commands in the examples puts the prefix 1 (used in the United States to indicate a long-distance number) and an area code in front of the remaining numbers to complete the destination pattern. You need to include similar codes for your country if the VoIP equipment needs to establish a connection to the PSTN.
|
Note The IOS software does not check the validity of the telephone number. It accepts any string of permitted characters as a valid number. |
The business that owns the West MRP also has a branch office in the East. Figure 5-4 shows the East office network, and Table 5-13 lists the phone numbers, voice ports, and dial-peer tags for this office.
| Telephone Number | Destination Pattern | Voice Port | Dial-Peer Tag |
|---|---|---|---|
919 555-8282 | 19195558282 | 1/0 | 901 |
919 555-9595 | 19195559595 | 1/1 | 902 |
Enter the following commands to configure the local ports on the East MRP with the dial-peer information in Table 5-13:
| Command | Purpose | |
Step 1 | East (config)# dial-peer voice 901 pots | Enter dial-peer configuration mode |
Step 2 | East (config-dial-peer)# destination-pattern 19195558282 | Define the destination telephone number associated with the POTS dial peer. |
Step 3 | East (config-dial-peer)# port 1/0 | Enter the port/slot number associated with this POTS dial peer. |
Step 4 | East (config)# dial-peer voice 902 pots | Enter dial-peer configuration mode |
Step 5 | East (config-dial-peer)# destination-pattern 19195559595 | Define the destination telephone number associated with the POTS dial peer. |
Step 6 | East (config-dial-peer)# port 1/1 | Enter the port and slot number associated with this POTS dial peer. |
Step 7 | East (config-dial-peer)# exit | Return to configuration mode. |
These commands are summarized in Figure 5-5.
If you configured POTS dial peers on your MRP by following these examples, you can place calls between telephones connected to the same MRP. You can also use the show dial-peer voice command to verify that the data you configured is correct.
|
Note If the voice port is offline, use the interface configuration no shutdown command to enable it. |
|
Note Although placing calls directly between ports on the same MRP helps to verify your configuration, it is not recommended for general telecommunications use. |
To enable the West and East offices to send voice traffic to each other over the same IP network they use for data traffic, use a WIC on each MRP to provide a connection to the IP network as shown in Figure 5-6.
Look at the connection between the West MRP and the IP network. This connection does not include a voice port or an attached telephone—it leads from a WAN interface to a remote destination somewhere on the IP network. IP MRPs know how to locate IP addresses on the network, but they do not know how to locate telephone numbers. To route an outgoing voice call over this connection, the West MRP has to associate a telephone number in the East office with the IP address of the East MRP.
Table 5-14 assigns a dial-peer tag to each telephone number and its associated IP address on the West MRP. This type of dial peer is called a remote dial peer or VoIP dial peer. (Remember, the dial-peer tags are arbitrary.) The term VoIP means that the dial peer associates a telephone number with an IP address.
| Remote Location | Telephone Number | Destination Pattern | IP Address | Dial-Peer Tag |
|---|---|---|---|---|
East | 919 555-8282 | 19195558282 | 192.168.11.3 | 501 |
East | 919 555-9595 | 19195559595 | 192.168.11.3 | 502 |
Create a VoIP dial peer on the West MRP for every telephone on the East MRP, all associated with the same IP address. But it is much easier to use periods as wildcards, as shown in Table 5-15.
| Remote Location | Telephone Number | Destination Pattern | IP Address | Dial-Peer Tag |
|---|---|---|---|---|
East | 919 555-xxxx | 1919555.... | 192.168.11.3 | 501 |
Construct a table similar to Table 5-15 for your own MRPs, assigning your own telephone numbers, IP addresses, and dial-peer tags.
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Note The IP addresses shown in this guide are meant only as examples. When you configure your network, be sure to substitute your own IP addresses. |
Enter the following information on the West MRP to create the dial-peer configuration given in Table 5-15:
| Command | Purpose | |
|---|---|---|
Step 1 | West (config)# dial-peer voice 501 voip | Enter dial-peer configuration mode. |
Step 2 | West (config-dial-peer)# destination-pattern 1919555.... | Define the destination telephone number associated with the POTS dial peer. |
Step 3 | West (config-dial-peer)# session target ipv4:192.168.11.3 | Specify a destination IP address for this dial peer. |
IOS software describes the remote network as the session target. This command is followed by the IP address of the remote MRP. The prefix ipv4 means IP version 4. Alternatively, you can use the prefix dns followed by the Domain Name System (DNS) name as follows:
West(config-dial-peer)# session target dns:voice.eastMRP.com
Configure a dial peer on each MRP for each telephone number on every other MRP connected to it.
Make things easier by configuring number expansion for East MRP telephone numbers on the West MRP. For details on num-exp command, refer to the "Configuring Number Expansion" section.
West(config)# num-exp 5.... 1919555....
Now users can dial a five-digit extension beginning with 5 from a telephone on the West MRP to reach a telephone on the East MRP.
These commands are summarized in Figure 5-7.
The West MRP is now configured to send calls to the East MRP.
Table 5-16 shows how to configure the East MRP to send calls to the West MRP.
| Remote Location | Telephone Number | IP Address | Dial-Peer Tag |
|---|---|---|---|
West | 408 555-xxxx | 192.168.19.27 | 801 |
Enter the following information on the East MRP to create the dial-peer configuration given in Table 5-16:
| Command | Purpose | |
|---|---|---|
Step 1 | East (config)# num-exp 5.... 1408555.... | Expand a five-digit extension beginning with numeral 5 by prefixing 140855 to it. |
Step 2 | East (config)# dial-peer voice 801 voip | Enter dial-peer configuration mode. |
Step 3 | East (config-dial-peer)# destination-pattern 1408555.... | Define the destination telephone number associated with the POTS dial peer. |
Step 4 | East (config-dial-peer)# session target ipv4:192.168.19.27 | Specify a destination IP address for this dial peer. |
These commands are summarized in Figure 5-8.
If the path between endpoints of a voice call runs through intermediate MRPs, configure those MRPs for VoIP traffic, as described in the section "Configuring FXS Interfaces" earlier in this guide.
You need to configure POTS or VoIP dial peers on an intermediate MRP only if that MRP also has voice devices attached to it.
If you configured VoIP dial peers on your MRP by following these examples, you can place calls from that MRP to telephones on the remote MRPs (using just the extension if you configured number expansion). If you have trouble placing calls, ping the remote MRP to make sure you have IP connectivity, or use the show dial-peer voice command to verify that the data you configured is correct.
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Note Refer to the section "Configuring Quality of Service" if you need to improve the quality of voice connections. |
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Note To prevent the loss of MRP configuration, refer to the "Saving Configuration Changes" section in this chapter. |
FXO interfaces provide a gateway from the VoIP network to the analog PSTN or to a PBX that does not support E&M signaling so that users can reach telephones and fax machines outside the VoIP network. Figure 5-9 shows a typical FXO gateway attached to the West MRP.
To create a POTS dial peer for an FXS interface as explained earlier, you enter the complete telephone number of the attached telephone as the destination pattern for incoming calls. However, to create a POTS dial peer for an FXO interface, the destination pattern refers to outgoing calls, and you can include wildcards in it because the PSTN performs the switching.
The VoIP feature can also remove digits that you do not want to send to the PSTN. For instance, to dial 9 to reach an outside line (that is, the analog PSTN), enter the following commands:
| Command | Purpose | |
|---|---|---|
Step 1 | | Enter enable mode. Enter the password. You have entered enable mode when the prompt changes to |
Step 2 | | Enter global configuration mode. You have entered global configuration mode when the prompt changes to |
Step 3 | West (config) # dial-peer voice 201 pots | Enter dial-peer configuration mode. |
Step 4 | West (config-dial-peer)# destination-pattern 9 | Define the destination telephone number associated with the POTS dial peer. |
Step 5 | West (config-dial-peer)# port 0/0 | Enter the port/slot number associated with this POTS dial peer. |
When you dial 9, the MRP makes a connection to the PSTN through voice port 0/0. The PSTN then provides a dial tone, and any digits you enter on the telephone thereafter are interpreted on the PSTN.
To enable East MRP users to make calls over the West MRP local PSTN, enter the following commands:
| Command | Purpose | |
|---|---|---|
Step 1 | East (config)# dial-peer voice 701 voip | Enter dial-peer configuration mode. |
Step 2 | East (config-dial-peer)# destination-pattern 7 | Define the destination telephone number associated with the POTS dial peer. |
Step 3 | East (config-dial-peer)# session target ipv4:192.168.19.27 | Specify a destination IP address for this dial peer. |
Step 4 | West (config) # dial-peer voice 601 pots | Enter dial-peer configuration mode. |
Step 5 | West (config-dial-peer)# destination-pattern 7 | Define the destination telephone number associated with the POTS dial peer. |
Step 6 | West (config-dial-peer)# port 0/0 | Enter the port and slot number associated with this POTS dial peer. |
When you dial 7 on the East MRP, the call is connected to the PSTN on the West MRP. The PSTN then provides a dial tone, and any digits you enter on the telephone thereafter are interpreted on the PSTN.
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Note In this example, West MRP voice port 0/0 has two separate POTS dial peers associated with it. Dial peer 201 matches calls beginning with the digit 9 and handles PSTN calls originating from the West MRP. Dial peer 601 matches calls beginning with the digit 7 and handles calls to the PSTN originating from the East MRP. |
If you configured your FXO interface according to this example, you can place outgoing calls over the PSTN. If you have trouble placing calls, use the show voice port command to make sure that the VIC is installed correctly. Use the show dial-peer voice command to make sure that the data you configured is correct, and test the PSTN by connecting a handset directly to the PSTN outlet and placing a call.
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Note Refer to the section "Configuring Quality of Service" if you need to improve the quality of voice connections. |
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Note To prevent the loss of MRP configuration, refer to the "Saving Configuration Changes" section in this chapter. |
Figure 5-10 shows a company with two offices, West and East. Each office has a PBX to operate its internal telephone network, and the IP network carries voice traffic between the offices. Each PBX connects to an E&M VIC port in the MRP.
To configure E&M voice ports, you need to use the following commands beginning in privileged EXEC mode:
Command | Purpose |
|---|---|
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Both PBXs in this example use E&M interface Type 2, with four-wire operation and immediate-start signaling. The values for your configuration depend on your PBX and are available from your telecommunications department or the PBX manufacturer. For more information about E&M interface configuration commands, refer to the "VoIP Commands" chapter of the Cisco 1750 Router Voice-over-IP Configuration Guide.
In this example, West users can dial 5 and then a 4-digit extension to reach telephones in the East Office. East users can dial 5 and then a 4-digit extension to reach telephones in the West office.
The West MRP connects to the PBX through an E&M VIC port 0/0. This port is associated with a POTS dial peer for incoming calls. But you no longer need to associate every telephone number with its own port. Instead, you can configure a local dial peer as if all the West telephones (represented by a wildcard destination pattern) are connected directly to this port, as shown in the following commands:
| Command | Purpose | |
|---|---|---|
Step 1 | | Enter enable mode. Enter the password. You have entered enable mode when the prompt changes to |
Step 2 | | Enter global configuration mode. You have entered global configuration mode when the prompt changes to |
Step 3 | West (config)# dial-peer voice 111 pots | Enter dial-peer configuration mode. |
Step 4 | West (config-dial-peer)# destination-pattern 1408555.... | Define the destination telephone number associated with the POTS dial peer. |
Step 5 | West (config-dial-peer)# port 0/0 | Enter the port and slot number associated with this POTS dial peer. |
Configure VoIP dial peers for outgoing calls and associate destination phone numbers on the East MRP with that MRP IP address, as shown in Figure 5-11, and in the following commands:
| Command | Purpose | |
|---|---|---|
Step 1 | West (config)# dial-peer voice 121 voip | Enter dial-peer configuration mode. |
Step 2 | West (config-dial-peer)# destination-pattern 1919555.... | Define the destination telephone number associated with the VoIP dial peer. |
Step 3 | West (config-dial-peer# session target ipv4:192.168.11.3 | Enter the port and slot number associated with this VoIP dial peer. |
Step 4 | West (config-dial-peer)# exit | Return to configuration mode. |
Now configure number expansion so that numbers beginning with 5 (belonging to the East office) and sent by the West PBX to the West MRP are expanded into the full destination pattern:
West(config)# num-exp 5.... 1919555....
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Note You do not need to configure number expansion for calls from one West telephone to another West telephone because the PBX switches those calls. |
Finally, enter the following global configuration voice-port command to configure the E&M port:
| Command | Purpose | |
|---|---|---|
Step 1 | West (config)# voice-port 0/0 | Enter voice port configuration mode. |
Step 2 | West (config-voice-port)# signal immediate | Select the signal type for this interface. |
Step 3 | West (config-voice-port)# operation 4-wire | Select the appropriate cabling scheme for this voice port. |
Step 4 | West (config-voice-port)# type 2 | Select the appropriate E&M interface type. |
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Note For these commands to take effect, you have to cycle the port by using the shutdown and no shutdown commands. |
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Note Configure the PBX to pass all dual-tone multifrequency (DTMF) signals to the MRP. |
Configure the East MRP similar to the West MRP. The East MRP connects to the PBX through an E&M VIC port 0/1. Enter the following commands to configure a POTS dial peer for all East telephones:
| Command | Purpose | |
|---|---|---|
Step 1 | East (config)# dial-peer voice 211 pots | Enter dial-peer configuration mode. |
Step 2 | East (config-dial-peer)# destination-pattern 1919555.... | Define the destination telephone number associated with the POTS dial peer. |
Step 3 | East (config-dial-peer)# port 0/1 | Enter the port/slot number associated with this POTS dial peer. |
Enter the following commands to configure a VoIP dial peer for telephones on the West MRP:
| Command | Purpose | |
|---|---|---|
Step 1 | East (config)# dial-peer voice 221 voip | Enter dial-peer configuration mode. |
Step 2 | East (config-dial-peer)# destination-pattern 1408555.... | Define the destination telephone number associated with the POTS dial peer. |
Step 3 | East (config-dial-peer)# session target ipv4:192.168.19.27 | Enter the port/slot number associated with this POTS dial peer. |
Step 4 | East (config-dial-peer)# exit | Return to configuration mode. |
Enter the following commands to configure number expansion and to make it easy for East users to dial numbers on the West MRP:
West(config)# num-exp 5.... 1408555....
Finally, configure the E&M port:
| Command | Purpose | |
|---|---|---|
Step 1 | East (config)# voice-port 0/1 | Enter voice port configuration mode. |
Step 2 | East (config-voice-port)# signal immediate | Select the signal type for this interface. |
Step 3 | East (config-voice-port)# operation 4-wire | Select the appropriate cabling scheme for this voice port. |
Step 4 | East (config-voice-port)# type 2 | Select the appropriate E&M interface type. |
If you configured the E&M interfaces correctly, you can place calls from a telephone served by one PBX to a telephone served by the other PBX (using just the extension, if you configured number expansion). If you have trouble placing calls, ping the remote MRP to make sure you have IP connectivity.
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Note Refer to the section "Configuring Quality of Service" if you need to improve the quality of voice connections. |
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Note To prevent the loss of MRP configuration, refer to the "Saving Configuration Changes" section in this chapter. |
Configure the multiflex T1 (MFT-T1) interface card (also referred to as T1) as a VIC (voice) in a channelized mode. Connecting the VWIC to a PBX or PSTN enables 24 channels or 24 simultaneous calls at any given time. VWIC has the flexibility to combine channels to form channel groups with the same characteristics.
Channel-associated signaling (CAS) is the transmission of signaling information within the voice channel. Various types of CAS signaling are available in the T1 world. The most common forms of CAS signaling are loop-start, ground-start, and E&M. The main disadvantage of CAS signaling is its use of user bandwidth to perform signaling functions. CAS signaling is often referred to as robbed-bit signaling because user bandwidth is being "robbed" by the network for other purposes. In addition to receiving and placing calls, CAS signaling processes the receipt of DNIS and ANI information, which is used to support authentication and other functions.
T1 CAS capabilities have been implemented on the Cisco ICS 7750 to enhance and integrate T1 CAS capabilities on common central office (CO) and PBX configurations for voice calls. The service provider application for T1 CAS includes connectivity to the public network using T1 CAS from the Cisco ICS 7750 to the end office switch.
VoIP for the Cisco ICS 7750 supports the following T1 CAS signaling systems:
Internet service providers can provide switched 56-kbps access to their customers using the Cisco ICS 7750. The subset of T1 CAS (robbed-bit) supported signaling commands are as follows:
Supervisory: line side
Supervisory: trunk side
Digital T1 ports require not only that you set timing, but that you consider the source of the timers. You must configure the time division multiplexing (TDM) clock to specify the clock source. You can specify up to two external clock sources for each MRP, which means that only two of the T1 ports can use line as the clock source. The clock source is selected by using the tdm clock global configuration command.
The MRP supports the following T1 port configurations:
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Note Define the port exporting the clock before importing a clock source from it. |
To configure the tdm clock, use the following commands beginning in privileged EXEC mode:
| Command | Purpose | |
|---|---|---|
Step 1 | | Enter global configuration mode. |
Step 2 | | Query the tdm clock commands by typing a question mark after the command. |
Step 3 | | Specify the slot and port number of the T1 interface to configure. These numbers can be 0/0, 0/1, 1/0, or 1/1. |
Step 4 | | Select T1 interface payload type. The parameters represent as follows: Query for more options by typing a question mark. |
Step 5 | | The parameters represent as follows: |
Step 6 | | Specify that Network Interface (NI) provides clock to the line to which this port is connected. |
Step 7 | | Specify that onboard clock provides the clock to the line to which this port is connected. |
Step 8 | | Specify which end of the circuit provides the clock to the TDM switch. If internal is selected, the onboard clock provides the clocking. If line is selected, the other end of the circuit (PSTN) provides the clocking. |
Step 9 | | Specify the clocking that is provided from one T1 slot and port to the other T1 slot and port on the same MRP. If line is selected, the first T1 is the backup clock source for the second T1. |
Step 10 | | If internal is selected, the onboard clock source is the only backup clock source for the first T1. |
Step 11 | | Specify that the clocking to the T1 0/0 is provided by the onboard clock. |
To understand the export and import options of the tdm clock command, it is necessary to understand the clocking inside the MRP. Use the export option on the port that drives the PLL (phase lock loop) inside the MRP. Use the import option on the port that is driven by the PLL.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice export lineMRP(config)#tdm clock T1 0/1 voice import T1 0/0 internal
Port 0/0 takes the clock source from the PSTN (line) and exports it to the PLL. Port 0/1 then takes the clock source from the PLL (internal) and imports it to the PBX.
Default ports on the MRP have the following features:
Depending on the VICs inserted in the two ports, there are different default tdm clock values on power up. (See Table 5-17.)
| VIC in Slot 0 | VIC in Slot 1 | Default tdm clock Values |
|---|---|---|
2MFT-T1 | 2MFT-T1 | tdm clock T1 0/0 both export line tdm clock T1 0/1 voice import T1 0/0 line (ports 1/0 and 1/1 are undefined) |
Either empty or have an analog VIC | 2MFT-T1 | tdm clock T1 1/0 both export line tdm clock T1 1/1 voice import T1 1/0 line |
1MFT-T1 | 2MFT-T1 | tdm clock T1 0/0 both export line tdm clock T1 1/0 voice import T1 0/0 line (port 1/1 is undefined) |
Either empty or have a serial WIC | 2MFT-T1 (No DSP PVDM present) | tdm clock T1 1/0 data export line (port 1/1 undefined) |
Enter the show tdm clock command to verify your configuration.
MRP# show tdm clock Controller Payload State tdm clock-type 0/0 Voice up Import 0/1 Line 0/1 Voice up Export Line 1/0 Data up Export Line tdm clock Sourced Feed Back-Up 0 0/1 0/0 0/0
To configure T1 CAS for VoIP on the MRP, use the following commands beginning in privileged EXEC mode:
| Command | Purpose | |
|---|---|---|
Step 1 | | Enter global configuration mode. |
Step 2 | | Enter controller configuration mode to configure your controller port. Enter the controller ports as 0/0, 0/1, 1/0 or 1/1. |
Step 3 | | Enter the framing type designated by your telephone company. |
Step 4 | Enter the linecode type designated by your telephone company. | |
Step 5 |
| Enter a number for the ds0 group. Enter 1-24 for range on the T1 port. Configure all channels for E&M, FXS, and FXO analog signaling. Signaling types include e&m-immediate-start, e&m-delay-dial, e&m-wink-start, fxs-ground-start, fxs-loop-start, fxo-ground-start, and fxo-loop-start. You must use the same type of signaling that your central office uses. Repeat Steps 2 through 5 to configure the other three controllers. |
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Note Super Frame AMI 64K timeslot is not reliable because T1 interface requires one's density. |
This section describes the timing scenarios that can occur when different combinations of WICs/VICs are used on the two slots of the MRP. All digital T1 interfaces are connected to a PBX, CO, or both. In all of the examples below, the PSTN (or Central Office) and the PBX can both provide and receive clocking.
The MRP has two on-board PLL (Phase-Lock Loop) chips that can derive clock source from any T1 interface on the system. The clock source is selected by using the tdm clock global configuration command. Both PLL can either provide a clock source to both T1 interfaces or receive clocking that can drive the second T1 interface.
The T1 interface payload type can be defined as either voice, data, or both. In the following scenarios, voice is the most commonly used payload type.
Configuration of TDM clocks affect the DSP groupings. (For information on DSP groups, refer to the "DSP Groups" section.) If only one clock source is used, the DSPs on both the PVDMs can be considered a single pool of DSP resource. If two clock sources are used, each PVDM constitutes a separate pool of DSP resource. If a port is used for an analog VIC, a single PVDM constitutes the DSP resource. Therefore, depending on whether one or two clock sources are defined, the bindings between the DSP resources and the set of ports that they can service can vary.
The MRP supports the following T1 configurations:
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Note These scenarios apply to both data and voice MFT-T1 VWICs. |
Table 5-18 describes the timing scenarios that can occur when slot 0 is a T1 interface and slot 1 is empty.
| Topology | Slot 0 | Slot 1 | PVDM 0 | PVDM 1 | Clocking slot 0/port 0 | Clocking slot 0/port 1 | Clocking slot 1/port 0 | Clocking slot 1/port 1 |
|---|---|---|---|---|---|---|---|---|
Single T1 port 0/0 provides the clock | T1 | None | PVDM-20 | None | Import onboard | None | None | None |
Single T1 port 0/0 receives the clock from the line | T1 | None | PVDM-20 | None | Export line | None | None | None |
Dual T1 ports. Both ports 0/0 and 0/1 receives the clock from the line | T1 | None | PVDM-20 | None | Export line | Export line | None | None |
Dual T1 ports. Both ports 0/0 and 0/1 receive the clock from the line and one is in the loop-timed | T1 | None | PVDM-20 | None | Export line | Import T1 port 0/0 line | None | None |
Dual T1ports. Port 0/0 receives the clock and port 0/1 provides the clock to the line | T1 | None | PVDM-20 | None | Export line | Import T1 port 0/0 internal | None | None |
Dual T1 ports. Both ports 0/0 and 0/1 provide the clock to the line | T1 | None | PVDM-20 | None | Import onboard internal | Import onboard internal | None | None |
Each of the topologies described in Table 5-18 are illustrated below.
In this scenario, the digital T1 port 0/0 is the clock source for the connected device. The PLL generates the clock internally and drives the clocking on the T1 line.
The following configuration sets up this clocking method:
MRP(config)# tdm clock T1 0/0 voice import onboard
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Note Generally this method is useful only when connecting to a PBX, key system or channel bank. A Cisco VoIP Gateway rarely provides clocking to the CO, because CO clocking provides a higher Stratum level. |
In this scenario, the digital T1 port 0/0 receives clocking from the connected device (CO). The PLL clocking is driven by the clock reference on the receive (Rx) side of the T1 connection.
The following configuration sets up this clocking method:
MRP(config)# tdm clock T1 0/0 voice export line
In this scenario, there are two reference clocks. Both T1 ports 0/0 and 0/1 receive clock from the CO.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice export lineMRP(config)#tdm clock T1 0/1 voice export line
In this scenario, T1 port 0/0 receives clocking for the PLL from the CO and puts the T1 port 0/1 connected to the CO into looped-time mode. This is usually the best method because the CO provides an excellent clock source.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice export lineMRP(config)#tdm clock T1 0/1 voice import t1 0/0 line
In this scenario, the digital T1 port 0/0 receives clocking for the PLL from the CO and uses this clock as a reference to clock T1 port 0/1. If T1 port 0/0 fails, the PLL internally generates the clock reference to drive T1 port 0/1.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice export lineMRP(config)#tdm clock T1 0/1 voice import t1 0/0 internal
In this scenario, the MRP generates the clock for the PLL and provides clocking to both T1 port 0/0 connected to a CO and T1 port 0/1 connected to a PBX.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice import onboard internalMRP(config)#tdm clock T1 0/1 voice import onboard internal
Table 5-19 describes the timing scenarios that can occur when both slot 0 and slot 1 are T1 interfaces.
| Topology | Slot 0 | Slot 1 | PVDM 0 | PVDM 1 | Clocking slot 0/port 0 | Clocking slot 0/port 1 | Clocking slot 1/port 0 | Clocking slot 1/port 1 |
|---|---|---|---|---|---|---|---|---|
Dual T1 ports. Both ports 0/0 and 0/1 receives the clock from two different line sources. | T1 | T1 | PVDM-20 | PVDM-20 | Export line | None | Export line | None |
Dual T1 ports. Both ports 0/0 and 0/1 receive the clock from the line and one is in the loop-timed mode. | T1 | T1 | PVDM-20 | PVDM-20 | Export line | None | Import T1 port 0/0 line | None |
Dual T1ports. Port 0/0 receives the clock and port 0/1 provides the clock to the line. | T1 | T1 | PVDM-20 | PVDM-20 | Export line | None | Import T1 port 0/0 internal | None |
Dual T1 ports. Both ports 0/0 and 0/1 provide the clock to the line. | T1 | T1 | PVDM-20 | PVDM-20 | Import onboard | None | Import onboard | None |
Each of the topologies described in Table 5-19 are illustrated below.
In this scenario, there are two reference clocks. Both T1 ports 0/0 and 1/0 receive clock from the CO.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice export lineMRP(config)#tdm clock T1 1/0 voice export line
In this scenario, T1 port 0/0 receives clocking for the PLL from the CO and puts the T1 port 1/0 connected to the CO into looped-time mode. This is usually the best method because the CO provides an excellent clock source.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice export lineMRP(config)#tdm clock T1 1/0 voice import t1 0/0 line
In this scenario, the digital T1 port 0/0 receives clocking for the PLL from the CO and uses this clock as a reference to clock T1 port 1/0. If T1 port 0/0 fails, the PLL internally generates the clock reference to drive T1 port 1/0.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice export lineMRP(config)#tdm clock T1 1/0 voice import t1 0/0 internal
In this scenario, the MRP generates the clock for the PLL and provides clocking to both T1 port 0/0 connected to a CO and T1 port 1/0 connected to a PBX.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice import onboardMRP(config)#tdm clock T1 0/1 voice import onboard
Table 5-20 describes the timing scenarios that can occur when slot 0 is a T1 interface and slot 1 is an analog VIC interface.
| Topology | Slot 0 | Slot 1 | PVDM 0 | PVDM 1 | Clocking slot 0/port 0 | Clocking slot 0/port 1 | Clocking slot 1/port 0 | Clocking slot 1/port 1 |
|---|---|---|---|---|---|---|---|---|
Dual T1 ports. Both ports 0/0 and 0/1 receive the clock from the line and one is in the loop-timed mode. | T1 | VIC | PVDM-20 | PVDM-4 | Export line | Import T1 port 0/0 line | None | None |
Dual T1ports. Port 0/0 receives the clock and port 0/1 provides the clock to the line. | T1 | VIC | PVDM-20 | PVDM-4 | Export line | Import T1 port 0/0 internal | None | None |
Dual T1 ports. Both ports 0/0 and 0/1 provide the clock to the line. | T1 | VIC | PVDM-20 | PVDM-4 | Import onboard | Import onboard | None | None |
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Note The MRP does not support the scenario in which both T1 ports in slot 0 are receiving clock from the line and there is a VIC installed in slot 1. The VIC utilizes one clock source and therefore the two T1 ports cannot receive two other clock sources because there is a limit of two clock sources in the MRP. |
Each of the topologies described in Table 5-20 are illustrated below.
In this scenario, there are two reference clocks. Both T1 ports 0/0 and 0/1 receive clock from the CO.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice export lineMRP(config)#tdm clock T1 0/1 voice export line
In this scenario, T1 port 0/0 receives clocking for the PLL from the CO and puts the T1 port 0/1 connected to the CO into looped-time mode. This is usually the best method because the CO provides an excellent clock source.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice export lineMRP(config)#tdm clock T1 0/1 voice import t1 0/0 line
In this scenario, the digital T1 port 0/0 receives clocking for the PLL from the CO and uses this clock as a reference to clock T1 port 0/1. If T1 port 0/0 fails, the PLL internally generates the clock reference to drive T1 port 0/1.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice export lineMRP(config)#tdm clock T1 0/1 voice import t1 0/0 internal
In this scenario, the MRP generates the clock for the PLL and provides clocking to both T1 port 0/0 connected to a CO and T1 port 0/1 connected to a PBX.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice import onboard internalMRP(config)#tdm clock T1 0/1 voice import onboard internal
Table 5-21 describes the timing scenarios that can occur when slot 0 is a T1 interface and slot 1 is an analog WIC interface.
| Topology | Slot 0 | Slot 1 | PVDM 0 | PVDM 1 | Clocking slot 0/port 0 | Clocking slot 0/port 1 | Clocking slot 1/port 0 | Clocking slot 1/port 1 |
|---|---|---|---|---|---|---|---|---|
Dual T1 ports. Both ports 0/0 and 0/1 receives the clock from two different line sources. | T1 | WIC | PVDM-20 | None | Export line | Export line | None | None |
Dual T1 ports. Both ports 0/0 and 0/1 receive the clock from the line and one is in the loop-timed mode. | T1 | WIC | PVDM-20 | None | Export line | Import T1 port 0/0 line | None | None |
Dual T1ports. Port 0/0 receives the clock and port 0/1 provides the clock to the line. | T1 | WIC | PVDM-20 | None | Export line | Import T1 port 0/0 internal | None | None |
Dual T1 ports. Both ports 0/0 and 0/1 provide the clock to the line. | T1 | WIC | PVDM-20 | None | Import onboard | Import onboard | None | None |
|
Note T1 data should not be configured on port 1 of either slot when a WIC is present on the same MRP because WIC takes the same data path as port 1 of the other slot. |
Each of the topologies described in Table 5-21 are illustrated below.
In this scenario, there are two reference clocks. Both T1 ports 0/0 and 0/1 receive clock from the CO.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice export lineMRP(config)#tdm clock T1 0/1 voice export line
In this scenario, T1 port 0/0 receives clocking for the PLL from the CO and puts the T1 port 0/1 connected to the CO into looped-time mode. This is usually the best method because the CO provides an excellent clock source.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice export lineMRP(config)#tdm clock T1 0/1 voice import t1 0/0 line
In this scenario, the digital T1 port 0/0 receives clocking for the PLL from the CO and uses this clock as a reference to clock T1 port 0/1. If T1 port 0/0 fails, the PLL internally generates the clock reference to drive T1 port 0/1.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice export lineMRP(config)#tdm clock T1 0/1 voice import t1 0/0 internal
In this scenario, the MRP generates the clock for the PLL and provides clocking to both T1 port 0/0 connected to a CO and T1 port 0/1 connected to a PBX.
The following configuration sets up this clocking method:
MRP(config)#tdm clock T1 0/0 voice import onboard internalMRP(config)#tdm clock T1 0/1 voice import onboard internal
To verify that your controller is up and running and no alarms have been reported, enter the show controller t1 command and specify the port number.
MRP# show controller t1 0/0
T1 0/0 is up.
No alarms detected.
Version info of slot 0: HW: 2, Firmware: 16, PLD Rev: 0
Manufacture Cookie Info:
EEPROM Type 0x0001, EEPROM Version 0x01, Board ID 0x42,
Board Hardware Version 1.0, Item Number 73-2217-4,
Board Revision A0, Serial Number 06467665,
PLD/ISP Version 0.0, Manufacture Date 14-Nov-1997.
Framing is ESF, Line Code is B8ZS, Clock Source is Internal.
Data in current interval (269 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
Note the following:
For troubleshooting information on T1 controller command, refer to the Cisco ICS 7750 Administration and Troubleshooting Guide.
|
Note To prevent the loss of MRP configuration, refer to the "Saving Configuration Changes" section in this chapter. |
Use a dial plan to map the destination telephone numbers with the voice ports on the MRP. In North America, the North American Numbering Plan (NANP) is used, which consists of an area code, an office code, and a station code. Area codes are assigned geographically, office codes are assigned to specific switches, and station codes identify a specific port on that switch. The format in North America is 1Nxx-Nxx-xxxx, with N = digits 2 through 9 and x = digits 0 through 9. Internationally, each country is assigned a one- to three-digit country code; the country's dialing plan follows the country code.
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 E.164 telephone number. VoIP can be configured to recognize extension numbers and expand them into their full E.164 dialed number by using two commands in tandem: destination-pattern and num-exp. Before you configure these two commands, it helps to map individual telephone extensions with their full E.164 dialed numbers. This can be done easily by creating a number expansion table.
For Cisco voice implementations, two types of dial peers are used to match a dialed number to either a local telephony port or a remote IP address:
West(config)# dial-peer voice 401 pots West(config-dial-peer)# destination-pattern 14085553737 West(config-dial-peer)# port 0/0
West(config)# dial-peer voice 501 voip West(config-dial-peer)# destination-pattern 1919555.... West(config-dial-peer)# session target ipv4:192.168.11.3
Use the dial-peer voice command to define dial peers and change to dial-peer configuration mode. To see an example, refer to the individual sections on configuring FXS, FXO, and E&M interfaces earlier in this chapter.
In Figure 5-31, a small company decides to use VoIP to integrate its telephony network with its existing IP network. The destination pattern (or expanded telephone number) associated with MRP 1 (left of the IP cloud) is (408) 555-xxxx, where xxxx identifies the individual dial peers by extension. The destination pattern (or expanded telephone number) associated with MRP 2 (right of the IP cloud) is (729) 555-xxxx.
Table 5-22 shows the number expansion table for this scenario.
| Extension | Destination Pattern | Num-Exp Command Entry | Description |
|---|---|---|---|
1... | 14085551... | num-exp 1... 14085551... | To expand a 4-digit extension beginning with the numeral 1 by prefixing 1408555 to it. |
2... | 14085552... | num-exp 2... 14085552... | To expand a 4-digit extension beginning with the numeral 2 by prefixing 1408555 to it. |
3... | 17295553... | num-exp 3... 17295553... | To expand a 4-digit extension beginning with the numeral 3 by prefixing 1729555 to it. |
4... | 17295554... | num-exp 4... 17295554... | To expand a 4-digit extension beginning with the numeral 4 by prefixing 1729555 to it. |
|
Note You can use a period (.) to represent variables (such as extension numbers) in a telephone number. A period is similar to a wildcard, which matches any entered digit. |
The information included in this example needs to be configured on both MRP 1 and MRP 2. In this configuration, MRP 1 can call any number string that begins with the digits 17295553 or 17295554 to connect to MRP 2. Similarly, MRP 2 can call any number string that begins with the digits 14085551 and 14085552 to connect to MRP 1.
To define how to expand an extension number into a particular destination pattern, use the following global configuration command:
East(config)# num-exp extension-number extension-string
Use the show num-exp command to verify that you have mapped the telephone numbers correctly.
East# show num-exp dialed-number
After you have configured dial peers and assigned destination patterns to them, use the show dialplan number command to see how a telephone number maps to a dial peer.
East# show dialplan number dialed-number
|
Note You must still configure each telephone number in full on a local dial peer so that the MRP can find the voice port it belongs to. |
You need to take certain factors into consideration when configuring VoIP so that it runs smoothly over Frame Relay. A public Frame Relay cloud provides no guarantees for QoS. For real-time traffic to be transmitted in a timely manner, the data rate must not exceed the committed information rate (CIR), or packets might be dropped. In addition, Frame Relay traffic shaping and RSVP are mutually exclusive. This is particularly important to remember if multiple data-link connection identifiers (DLCIs) are carried on a single interface.
For Frame Relay links with slow output rates (less than or equal to 64 kbps), where data and voice are being transmitted over the same permanent virtual circuit (PVC), we recommend the following solutions:
|
Note Lowering the MTU size affects data throughput speed. |
In IOS Release 12.0, Frame Relay traffic shaping is not compatible with RSVP. We suggest one of the following workarounds:
interface Serial 0/0 mtu 300 no ip address encapsulation frame-relay no ip route-cache no ip mroute-cache fair-queue 64 256 1000 frame-relay ip rtp header-compression interface Serial 0/0.1 point-to-point mtu 300 ip address 40.0.0.7 255.0.0.0 ip rsvp bandwidth 48 48 no ip route-cache no ip mroute-cache bandwidth 64 traffic-shape rate 32000 4000 4000 frame-relay interface-dlci 16 frame-relay ip rtp header-compression
In this configuration example, the main interface is configured as follows:
The subinterface is configured as follows:
|
Note When traffic bursts over the CIR, the output rate is held at the speed configured for the CIR (for example, traffic will not go beyond 32 kbps if CIR is set to 32 kbps). |
For more information about configuring Frame Relay for VoIP, refer to the "Configuring Frame Relay" chapter in the Wide-Area Networking Configuration Guide for IOS Release 12.0.
To prevent the loss of MRP configuration, refer to the "Saving Configuration Changes" section in this chapter.
You need to have a well-engineered, end-to-end network when running delay-sensitive applications such as VoIP. Voice traffic is much more sensitive to timing variations than data traffic. For good voice performance, you need to configure your data network so that voice packets are not lost or delayed. Fine-tuning your network to adequately support VoIP involves a series of protocols and features to improve quality of service (QoS). It is beyond the scope of this document to explain the specific details relating to wide-scale QoS deployment. Cisco IOS software provides many tools for enabling QoS on your backbone, such as Random Early Detection (RED), Weighted Random Early Detection (WRED), Fancy Queuing (meaning custom, priority, or weighted fair queuing), and IP precedence. To configure your IP network for real-time voice traffic, you must take into consideration the entire scope of your network and then select the appropriate QoS tool or tools.
|
Note QoS measures the level of network performance. It does not directly measure the quality of the voice signal. |
The important thing to remember is that QoS must be configured throughout your network—not just on your MRP running VoIP—to improve voice network performance.
On a relatively low-bandwidth connection, such as a PPP or High-Level Data Link Control (HDLC) serial link, you should consider using methods to ensure QoS. If you have a high-bandwidth network, such as Ethernet or Fast Ethernet, and voice and data traffic together occupy only a small fraction of the bandwidth available, you might not need to provide QoS. (See Figure 5-32.)
Although not mandatory, some QoS tools can be valuable in fine-tuning your network to support real-time voice traffic. To configure your IP network for QoS, perform one or more of the tasks in the following sections:
16384 + (4 x number of voice ports in the MRP)
Custom Queuing and other methods for identifying high priority streams should be configured for these port ranges. For more information about custom queuing, refer to the "Managing System Performance" chapter in the Configuration Fundamentals Configuration Guide for IOS Release 12.0.
In general, weighted fair queuing is used in conjunction with multilink PPP with interleaving and RSVP or IP precedence to ensure voice packet delivery. Use weighted fair queuing with multilink PPP to define how data is managed; use RSVP or IP precedence to give priority to voice packets. For more information about weighted fair queuing, refer to the "Managing System Performance" chapter in the Configuration Fundamentals Configuration Guide for IOS Release 12.0.
MRP> enable Password: MRP# configure terminal MRP(config)# dial-peer voice 221 voip MRP(config-dial-peer)# ip precedence 5
The ip precedence command should also be used if RSVP is not enabled and you want to give voice packets a higher priority over other IP data traffic.
Resource Reservation Protocol (RSVP) enables MRPs to reserve enough bandwidth on an interface for reliability and quality performance. RSVP allows end systems to request a particular QoS from the network. Real-time voice traffic requires network consistency. Without consistent QoS, real-time traffic can experience jitter, insufficient bandwidth, delay variations, or information loss. RSVP works in conjunction with current queuing mechanisms. It is up to the interface queuing mechanism (such as weighted fair queuing or WRED) to implement the reservation.
RSVP works well on PPP, HDLC, and similar serial line interfaces. It does not work well on multi-access LANs. RSVP can be equated to a dynamic access list for packet flows.
You should configure RSVP to ensure QoS if the following conditions describe your network:
To minimally configure RSVP for voice traffic, you must enable RSVP on each interface where priority must be set.
By default, RSVP is disabled so that it is backwards compatible with systems that do not implement RSVP. To enable RSVP for IP on an interface, use the following interface configuration command:
MRP(config-if)# ip rsvp bandwidth [interface-kbps] [single-flow-kbps]
This command starts RSVP and sets the bandwidth and single-flow limits. The default maximum bandwidth is up to 75 percent of the bandwidth available on the interface. By default, the amount reservable by a flow can be up to the entire reservable bandwidth.
On subinterfaces, RSVP applies to the more restrictive of the available bandwidths of the physical interface and the subinterface.
Reservations on individual circuits that do not exceed the single flow limit normally succeed. However, if reservations are made on other circuits adding up to the line speed, and a reservation is made on a subinterface that has enough remaining bandwidth, reservation will still be refused because the physical interface lacks supporting bandwidth.
A MRP running VoIP and configured for RSVP requests allocations using the following formula:
bps=packet_size+ip/udp/rtp header size * 50 per second
For G.729, the allocation works out to be 24,000 bps. For G.711, the allocation is 80,000 bps.
For more information about configuring RSVP, refer to the "Configuring RSVP" chapter of the Network Protocols Configuration Guide, Part 1 for Cisco IOS Release 12.0.
The following example enables RSVP and sets the maximum bandwidth to 100 kbps and the maximum bandwidth per single request to 32 kbps (the example assumes that both VoIP dial peers are configured):
MRP(config)# interface serial 0 MRP(config-if)# ip rsvp bandwidth 100 32 MRP(config-if)# fair-queue MRP(config-if)# end
After enabling RSVP, you must also use the req-qos dial-peer configuration command to request an RSVP session on each VoIP dial peer. Otherwise, no bandwidth is reserved for voice traffic.
MRP(config)# dial-peer voice 211 voip MRP(config-dial-peer)# req-qos controlled-load MRP(config)# dial-peer voice 212 voip MRP(config-dial-peer)# req-qos controlled-load
In general, multilink PPP with interleaving is used in conjunction with weighted fair queuing and RSVP or IP precedence to ensure voice packet delivery. Use multilink PPP with interleaving and weighted fair queuing to define how data is managed; use RSVP or IP precedence to give priority to voice packets.
You should configure multilink PPP if the following conditions describe your network:
|
Note Do not use multilink PPP on links greater than 2 Mbps. |
Multilink PPP support for interleaving can be configured on virtual templates, dialer interfaces, and ISDN BRI or PRI interfaces. To configure interleaving, you need to complete the following tasks:
To configure multilink PPP and interleaving on a configured and operational interface or virtual interface template, use the following interface configuration commands:
| Command | Purpose | |
|---|---|---|
Step 1 | Enable Multilink PPP. | |
Step 2 | Enable real-time packet interleaving. | |
Step 3 | ppp multilink fragment-delay milliseconds | Optionally, configure a maximum fragment delay of 20 milliseconds. |
Step 4 | ip rtp reserve lowest-UDP-port range-of-ports [maximum-bandwidth] | Reserve a special queue for real-time packet flows to specified destination UDP ports, allowing real-time traffic to have higher priority than other flows. This only applies if you have not configured RSVP. |
|
Note You can use the ip rtp reserve command instead of configuring RSVP. If you configure RSVP, this command is not required. |
For more information about multilink PPP, refer to the "Configuring Media-Independent PPP and Multilink PPP" chapter in the Dial Solutions Configuration Guide for Cisco IOS Release 12.0.
The following example defines a virtual interface template that enables multilink PPP with interleaving and a maximum real-time traffic delay of 20 milliseconds and then applies that virtual template to the multilink PPP bundle:
MRP(config)# interface virtual-template 1 MRP(config-if)# ppp multilink MRP(config-if)# encapsulated ppp MRP(config-if)# ppp multilink interleave MRP(config-if)# ppp multilink fragment-delay 20 MRP(config-if)# ip rtp reserve 16384 100 64 MRP(config)# multilink virtual-template 1
Real-Time Transport Protocol (RTP) is used for carrying audio traffic in packets over an IP network. RTP header compression compresses the IP/UDP/RTP header in an RTP data packet from 40 bytes to approximately 2 to 4 bytes (most of the time), as shown in Figure 5-33.
This compression feature is beneficial if you are running VoIP over slow links. Enabling compression on both ends of a low-bandwidth serial link can greatly reduce the network overhead if there is a lot of RTP traffic on that slow link.
Typically, an RTP packet has a payload of approximately 20 to 160 bytes for audio applications that use compressed payloads. RTP header compression is especially beneficial when the RTP payload size is small (for example, compressed audio payloads between 20 and 50 bytes).
You should configure RTP header compression if the following conditions describe your network:
|
Note Do not use RTP header compression on links greater than 2 Mbps. |
Perform the following tasks to configure RTP header compression for VoIP. The first task is required; the second task is optional.
You need to enable compression on both ends of a serial connection. To enable RTP header compression, use the following interface configuration command:
MRP(config-if)# ip rtp header-compression [passive]
If you include the passive keyword, the software compresses outgoing RTP packets only if incoming RTP packets on the same interface are compressed. If you use the command without the passive keyword, the software compresses all RTP traffic.
By default, the software supports a total of 16 RTP header compression connections on an interface. To specify a different number of RTP header compression connections, use the following interface configuration command:
MRP(config-if)# ip rtp compression connections number
The following example enables RTP header compression for a serial interface:
MRP(config)# interface serial0 MRP(config-if)# ip rtp header-compression MRP(config-if)# encapsulation ppp MRP(config-if)# ip rtp compression-connections 25
For more information about RTP header compression, see the "Configuring IP Multicast Routing" chapter of the Network Protocols Configuration Guide, Part 1 for Cisco IOS Release 12.0.
|
Note When you enter the show running-config command, the format of the ip rtp header-compression command will change to ip rtp header-compression iphc-format. |
After configuring the new interface, you can perform the following tests to verify that the new interface is operating correctly:
MRP_Slot1#show version Cisco Internetwork Operating System Software IOS (tm) C1700 Software (C1700-SV3Y-M), Experimental Version 12.0(20000119:032355) [jlu-jlu-test_0118 100] Copyright (c) 1986-2000 by cisco Systems, Inc. Compiled Tue 18-Jan-00 19:23 by jlu Image text-base:0x80008088, data-base:0x807C2C10 ROM:System Bootstrap, Version 12.0(3)T, RELEASE SOFTWARE (fc1) MRP_Slot1 uptime is 1 day, 1 hour, 50 minutes System returned to ROM by reload Running default software cisco 1750T (MPC860) processor (revision 0x00) with 36864K/12288K bytes of memory. Processor board ID 0000 (1314672220), with hardware revision 0000 M860 processor:part number 0, mask 32 Bridging software. X.25 software, Version 3.0.0. Primary Rate ISDN software, Version 1.1. 1 FastEthernet/IEEE 802.3 interface(s) 1 Serial(sync/async) network interface(s) 2 Channelized T1/PRI port(s) 32K bytes of non-volatile configuration memory. Configuration register is 0x0
MRP_Slot1#show controller t1 0/0
T1 0/0 is up.
Applique type is Channelized T1
Cablelength is long gain36 0db
No alarms detected.
Version info Firmware:19990616, FPGA:6
Framing is ESF, Line Code is B8ZS, Clock Source is Line.
Data in current interval (21 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
MRP_Slot1#show protocols Global values: Internet Protocol routing is enabled FastEthernet0 is up, line protocol is up Internet address is 1.16.164.201/16 Serial0 is down, line protocol is down Internet address is 199.1.1.1/24
MRP_Slot1#show interfaces
FastEthernet0 is up, line protocol is up
Hardware is PQUICC_FEC, address is 0050.73ff.727f (bia 0050.73ff.727f)
Internet address is 1.16.164.201/16
MTU 1500 bytes, BW 100000 Kbit, DLY 100 usec,
reliability 255/255, txload 1/255, rxload 1/255
Encapsulation ARPA, loopback not set
Keepalive set (10 sec)
Full-duplex, 100Mb/s, 100BaseTX/FX
ARP type:ARPA, ARP Timeout 04:00:00
Last input 00:00:00, output 00:00:09, output hang never
Last clearing of "show interface" counters 00:00:07
Queueing strategy:fifo
Output queue 0/40, 0 drops; input queue 0/75, 0 drops
5 minute input rate 4000 bits/sec, 4 packets/sec
5 minute output rate 0 bits/sec, 0 packets/sec
42 packets input, 5003 bytes
Received 42 broadcasts, 0 runts, 0 giants, 0 throttles
0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored
0 watchdog, 0 multicast
0 input packets with dribble condition detected
0 packets output, 0 bytes, 0 underruns
0 output errors, 0 collisions, 0 interface resets
0 babbles, 0 late collision, 0 deferred
0 lost carrier, 0 no carrier
0 output buffer failures, 0 output buffers swapped out
Serial0 is down, line protocol is down
Hardware is PowerQUICC Serial
Internet address is 199.1.1.1/24
MTU 1500 bytes, BW 1544 Kbit, DLY 20000 usec,
reliability 255/255, txload 1/255, rxload 1/255
Encapsulation FRAME-RELAY, loopback not set
Keepalive not set
FR SVC disabled, LAPF state down
Broadcast queue 0/64, broadcasts sent/dropped 0/0, interface broadcasts 0
Last input 05:34:50, output 05:34:57, output hang never
Last clearing of "show interface" counters 00:00:09
Input queue:0/75/0 (size/max/drops); Total output drops:0
Queueing strategy:weighted fair
Output queue:0/1000/64/0 (size/max total/threshold/drops)
Conversations 0/1/256 (active/max active/max total)
Reserved Conversations 0/0 (allocated/max allocated)
5 minute input rate 0 bits/sec, 0 packets/sec
5 minute output rate 0 bits/sec, 0 packets/sec
0 packets input, 0 bytes, 0 no buffer
Received 0 broadcasts, 0 runts, 0 giants, 0 throttles
0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort
0 packets output, 0 bytes, 0 underruns
0 output errors, 0 collisions, 0 interface resets
0 output buffer failures, 0 output buffers swapped out
0 carrier transitions
DCD=up DSR=up DTR=down RTS=down CTS=up
MRP_Slot1#show running-config Building configuration... Current configuration: ! version 12.0 service timestamps debug uptime service timestamps log uptime no service password-encryption ! hostname MRP_Slot1 ! no logging console ! ! memory-size iomem 25 tdm clock T1 0/0 voice export line tdm clock T1 0/1 voice import T1 0/0 internal ip subnet-zero no ip domain-lookup ! ! !controller T1 0/0 ds0-group 0 timeslots 1-24 type e&m-wink-start ! controller T1 0/1 ds0-group 1 timeslots 1-24 type e&m-wink-start ! ! voice-port 0/0:0 ! voice-port 0/1:1 ! dial-peer voice 102 voip destination-pattern 250071100. codec g711ulaw session target ipv4:199.1.1.2 ! dial-peer voice 1021 voip destination-pattern 250072900. session target ipv4:1.16.164.105 ! dial-peer voice 200 pots destination-pattern 140071100. port 0/0:0 prefix 140071100 ! process-max-time 200 ! interface FastEthernet0/0 ip address 1.9.28.201 255.255.0.0 speed auto ! interface Serial1/0 bandwidth 1544 ip address 199.1.1.1 255.255.255.0 encapsulation ppp no keepalive clockrate 4000000 ip rsvp bandwidth ip classless no ip http server ! snmp-server engineID local 000000090200005073FF727F snmp-server community xena RO snmp-server community hercules RW ! line con 0 exec-timeout 0 0 transport input none line aux 0 line vty 0 4 login ! ! no scheduler allocate end
If an interface is down and you configured it as up, or if the displays indicate that the hardware is not functioning properly, make sure that the new interface is properly connected and configured.
At this point you can proceed to the following:
Posted: Mon Oct 2 14:00:39 PDT 2000
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