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Cisco IOS Multiservice Applications Configuration Guide, Release 12.1
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About the Cisco IOS Software Documentation
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Using Cisco IOS Software
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Multiservice Applications Overview
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Part 1: Voice
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Configuring Voice over IP
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Configuring Voice Ports for Voice over IP
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Configuring Gatekeepers (Multimedia Conference Manager)
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Configuring Interactive Voice Response
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Configuring Debit Card for Packet Telephony
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Configuring Settlement for Packet Telephony
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Managing Cisco AS5300 Voice Feature Cards
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Configuring Voice over Frame Relay
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Configuring Voice over ATM
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Configuring Voice over HDLC
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Configuring Voice-Related Support Features
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Configuring PBX Signaling
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Configuring Store and Forward Fax
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- Part 2: Video
- Part 3: Broadband
- Part 4: Appendix
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Table Of Contents
Configuring Voice over Frame Relay
Voice over Frame Relay Overview
Cisco-Trunk (Private Line) Calls
End-to-End FRF.12 Fragmentation
Frame Relay Fragmentation Using FRF.11 Annex C
Cisco Proprietary Voice Encapsulation
Frame Relay Fragmentation Conditions and Restrictions
FRF.ll Implementation Agreement Support
Voice Ports on the Cisco 2600 and 3600 Series Routers
Voice Ports on the Cisco MC3810
Configuring FXO or FXS Voice Ports
Verifying the FXO or FXS Voice Port Configuration
Fine-Tuning FXO and FXS Voice Ports
Verifying the E&M Voice Port Configuration
Configuring Frame Relay to Support VoFR
Configuring a Map Class to Support VoFR
Configuring a Frame Relay Map Class to Support Voice Traffic
Configuring a Frame Relay Map Class to Support FRF.12 Fragmentation
Configuring a Frame Relay Map Class for Traffic Shaping Parameters
Verifying Your Frame Relay Configuration
Preparing to Configure Voice Dial Peers
Organizing Voice Network Information
Creating a Peer Configuration Table
Configuring Dial Plan Options for the POTS Dial Peer
Configuring VoFR Dial Peers for Switched Calls
Configuring VoFR Dial Peers for Cisco-Trunk (Private Line) Calls
Configuring VoFR Dial Peers for FRF.11 Trunk Calls
Configuring VoFR Dial Peers for Tandem Nodes
Disabling Dial-Peer Hunting on a Dial Peer (Cisco MC3810 Only)
Overview of VoFR Connection Types
Configuring Switched Calls (User Dialed or Auto-Ringdown)
Configuring Switched Calls on Cisco 2600, 3600, and 7200 Series Routers
Configuring Switched Calls on Cisco 2600, 3600, or 7200 Series Routers to a Cisco MC3810
Configuring Switched Calls on a Cisco MC3810
Configuring Cisco-Trunk (Private Line) Permanent Calls
Configuring Cisco Trunk Permanent Calls on Cisco 2600, 3600, and 7200 Series Routers
Configuring Cisco Trunk Permanent Calls on Cisco 2600 and 3600 Routers to a Cisco MC3810
Configuring Cisco Trunk Permanent Calls on a Cisco MC3810
Configuring FRF.11 Trunk (Private Line) Calls
Configuring Connections for Tandem Nodes
Verifying Your Voice Connections
Monitoring and Maintaining Your VoFR Configuration
Dial Planning and Dial-Peer Digit Manipulation Options
Forward Digits and Voice Default Routes
Hunt Groups and Preference Configuration
Two Routers Using Frame Relay Fragmentation Example
Two Routers Using a VoFR PVC Example
Router Using a VoFR PVC to a Cisco MC3810 Example
Cisco MC3810 Concentrators Using Different VoFR Configuration Methods Example
Cisco-Trunk (Private Line) Calls Between Two Routers Example
FRF.11 Trunk Calls Between Two Routers Example
Tandem Configuration with Three Routers for Switched Calls Example
Tandem Configuration with a Cisco MC3810 Tandem Node for Switched Calls Example
Tandem Configuration with a Cisco MC3810 Endpoint Node for Switched Calls Example
Tandem Configuration with a Cisco MC3810 Endpoint Node for Cisco-Trunk (Private Line) Calls Example
Tandem Configuration with All Cisco MC3810 Concentrators for Switched Calls Example
Cisco-Trunk Call with Hunt Groups Example
Configuring Voice over Frame Relay
This chapter shows you how to configure Voice over Frame Relay. This chapter contains the following sections:
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Voice over Frame Relay Overview
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For a description of the commands used to configure Voice over Frame Relay, refer to the Cisco IOS Multiservice Applications Command Reference publication.
Voice over Frame Relay Overview
Voice over Frame Relay enables a router to carry voice traffic (for example, telephone calls and faxes) over a Frame Relay network.
This chapter describes the commands to specifically configure Voice over Frame Relay on a router. It is assumed you have already configured your Frame Relay backbone network. As part of your Frame Relay configuration, you need to configure the map class, and the Local Management Interface (LMI) among other Frame Relay functionality. For more information about Frame Relay configuration, refer to the Cisco IOS Wide-Area Networking Configuration Guide.
The Cisco VoFR implementation allows the following types of VoFR calls:
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This section describes the setup of Cisco VoFR calls. In addition, the following functionality is described:
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Static FRF.11 trunks and permanent switched trunks are used to create fixed point-to-point connections, which are typically used to connect two PBXs. In this case, the VoFR system simply provides transportation of the voice connection channels, but does not provide dial-plan-based telephone call switching. This functionality is sometimes referred to as tie-line emulation. In this scenario, all telephone call switching is performed by the PBXs.
With dynamic switched calls, the VoFR system includes dial-plan information that is used to process and route the calls based on the telephone numbers dialed by the callers. The dial-plan information is contained within dial-peer entries.
Static FRF.11 Trunks
FRF.11 trunks allow for standards-based vendor interoperability by specifying the frame format and coder types to be used when sending voice traffic through a Frame Relay network; however, FRF.11 includes no specifications for end-to-end negotiation, call setup process, or any other form of communication between the Frame Relay nodes. As a result, static FRF.11 trunks must be set up by manually configuring each router within the voice trunk path with compatible parameters—a voice port and a specific subchannel on a data-link connection identifier (DLCI) are explicitly bound on each end router. Signalling information is packed and sent transparently end to end.
There is no possibility of automatic enforcement of compatible configuration parameters between the two ends of an FRF.11-based call. For example, it is possible to incorrectly configure the two ends of an FRF.11 call using different and incompatible speech compression CODECs. In this situation, the call will exist and voice packets will be sent and received, but no usable voice path will be created.
When configured, a static FRF.11 trunk remains up until the voice port or serial port is shut down, or until a network disruption occurs. The FRF.11 specification does not include any standardized methods for performing Operation, Administration, and Maintenance (OAM) functions. There is no standard protocol for detecting faults and providing rerouting of connection paths.
FRF.11 allows up to 255 subchannels to be multiplexed onto a single Frame Relay DLCI. The current implementation supports the multiplexing of a single data channel with many voice channels. (Subchannels 0 to 3 are reserved and cannot be configured either for voice or data.)
FRF.11 can only be used when an end-to-end permanent virtual circuit (PVC) is available between the voice ports at each end of the connection. At intermediate Frame Relay nodes, you must route the entire PVC; connection ID-based routing (individual Channel ID switching) is not supported. Because the entire PVC is routed, no prioritization of voice packets is possible at the intermediate Frame Relay nodes.
The connection trunk voice-port configuration command is used to establish a static FRF.11 trunk; the dial peer is configured using the session protocol frf11-trunk command, which invokes the FRF.11-compliant session protocol.
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Cisco-Switched VoFR
Cisco addresses the lack of end-to-end call parameter negotiation and call setup syntax in FRF.11 by implementing a proprietary Q.931-like session protocol running on a user-configurable channel identifier (CID) of an FRF.11-format multiplexed DLCI. The Cisco-switched VoFR protocol handles call setup and parameter negotiation for both endpoints and intermediate nodes within the (multihop) call path. The call setup mechanism originally implemented in the Cisco MC3810 is used; this mechanism can be used for either permanent switched (Cisco-trunk) or dynamic switched calls. The Cisco-switched VoFR protocol includes forwarding of the called telephone number and supports tandem switching of the call over multiple Frame Relay PVC hops.
A tandem node is an intermediate router node within the Frame Relay call path. Its purpose is to switch the frames from one PVC subchannel to another (from one VoFR dial peer onto another VoFR dial peer) as the frames traverse the network. Use of tandem router nodes also avoids the need to have complete dial-plan information present on every router.
The Cisco MC3810 also supports Voice over ATM (VoATM) and Voice over HDLC (VoHDLC). The Cisco MC3810 is able to tandem switch voice calls between VoFR, VoATM, and VoHDLC call legs. The Cisco 2600 series and 3600 series routers also support VoIP. It is not possible in this release to translate from the VoIP transport protocol to other protocols such as VoFR. As a result, a call coming in on a VoIP connection may not be (tandem) switched to a VoFR connection.
Dynamic Switched Calls
Dynamic switched calls are regular telephone calls in which the dial-plan-based call switching is performed by the Cisco router. The destination endpoint of the call is selected by the router based on the telephone number dialed and the dial-plan information contained in the dial-peer configuration entries. Contrast this implementation with permanent calls (Cisco-trunk calls), where the call endpoints are permanently fixed at configuration time.
The dial peer is configured using the session protocol cisco-switched dial-peer configuration command, which uses the Cisco proprietary session protocol.
Cisco-Trunk (Private Line) Calls
A Cisco-trunk (private line) call is basically a normal dynamic switched call of indefinite duration that uses a fixed destination telephone number and includes optional transparent end-to-end signalling. The telephone number of the destination endpoint is permanently configured into the router so that it always selects a fixed destination. Once established, either at boot-up or when configured, the call stays up until one of the voice ports or network ports is shut down, or until a network disruption occurs.
The connection trunk voice-port command is used to establish a Cisco-trunk call; the dial peer is configured using the session protocol cisco-switched command, which invokes the Cisco proprietary session protocol.
Frame Relay Fragmentation
Cisco has developed three methods of performing Frame Relay fragmentation, which are described in the following sections:
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Frame Relay fragmentation can be configured in conjunction with VoFR or independently of it.
End-to-End FRF.12 Fragmentation
FRF.12 fragmentation is defined by the FRF.12 standard. The FRF.12 Implementation Agreement was developed to allow long data frames to be fragmented into smaller pieces and interleaved with real-time frames. In this way, real-time voice and non-real-time data frames can be carried together on lower speed links without causing excessive delay to the real-time traffic.
Use this fragmentation type when the PVC is not carrying voice, but is sharing the link with other PVCs that are carrying voice. The fragmentation header is only included for frames that are greater than the fragment size configured. As a result, FRF.12 is the recommended fragmentation to be used by VoIP.
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The Cisco 2600 series, 3600 series, and 7200 series routers and the Cisco MC3810 multiservice access concentrator support end-to-end fragmentation on a per-PVC basis. Fragmentation is configured through a map class, which can apply to one or many PVCs, depending on how the class is applied.
Frame Relay Fragmentation Using FRF.11 Annex C
When VoFR (FRF.11) and fragmentation are both configured on a PVC, the Frame Relay fragments are sent in the FRF.11 Annex C format.
This fragmentation is used when FRF.11 voice traffic is sent on the PVC, and it uses the FRF.11 Annex C format for data.
With FRF.11, all data packets contain fragmentation headers regardless of size. This form of fragmentation is not recommended for use with Voice over IP (VoIP).
Cisco Proprietary Voice Encapsulation
Cisco proprietary voice fragmentation was implemented on earlier releases of the Cisco MC3810 multiservice access concentrator. This fragmentation type is used on data packets on a PVC that is also used for voice traffic. When the vofr cisco command is configured on a DLCI and fragmentation is enabled on a map class, the Cisco 2600 series, 3600 series, and 7200 series routers can interoperate as tandem nodes (but cannot perform call termination) with Cisco MC3810 concentrators running Cisco IOS releases prior to 12.0(3)XG or 12.0(4)T.
On the Cisco 2600, 3600, and 7200 series routers, entering the vofr cisco command is the only method for configuring Cisco proprietary voice encapsulation. You must then configure a map class to enable voice traffic on the PVCs.
On the Cisco MC3810, you have two methods for configuring Cisco proprietary voice encapsulation:
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When using this method, you must configure a map class to enable voice traffic.
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When using this method, you can configure either an individual DLCI or configure a map class to enable voice traffic. This command is not supported on the Cisco 2600 and 3600 series routers.
These commands are mutually exclusive on the interface and each provides different advantages. The vofr cisco command uses weighted fair queueing for controlling the data flows. The frame-relay interface-dlci voice-encap provides only simple first-in, first-out (FIFO) queueing and does not provide prioritization between nonvoice (data) flows. The command does provide prioritization for voice flows.
Because the frame-relay interface-dlci voice-encap command provides only FIFO queueing, this method has less overhead than the vofr cisco command. However, the vofr cisco command provides greater control over the queueing mechanism.
Frame Relay Fragmentation Conditions and Restrictions
When Frame Relay fragmentation is configured, the following conditions and restrictions apply:
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FRF.ll Implementation Agreement Support
VoFR provides support for different FRF.11 forum features depending on the hardware platform used (see Table 21).
Prerequisite Tasks
Before you can configure your router to use VoFR, you must first perform the following tasks:
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After you have analyzed your dial plan and decided how to integrate it into your existing Frame Relay network, you are ready to configure your network devices to support VoFR.
VoFR Configuration Task List
VoFR has a set of core tasks. To configure VoFR, perform the following tasks:
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Configuring Voice Ports
This section describes how to configure voice ports to support VoFR.
This section is divided into the following procedures:
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Voice ports provide support for three basic voice signalling formats:
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In general, voice-port commands define the characteristics associated with a particular voice-port signalling type. Under most circumstances, the default voice-port command values are adequate to configure FXO and FXS ports to transport voice data using the Cisco MC3810. Because of the inherent complexities involved with PBX networks, E&M ports might need specific voice-port values configured, depending on the specifications of the devices in your telephony network.
Voice Ports on the Cisco 2600 and 3600 Series Routers
The Cisco 2600 and 3600 series routers provide analog and digital voice ports. The type of signalling associated with these analog voice ports depends on the interface module installed into the device. The Cisco 3600 series router supports either a 2-port or 4-port voice network module (VNM); VNMs can hold either two or four voice interface cards (VICs).
Each VIC is specific to a particular signalling type; therefore, VICs determine the type of signalling for the voice ports on that particular VNM. Even though VNMs can hold multiple VICs, each VIC on a VNM must conform to the same signalling type. For more information about the physical characteristics of VNMs and VICs or how to install them, refer to the installation document, Voice Network Module and Voice Interface Card Configuration Note, that came with your VNM.
Voice Ports on the Cisco MC3810
The Cisco MC3810 hardware features different hardware configuration options for voice ports:
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Table 22 lists the valid slot and port numbers for the different voice interfaces.
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Configuring FXO or FXS Voice Ports
Under most circumstances the default voice-port values are adequate for both FXO and FXS voice ports. To configure FXO and FXS voice ports, use the following commands beginning in global configuration mode:
Step 1
For Cisco 2600 and 3600 series analog voice ports:
router(config)# voice-port slot/subunit/port
For Cisco 2600 and 3600 series digital voice ports:
router(config)# voice-port slot/port:ds0-group
For Cisco MC3810 series analog voice ports:
router(config)# voice-port slot/port
For Cisco MC3810 series digital voice ports:
router(config)# voice-port slot:ds0-group
Identifies the voice port you want to configure and enters voice-port configuration mode.
Step 2
Router(config-voiceport)# dial-type {pulse | dtmf}
(For FXO ports only) Selects the appropriate dial type for out-dialing. The default is dtmf.
Step 3
Router(config-voiceport)# signal {loop-start | ground-start}
Configures the signalling type for analog FXO and FXS voice ports. The default is loop-start.
Step 4
Router(config-voiceport)# compand-type {u-law | a-law}
(Digital voice ports only) Configures the companding standard used to convert between analog and digital signals in pulse code modulation (PCM) systems.
Step 5
Router(config-voiceport)# cptone country
Configures the appropriate call progress tone for the local region.
The default for this command is us. For a list of supported countries, refer to the Cisco IOS Multiservice Applications Command Reference publication.
Step 6
Router(config-voiceport)# ring number number
(For FXO ports only) Configures the number of rings detected before a connection is closed on the FXO port.
Step 7
Router(config-voiceport)# ring frequency number
(For FXS ports only) Specifies the local ring frequency for the FXS voice port.
Step 8
Router(config-voiceport)# music-threshold number
(Optional) Specifies the threshold (in decibels) for on-hold music. Valid entries are from -70 to -30.
Step 9
Router(config-voiceport)# connection {plar | tie-line | plar-opx} string
(Optional) Configures the voice-port connection mode type and the destination telephone number.
The plar value is used for private line auto ringdown (PLAR) connections. The tie-line value on the Cisco MC3810 is used for a tie-line connection to a PBX. The plar-opx value on the Cisco MC3810 is used for PLAR OPX, to allow the local voice port to provide a local response before the remote voice port receives an answer. If you select the plar-opx value, you must also configure the voice confirmation-tone command.
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Step 10
Router(config-voiceport)# description string
(Optional) Enters a string description for the voice port. The string describes the voice port in displays. You can use the description command to note the voice port's location or use.
Step 11
Router(config-voiceport)# codec {g729r8 | g729ar8 | g726r32 | g711alaw | g711ulaw}
(Optional for Cisco MC3810 only) Configures the voice-port compression mode. The g729ar8 value is the default and is recommended.
The g729ar8 compression mode can support a maximum of 24 simultaneously active on-net voice calls and the g729r8 value can only support a maximum of 12. The g729 compression modes have a nominal data rate of 8 kbps.
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Step 12
Router(config-voiceport)# vad
(Optional for Cisco MC3810 only) Enables voice activity detection (VAD).
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Step 13
Router(config-voiceport)# comfort-noise
(Optional) Specifies that background noise will be generated if you have VAD activated.
Step 14
Router(config-voiceport)# voice confirmation-tone
(Optional for Cisco MC3810 only) If the voice port is configured for connection plar-opx for OPX, disables the two-beep confirmation tone that a caller hears when picking up the handset.
Verifying the FXO or FXS Voice Port Configuration
You can check the validity of your voice-port configuration by performing the following tasks:
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Troubleshooting Tips
If you are having trouble connecting a call and you suspect the problem is associated with voice-port configuration, you can try to resolve the problem by performing the following tasks:
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Fine-Tuning FXO and FXS Voice Ports
Depending on your particular network, you may need to adjust voice parameters involving timing, input gain, and output attenuation for FXO or FXS voice ports. Collectively, these commands are referred to as voice-port tuning commands.
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To fine-tune FXO or FXS voice ports, use the following commands beginning in global configuration mode:
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Configuring E&M Voice Ports
Unlike FXO and FXS voice ports, the default E&M voice-port parameters most likely will not be sufficient to enable voice data transmission over your network.
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To configure E&M voice ports , use the following commands beginning in global configuration mode:
Step 1
For Cisco 2600 and 3600 series analog voice ports:
router(config)# voice-port slot/subunit/port
For Cisco 2600 and 3600 series digital voice ports:
router(config)# voice-port slot/port:ds0-group
For Cisco MC3810 series analog voice ports:
router(config)# voice-port slot/port
For Cisco MC3810 series digital voice ports:
router(config)# voice-port slot:ds0-group
Identifies the voice port you want to configure and enters voice-port configuration mode.
Step 2
Router(config-voiceport)# dial-type {dtmf | pulse}
Selects the appropriate dial type for out-dialing.
Step 3
Router(config-voiceport)# operation {2-wire | 4-wire}
Selects the appropriate cabling scheme for this voice port.
Step 4
Router(config-voiceport)# type {1 | 2 | 3 | 5}
Selects the appropriate E&M interface type.
Type 1 indicates the following lead configuration:
E—output, relay to ground
M—input, referenced to groundType 2 indicates the following lead configuration:
E—output, relay to SG
M—input, referenced to ground
SB—feed for M, connected to -48V
SG—return for E, galvanically isolated from groundType 3 indicates the following lead configuration:
E—output, relay to ground
M—input, referenced to ground
SB—connected to -48V
SG—connected to groundType 5 indicates the following lead configuration:
E—output, relay to ground
M—input, referenced to -48VStep 5
Router(config-voiceport)# signal {wink-start | immediate | delay-dial}
Configures the signalling type for E&M voice ports. The default is wink-start.
Step 6
Router(config-voiceport)# impedance {600c | 600r | 900c | complex1 | complex2}
Specifies a terminating impedance. This value must match the specifications from the telephony system to which this voice port is connected.
Step 7
Router(config-voiceport)# connection {plar | tie-line | plar-opx} string
(Optional) Configures the voice-port connection mode type and the destination telephone number.
The plar value is used for PLAR connections. The tie-line value on the Cisco MC3810 is used for a tie-line connection to a PBX. The plar-opx value on the Cisco MC3810 is used for PLAR OPX, to allow the local voice port to provide a local response before the remote voice port receives an answer. If you select the plar-opx value, you must also configure the voice confirmation-tone command.
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Step 8
Router(config-voiceport)# compand-type {u-law | a-law}
(Digital voice ports only) Configures the companding standard used to convert between analog and digital signals in PCM systems.
Step 9
Router(config-voiceport)# cptone locale
Configures the appropriate call progress tone for the local region.
The default for this command is the us keyword. For a list of supported countries, refer to the Cisco IOS Multiservice Applications Command Reference.
Step 10
Router(config-voiceport)# music-threshold number
(Optional) Specifies the threshold (in decibels) for on-hold music. Valid entries are from -70 to -30.
Step 11
Router(config-voiceport)# description string
(Optional) Attaches descriptive text about this voice port connection.
Step 12
Router(config-voiceport)# codec {g729r8 | g729ar8 | g726r32 | g711alaw | g711ulaw}
(Optional for Cisco MC3810 only) Configures the voice-port compression mode. The g729ar8 value is the default and is recommended.
The g729ar8 compression mode can support a maximum of 24 simultaneously active on-net voice calls and the g729r8 value can only support a maximum of 12. The g729 compression modes have a nominal data rate of 8 kbps.
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Step 13
Router(config-voiceport)# vad
(Optional for Cisco MC3810 only) Enables voice activity detection (VAD).
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Step 14
Router(config-voiceport)# voice confirmation-tone
(Optional for Cisco MC3810 only) If the voice port is configured for connection plar-opx for OPX, disables the two-beep confirmation tone that a caller hears when picking up the handset.
Verifying the E&M Voice Port Configuration
You can verify your voice-port configuration by performing the following tasks:
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Troubleshooting Tips
If you are having trouble connecting a call and you suspect the problem is associated with voice-port configuration, you can try to resolve the problem by performing the following tasks:
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Fine-Tuning E&M Voice Ports
Depending on your particular network, you may need to adjust voice parameters involving timing, input gain, and output attenuation for E&M voice ports. Collectively, these commands are referred to as voice-port tuning commands.
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To fine-tune E&M voice ports , use the following commands beginning in global configuration mode:
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Activating the Voice Port
After you have configured the voice port, you need to activate the voice port to bring it online. In fact it is a good idea to cycle the port—meaning to shut the port down and then bring it online again.
To activate a voice port, use the following command in voice-port configuration mode:
To cycle a voice port, use the following commands in voice-port configuration mode:
Step 1
Router(config-voiceport)# shutdown
Deactivate the voice port.
Step 2
Router(config-voiceport)# no shutdown
Activate the voice port.
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Configuring Frame Relay to Support VoFR
This section describes preliminary Frame Relay configuration tasks that are necessary to support VoFR:
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Configuring a Map Class to Support VoFR
Before configuring a Frame Relay DLCI for voice traffic, you must create a Frame Relay map class and configure it to support voice traffic. Configuring a Frame Relay map class is required because the voice bandwidth, fragmentation size, and traffic shaping attributes are configured on the map class. These attributes are required for sending voice traffic on the PVC.
This section is divided into the procedures described in the following sections:
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A map class applies to a single DLCI or to a group of DLCIs, depending on how the class has been applied to the virtual circuit. If you have a large number of PVCs to configure, you can assign the PVCs the same traffic shaping properties without statically defining the values for each PVC. You can create multiple map classes with different variables for each map class.
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Configuring a Frame Relay Map Class to Support Voice Traffic
To configure a Frame Relay map class to support voice traffic on a single DLCI or a group of DLCIs, use the following commands beginning in global configuration mode:
To configure the map class to support FRF.12 fragmentation, see the "Configuring a Frame Relay Map Class to Support FRF.12 Fragmentation" section later in this chapter. To configure the map class to support traffic shaping if you want to send both voice traffic and data traffic on the same PVC, see the "Configuring a Frame Relay Map Class for Traffic Shaping Parameters" section later in this chapter.
Calculating Voice Bandwidth
The frame-relay voice-bandwidth map-class command is used to configure how much bandwidth is reserved for voice traffic. If not enough reserved voice bandwidth remains on the PVC, then any new call attempted will be rejected.
When you calculate the amount of voice bandwidth to allocate to voice, the overall bandwidth calculation must include the voice packetization overhead and not just the raw compressed speech codec bandwidth. For VoFR voice packets, there are a total of 6 or 7 bytes total overhead per packet (including standard Frame Relay headers and flags). For subchannels (CIDs) less than number 64, the overhead is 6 bytes. For subchannels greater than or equal to number 64, the overhead is 7 bytes. Add one byte if voice sequence numbers are enabled in the voice packets.
To determine the required voice bandwidth, use the following calculation:
required_bandwidth = codec_bandwidth * (1 + overhead/payload_size)
This calculation addresses the amount of bandwidth consumed on the physical network interface. This figure does not necessarily represent the amount of connection bandwidth used within the Frame Relay network itself, which may be higher due to the overhead of switching small packets.
When using 30-millisecond (ms) duration voice packets, an approximate general rule is to add 2000 bps overhead to the raw voice compressed speech codec rate. With the 32 kbps G.726 ADPCM speech coder, a 30-ms speech frame uses 120 bytes voice payload plus 6 to 7 bytes overhead, and the overall bandwidth requirement is around 34 kbps for each call.
The codec command is configured as part of the dial peer configuration procedures in the "Configuring Dial Peers" section later in this chapter.
Configuring a Frame Relay Map Class to Support FRF.12 Fragmentation
To configure the map class to support FRF.12 fragmentation, use the following commands in map-class configuration mode:
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To configure the map class to support traffic shaping if you want to send both voice traffic and data traffic on the same PVC, see the next section, "Configuring a Frame Relay Map Class for Traffic Shaping Parameters."
Configuring a Frame Relay Map Class for Traffic Shaping Parameters
When you configure a Frame Relay PVC to support voice traffic, you must ensure that the carrier can accommodate the traffic rate or profile sent on the PVC. If too much traffic is sent at once, the carrier might discard frames, which causes disruptions to real-time voice traffic. The carrier might also deal with traffic bursts by queueing up the bursts and delivering them at a metered rate. Excessive queueing also causes disruption to real-time voice traffic.
To compensate for this condition, traffic shaping is required if both voice traffic and data traffic are sent over the same PVC.
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To configure a Frame Relay map class to determine the traffic shaping characteristics for one or more DLCIs, use the following commands in map-class configuration mode:
If your Frame Relay map class configuration is complete, see the "Verifying Your Frame Relay Configuration" section later in this chapter. To begin your dial peer configuration, see either the "Preparing to Configure Voice Dial Peers" later in this chapter, or the "Configuring Dial Peers" section later in this chapter.
Verifying Your Frame Relay Configuration
You can check the validity of your Frame Relay configuration by performing the following tasks:
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Preparing to Configure Voice Dial Peers
After you have analyzed your dial plan and decided how to integrate it into your existing network, you are ready to configure your network devices to support VoFR. The actual configuration procedure depends entirely upon the topology of your voice network, but, in general, you need to perform the tasks described in the following sections:
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Organizing Voice Network Information
After you have configured your Frame Relay network, you should collect all of the data directly related to each dial peer by creating a peer configuration table to prepare for configuring VoFR.
Creating a Peer Configuration Table
Specific information relative to each dial peer needs to be identified before you can configure VoFR. One way to identify this information is to create a peer configuration table.
Figure 65 shows a diagram of a small voice network in which Router No. 1 connects a small sales branch office to the main office through Router No. 2. Only two devices in the sales branch office that need to be established as dial peers: a telephone and a fax machine. Router No. 2 is the primary gateway to the main office; as such, it needs to be connected to the company's PBX. Two telephones and one fax machine connected to the PBX need to be established as dial peers in the main office.
Table 23 shows the peer configuration table for the example illustrated in Figure 65.
Figure 65 Sample VoFR Network
The dial plan shown in Table 23 lists a simple dial-peer configuration table. No configuration for forwarding digits to a PBX is shown. Additional configuration for forwarding digits is required.
Configuring Dial Peers
Dial peers describe the entities to and from which a call is established. Dial-peer configuration tasks define the address or set of addresses serviced by that dial peer and the call parameters required to establish a call to and from that dial peer.
Two different kinds of dial peers pertain to VoFR configurations:
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POTS peers associate a telephone number with a particular voice port so that incoming calls for that port can be received. VoFR peers point to specific voice-network devices (by associating destination telephone numbers with a specific Frame Relay DLCI) so that outgoing calls can be placed. Both POTS and VoFR dial peers are needed to establish VoFR connections if you want both to send and receive calls.
For tandem voice nodes, POTS dial peers are not configured.
Establishing two-way communication using VoFR requires establishing a specific voice connection between two defined endpoints. As shown in Figure 66, for outgoing calls (from the perspective of the voice-telephony dial peer 1), the voice-telephony dial peer establishes the source (the originating telephone number and voice port) of the call. The voice-network dial peer establishes the destination by associating the destination telephone number with a specific Frame Relay DLCI.
Figure 66 Calls from the Perspective of Router No.1
In the example, the destination pattern 14085554000 string maps to a U.S. telephone number 555-4000, with the digit 1 plus the area code (408) preceding the number. When you configure the destination pattern, set the dial string to match the local dial conventions.
To complete the two-way communications loop, you need to configure VoFR dial peer 4 as shown in Figure 67.
Figure 67 Calls from the Perspective of Router No. 2
The only exception to this example is when both POTS dial peers are connected to the same router, as shown in Figure 68. In this circumstance, you would not need to configure a VoFR dial peer. Figure 68 shows an example for switched calls only.
Figure 68 Communication Between Dial Peers Sharing the Same Router
When you configure dial peers, ensure that you understand the relationship between the destination pattern and the session target. The destination pattern is the telephone number of the voice device attached to the voice port. The session target represents the route to a serial port on the peer router at the other end of the Frame Relay connection. Figure 69 and Figure 70 show the relationship between the destination pattern and the session target, as seen from the perspective of both routers in a VoFR configuration. These examples show a topology for switched calls only.
Figure 69 Relationship Between Destination Pattern and Session Target from the Perspective of Router No. 1
Figure 70 Relationship Between Destination Pattern and Session Target from the Perspective of Router No. 2
The following sections describe how to configure POTS peers and VoFR peers.
Configuring POTS Dial Peers
To configure a POTS dial peer, you need to uniquely identify the peer (by assigning it a unique tag number), define its telephone number, and associate it with a voice port through which calls will be established. Under most circumstances, the default values for the remaining dial-peer configuration commands are sufficient to establish connections.
Depending on your dial plan configuration, you might need to consider how to configure voice networks with variable-length dial plans, number expansion, excess digit playout, forward digits, and default voice routes, or use hunt groups with dial-peer preferences.
Note
To configure POTS dial peers, use the following commands beginning in global configuration mode:
Configuring Dial Plan Options for the POTS Dial Peer
When you configure the dial plan, you have different options for how the dial plan is designed.
To configure dial plan options, use the following commands in dial-peer configuration mode:
To configure the next POTS dial peer, exit dial-peer configuration mode by entering exit, and repeat the previous steps. To configure VoFR dial peers, see the next section, "Configuring VoFR Dial Peers."
Configuring VoFR Dial Peers
To configure a VoFR dial peer, you need to uniquely identify the peer (by assigning it a unique tag number) and define the outgoing serial port number and the virtual circuit number.
Depending on your dial plan configuration, you might need to consider how to configure voice networks with variable-length dial plans, number expansion, excess digit playout, forward digits, and default voice routes, or use hunt groups with dial peer preferences.
VoFR dial peer configuration procedures are described in the following sections:
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Figure 71 shows an example of a Frame Relay network with switched calls. In the example, there are two routers (Routers No. 1 and No. 2) with telephone devices at the endpoints. In between the two endpoint routers are tandem routers (Routers A, B, and C) with switched connections in between. Standard VoFR dial peers are configured on the intermediate nodes in the Frame Relay network. Support for switched calls are configured on the dial peers on both Router No. 1 and Router No 2.
Figure 71 Endpoint Nodes and Tandem Nodes in VoFR Network for Switched Calls
In this example, different types of VoFR dial peers must be configured on the different routers.
VoFR dial peers for switched connections must be configured between Router No. 1 and Router No. 2 (the endpoints for the voice connection).
On the tandem routers (Routers A, B, and C) VoFR dial peers for switched calls must be configured for the following:
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Configuring VoFR Dial Peers for Switched Calls
If you will be sending switched calls over the Frame Relay network, you must configure the VoFR dial peers to specifically support switched calls.
To configure a VoFR dial peer to support switched calls, use the following commands beginning in global configuration mode:
To configure another VoFR dial peer, exit dial-peer configuration mode and repeat Steps 1 through 10.
To configure VoFR dial peers for tandem nodes, see the "Configuring VoFR Dial Peers for Tandem Nodes" later in this chapter.
For procedures on how to configure switched calls, see the "Configuring Switched Calls (User Dialed or Auto-Ringdown)" section later in this chapter. For information on configuring all call types, see the "Overview of VoFR Connection Types" later in this chapter.
Configuring VoFR Dial Peers for Cisco-Trunk (Private Line) Calls
If you will be sending Cisco-trunk (private line) calls over the Frame Relay network, you must configure the VoFR dial peers to specifically support Cisco-trunk (private line) calls. Cisco-trunk (private line) calls are permanent calls.
One key task when you configure Cisco-trunk (private line) connections is to configure the signal type for the dial peer. Depending on the router you are using, you might have several options. The signal-type dial-peer command supports the following options:
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The signal type normally must be configured such that the signal type selected in the dial peers on the routers at both ends of the permanent voice call are the same. When you configure a permanent connection between a T1/E1 Cisco MC3810 and an analog voice port on a Cisco 2600 or Cisco 3600, the signal type normally should be set to cas, which is the default.
Cisco 2600 and Cisco 3600 analog voice ports do not support the cept or transparent signal types. The T1/E1 Cisco MC3810 can also be set to transparent, which simply passes the signalling through from the Cisco 2600/3600 without interpretation. However, when transparent is used, the Cisco MC3810 makes no assumptions regarding the on-hook and off-hook state of the call. By default, when configured using transparent, the Cisco MC3810 operates the voice path in the permanently open state so that voice packets are sent (and network bandwidth consumed) regardless of the state of the call. For more information on controlling the flow of voice packets with transparent signalling, see the "Verifying Your Frame Relay Configuration" section earlier in this chapter.
To configure a VoFR dial peer to support Cisco-trunk permanent (private line) calls, use the following commands beginning in global configuration mode:
To configure another VoFR dial peer, exit dial-peer configuration mode and repeat steps 1 through 11.
To configure VoFR dial peers for tandem nodes, see the "Configuring VoFR Dial Peers for Tandem Nodes" section later in this chapter.
For procedures on how to configure Cisco-trunk permanent (private line) calls, see the "Configuring Cisco-Trunk (Private Line) Permanent Calls" section later in this chapter. For information on configuring all call types, see the "Overview of VoFR Connection Types" section later in this chapter.
Configuring VoFR Dial Peers for FRF.11 Trunk Calls
If you will be sending FRF.11 trunk calls over the Frame Relay network, you must configure the VoFR dial peers to specifically support FRF.11 trunk calls. For FRF.11 trunk calls, you must statically configure the VoFR dial peers on both sides of the FRF.11 trunk.
Note
To configure a VoFR dial peer to support FRF.11 trunk calls, use the following commands beginning in global configuration mode:
To configure another VoFR dial peer, exit dial-peer configuration mode and repeat Steps 1 through 12.
Repeat this procedure on the destination router on the other side of the FRF.11 trunk.
To configure VoFR dial peers for tandem nodes, see the "Configuring VoFR Dial Peers for Tandem Nodes" section later in this chapter.
For procedures on how to configure FRF.11 trunk calls, see the "Configuring FRF.11 Trunk (Private Line) Calls" section later in this chapter. For information on configuring all call types, see the 'Overview of VoFR Connection Types" section later in this chapter.
Configuring VoFR Dial Peers for Tandem Nodes
You configure standard VoFR dial peers for switched calls on the tandem routers in the network topology. Tandeming is not allowed when the call type is an FRF.11 trunk call.
You can configure VoFR dial peers for tandem routers on the Cisco MC3810 and on the Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series routers.
Note
To configure VoFR dial peers on tandem routers, use the following commands beginning in global configuration mode:
To configure the next VoFR dial peer, exit dial-peer configuration mode by entering exit, and repeat the previous steps. On tandem nodes, at least two VoFR dial peers are required, one for each call leg in the router.
Disabling Dial-Peer Hunting on a Dial Peer (Cisco MC3810 Only)
To disable dial-peer hunting on a dial peer, enter the following commands beginning in global configuration mode:
To reenable dial-peer hunting on a dial peer, enter the following commands beginning in global configuration mode:
Configuring VoFR Connections
After you have configured the Frame Relay DLCI settings and you have configured your dial plan, you are ready to configure specific VoFR connections.
There are many different scenarios for VoFR connections. For information on the different connection types, see the next section, "Overview of VoFR Connection Types."
Procedures on how to configure the different connection types are described in the following sections:
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In addition, special consideration is required for configuring calls for tandem nodes. For more information, see the "Configuring Connections for Tandem Nodes" section later in this chapter.
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Overview of VoFR Connection Types
When you configure VoFR connections, many different connection types are possible depending on the hardware platform, whether the call is to be a regular switched (user dialed or auto-ringdown) call, or whether the call is a permanent call (Cisco-trunk or FRF.11-trunk). You configure these specific connection types using combinations of several commands.
Table 24 lists the different connection types for VoFR connections supported on the Cisco 2600 and 3600 series, and the combinations of commands to enter for each call type.
Table 25 lists the different connection types for VoFR connections supported on the Cisco MC3810 series, and the combinations of commands to enter for each call type.
Table 24 VoFR Connection Types Supported on the Cisco 2600/3600 Series and 7200 Series
Switched call
(user dialed or auto-ringdown) to other Cisco 2600/3600 or 7200 routersvofr [data cid]
[call-control [cid]]1FRF.11 Annex C
session protocol cisco-switched2
For user dialed calls: none
For auto-ringdown calls:
connection plar destination-stringSwitched call
(user dialed or auto-ringdown)
to a Cisco MC3810vofr cisco3
Cisco proprietary4
session protocol cisco-switched
For user dialed calls: none
For auto-ringdown calls:
connection plar destination-stringCisco-trunk
permanent call (private-line) to other Cisco 2600/3600 or 7200 routersvofr data cid
call-control cidFRF.11 Annex C
session protocol cisco-switched
connection trunk destination-string
[answer mode]Cisco-trunk
permanent call
(private-line)
to a Cisco MC3810vofr cisco
Cisco proprietary
session protocol cisco-switched
connection trunk destination-string
[answer mode]FRF.11 trunk call (private-line)
vofr [data cid] [call-control cid]5
FRF.11 Annex C
session protocol frf11-trunk
connection trunk destination-string
[answer mode]
1 The recommended use of this command is vofr data 4 call-control 5.
2 The session protocol cisco-switched option is the default setting. If you do not enter this command, the setting will still apply.
3 This command consumes data CID 4 and call-control CID 5.
4 Cisco proprietary fragmentation is based on an early draft of FRF.12, and is compatible with Cisco MC3810 concentrators running software versions prior to Cisco IOS Release 12.0(3)XG or 12.0(4)T.
5 For FRF.111 trunk calls, the call-control option is not required. It is only required if you mix FRF.11 trunk calls with other types of voice calls on the same PVC.
Table 25 VoFR Call Types Supported on the Cisco MC3810
Switched call
(user dialed or auto-ringdown)vofr cisco1
or
frame-relay interface-dlci dlci voice-encap size [voice-cir cir]2
Cisco proprietary3
session protocol cisco-switched
For user dialed calls: none
For auto-ringdown calls:
connection plar destination-stringFor tie-line connections:
connection tie-line
destination-stringCisco-trunk
permanent call (private line)vofr cisco
or
frame-relay interface-dlci dlci voice-encap size [voice-cir cir]
Cisco proprietary
session protocol cisco-switched
connection trunk destination-string
[answer mode]FRF.11 trunk call (private-line)
vofr [data cid]4
FRF.11 Annex C
session protocol frf11-trunk
connection trunk destination-string
[answer mode]
1 This command consumes data CID 4 and call-control CID 5.
2 The voice-encap value is required, but the voice-cir value is optional.
3 Cisco proprietary fragmentation is based on an early draft of FRF.12, and is compatible with Cisco MC3810 concentrators running software versions prior to Cisco IOS Release 12.0(3)XG or 12.0(4)T.
4 If the vofr command is entered without the cisco option, only FRF.11 trunks are supported. You cannot mix FRF.11 trunk calls with other call types on the same PVC on the Cisco MC3810.
Configuring Switched Calls (User Dialed or Auto-Ringdown)
This section describes how to configure switched calls (user dialed or auto-ringdown) on the different router platforms. This section is divided into the procedures described in the following sections:
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Configuring Switched Calls on Cisco 2600, 3600, and 7200 Series Routers
You can configure switched calls on Cisco 2600, 3600, and 7200 series routers. To configure switched calls on these routers, use the following commands beginning in interface configuration mode:
This configuration uses standard FRF.11 Annex C fragmentation.
Configuring Switched Calls on Cisco 2600, 3600, or 7200 Series Routers to a Cisco MC3810
On the Cisco 2600, 3600, and 7200 series routers, you can configure switched calls to a Cisco MC3810. However, the configuration is different from standard switched calls because the earlier Cisco MC3810 versions used the Cisco proprietary version of FRF.12.
Note
To configure switched calls to a Cisco MC3810, use the following commands beginning in interface configuration mode:
This configuration uses Cisco proprietary data fragmentation.
Configuring Switched Calls on a Cisco MC3810
On the Cisco MC3810 you can configure switched calls to another Cisco MC3810 or to Cisco 2600, Cisco 3600, and Cisco 7200 series routers. However, the configuration is different from switched calls on other routers because the earlier Cisco MC3810 versions used the Cisco proprietary version of FRF.12.
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When configuring switched calls on a Cisco MC3810, you must enter one of the following two commands:
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This command was added in Cisco IOS Release 12.0(4)T, and uses WFQ at the PVC level. Using the vofr cisco command, you cannot disable support for WFQ.
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The voice-encap option for the frame-relay interface-dlci command was introduced in Cisco IOS Release 11.3(1)MA and does not support WFQ. However, you also have the option to configure fancy queueing with the voice-encap option by entering the no frag-pre-queuing command and configuring the appropriate fancy queueing commands.
The procedures for using each command differ. See the appropriate procedure for the command you plan to use. These configurations both use Cisco proprietary data fragmentation.
Note
To configure switched calls on a Cisco MC3810 using the vofr cisco command, use the following commands beginning in interface configuration mode:
To configure switched calls on a Cisco MC3810 using the frame-relay interface-dlci voice-encap command, use the following commands beginning in interface configuration mode:
Step 1
Router(config-if)# frame-relay interface-dlci dlci voice-encap size [voice-cir cir]
Configures the Frame Relay DLCI, enters DLCI configuration mode, and specifies that the DLCI will support voice traffic. For recommended fragmentation sizes to use with the voice-encap option, see Table 26.
In this configuration, the voice-encap value is required, but the voice-cir value is optional. The default voice-cir value is the CIR configured for the Frame Relay map class.
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Step 2
Router(config-if)# no frag pre-queuing
(Optional) Configures the interface to disable first-come, first-served (FCFS) queueing.
By default, the frag pre-queuing command is enabled. This command supports only FCFS queueing, and performs the queueing before the data fragmentation takes place.
If you want to configure fancy queueing (WFQ, priority queueing, or custom queueing) on the interface, enter the no frag pre-queuing command. Note that after you enter the no frag pre-queuing command, you still must configure the commands to support the fancy queueing method desired.
Note
Step 3
For Cisco MC3810 series analog voice ports:
router(config)# voice-port slot/port
For Cisco MC3810 series digital voice ports:
router(config)# voice-port slot:ds0-group
Enters voice-port configuration mode.
Step 4
Router(config-voiceport)# connection plar destination-string
or
For auto-ringdown calls, configures the PLAR connection, specifying the telephone number in the destination-string argument.
Router(config-voiceport)# connection tie-line destination-string
For tie-line calls, configures the tie-line connection, specifying the telephone number in the destination-string argument.
Table 26 lists recommended data fragmentation sizes to use when configuring the voice-encap option for the frame-relay interface-dlci command.
Table 26 Recommended Data Fragmentation Sizes
64 kbps
80 bytes
128 kbps
160 bytes
256 kbps
320 bytes
512 kbps
640 bytes
1536 kbps (full T1)
1600 bytes
2048 kbps (full E1)
1600 bytes
1 The data fragmentation size is based on back-to-back Frame Relay. If you are sending traffic through an IGX node with standard Frame Relay, subtract 6 bytes from the recommended data fragmentation size.
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Configuring Cisco-Trunk (Private Line) Permanent Calls
This section describes how to configure Cisco-trunk (private line) calls on the different router platforms. This section is divided into the procedures described in the following sections:
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Configuring Cisco Trunk Permanent Calls on Cisco 2600, 3600, and 7200 Series Routers
You can configure Cisco trunk permanent calls on Cisco 2600, 3600, and 7200 series routers. To configure Cisco trunk permanent calls on these routers, use the following commands beginning in interface configuration mode:
This configuration uses standard FRF.11 Annex C fragmentation.
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Configuring Cisco Trunk Permanent Calls on Cisco 2600 and 3600 Routers to a Cisco MC3810
To configure Cisco trunk permanent calls to a Cisco MC3810 from a 2600 series or 3600 series router, use the following commands beginning in interface configuration mode:
This configuration uses Cisco proprietary data fragmentation.
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Configuring Cisco Trunk Permanent Calls on a Cisco MC3810
When configuring Cisco trunk permanent calls on a Cisco MC3810 interface, you must enter one of the following two commands:
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This command was added in Cisco IOS Release 12.0(3)XG, and uses WFQ at the PVC level. Using the vofr cisco command, you cannot disable support for WFQ.
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The voice-encap option for the frame-relay interface-dlci command was introduced in Cisco IOS Release 11.3(1)MA and does not support WFQ. However, you also have the option to configure fancy queueing with the voice-encap option by entering the no frag-pre-queuing command and configuring the appropriate fancy queueing commands.
The procedures for using each command differ. See the appropriate procedure for the command you plan to use. These configurations both use Cisco proprietary data fragmentation.
Note
To configure Cisco trunk permanent calls on a Cisco MC3810 using the vofr cisco command, use the following commands from interface configuration mode:
Note
To configure Cisco trunk permanent calls on a Cisco MC3810 using the frame-relay interface-dlci voice-encap command, use the following commands beginning in interface configuration mode:
Step 1
Router(config-if)# frame-relay interface-dlci dlci voice-encap size [voice-cir cir]
Configures the Frame Relay DLCI, enters DLCI configuration mode, and specify that the DLCI will support voice traffic.
For recommended fragmentation sizes to use with the voice-encap option, see Table 26.
In this configuration, the voice-encap value is required, but the voice-cir value is optional. The default voice-cir value is the CIR configured for the Frame Relay map class
Note
Step 2
Router(config-if)# no frag pre-queuing
(Optional) Enters this command if you want to disable FCFS queueing on the interface.
By default, the frag pre-queuing command is enabled. This command supports only FCFS queueing, and performs the queueing before the data fragmentation takes place.
If you want to configure fancy queueing (WFQ, priority queueing, or custom queueing) on the interface, enter the no frag pre-queuing command. Note that after you enter the no frag pre-queuing command, you still must configure the commands to support the fancy queueing method desired.
Note
Step 3
For Cisco MC3810 series analog voice ports:
router(config)# voice-port slot/port
For Cisco MC3810 series digital voice ports:
router(config)# voice-port slot:ds0-group
Enters voice-port configuration mode.
Step 4
Router(config-voiceport)# connection trunk destination-string [answer-mode]
For private line calls, configures the trunk connection, specifying the telephone number in the destination-string argument.
When you configure Cisco trunk permanent calls, one side must be the call initiator (master) and the other side is normally the call answerer (slave). By default, the voice port is the master. Enter the answer-mode keyword to specify that the voice port will be the slave.
Step 5
Router(config-voiceport)# shutdown
Shuts down the voice port.
Step 6
Router(config-voiceport)# no shutdown
Reactivates the voice port to enable the trunk connection to take effect.
Note
Configuring FRF.11 Trunk (Private Line) Calls
On the Cisco MC3810 and on Cisco 2600 and 3600 series routers, you can configure FRF.11 trunk calls to a second router.
You cannot configure FRF.11 trunk calls for tandem VoFR configurations.
Note
To configure FRF.11 trunk (private line) calls, use the following commands beginning in interface configuration mode:
This configuration uses FRF.11 Annex C data fragmentation.
Note
Configuring Connections for Tandem Nodes
Tandeming is the switching of an incoming VoFR call on a Frame Relay DLCI to an outgoing VoFR enabled DLCI. Tandeming works for switched calls and Cisco-trunk permanent calls only. You cannot tandem FRF.11 trunk calls over a multihop network.
Tandeming is supported on the Cisco MC3810 and on Cisco 2600, 3600, and 7200 series routers. The Cisco 7200 series can only act as a tandem router in a VoFR network.
Note
Depending on which router is used as the end node and which router is used as the tandem node, you must use the correct Frame Relay PVC type when configuring your connections. Table 27 shows the different combinations of routers that can serve as end nodes and tandem nodes, and the Frame Relay PVC type required.
When you configure a tandem node, you must configure two VoFR dial peers, one for each tandem connection.
Verifying Your Voice Connections
Verify that the voice connection for switched calls is working by performing the following tasks:
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Verify that the voice connection for FXO-FXS trunk calls from a telephone to a remote PBX is working by performing the following tasks:
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Verify your dial-peer and voice-port configuration by performing the following tasks:
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Verify your VoFR configuration on the DLCI by performing the following tasks:
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Troubleshooting Tips
If you are having trouble connecting a call, you can try to resolve the problem by performing the following tasks:
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Monitoring and Maintaining Your VoFR Configuration
To monitor and maintain your VoFR configuration, use the following commands in EXEC mode:
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Dial Planning and Dial-Peer Digit Manipulation Options
Depending on the configuration, you may need to consider different strategies for how you configure the dial plan and dial-peer digit manipulation, and how you configure the dial peers. Dial peer digit-manipulation tasks that you may need to consider are described in the following sections:
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Excess Digits Playout
Excess digits are defined as received digits that are beyond the length of the destination pattern on a terminating router. A terminating router will forward excess digits to the telephony interface. For example, if the digits "123456789" are matched on a terminating router with a destination pattern of "1.....," the "6789" are excess digits and will be forwarded.
A router that is originating a call will only collect digits up to the length of a defined destination pattern. When a number is dialed that is longer than the destination pattern, after the last digit in the destination pattern is dialed, the call is immediately placed, and the additional digits are not collected by the router.
For example, if the digits "123456789" are dialed on an originating router with a destination pattern of 1...., then the digits "6789" are not collected. The call is placed immediately after the digit "5" is dialed. The additional digits "6789" are not collected, but are passed through the audio path.
In configurations where a phone on an FXS port on the router is connected to the PSTN through an FXO port on the same router, additional steps are necessary to enable a call to go through. Because the uncollected excess digits are passed through the audio path, the excess digits might not go through if the audio path is not ready when the excess digit is dialed. When dialing a destination pattern in this situation, pause and wait for a second dial tone before dialing the excess digits. This will give the audio path time to accept the excess digits.
Forward Digits and Voice Default Routes
Voice default routing for fixed-length dial plans can be used with larger PBX configurations to forward digits to router telephony interfaces.
Figure 72 shows an example of routing voice calls through a PBX using forward digits. In the configuration, the Cisco MC3810-t and Cisco MC3810-T are tandem nodes, and are required to support forwarding digits so that calls from Cisco MC3810s 0, 5, or 9 can make a call to extension "8208."
Figure 72 Routing Voice Calls Among Cisco MC3810s Through a PBX
On the two tandem nodes, the forward-digits command is required. This command specifies to forward all digits matched with the destination "8..." to the appropriate port. For a call from Cisco MC3810-0 to reach extension 8208, the call must first terminate at Cisco MC3810-T, which plays out the digits "8208" to the voice port connected to the PBX. The PBX then routes the voice call to Cisco MC3810-t. On the tandem nodes, although the forward-digits all command is used, the forward-digits 4 command can also be used in this example.
The following are the dial-peer configurations on each Cisco MC3810 required for this configuration:
Cisco MC3810-t Configuration
dial-peer voice 8 pots destination-pattern 8208 session-target s0 1dial-peer voice 1000 pots destination-pattern 8... forward-digits all port 1/1dial-peer voice 9999 pots destination-pattern .... forward-digits all port 1/1Cisco MC3810-T Configuration
dial-peer voice 1 vofr destination-pattern 8200 session-target s0 1dial-peer voice 6 vofr destination-pattern 8209 session-target s0 6dial-peer voice 10 vofr destination-pattern 8209 session-target s0 10dial-peer voice 1 pots destination-pattern 8... forward-digits all port 1/1Cisco MC3810-0 Configuration
dial-peer voice 1 pots destination-pattern 8200 port 1/1dial-peer voice 1000 vofr destination-pattern8... session-target s0 1For Cisco MC3810-5:dial-peer voice 5 pots destination-pattern 8205 port 1/1dial-peer voice 1000 vofr destination-pattern 8... session-target s0-1Cisco MC3810-9 Configuration
dial-peer voice 9 pots destination-pattern 8209 port 1/1dial-peer voice 1000 vofr destination-pattern 8... session-target s0 1The concept of voice default routes is also shown in Figure 72 and in the configuration. In the example, the configurations for the destination dial plans "8..." in both the tandem nodes are voice default routes because all voice calls dialed that start with 8 followed by three digits will either match on 8208 or end up with 8...., which is the last resort voice route used by Cisco MC3810-t if no other route is discovered.
Hunt Groups and Preference Configuration
When configuring the destination pattern for voice-telephony dial peers (such as to a PBX), if you configure each dial peer with a different destination pattern, then you may have limitations on the availability of calls getting through. Because you configure each dial peer with a different destination pattern, the destination is a single DS0 time slot on the voice connection to the PBX. If that time slot gets busy, then a call attempt to the dial peer mapped to that specific DS0 time slot will fail.
For example, in the following dial-peer configuration, four POTS dial peers are configured, all with a different destination pattern on the same PBX:
dial-peer voice 1 pots destination-pattern 3001 port 1/1dial-peer voice 2 pots destination-pattern 3002 port 1/2dial-peer voice 3 pots destination-pattern 3003 port 1/3dial-peer voice 4 pots destination-pattern 3004 port 1/4Because each dial peer has a different destination pattern configured, no backup is available if the DS0 time slot mapped to the dial peer is busy with another call.
However, the router supports the concept of hunt groups, in which you configure a group of dial peers on the same PBX with the same destination pattern. With a hunt group, if a call attempt is made to a dial peer on a specific DS0 time slot, if that time slot is busy, the router hunts for another time slot on that channel until an available time slot is found. In this case, each dial peer is configured using the same destination pattern of 3000, forming a dial pool to that destination pattern.
To provide specific dial peers in the pool with a preference over other dial peers, you can configure the preference order for each dial peer using the preference command. The following is an example of the dial-peer configuration with all dial peers having the same destination pattern, but different preference orders:
dial-peer voice 1 pots destination pattern 3000 port 1/1 preference 0dial-peer voice 2 pots destination pattern 3000 port 1/2 preference 1dial-peer voice 3 pots destination pattern 3000 port 1/3 preference 2dial-peer voice 4 pots destination pattern 3000 port 1/4 preference 3Note that when the preference order is set, the lower the preference number, the higher the priority. The highest priority is given to the dial peer with preference order 0.
You can also set the preference order on the network side, for voice network dial peers. However, you cannot mix the preference orders for POTS dial peers (local telephone devices) and voice-network peers (devices across the WAN backbone). You can set a separate preference order for each dial-peer type, but the preference order will not work on both at the same time. For example, you can configure preference order 0, 1, and 2 for voice telephony dial peers, and you can configure a separate preference order 0, 1, and 2 for the voice network dial peers, but the two preference orders are separate. The system only resolves the preference among dial peers of the same type; it does not resolve preferences between the two separate preference order lists.
The hunt group feature hunts for dial peers in the following order:
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Variable-Length Dial Plans
In most voice configurations, fixed-length dialing plans, in which all the dial-peer destination patterns have the same length, are sufficient because the telephone number strings are all the same length. However, in some voice network configuration, variable-length dial plans are required, especially if the network crosses national boundaries and the telephone number strings are of different lengths.
If you enter the "T" timer character in the destination pattern, the router can be configured to accept a fixed-length dial string, and then wait for additional dialed digits. For example, the following dial-peer configuration shows how the T character can be set to allow variable-length dial strings.
dial peer voice 1 potsdestination-pattern 2222Tport 1/1In this example, the router accepts the digits 2222, and then waits for an unspecified number of dialed digits. If digits continue to be entered before the interdigit timeout expires, then the router will gather up to 31 additional digits. Once the interdigit timeout expires, the router places the call. The interdigit timeout is set by the timeouts inter-digit voice-port command.
The interdigit timeout timer can be immediately terminated when you enter the "#" character. If the # character is entered while the router is waiting for the additional digits, the # character is treated as an end-dial accelerator. The # character is not treated as an actual digit in the destination pattern and is not sent as part of the dialed string across the network.
However, if the # character is entered before the router is waiting for the additional digits (meaning before the T character is entered in the destination pattern), then the # character is treated as a dialed digit. For example, if a destination pattern is configured with the string 222...T, then the digits 2222####1234567 can be gathered, but the digits 2222###1234#67 cannot be gathered because the final # character is treated as a terminator.
The default value for the interdigit timeout is 1 second. If the duration is not changed, using the T timer adds 10 seconds to each call setup time because the call is not attempted until the timer expires (unless the # character is used as a terminator). Because of this dependency, if you use variable-length dial plans, the voice-port interdigit timeout should be reduced to reduce the call setup time.
VoFR Configuration Examples
This section provides specific configuration examples for different VoFR connections and call type scenarios. This section includes the following examples:
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The examples do not provide complete configurations, but show the required commands to configure VoFR.
Two Routers Using Frame Relay Fragmentation Example
Figure 73 shows an example of Frame Relay fragmentation between two routers.
Figure 73 Two Routers Using Frame Relay Fragmentation
This configuration uses FRF.12 fragmentation.
Two Routers Using a VoFR PVC Example
Figure 74 shows an example of two routers with connections using a VoFR PVC.
Figure 74 Two Routers Using a VoFR PVC
This configuration uses FRF.11 Annex C fragmentation.
Router Using a VoFR PVC to a Cisco MC3810 Example
Figure 75 shows an example of a Cisco 3600 series router with connections to a Cisco MC3810 using a VoFR PVC. In this example, the VoFR interface on the Cisco MC3810 is configured using the vofr cisco command.
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Figure 75 Router Using a VoFR PVC to a Cisco MC3810
This configuration uses FRF.11 Annex C fragmentation.
Cisco MC3810 Concentrators Using Different VoFR Configuration Methods Example
On the Cisco MC3810, there are two different configuration methods for configuring VoFR on an interface. You can either use the vofr cisco command or the frame-relay interface-dlci dlci voice-encap size command. The first configuration method uses the vofr cisco command, which automatically provides support for WFQ and matches the configuration method supported on the Cisco 2600 and Cisco 3600 series routers.
The second configuration method uses the frame-relay interface-dlci voice-encap command, which does not automatically provide support for any queueing mechanism. However, if you enter the no frag-pre-queuing command, you can then statically configure either WFQ, priority queueing, or custom queueing on the interface. This second configuration method can be used to configure VoFR connections to Cisco 2600 and Cisco 3600 series routers.
You must choose one configuration method, because these commands are mutually exclusive on an interface.
Figure 76 shows an example of a VoFR PVC between two Cisco MC3810 concentrators, with each concentrator using a different VoFR configuration method. In this example, the VoFR PVC on the first Cisco MC3810 is configured using the vofr cisco command; on the second concentrator, the VoFR PVC is configured using the frame-relay interface-dlci voice-encap command.
Figure 76 VoFR PVC Between Two Cisco MC3810 Concentrators Using Different Configuration Methods
This configuration uses FRF.11 Annex C fragmentation.
Cisco-Trunk (Private Line) Calls Between Two Routers Example
Figure 77 shows an example of VoFR Cisco-trunk (private line) calls between two routers.
Figure 77 Permanent Switched Calls Between Two Routers
FRF.11 Trunk Calls Between Two Routers Example
Figure 78 shows an example of FRF.11 trunk calls configured between two routers.
Figure 78 FRF.11 Trunk Calls Between Two Routers
Tandem Configuration with Three Routers for Switched Calls Example
Figure 79 shows an example of a tandem configuration with two Cisco 3600 routers as endpoints, and a third Cisco 3600 as a tandem node.
Figure 79 Tandem Configuration with Three Routers for Switched Calls
Tandem Configuration with a Cisco MC3810 Tandem Node for Switched Calls Example
Figure 80 shows an example of a tandem configuration with a Cisco MC3810 acting as a tandem node.
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Figure 80 Tandem Configuration with a Cisco MC3810 Tandem Node for Switched Calls
Tandem Configuration with a Cisco MC3810 Endpoint Node for Switched Calls Example
Figure 81 shows an example of a tandem configuration with a Cisco MC3810 acting as an endpoint node.
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Figure 81 Tandem Configuration with a Cisco MC3810 Endpoint Node
Tandem Configuration with a Cisco MC3810 Endpoint Node for Cisco-Trunk (Private Line) Calls Example
Figure 82 shows an example of a tandem configuration with a Cisco MC3810 acting as an endpoint node for Cisco-trunk (private line) calls. When a Cisco MC3810 is on a VoFR network, the configuration for connections to and from the Cisco MC3810 is slightly different than for other routers that support VoFR. The vofr cisco command is required for those connections.
Figure 82 Tandem Configuration with a Cisco MC3810 Endpoint Node for Permanent Switched Call
Tandem Configuration with All Cisco MC3810 Concentrators for Switched Calls Example
Figure 83 shows an example of a tandem configuration with Cisco MC3810 concentrators as both endpoint and tandem nodes.
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Figure 83 Tandem Configuration with all Cisco MC3810 Concentrators
Cisco-Trunk Call with Hunt Groups Example
Figure 84 shows an example of a Cisco-trunk (private line) call with hunt groups configured. In this example, the two routers are in master-slave mode with a backup path. Router B is configured as a slave and Router A is configured as the master. The master makes periodic attempts to establish the trunk until the trunk is established. Two dial peers match the destination string configured in the voice port, but because one dial peer has a higher preference, the call setup is attempted through that dial peer. If the call setup fails, the master can continue attempting call setups using the next available dial peer. After all dial peers are exhausted, the master can continue following the list cyclically by starting again from the dial peer with the highest preference.
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Figure 84 Cisco Trunk (Private Line) Call with Hunt Groups
