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This chapter shows you how to configure voice ports for Voice over IP. This chapter contains the following sections:
For a complete description of the commands used in this chapter, refer to the Cisco IOS Multiservice Applications Command Reference publication. To locate documentation of other commands that appear in this chapter, use the command reference master index or search online.
Analog voice signalling in VoIP is sent via an analog voice port. Analog voice ports support three basic voice signalling types:
The VMN or VFC installed in your Cisco device determines the type of analog signalling a voice port sends.
In general, voice-port commands define the characteristics associated with a particular voice-port signalling type. Under most circumstances, the default voice-port configuration command values are adequate to configure FXO and FXS ports to transport voice data over your existing IP network. Because of the inherent complexities involved with PBX networks, E&M ports usually need specific values configured, depending on the specifications of the PBX devices in your telephony network.
To configure the analog voice ports in Voice over IP, perform the following tasks:
| Command | Purpose | |||
|---|---|---|---|---|
Step 7 | Router#configure terminal | Enters global configuration mode. | ||
Step8 | Router(confi g)#voice-port slot | Identifies the voice port you want to configure and enters voice-port configuration mode.
| ||
Step9 | Router(config-voiceport)#dial-type {dtmf | pulse}
| (For FXO ports only) Selects the appropriate dial type for out-dialing, either touchtone (DTMF) or pulse. | ||
Step10 | Router(config-voiceport)#signal {loop-start |
ground-start}
| Selects the appropriate signal type for this interface. With the loop-start keyword, only one side of a connection can hang up. (The default signalling type is loop-start.) With ground-start signalling, both sides of a connection can place calls and hang up. | ||
Step11 | Router(config-voiceport)#cptone country | Selects the appropriate voice call progress tone for this interface. For a list of supported countries, refer to the CiscoIOS Multiservice Applications Command Reference publication. | ||
Step12 | Router(config-voiceport)#ring frequency {25 | 50}
| (For FXS ports only) Selects the appropriate ring frequency (in Hertz) specific to the equipment attached to this voice port. | ||
Step13 | Router(config-voiceport)#ring number number | (For FXO ports only) Specifies the maximum number of rings to be detected before answering a call. | ||
Step14 | Router(config-voice-port)# connection {plar | trunk}
string
| (Optional) Sets up a connection mode for the voice port. The plar keyword specifies a private line, automatic ring down (PLAR) connection, which rings a remote telephone when the dial peer goes off hook. The trunk keyword specifies a straight tie-line connection to a PBX. The string argument specifies the remote telephone number or significant start digits of the number. | ||
Step15 | Router(config-voiceport)#music-threshold number | (Optional) Specifies the threshold (in decibels) for on-hold music. Valid entries are from -70 to -30. | ||
Step16 | Router(config-voiceport)#description string | (Optional) Attaches descriptive text about this voice-port connection. | ||
Step17 | Router(config-voiceport)#comfort-noise | (Optional) Specifies that background noise will be generated. | ||
Step18 | Router(config-voiceport)#no shutdown | Activates the voice port. |
 |
NoteAfter you change any voice-port command, it is a good idea to cycle the port by using the shutdown and no shutdown commands. |
|
NoteIn most cases, the default values for voice-port tuning commands will be sufficient. |
To fine-tune FXO or FXS voice ports, use the following commands beginning in privileged EXEC mode:
:
| Command | Purpose | |||
|---|---|---|---|---|
Step1 | Router#configure terminal | Enters global configuration mode. | ||
Step2 | Router(config)#voice-port slot | Identifies the voice port you want to configure and enters voice-port configuration mode.
| ||
Step3 | Router(config-voiceport)#input gain value | Specifies (in decibels) the amount of gain to be inserted at the receiver side of the interface. Acceptable values are from -6 to 14. | ||
Step4 | Router(config-voiceport)#output attenuation value | Specifies (in decibels) the amount of attenuation at the transmit side of the interface. Acceptable values are from 0 to 14. | ||
Step5 | Router(config-voiceport)#echo-canel enable | Enables echo-cancellation of voice that is sent out the interface and received back on the same interface. | ||
Step6 | Router(config-voiceport)#echo-canel coverage value | Adjusts the size (in milliseconds) of the echo-cancel. Acceptable values are 16, 24, and 32. | ||
Step7 | Router(config-voiceport)#non-linear | Enables nonlinear processing, which shuts off any signal if no near-end speech is detected. (Nonlinear processing is used with echo-cancellation.) | ||
Step8 | Router(config-voiceport)#timeouts initial seconds | Specifies the number of seconds the system will wait for the caller to input the first digit of the dialed digits. Valid entries for this command are from 0to120. | ||
Step9 | Router(config-voiceport)#timeouts interdigits seconds | Specifies the number of seconds the system will wait (after the caller has input the initial digit) for the caller to input a subsequent digit. Valid entries for this command are from 0 to 120. | ||
Step10 | Router(config-voiceport)#timing digits milliseconds | If the voice-port dial type is DTMF, configures the DTMF digit signal duration. The range of the DTMF digit signal duration is from 50to100. The default is 100. | ||
Step11 | Router(config-voiceport)#timing inter-digits milliseconds | If the voice-port dial type is DTMF, configures the DTMF interdigit signal duration. The range of the DTMF interdigit signal duration is from 50 to 500. The default is 100. | ||
Step12 | Router(config-voiceport)#timing pulse digit milliseconds | (FXO ports only) If the voice-port dial type is pulse, configures the pulse digit signal duration. The range of the pulse digit signal duration is from 10 to 20. The default is 20. | ||
Step13 | Router(config-voiceport)#timing pulse-inter-digit milliseconds | (FXO ports only) If the voice-port dial type is pulse, configures the pulse interdigit signal duration. The range of the pulse interdigit signal duration is from 100 to 1000. The default is 500. | ||
Step14 | Router(config-voiceport)#no shutdown | Activates the voice port. |
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NoteAfter you change any voice-port command, it is a good idea to cycle the port by using the shutdown and no shutdown commands. |
You can check the validity of your voice-port configuration by performing the following tasks:
To configure E&M voice ports, use the following commands beginning in privileged EXEC mode:
| Command | Purpose | |||
|---|---|---|---|---|
Step1 | Router#configure terminal | Enters global configuration mode. | ||
Step2 | Router(config)#voice-port slot | Identifies the voice port you want to configure and enters voice-port configuration mode.
| ||
Step3 | Router(config-voiceport)#dial-type {dtmf | pulse}
| Selects the appropriate dial type for out-dialing, either touchtone (DTMF) or pulse. | ||
Step4 | Router(config-voiceport)#signal {wink-start |
immediate | delay-dial}
| Selects the appropriate signal type for this interface. The wink-start keyword indicates that the calling side seizes the line by going off-hook on its E lead, then waits for a short off-hook "wink" indication on its M lead from the called side before sending address information as DTMF digits. The immediate keyword indicates that the calling side seizes the line by going off-hook on its E lead and sends address information as DTMF digits. Immediate signalling is used for E&M tie trunk interfaces. The delay-dial keyword indicates that the calling side seizes the line by going off-hook on its E lead. After a timing interval, the calling side looks at the supervision from the called side. If the supervision is on-hook, the calling side starts sending information as DTMF digits; otherwise the calling side waits until the called side goes on-hook and then starts sending address information. Delay-dial signalling is used for E&M tie trunk interfaces. | ||
Step5 | Router(config-voiceport)#cptone country | Selects the appropriate voice call progress tone for this interface. For a list of supported countries, refer to the CiscoIOS Multiservice Applications Command Reference publication. | ||
Step6 | Router(config-voiceport)#operation {2-wire |
4-wire}
| Selects the appropriate cabling scheme for this voice port. | ||
Step7 | Router(config-voiceport)#type {1 | 2 | 3 | 5}
| Selects the appropriate E&M interface type. Type 1 indicates the following lead configuration:
Type 2 indicates the following lead configuration:
Type 3 indicates the following lead configuration:
Type 5 indicates the following lead configuration:
| ||
Step8 | 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. | ||
Step9 | Router(config-voice-port)# connection {plar | trunk}
string
| (Optional) Sets up a connection mode for the voice port. The plar keyword specifies a PLAR connection, which rings a remote telephone when the dial peer goes off-hook. The trunk keyword specifies a straight tie-line connection to a PBX. The string argument specifies the remote telephone number or significant start digits of the number. | ||
Step10 | Router(config-voiceport)#music-threshold number | (Optional) Specifies the threshold (in decibels) for on-hold music. Valid entries are from -70 to -30. | ||
Step11 | Router(config-voiceport)#description string | (Optional) Attaches descriptive text about this voice-port connection. | ||
Step12 | Router(config-voiceport)#comfort-noise | (Optional) Specifies that background noise will be generated. | ||
Step13 | Router(config-voiceport)#no shutdown | Activates the voice port. |
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NoteAfter you change any voice-port command, it is a good idea to cycle the port by using the shutdown and no shutdown commands. |
|
NoteIn most cases, the default values for voice-port tuning commands will be sufficient. |
To fine-tune E&M voice ports, use the following commands beginning in privileged EXEC mode:
| Command | Purpose | |||
|---|---|---|---|---|
Step1 | Router#configure terminal | Enters global configuration mode. | ||
Step2 | Router(config)#voice-port slot | Identifies the voice port you want to configure and enters voice-port configuration mode.
| ||
Step3 | Router(config-voiceport)#input gain value | Specifies (in decibels) the amount of gain to be inserted at the receiver side of the interface. Acceptable values for the value argument are from -6 to 14. | ||
Step4 | Router(config-voiceport)#output attenuation value | Specifies (in decibels) the amount of attenuation at the transmit side of the interface. Acceptable values for the value argument are from 0 to 14. | ||
Step5 | Router(config-voiceport)#echo-cancel enable | Enables echo-cancellation of voice that is sent out the interface and received back on the same interface. | ||
Step6 | Router(config-voiceport)#echo-cancel coverage value | Adjusts the size (in milliseconds) of the echo-cancel. Acceptable values for the value argument are 16, 24, and 32. | ||
Step7 | Router(config-voiceport)#non-linear | Enables nonlinear processing, which shuts off any signal if no near-end speech is detected. (Nonlinear processing is used with echo-cancellation.) | ||
Step8 | Router(config-voiceport)#timeouts initial seconds | Specifies the number of seconds the system will wait for the caller to input the first digit of the dialed digits. Valid entries for the seconds argument are from 0to120. | ||
Step9 | Router(config-voiceport)#timeouts interdigit seconds | Specifies the number of seconds the system will wait (after the caller has input the initial digit) for the caller to input a subsequent digit. Valid entries for the seconds argument are from 0 to 120. | ||
Step10 | Router(config-voiceport)#timing clear-wait milliseconds | Specifies the minimum amount of time between the inactive seizure signal and the call being cleared. Valid entries for the milliseconds argument are from 200 to 2000 milliseconds. | ||
Step11 | Router(config-voiceport)#timing delay-duration milliseconds | Specifies the delay signal duration for delay dial signalling. Valid entries for the milliseconds arguments are from 100 to 5000 milliseconds. | ||
Step12 | Router(config-voiceport)#timing delay-start milliseconds | Specifies the minimum delay time from outgoing seizure to outdial address. Valid entries for the milliseconds argument are from 20 to 2000milliseconds. | ||
Step13 | Router(config-voiceport)#timing dial-pulse min-delay milliseconds | Specifies the time between generation of wink-like pulses. Valid entries for the milliseconds argument are from 0 to 5000 milliseconds. | ||
Step14 | Router(config-voiceport)#timing digit milliseconds | If the voice-port dial type is DTMF, configures the DTMF digit signal duration. Valid entries for the milliseconds argument are from 50 to 100milliseconds. | ||
Step15 | Router(config-voiceport)#timing inter-digit milliseconds | If the voice-port dial type is DTMF, specifies the DTMF interdigit duration. Valid entries for the milliseconds argument are from 50 to 500milliseconds. | ||
Step16 | Router(config-voiceport)#timing pulse pulse-per-second | If the voice-port dial type is pulse, specifies the pulse dialing rate. Valid entries for the pulse-per-second argument are from 10 to 20 pulses per second. | ||
Step17 | Router(config-voiceport)#timing pulse-inter-digit milliseconds | If the voice-port dial type is pulse, specifies the pulse dialing interdigit timing. Valid entries for the milliseconds argument are 100 to 1000 milliseconds. | ||
Step18 | Router(config-voiceport)#timing wink-duration milliseconds | Specifies the maximum wink signal duration. Valid entries for the milliseconds argument are from 100 to 400 milliseconds. | ||
Step19 | Router(config-voiceport)#timing wink-wait milliseconds | Specifies the maximum wink-wait duration for a wink start signal. Valid entries for the milliseconds argument are from 100 to 5000 milliseconds. | ||
Step20 | Router(config-voiceport)#no shutdown | Activates the voice port. |
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NoteAfter you change any voice-port command, it is a good idea to cycle the port by using the shutdown and no shutdown commands. |
You can check the validity of your voice-port configuration by performing the following tasks:
The following sections include tasks for configuring digital voice port types:
Digital T1 packet voice trunk network modules for Cisco 2600 and 3600 series routers allow enterprises or service providers, using the equipped routers as CPE, to deploy digital voice and fax relay. These modules receive constant bit-rate telephony information over T1 interfaces and can convert that information to a compressed format, so that it can be sent as VoIP. The digital T1 packet voice trunk network modules can connect to either a PBX (or similar telephony device) or to a central office (CO) in order provide PSTN connectivity.
T1 digital VoIP includes the following functionality:
You must set timing, signalling, framing, and line encoding as follows:
This section describes the five basic timing scenarios that can occur when a digital T1 packet voice trunk network module is connected to a PBX, CO, or both. In all of the following examples, the PSTN (or CO) and the PBX are interchangeable for the purposes of providing or receiving clocking.
The digital T1 module has an on-board Phase-Lock Loop (PLL) chip that can either provide a clock source to both T1 lines or receive clocking that can drive the second T1 line. All timing commands are T1 controller configuration commands.
In this scenario, the digital T1 module is the clock source for the connected device. The PLL generates the clock internally and drives the clocking on the T1 line. Figure 13 shows how the single T1 port provides clocking for the PBX.
The following configuration sets up this clocking method:
controller T1 1/0 framing esf linecoding b8zs clock source internal ds0-group 1 timeslots 1-24 type e&m-wink
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NoteGenerally 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. |
In this scenario, the digital T1 module receives clocking from the connected device (CO or PBX). The PLL clocking is driven by the clock reference on the receive (Rx) side of the T1 connection. Figure 14 shows how the single T1 port receives clocking from the line.
The following configuration sets up this clocking method:
controller T1 1/0 framing esf linecoding b8zs clock source line ds0-group 1 timeslots 1-24 type e&m-wink
In this scenario, the digital T1 has two reference clocks, one from the PBX and another from the CO. Because the PLL can only derive clocking from one source, this case is more complex than the two preceding examples.
Before looking at the details, consider two important concepts that underlie the clocking method:
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NotePhysical layer issues, such as bad cabling or faulty clocking references, can also cause slips. |
Figure 15 shows how the dual T1 ports receive clocking from the line.
As shown in Figure 15, the PLL derives clocking from the CO and puts the T1 port connected to the PBX into looped-time mode. This is usually best because the CO provides an excellent clock source (and usually requires that it provide that source) and a PBX usually must receive clocking from the other T1.
The following configuration sets up this clocking method:
! The following T1 port is connected to the CO. controller T1 1/0 framing esf linecoding b8zs clock source line primary ds0-group 1 timeslots 1-24 type e&m-wink ! ! The following T1 port is connected to the PBX. controller T1 1/1 framing esf linecoding b8zs clock source line ds0-group 1 timeslots 1-24 type e&m-wink
The clock source line primary command tells the router to use this T1 port to drive the PLL. All other T1 ports configured as clock source line are then put into an implicit loop-timed mode. If the primary T1 port fails or goes down, the other T1 instead receives the clock that drives the PLL. In this configuration, T1 1/1 may see controlled slips, but these slips should not force the line down. This method prevents the PBX from seeing slips.
In this scenario, the digital T1 module receives clocking for the PLL from T1 0 and uses this clock as a reference to clock T1 1. If T1 0 fails, the PLL internally generates the clock reference to drive T1 1.
Figure 16 shows dual T1 ports where one T1 port receives clocking from the line and one T1 port provides clocking.
The following configuration sets up this clocking method:
controller T1 1/0 framing esf linecoding b8zs clock source line ds0-group 1 timeslots 1-24 type e&m-wink ! controller T1 1/1 framing esf linecoding b8zs clock source internal ds0-group 1 timeslots 1-24 type e&m-wink
In this scenario, the router generates the clock for the PLL and therefore for both T1s.
Figure 17 shows how dual T1 ports both receive clocking from the router.
The following configuration sets up this clocking method:
controller T1 1/0 framing esf linecoding b8sz clock source internal ds0-group 1 timeslots 1-24 type e&m-wink ! controller T1 1/1 framing esf linecoding b8zs clock source internal ds0-group 1 timeslots 1-24 type e&m-wink
There are three types of signalling that you should consider for digital T1:
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NoteDigital T1 packet voice trunk network modules support T1 CAS for this Cisco IOS release. Future Cisco IOS releases will support E1, PRI, R2, and common channel signalling (CCS). The digital T1 module can support E&M wink-start, immediate-start, and delay-start signalling, and FXS and FXO ground-start and loop-start signalling. |
controller T1 1/0 ds0-group 1 timeslots 1-24 type e&m-wink-start
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NoteCurrently, wink-start signalling provides only the functionality of Feature Group B and not that of Feature Group D, which will be supported in later Cisco IOS releases. |
controller T1 1/0 ds0-group 1 timeslots 1-24 type fxo-ground-start
controller T1 1/0 ds0-group 1 timeslots 1-24 type fxs-loop-start
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NoteAlthough some switches (CO or PBX) can specify both an inbound and outbound signalling method, Cisco VoIP gateway routers can only specify one signalling type for both inbound and outbound calls. The switch inbound and outbound signalling types must match, or calls may only work in one direction. |
Digital T1 packet voice trunk network modules support two types of framing for T1 CAS: ESF and SF (also called D4 framing). The framing type of the router and switch (CO or PBX) must match. The framing controller configuration command defines T1 framing, as in the following examples:
controller T1 1/0 framing esf
or
controller T1 1/0 framing sf
Digital T1 packet voice trunk network modules support two types of framing for T1 CAS: B8ZS and AMI. The line encoding of the router and switch (CO or PBX) must match. The linecoding controller configuration command defines T1 framing, as in the following examples:
controller T1 1/0 linecoding b8zs
or
controller T1 1/0 linecoding ami
The following restrictions apply to digital T1 packet voice trunk network module configuration:
Digital T1 packet voice requires specific service, software, and hardware as follows:
|
NoteYou can install one module in a Cisco 2600 series router or a Cisco 3620 router. A Cisco3640 router can support three modules, and you can install as many as six modules in a Cisco3660 router. |
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NoteEach PVDM holds three DSPs. With five PVDM slots populated, a total of 15 DSPs are provided. High-complexity codecs support two simultaneous calls on each DSP, while medium-complexity codecs support four calls on each DSP. |
To specify codec settings for voice cards and set up T1 controllers for clocking and other T1 parameters, and for DS0 groups that define the channels for compressed voice and TDM groups for drop-and-insert capability, use the following commands, beginning in privileged EXEC mode:
| Command | Purpose | |||
|---|---|---|---|---|
Step1 | Router# configure terminal | Enters global configuration mode. | ||
Step2 | Router(config)# voice-card slot | Enters voice card interface configuration mode and specifies the slot location by using a value from 0 to 5, depending upon your router. | ||
Step3 | Router(config-voice-ca)# codec complexity {high |
medium}
| Specifies the codec complexity based on the codec standard you are using. High-complexity codecs support lower call density than do medium-complexity codecs. The number of channels supported is based on the number of PVDMs installed and the codec complexity. Here are guidelines:
All voice cards in a router must use the same codec complexity setting. The keyword that you specify for codec complexity command affects the choice of codecs available using the codec dial-peer configuration command. For more information about applying codecs to dial peers, see the "Configuring Dial Peers" section later in this chapter.
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Step4 | Router(config)# controller T1 slot/port | Enters controller configuration mode for the T1 controller at the specified slot/port location. Valid values for the slot and port arguments are 0 and 1. | ||
Step5 | Router(config-controller)# clock source {line
[primary] | internal}
| Configures controller T1 1/0 to specify the clock source. The line keyword specifies that the clock source is derived from the active line---rather than from the free-running internal clock. This is the default setting and is generally more reliable. These rules apply to clock sourcing on the T1 controller ports:
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Step6 | Router(config-controller)# | Sets the framing according to your service provider instructions. Use the sf keyword to select SF format and the esf keyword to select the ESF format. | ||
Step7 | Router(config-controller)# | Sets the line encoding according to the instructions given by your service provider. Use the b8zs keyword to select B8ZS encoding, which encodes a sequence of eight zeros in a unique binary sequence to detect line-coding violations. Use the ami keyword to select AMI encoding, which represents zeros using a 01 during each bit cell, and ones are represented by 11 or 00, alternately, during each bit cell. AMI requires that the sending device maintain ones density. Ones density is not maintained independent of the data stream. | ||
Step8 | Router(config-controller)#cablelength long {gain26
| gain36}{-15db | -22.5db | -7.5db | 0db}
| (T1 interfaces only) Sets the cable length. The cable length setting must conform to the actual cable length you are using. For example, if you attempt to enter the cablelength short command on a long-haul T1 link, the command is rejected. To set a cable length longer than 655 feet for a T1 link, enter the cablelength long command. The keywords are as follows:
To set a cable length 655 feet or less for a T1 link, enter the cablelength short command. There is no default for cablelength short. The keywords are as follows:
If you do not set the cable length, the system defaults to a setting of cablelength long gain26 0db. | ||
Step9 | Router(config-controller)# ds0-group ds0-group-no
timeslots timeslot-list type {e&m-immediate |
e&m-delay | e&m-wink | fxs-ground-start |
fxs-loop-start | fxo-ground-start | fxo-loop-start}
| Defines the T1 channels for use by compressed voice calls and the signalling method the router uses to connect to the PBX or CO. You should set up DS0 groups after you have specified codec complexity in voice-card configuration. If you modify the codec complexity command parameters, you must first remove any existing DS0 groups, then reinstate them after the change to the codec complexity. The ds0-group-no argument is a value from 0 to 23 that identifies the DS0 group.
The timeslot-list argument is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen to indicate a range of time slots. For T1, allowable values are from 1 to 24. To map individual DS0 time slots, define additional groups. The system maps additional voice ports for each defined group. The signalling method selection for the type keyword depends on the connection that you are making:
| ||
Step10 | Router(config-controller)# tdm-group tdm-group-no timeslots timeslot-list type [e&m | fxs [loop-start | ground-start] fxo [loop-start | ground-start]] | (Optional) Defines TDM channel groups for the drop-and-insert (also called TDM Cross-Connect) function with a two-port T1 multiflex trunk interface card. The tdm-group-no argument specifies a value from 0 to 23 that identifies the channel group. The timeslot-list argument is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen to indicate a range of time slots. For T1, allowable values are from 1 to 24. The signalling method selection for the type keyword depends on the connection that you are making. The fxs and fxo options allow you to specify a ground-start or loop-start line.
| ||
Step11 | Router(config-controller)# no shutdown | Activates the controller. | ||
Step12 | Router(config-controller)# exit | Exits controller configuration mode. | ||
Step13 | Router(config)# connect id T1 slot/port tdm-group-no-1 T1 slot/port tdm-group-no-2 | (Optional) Sets up the connection between two T1 TDM groups of time slots on the trunk interfaces---for the drop-and-insert capability. The id argument is a name for the connection. Identify each T1 controller by its slot/port location. Valid values for the slot and port arguments are 0 and 1. The tdm-group-no-1 and tdm-group-no-2 arguments identify the TDM group numbers (from 0 to 23) on the specified controller. |
Repeat Steps 2 and 3 for each voice card.
Repeat Steps 4 through 12 for each controller.
To configure voice port parameters, use the following commands, beginning in global configuration mode:
| Command | Purpose | |||
|---|---|---|---|---|
Step1 | Router(config)# voice-port slot/port:ds0-group-no | Enters voice-port configuration mode. The slot argument is the router location where the voice module is installed. Valid entries are from 0 to3. The port argument indicates the VIC location. Valid entries are 0 or 1. Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1 card. | ||
Step2 | Router(config-voice-port)# busyout monitor interface interface number | (Optional) Specifies a LAN or WAN interface that will be monitored, and, when it is down, trigger a busyout (off-hook) state on the voice port. This allows rerouting of calls. For example, if you specify Serial 1/0 as the interface and number, the voice port sends a busyout signal when that interface is down. You can issue the command repeatedly to specify as many interfaces, virtual interfaces, and subinterfaces as are required for a voice port. If you issue the command three times so that three interfaces are monitored, the voice port only goes into busyout state when all three interfaces are down. When any one of the interfaces is operational, the busyout state is removed. | ||
Step3 | Router(config-voice-port)# comfort-noise | (Optional) Enables comfort noise. (This parameter is enabled by default.) It creates subtle background noise to fill silent gaps during calls when VAD is enabled on voice dial peers. If comfort noise is not generated, the silence can be unsettling to callers. | ||
Step4 | Router(config-voice-port)# echo-cancel enable | (Optional) Enables echo cancellation. (This setting is enabled by default.) Echo cancellation adds to the quality of voice transmissions by adjusting the echo that occurs on the interface due to impedance mismatches. Some echo is reassuring; echo over 25milliseconds long can cause problems. | ||
Step5 | Router(config-voice-port)# echo-cancel coverage {16
| 24 |32 | 8}
| (Optional) Adjusts the echo canceller by the specified number of milliseconds; the default is 16. | ||
Step6 | Router(config-voice-port)# connection {plar |trunk}
string
| (Optional) Sets up a connection mode for the voice port. The plar keyword specifies a private line auto ring down (PLAR) connection, which rings a remote telephone when the dial peer goes off hook. The trunk keyword specifies a straight tie-line connection to a PBX. The string argument specifies the remote telephone number or significant start digits of the number. | ||
Step7 | Router(config-voice-port)# timeouts interdigit seconds | (Optional) Sets the number of seconds the system waits---after the caller has input the initial digit---for a subsequent digit of the dialed string. If the timeout ends before the destination is identified, a tone sounds and the call ends. The default value is 10seconds, and the timeout can be set from 0 to 120seconds.
| ||
Step8 | Router(config-voice-port)# exit | Exits voice-port configuration mode. |
Repeat Steps 2 through 8 for each DS0 group you create.
You can check the validity of your digital T1 packet VTNM configuration by performing the following tasks:
Cisco T1/E1 Multiflex VWICs support voice and data applications in Cisco 2600 and 3600 series routers. The VWICs offer WIC and VIC functionality in a variety of applications for enterprises and for service providers that supply CPE.
Figure 18 shows how T1/E1 Multiflex VWIC are used where VWIC ports are assigned to a PBX and a CO in an network environment where there is no WAN connectivity.
Multiflex VWICs support the following applications:
The following multiflex VWICs are available:
Multiflex VWIC features include the following:
|
NoteT1/E1 channels can be used either for drop-and-insert or VoIP, but not both. |
The following restrictions apply to T1/E1 multiflex VWIC configurations:
T1/E1 multiflex VWICs require the following specific service, software, and hardware:
|
NoteYou can install one digital T1 packet voice trunk network module in a Cisco 2600 series router or a Cisco 3620 router. A Cisco 3640 router can support three modules, and you can install as many as six modules in a Cisco3660 router. |
If you are configuring T1 multiflex VWICs installed in digital T1 packet voice trunk network modules for voice, use the following commands beginning in privileged EXEC mode:
| Command | Purpose | |||
|---|---|---|---|---|
Step1 | Router# configure terminal | Enters global configuration mode. | ||
Step2 | Router(config)# voice-card slot | Enters voice card interface configuration mode. The slot argument specifies the card location by using a value from 0 to 5, depending upon your router. | ||
Step3 | Router(config-voice-ca)# codec complexity {high |
medium}
| Specifies the codec complexity based on the codec standard you are using. High-complexity codecs support lower call density than do medium-complexity codecs. The number of channels supported is based on the number of PVDMs installed and the codec complexity. Here are guidelines:
All voice cards in a router must use the same codec complexity setting. The keyword that you specify for codec complexity affects the choice of codecs available using the codec dial-peer configuration command. For more information about applying codecs to dial peers, see the "Configuring Dial Peers" section later in this chapter. You cannot change codec complexity while DS0 groups are defined. If they are already set up, use the no ds0-group command before resetting the codec complexity. | ||
Step4 | Router(config)# controller T1 slot/port | Enters controller configuration mode for the VWIC. Valid values for the slot argument are 0 through 5 and for the port argument are 0 and1. | ||
Step5 | Router(config-controller)# ds0-group ds0-group-no
timeslots timeslot-list type {e&m-immediate |
e&m-delay |e&m-wink | fxs-ground-start |
fxs-loop-start | fxo-ground-start | fxo-loop-start}
| (Voice only) Defines the T1 channels for use by compressed voice calls and the signalling method the router uses to connect to the PBX or CO. Set up DS0 groups after you have specified codec complexity in voice-card interface configuration mode. If you modify the codec complexity command parameters, you must first remove any existing DS0 groups, then reinstate them after the change to the codec complexity. The ds0-group-no argument is a value from 0 to 23 that identifies the DS0 group.
The timeslot-list argument is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen to indicate a range of time slots. For T1, allowable values are from 1 to 24. To map individual DS0 time slots, define additional groups. The system maps additional voice ports for each defined group. The signalling method selection for the type keyword depends on the connection that you are making:
|
To configure T1 and E1 controllers, use the following commands beginning in global configuration mode:
| Command | Purpose | |
|---|---|---|
Step1 | Router(config)# controller {T1 | E1} slot/port
| Enters controller configuration mode for the T1 or E1 controller at the specified slot/port location. |
Step2 | Router(config-controller)# loopback {diagnostic |
local {payload | line}| remote {iboc | esf {payload |
line}}
| (Optional) Generates a local loopback test at the line or payload level, or a remote loopback. |
Step3 | Router(config-controller)# clock source {line
[primary] | internal}
| Specifies the clock source. The line keyword specifies that the clock source is derived from the active line---rather than from the free-running internal clock. This is the default setting and is generally more reliable. These rules apply to clock sourcing:
|
Step4 | Router | Sets the framing to SF or ESF format, according to service provider requirements. Sets the framing to cyclic redundancy check 4 (CRC4) or no CRC4, according to service provider requirements. The australia optional keyword specifies Australian Layer 1 Homologation for E1 framing. |
Step5 | Router | Sets the line encoding according to your service provider's instructions. Use the b8zs keyword to specify B8ZS line encoding. B8ZS, available only for T1 lines, encodes a sequence of eight zeros in a unique binary sequence to detect line coding violations. Use the ami keyword to specify AMI encoding. AMI, available for T1 or E1 lines, represents zeros using a 01 during each bit cell, and ones are represented by 11 or 00, alternately, during each bit cell. AMI requires that the sending device maintain ones density. Ones density is not maintained independent of the data stream. For E1, set the line-coding to either AMI or high-density bipolar 3 (HDB3), which is the default. |
Step6 | Router(config-controller)# line-termination {75-ohm
| 120-ohm}
| (E1 only) Enters a line-termination value. This command specifies the impedance (amount of wire resistance and reactivity to current) for the E1 termination. Impedance levels are maintained to avoid data corruption over long-distance links. Specify the 120-ohm keyword to match the balanced 120-ohm interface. This is the default. Specify the 75-ohm keyword for an unbalanced BNC 75-ohm interface. |
Step7 | Router(config-if)# fdl {att | ansi | both}
| (T1 interfaces only) Sets the Facility Data Link (FDL) exchange standard for the CSU controllers. The FDL is a 4-kbps channel used with the ESF framing format to provide out-of-band messaging for error-checking on a T1 link. You typically leave this setting at the default, ansi, which follows the American National Standards Institute (ANSI) T1.403 standard for extended superframe facilities data-link exchange support. Changing it allows improved management in some cases but can cause problems if your setting is not compatible with that of your service provider. Use the att keyword to select the AT&T TR54016 standard for ESF facilities data-link exchange support. Use the both keyword to enable both of the described standards. |
Step8 | Router(config-controller)#cablelength long {gain26
| gain36} {-15db | -22.5db | -7.5db | 0db}
| (T1 interfaces only) Sets the cable length. The cable length setting must conform to the actual cable length you are using. For example, if you attempt to enter the cablelength short command on a long-haul T1 link, the command is rejected. To set a cable length longer than 655 feet for a T1 link, enter the cablelength long command. The keywords are as follows:
To set a cable length 655 feet or less for a T1 link, enter the cablelength short command. There is no default for cablelength short. The keywords are as follows
If you do not set the cable length, the system defaults to a setting of cablelength long gain26 0db. |
Repeat the steps following Step 4 for each controller.
(Optional) To set up drop-and-insert, use the following commands, beginning in controller configuration mode:
| Command | Purpose | |||
|---|---|---|---|---|
Step1 | Router(config-controller)# tdm-group tdm-group-no timeslots timeslot-list type [e&m | fxs [loop-start | ground-start] fxo [loop-start | ground-start] | Sets up TDM channel groups for the drop-and-insert function with a 2-port multiflex VWIC. The tdm-group-no argument identifies the TDM group and is a value from 0 to 23 for T1 and from 0 to 30 for E1. The timeslot-list argument indicates a range of time slots and is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen. The valid range is from 1 to 24 for T1. For E1, the range is from 1 to 31. The signalling method selection for the type keyword depends on the connection that you are making. The fxs and fxo options allow you to specify a ground-start or loop-start line.
| ||
Step2 | Router(config-controller)# channel-group channel-group-no timeslots timeslot-list [speed [48 | 56 | 64 ]] | (Optional) Sets up channel groups for WAN data services with a 2-port multiflex drop-and-insert VWIC. The channel-group-no argument identified the channel group and is a value from 0 to 23 for T1 and from 0 to 30 for E1; because there can be only one channel group on a 1- or 2-port multiflex VWIC, 0 is always the value. The timeslot-list argument indicates a range of time slots and is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen. The valid range is from 1 to 24 for T1. For E1, the range is from 1 to 31. The optional speed keyword defaults to 56 kbps for T1 and 64 kbps for E1.
| ||
Step3 | Router(config-controller)# no shutdown | Activates the controller. | ||
Step4 | Router(config-controller)# exit | Exits controller configuration mode. | ||
Step5 | Router(config)# connect id {T1 | E1} slot/port-1
tdm-group-no-1 {T1 | E1} slot/port-2 tdm-group-no-2
| Sets up the connection between two T1 or E1 TDM groups of time slots on the WVIC---for drop-and-insert. Use the id argument to define a name for the connection. Use the slot/port argument to identify each controller by its location. Use the tdm-group-no-1 and tdm-group-no-2 arguments to identify the TDM group numbers (from 0 to 23 or 30) on the specified controller. |
To configure voice port parameters to support local and remote stations, use the following commands beginning in global configuration mode:
| Command | Purpose | |
|---|---|---|
Step1 | Router(config)# voice-port slot/port:ds0-group-no | Enters voice-port configuration mode. The slot argument identifies the router location where the voice module is installed. Valid entries are from 0to 3. The port argument indicates the multiflex VWIC location. Valid entries are 0 or 1. Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1 card. |
Step2 | Router(config-voice-port)# busyout monitor interface interface number | (Optional) Specifies a LAN or WAN interface that will be monitored, and, when it is down, trigger a busyout (off-hook) state on the voice port. This allows rerouting of calls. For example, if you specify Serial 1/0 as the interface and number, the voice port sends a busyout signal when that interface is down. You can issue the command repeatedly to specify as many interfaces, virtual interfaces, and subinterfaces as are required for a voice port. For example, if you issue the command three times so that three interfaces are monitored, the voice port only goes into busyout state when all three interfaces are down. When any one of the interfaces is operational, the busyout state is removed. |
Step3 | Router(config-voice-port)# comfort-noise | (Optional) Enables comfort noise. (This parameter is enabled by default.) It creates subtle background noise to fill silent gaps during calls when VAD is enabled on voice dial peers. If comfort noise is not generated, the silence can be unsettling to callers. |
Step4 | Router(config-voice-port)# echo-cancel enable | (Optional) Enables echo cancellation. (This setting is enabled by default.) Echo cancellation adds to the quality of voice transmissions by adjusting the echo that occurs on the interface due to impedance mismatches. Some echo is reassuring; echo over 25milliseconds long can cause problems. |
Step5 | Router(config-voice-port)# echo-cancel coverage {16
| 24 |32 | 8}
| (Optional) This command adjusts the echo canceller by the specified number of milliseconds; the default is 16. |
Step6 | Router(config-voice-port)# connection {plar | trunk}
string
| (Optional) Sets up a connection mode for the voice port. The plar keyword specifies a PLAR connection, which rings a remote telephone when the dial peer goes off hook. The trunk keyword specifies a straight tie-line connection to a PBX. The string argument specifies the remote telephone number or significant start digits of the number. |
Repeat Steps 1 through 8 for each DS0 group you create.
You can check the validity of your digital T1/EI multiflex interface configuration by performing the following tasks:
VoIP enables the Cisco 2600 and Cisco 3600 series of modular routers to carry voice traffic simultaneously with data traffic over an IP network. VoIP is primarily a software feature, supporting both voice and fax calls. Support for the ISDN BRI signalling type allows a Cisco 2600 or Cisco 3600 series router to provide voice access connectivity to either an ISDN telephone network or a digital interface on PBX and key communications system. The voice or data also crosses an IP network to which the router connects, allowing branch offices and enterprises to route incoming PSTN ISDN BRI calls over an IP network or send outgoing digital fax and voice calls via an IP network.
ISDN BRI VoIP offers direct ISDN network connectivity and connectivity to the digital interfaces of PBX and key communications systems. Prior to the introduction of this feature, VoIP was available only for FXS connection to a POTS telephone or other telephony equipment, FXO for connection to a POTS PBX or key system, or E&M for 2-wire and 4-wire telephone and trunk interfaces---typically used to connect remote calls from an IP network to a PBX.
ISDN BRI VoIP provides the following toll-saving benefits for enterprises and branch offices:
Figure 19 shows a home-office user dialing directly in to a local router via the PSTN, and reaching headquarters through an IP network, saving the cost of a long-distance call. In another example, Figure 19 shows how an extension at headquarters makes a fax or voice call to a branch office in a different area code using a corporate IP network only.
Before you can configure your Cisco 2600 or Cisco 3600 series router for VoIP on a BRI interface, you must perform the following tasks:
To configure BRI interfaces, use the following commands beginning in privileged EXEC mode:
| Command | Purpose | |
|---|---|---|
Step1 | Router# configure terminal | Enters global configuration mode. |
Step2 | Router(config)# isdn switch-type switch-type | Configures the global ISDN switch type to match the service provider switch type. For a list of keywords, see Table 8. |
Step3 | Router(config)# interface bri slot/port | Enters interface configuration mode to configure parameters for the specified interface. The slot argument specifies the location of the VNM in the router. The port argument specifies the location of the BRI VIC in the VNM. Valid values are 0 or 1. |
Step4 | Router(config-if)# no ip address | Specifies that there is no IP address for this interface. |
Step5 | Router(config-if)# no ip-directed broadcast | Disables the translation of directed broadcast to physical broadcasts. |
Step6 | Router(config-if)# isdn switch-type switch-type | (Optional) Configures the interface ISDN switch type to match the service provider switch type. The interface ISDN switch type overrides the global ISDN switch type on the interface. For a list of switch type keywords, see Table 8. |
Step7 | Router(config-if)# isdn spid1 spid-number [ldn] | Specifies a SPID and local directory number for the B1 channel. Currently, only the DMS-100 and NI-1 switch types require SPIDs. Although the Lucent 5ESS switch type might support a SPID, we recommend that you set up that ISDN service without SPIDs. |
Step8 | Router(config-if)# isdn spid2 spid-number [ldn] | Specifies a SPID and local directory number for the B2 channel. |
Step9 | Router(config-if)# isdn twait-disable | (Optional) Delays a National ISDN BRI switch a random time before activating the Layer 2 interface when the switch starts up. Use this command when the ISDN switch type is basic-nil. |
Step10 | Router(config-if)# isdn incoming-voice modem | Configures the port for incoming voice calls. |
Table 8 lists the available switch type keywords.
| Country | ISDN Switch Type | Description |
|---|---|---|
Australia | basic-ts013 | Australian TS013 switches |
Europe | basic-1tr6 | German 1TR6 ISDN switches |
| basic-nwnet3 | Norwegian NET3 ISDN switches (phase 1) |
| basic-net3 | NET3 ISDN switches (United Kingdom and others) |
| vn2 | French VN2 ISDN switches |
| vn3 | French VN3 ISDN switches |
Japan | ntt | Japanese NTT ISDN switches |
New Zealand | basic-nznet3 | New Zealand NET3 switches |
North America | basic-5ess | Lucent Technologies basic rate switches |
| basic-dms100 | NT DMS-100 basic rate switches |
| basic-ni1 | National ISDN-1 switches |
You can check the validity of your ISDN BRI configuration by performing the following tasks:
T1/E1 high-capacity digital voice port adapters for Cisco 7200 series routers allow enterprises or service providers, using the equipped routers as CPE, to deploy digital voice and fax relay. These port adapters receive constant bit-rate telephony information over T1 interfaces and can convert that information to a compressed format and be sent as VoIP.
T1/E1 digital voice over IP includes the following functionality:
The following restrictions apply to digital T1/E1 voice port adapter configuration:
Digital T1/E1 voice requires specific service, software, and hardware as follows:
To configure a DSPfarm interface, use the following commands beginning in global configuration mode:
| Command | Purpose | |||
|---|---|---|---|---|
Step1 | Router(config)#dspinterface dspfarm slot/port | Opens DSPfarm interface configuration mode to configure the DSP interface. | ||
Step2 | Router(config-dspfarm)# codec {high | medium | low}
1-30
| Specifies the codec complexity based on the codec standard you are using. High-complexity codecs support lower call density than do medium-complexity codecs. For example:
The keyword that you specify for codec affects the choice of codecs available using the codec dial-peer configuration command.
| ||
Step3 | Router(config-dspfarm)#no shutdown | Enables the interface. |
To specify codec settings for card types and set up T1 controllers for clocking and other T1 parameters, and for DS0 groups that define the channels for compressed voice and TDM groups for drop-and-insert capability, use the following commands beginning in privileged EXEC mode:
| Command | Purpose | |||
|---|---|---|---|---|
Step1 | Router# configure terminal | Enters global configuration mode. | ||
Step2 | Router(config)# card type {t1/e1} slot
| Enters T1 card type and specifies the slot location. Valid entries for the slot argument are 0 to 5, depending upon your router. | ||
Step3 | Router(config)# controller T1 slot/port | Enters controller configuration mode for the T1 controller at the specified slot/port location. Valid values for the slot and port arguments are 0 and 1. | ||
Step4 | Router(config-controller)# clock source {line
[primary] | internal}
| Configures controller T1 1/0 to specify the clock source. The line keyword specifies that the clock source is derived from the active line---rather than from the free-running internal clock. This is the default setting and is generally more reliable. These rules apply to clock sourcing on the T1 controller ports:
| ||
Step5 | Router(config-controller)# | Sets the framing according to the instructions from your service provider. Use the esf keyword to select the ESF framing format or the sf keyword for the SF framing format. | ||
Step6 | Router(config-controller)# | Sets the line encoding according to the instructions from your service provider. Use the b8zs keyword to specify B8ZS line encoding, which encodes a sequence of eight zeros in a unique binary sequence to detect line-coding violations. Use the ami keyword to specify AMI line encoding, which represents zeros using a 01 during each bit cell, and ones are represented by 11 or 00, alternately, during each bit cell. AMI requires that the sending device maintain ones density. Ones density is not maintained independent of the data stream. | ||
Step7 | Router(config-controller)#cablelength long {-15db |
-22.5db | -7.5db | 0db}
| (T1/E1 interfaces only) Configures the cable length. The cable length setting must conform to the actual cable length you are using. For example, if you attempt to enter the cablelength short command on a long-haul T1 link, the command is rejected. To set a cable length longer than 600 feet for a T1 link, use the cablelength long command. The keywords are as follows:
To set a cable length 600 feet or less for a T1 link, use the cablelength short command. There is no default for cablelength short. The keywords are as follows:
If you do not set the cable length, the system defaults to a setting of cablelength long 0db. | ||
Step8 | Router(config-controller)# ds0-group ds0-group-no
timeslots timeslot-list type {e&m-immediate |
e&m-delay | e&m-wink | fxs-ground-start |
fxs-loop-start | fxo-ground-start | fxo-loop-start}
| Defines the T1 channels for use by compressed voice calls and the signalling method the router uses to connect to the PBX or CO. You should set up DS0 groups after you have specified codec complexity in voice-card configuration mode. If you modify the codec complexity command parameters, you must first remove any existing DS0 groups, then reinstate them after the change to the codec complexity. The ds0-group-no argument identifies the DS0 group and is a value from 0 to 23.
The timeslot-list argument indicates a range of time slots and is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen. For T1 or E1, allowable values are from 1 to 24. To map individual DS0 time slots, define additional groups. The system maps additional voice ports for each defined group. The signalling method selection for the type keyword depends on the connection that you are making:
| ||
Step9 | Router(config-controller)# tdm-group tdm-group-no timeslots timeslot-list type [e&m | fxs [loop-start | ground-start] fxo [loop-start | ground-start]] | (Optional) Configures TDM channel groups for the drop-and-insert (also called TDM Cross-Connect) function with a 2-port T1 multiflex trunk interface card. The tdm-group-no argument identifies the channel group and is a value from 1 to 31. The timeslot-list argument indicates a range of time slots and is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen. For T1, allowable values are from 1 to 24. The signalling method selection for the type keyword depends on the connection that you are making. The fxs and fxo options allow you to specify a ground-start or loop-start line.
| ||
Step10 | Router(config-controller)# no shutdown | Activates the controller. | ||
Step11 | Router(config-controller)# exit | Exits controller configuration mode. | ||
Step12 | Router(config)# connect id T1 slot/port tdm-group-no-1 T1 slot/port tdm-group-no-2 | (Optional) Sets up the connection between two T1 TDM groups of time slots on the trunk interfaces---for the drop-and-insert capability. The id argument specifies the name for the connection. The slot/port argument identifies each T1/E1 controller by its location. Valid values for slot and port are 0 and 1. The tdm-group-no-1 and tdm-group-no-2 arguments identify the TDM group numbers (from 1 to 31) on the specified controller. |
Repeat Steps 2 and 3 for each card type.
Repeat Steps 4 through 12 for each controller.
To specify codec settings for card types and set up E1 controllers for clocking and other E1 parameters, as well as for DS0 groups that define the channels for compressed voice and TDM groups for drop-and-insert capability, use the following commands beginning in privileged EXEC mode:
| Command | Purpose | |||
|---|---|---|---|---|
Step1 | Router# configure terminal | Enters global configuration mode. | ||
Step2 | Router(config)# card type {t1/e1} slot
| Enters E1 card type and specifies the slot location by using a value from 0 to 5, depending upon your router. | ||
Step3 | Router(config-voice-ca)# codec {high | medium | low}
1-30
| Specifies the codec complexity based on the codec standard you are using. High-complexity codecs support lower call density than do medium-complexity codecs. For example:
The keyword that you specify for codec affects the choice of codecs available using the codec dial-peer configuration command.
| ||
Step4 | Router(config)# controller E1 slot/port | Enters controller configuration mode for the E1 controller at the specified slot/port location. Valid values for the slot and port arguments are 0 and 1. | ||
Step5 | Router(config-controller)# clock source {line
[primary] | internal}
| Configures controller E1 1/0 to specify the clock source. The line keyword specifies that the clock source is derived from the active line---rather than from the free-running internal clock. This is the default setting and is generally more reliable. These rules apply to clock sourcing on the E1 controller ports:
| ||
Step6 | Router(config-controller)# | Sets the framing according to the instructions from your service provider. Choose CRC4 format or No CRC4 format. | ||
Step7 | Router(config-controller)# | Sets the line encoding according to the instructions from your service provider. | ||
Step8 | Router(config-controller)# ds0-group ds0-group-no
timeslots timeslot-list type {e&m-immediate |
e&m-delay | e&m-wink | fxs-ground-start |
fxs-loop-start | fxo-ground-start | fxo-loop-start}
| Defines the E1 channels for use by compressed voice calls and the signalling method the router uses to connect to the PBX or CO. You should set up DS0 groups after you have specified codec complexity in voice-card configuration. If you modify the codec complexity command parameters, you must first remove any existing DS0 groups, then reinstate them after the change to the codec complexity. The ds0-group-no argument identifies the DS0 group and is a value from 0 to 23.
The timeslot-list argument indicate a range of time slots and is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen. For T1 or E1, allowable values are from 1 to 24. To map individual DS0 time slots, define additional groups. The system maps additional voice ports for each defined group. The signalling method selection for the type keyword depends on the connection that you are making:
| ||
Step9 | Router(config-controller)# tdm-group tdm-group-no timeslots timeslot-list type [e&m | fxs [loop-start | ground-start] fxo [loop-start | ground-start]] | (Optional) Configures TDM channel groups for the drop-and-insert (also called TDM Cross-Connect) function with a 2-port T1/E1 multiflex trunk interface card. The tdm-group-no argument identifies the channel group and is a value from 1 to 31. The timeslot-list argument indicates a range of time slots and is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen. For T1 or E1, allowable values are from 1 to 24. The signalling method selection for the type keyword depends on the connection that you are making. The fxs and fxo options allow you to specify a ground-start or loop-start line.
| ||
Step10 | Router(config-controller)# no shutdown | Activates the controller. | ||
Step11 | Router(config-controller)# exit | Exits controller configuration mode. | ||
Step12 | Router(config)# connect id E1 slot/port tdm-group-no-1 E1 slot/port tdm-group-no-2 | (Optional) Sets up the connection between two T1/E1 TDM groups of time slots on the trunk interfaces for the drop-and-insert capability. The id argument specifies a name for the connection. The slot/port argument identifies each E1 controller by its location. Valid values for slot and port are 0 and1. The tdm-group-no-1 and tdm-group-no-2 arguments identify the TDM group numbers (from 1 to 31) on the specified controller. |
Repeat Steps 2 and 3 for each card type.
Repeat Steps 4 through 12 for each controller.
To set up voice ports to support the local and remote stations, use the following commands beginning in privileged EXEC mode:
| Command | Purpose | |||
|---|---|---|---|---|
Step1 | Router# configure terminal | Enters global configuration mode. | ||
Step2 | Router(config)# voice-port slot/port:ds0-group-no | Enters voice-port configuration mode. The slot argument is the router location where the voice port adapter is installed. Valid entries are from 0 to 3. The port argument indicates the VIC location. Valid entries are 0 or 1. Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1/E1 card. | ||
Step3 | Router(config-voice-port)# busyout monitor interface interface number | (Optional) Specifies a LAN or WAN interface that will be monitored, and, when it is down, trigger a busyout (off-hook) state on the voice port. This allows rerouting of calls. For example, if you specify Serial 1/0 as the interface and number, the voice port sends a busyout signal when that interface is down. You can issue the command repeatedly to specify as many interfaces, virtual interfaces, and subinterfaces as are required for a voice port. For example, if you issue the command three times so that three interfaces are monitored, the voice port only goes into busyout state when all three interfaces are down. When any one of the interfaces is operational, the busyout state is removed. | ||
Step4 | Router(config-voice-port)# comfort-noise | (Optional) Creates subtle background noise to fill silent gaps during calls when VAD is enabled on voice dial peers. (This parameter is enabled by default.) If comfort noise is not generated, the resulting silence can fool the caller into thinking the call is disconnected instead of being merely idle. | ||
Step5 | Router(config-voice-port)# echo-cancel enable | (Optional) Enables echo cancellation. (This setting is enabled by default.) Echo cancellation adds to the quality of voice transmissions by adjusting the echo that occurs on the interface due to impedance mismatches. Some echo is reassuring; echo over 25milliseconds long can cause problems. | ||
Step6 | Router(config-voice-port)# echo-cancel coverage {16
| 24 |32 | 8}
| (Optional) Adjusts the echo canceller by the specified number of milliseconds; the default is 16. | ||
Step7 | Router(config-voice-port)# connection {plar |trunk}
string
| (Optional) Sets up a connection mode for the voice port. The plar keyword specifies a PLAR connection, which rings a remote telephone when the dial peer goes off-hook. The trunk keyword specifies a straight tie-line connection to a PBX. The string argument specifies the remote telephone number or significant start digits of the number. | ||
Step8 | Router(config-voice-port)# timeouts interdigit seconds | (Optional) Sets the number of seconds the system waits---after the caller has input the initial digit---for a subsequent digit of the dialed string. If the timeout ends before the destination is identified, a tone sounds and the call ends. The default value is 10seconds, and the timeout can be set from 0 to 120seconds.
| ||
Step9 | Router(config-voice-port)# exit | Exits voice-port configuration mode. |
Repeat Steps 2 through 9 for each DS0 group you create
You can check the validity of your T1/E1 high-capacity digital voice port configuration by performing the following tasks:
With ISDN PRI, signalling in VoIP for the Cisco AS5300 and AS5800 is handled by ISDN PRI group configuration. After ISDN PRI has been configured for both B and D channels for both ISDN PRI lines, you need to enter the isdn incoming-voice command on the serial interface (acting as the D channel) to ensure a dial tone.
Under most circumstances, the default voice-port command values are adequate to configure voice ports to transport voice data over your existing IP network. Because of the inherent complexities involved with PBX networks, you might need specific voice-port values configured, depending on the specifications of the devices in your telephony network.
To configure basic ISDN PRI parameters for the Cisco AS5300 or Cisco AS5800 access servers, use the following commands beginning in global configuration mode:
| Command | Purpose | |
|---|---|---|
Step1 | Router(config)#isdn switch-type switch-type | Defines the telephone company switch type. |
Step2 | Router(config)#controller T1 1/0/0 | Enables the T1 0 controller on the T1 card and enters controller configuration mode. or Enables the T1 1 controller on the T3 card and enters controller configuration mode. |
Step3 | Router(config-controller)# | Defines the framing characteristics. |
Step4 | Router(config-controller)# | Sets the line-code type to match that of your telephone company service provider. |
Step5 | Router(config-controller)# | Configures ISDN PRI. |
Step6 | Router(config-controller)# | Enables the T1 1 on the T1 card controller (CiscoAS5800). or Enables the T1 2 controller on the T3 card (CiscoAS5800). or Enables the T1 0 controller (Cisco AS5300). |
Step7 | Router(config-controller)# | Defines the framing characteristics. |
Step8 | Router(config-controller)# | Sets the line-code type to match that of your telephone company service provider. |
Step9 | Router(config-controller)# | Configures ISDN PRI. |
Step10 | Router(config-controller)# exit | Exits controller configuration mode. |
Step11 | Router(config)# | Configures the channel for the first ISDN PRI line on the T1 card. (The ISDN serial interface is the D channel.) (Cisco AS5800) or Configures the channel for the first ISDN PRI line on the T3 card. (The serial interface is the D channel.) (CiscoAS5800) or Configures the channel for the first ISDN PRI line. (The serial interface is the D channel.) (CiscoAS5300) |
Step12 | Router(config-if)# | Enables incoming ISDN voice calls. This command has two possible keywords: data and modem. You must use the modem keyword to enable voice calls. The modem keyword represents bearer capabilities of speech. |
Step13 | Router(config-if)# | Configures the channel for the second ISDN PRI line on the T1 card (Cisco AS5800). or Configures the channel for the second ISDN PRI line on the T3 card (Cisco AS5800). or Configures the channel for the second ISDN PRI line (CiscoAS5300). |
Step14 | Router(config-if)# | Enables incoming ISDN voice calls. This command has two possible keywords: data and modem. You must use the modem keyword to enable voice calls. The modem keyword represents bearer capabilities of speech. |
Step15 | Router(config-if)# exit | Exits interface configuration mode. |
As mentioned, under most circumstances, the default voice-port command values are adequate to configure voice ports to transport voice data over your existing IP network. To configure specific voice port parameters, use the following commands beginning in privileged EXEC mode:
| Command | Purpose | |
|---|---|---|
Step1 | Router#configure terminal | Enters global configuration mode. |
Step2 | Router(config)#voice-port {shelf/slot/port:D} |
{shelf/slot/parent:port:D}
| Identifies the voice port you want to configure and enters voice-port configuration mode (CiscoAS5800) or Identifies the voice port you want to configure and enters voice-port configuration mode (CiscoAS5300). |
Step3 | Router(config-voiceport)#cptone country | Selects the appropriate voice call progress tone for this interface. The default for this command is us. For a list of supported countries, refer to the Cisco IOS Multiservice Applications Command Reference publication. |
Step4 | Router(config-voiceport)#compand-type {a-law |
u-law}
| Selects a companding type for this voice port. |
Step5 | Router(config-voiceport)#connection {plar string |
trunk string}
| (Optional) Specifies either the trunk connection or the PLAR connection. The string argument specifies the destination telephone number. |
Step6 | Router(config-voiceport)#music-threshold number | (Optional) Specifies the threshold (in decibels) for on-hold music. Valid entries are from -70 to -30. |
Step7 | Router(config-voiceport)#description string | (Optional) Attaches descriptive text about this voice-port connection. |
Step8 | Router(config-voiceport)#input gain value | Specifies (in decibels) the amount of gain to be inserted at the receiver side of the interface. Acceptable values are from -6 to 14. |
Step9 | Router(config-voiceport)#output attenuation value | Specifies (in decibels) the amount of attenuation at the transmit side of the interface. Acceptable values are from 0 to 14. |
Step10 | Router(config-voiceport)#echo-canel enable | Enables echo cancellation of voice that is sent out the interface and received back on the same interface. |
Step11 | Router(config-voiceport)#echo-canel coverage value | Adjusts the size (in milliseconds) of the echo cancellation. Acceptable values are 16, 24, and 32. |
Step12 | Router(config-voiceport)#non-linear | Enables nonlinear processing, which shuts off any signal if no near-end speech is detected. (Nonlinear processing is used with echo cancellation.) |
Step13 | Router(config-voiceport)#playout-delay {maximum
milliseconds | nominal milliseconds}
| Specifies the amount of time in milliseconds configured for the playout delay buffer. |
Step14 | Router(config-voiceport)#timeouts initial seconds | Specifies the number of seconds the system will wait for the caller to input the first digit of the dialed digits. Valid entries for this command are from 0to120. |
Step15 | Router(config-voiceport)#timeouts interdigits seconds | Specifies the number of seconds the system will wait (after the caller has input the initial digit) for the caller to input a subsequent digit. Valid entries for this command are from 0 to 120. |
Step16 | Router(config-voiceport)#timeouts ringing {seconds
| infinity}
| Specifies the number of seconds the system will continue to ring the destination if there is no answer. |
Step17 | Router(config-voiceport)#timeouts wait-release
{seconds | infinity}
| Specifies the wait release timeout duration in seconds. |
Step18 | Router(config-voiceport)#translate {called number |
calling number}
| Defines translation rules pertaining to either the called or calling numbers. |
Step19 | Router(config-voiceport)# exit | Exits voice-port configuration mode. |
For more information on specific voice-port configuration commands or additional voice-port commands, refer to the Cisco IOS Multiservice Applications Command Reference publication.
You can check the validity of your voice port configuration by performing the following tasks:
The VoIP VNM for the Cisco AS5300 supports E1 R2 signalling and ISDN PRI. R2 signalling is an international signalling standard that is common to channelized E1 networks. However, there is no single signalling standard for R2. The ITU-T Q.400-Q.490 recommendation defines R2, but a number of countries and geographic regions implement R2 in entirely different ways. Cisco addresses this lack of standards by supporting many localized implementations of R2 signalling in its Cisco IOS software.
The Cisco E1 R2 signalling default is ITU, which supports the technology used in the following countries: Denmark, Finland, Germany, Russia (ITU variant), Hong Kong (ITU variant), and South Africa (ITU variant). The expression "ITU variant" means there are multiple R2 signalling types in the specified country, but Cisco supports the ITU variant.
Cisco also supports specific local variants of E1 R2 signalling in the following regions, countries, and corporations:
Of the local variants listed, the following local variants have been verified:
R2 signalling is channelized E1 signalling used in Europe, Asia, and South America. It is equivalent to channelized T1 signalling in North America. There are two types of R2 signalling: line signalling and interregister signalling. R2 line signalling includes R2 digital, R2 analog, and R2 pulse. R2 interregister signalling includes R2 compelled, R2 noncompelled, and R2 semicompelled. These signalling types are configured using the cas-group command.
Many countries and regions have their own E1 R2 variant specifications, which supplement the ITU-T Q.400-Q.490 recommendation for R2 signalling. Unique E1 R2 signalling parameters for specific countries and regions are set by entering the cas-custom command followed by the country command.
The Cisco implementation of R2 signalling has dialed number identification service (DNIS) support turned on by default. If you enable the automatic number identification (ani) option, the collection of DNIS information is still performed. Specifying the ani option does not disable DNIS collection. DNIS is the number being called. ANI is the number of the caller. For example, if you are configuring router A to call router B, then the DNIS number is assigned to router B; the ANI number is assigned to router A. ANI is similar to caller ID.
To configure E1 R2 signalling, use the following commands beginning in global configuration mode:
| Command | Purpose | |
|---|---|---|
Step1 | Router(config)# controller e1 number | Specifies the E1 controller that you want to configure with R2 signalling. |
Step2 | Router(config-controller)# cas-group channel
timeslots range type {r2-analog | r2-digital |
r2-pulse} [dtmf | r2-compelled [ani] |
r2-non-compelled [ani] | r2-semi-compelled [ani]]
| Configures R2 CAS on the E1 controller. For a complete description of the available R2 options, refer to the cas-group (controller e1) command in the Cisco IOS Dial Services Command Reference publication. |
Step3 | Router(config-controller)# cas-custom channel | Enters cas-custom configuration mode. In this mode, you can localize E1R2 signalling parameters, such as specific R2 country settings for Hong Kong. For the customization to take effect, the number used for the channel argument in the cas-custom command must match the channel number specified by the cas-group command. |
Step4 | Router(config-controller)# country name use-default | Specifies the local country, region, or corporation specification to use with R2 signalling. Replace the name argument with one of the supported country names. Refer to the cas-custom command in the Cisco IOS Dial Services Command Reference publication for the list of supported regions, countries, or corporation specifications. We strongly recommend that you include the use-defaults option, which engages the default settings for a specific country. The default setting for all countries is ITU. |
Step5 | ani-digitsanswer-signal caller-digits category default dnis-digits invert-abcd ka kd metering nc-congestion unused-abcd request-category | (Optional) Further customizes the R2 signalling parameters. Some switch types require you to fine-tune your R2 settings. Do not tamper with these commands unless you fully understand the requirements of your switch. For nearly all network scenarios, the country name use-defaults command fully configures the local settings for your country. You should not need to perform Step 5. Refer to the cas-custom command in the Cisco IOS Dial Services Command Reference publication for more information about each signalling command. |
Step6 | Router(config-controller)# exit | Exits controller configuration mode. |
Step7 | Router(config)# voice-port controller-number:channel-number | Enters voice-port configuration mode for the specified voice port. |
Step8 | Router(config-voice-port)# cptone country-code | Defines the country-specific pulse code modulation (PCM) encoding and tones. The PCM encoding type must match the country code defined by the cas-custom command. |
Step9 | Router(config-voice-port)# exit | Exits voice-port configuration mode. |
Step10 | Router(config)# exit | Exits global configuration mode. |
The E1 R2 signalling type (whether ITU, ITU variant, or local variant as defined by the cas-custom command) needs to match the appropriate PCM encoding type as defined by the cptone command. For countries for which a cptone value has not yet been defined, you can try the following:
For more information about configuring R2 signalling, refer to the Cisco IOS DialServices Configuration Guide: Terminal Services and the Cisco IOS DialServices Configuration Guide: Network Services publications.
You can check the validity of your E1 R2 signalling configuration by performing the following tasks:
If the connection does not come up, check for the following:
If you see errors on the line or the line is going up and down, check for the following:
CAS is the transmission of signalling information within the voice channel. Various types of CAS signalling are available in the T1 world. The most common forms of CAS signalling are loop-start, ground-start, and E&M. The main disadvantage of CAS signalling is its use of user bandwidth to perform signalling functions. CAS signalling is often referred to as robbed-bit signalling because user bandwidth is being "robbed" by the network for other purposes. In addition to receiving and placing calls, CAS signalling processes the receipt of DNIS and automatic number identification (ANI) information, which is used to support authentication and other functions.
T1 CAS capabilities have been implemented on the Cisco AS5300 VFC to enhance and integrate T1 CAS capabilities on common 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 CiscoAS5300 to the end office switch. In this configuration, the CiscoAS5300 captures the dialed-number or called-party number information and passes it along to the upper level applications for interactive voice response (IVR) script selection, modem pooling, and other applications. Service providers also require access to calling party number, ANI, for user identification, for billing account number, and in the future, for more complicated call routing.
Service providers that implement VoIP include traditional voice carriers, new voice and data carriers, and existing ISPs. Some of these service providers might use subscriber side lines for their VoIP connectivity to the PSTN; others might use tandem-type service provider connections.
Voice over IP for the AS5300 supports the following T1 CAS signalling systems:
ISPs can provide switched 56-kbps access to their customers using the CiscoAS5300. The subset of T1 CAS (robbed bit) supported features are as follows:
To configure T1 CAS for VoIP on the Cisco AS5300, use the following commands beginning in privileged EXEC mode:
| Command | Purpose | |
|---|---|---|
Step1 | Router#configure terminal | Enters global configuration mode. |
Step2 | Router(config)#controller t1 number | Enters controller configuration mode to configure your controller port. The controller ports are labeled 0 through 3 on the Quad T1/PRI and E1/PRI cards. |
Step3 | Router(config-controller)# | Specifies the framing type designated by your telephone company. |
Step4 | Router(config-controller)# | Configures the primary PRI clock source. Configure other lines as secondary or internal clock sources. Note that only one PRI can be clock source primary and one PRI can be clock source secondary. |
Step5 | Router(config-controller)# | Specifies the line-code type designated by your telephone company. |
Step6 | Router(config-controller)# | Configures all channels for E&M, FXS, and SAS analog signalling. Enter 1-24 for T1. If E1, enter 1-31.
You must use the same type of signalling that your CO uses. For E1 using the Anadigicom converter, use cas e&m-fgb signalling. |
Step7 | Router(config-controller)#controller t1 number | Enters controller configuration mode to configure the |
Step8 | Router(config-controller)# | Specifies the framing type designated by your telephone company. |
Step9 | Router(config-controller)# | Configures the secondary PRI clock source. Note that only one PRI can be clock source primary and one PRI can be clock source secondary. |
Step10 | Router(config-controller)# | Specifies the line-code type designated by your telephone company. |
Step11 | Router(config-controller)# | Configures all channels for E&M, FXS, and SAS analog signalling. Enter 1-24 for T1. If E1, enter 1-31. Signalling types for the signal argument include e&m-fgb, e&m-fgd, e&m-immediate-start, fxs-ground-start, fxs-loop-start, sas-ground-start, and sas-loop-start. You must use the same type of signalling that your CO uses. For E1 using the Anadigicom converter, use cas e&m-fgb signalling. |
Step12 | Router(config-controller)#controller t1 number | Enters controller configuration mode to configure the third controller port (there are a total of four controller ports). The controller ports are labeled 0 through 3 on the Quad T1/PRI and E1/PRI cards. |
Step13 | Router(config-controller)# | Specifies the framing type designated by your telephone company. |
Step14 | Router(config-controller)# | Configures the internal PRI clock source. Note that only one PRI can be clock source primary and one PRI can be clock source secondary. All other controller ports use an internal PRI clock source. |
Step15 | Router(config-controller)# | Specifies the line-code type designated by your telephone company. |
Step16 | Router(config-controller)# cas-group channel timeslots rangetype signal | Configures all channels for E&M, FXS, and SAS analog signalling. Enter 1-24 for T1. If E1, enter 1-31. Signalling types for the signal argument include e&m-fgb, e&m-fgd, e&m-immediate-start, fxs-ground-start, fxs-loop-start, sas-ground-start, and sas-loop-start. You must use the same type of signalling that your CO uses. For E1 using the Anadigicom converter, use cas e&m-fgb signalling. |
Repeat Steps 12 through 16 to configure the last controller.
You can check the validity of your T1 CAS configuration by entering the show controller t1 or show controller e1 command and specify the port number.
Make sure the show controller t1 output is not reporting alarms or violations.
The Busyout Monitor feature is one aspect of Call Admission Control (CAC) that allows network administrators to use both a data network and the PSTN to provide the best possible quality for VoIP calls. Although voice calls are routed across the data network whenever possible to take advantage of the cost savings provided by integrated applications, the Busyout Monitor allows network administrators to provide voice services through the PSTN in the event of a network interface failure.
If a locally connected LAN or WAN interface on a VoIP gateway fails, it busies out voice ports, which means that a connected PBX or key system reroutes the call through the local PSTN.
The Busyout Monitor CAC feature provides the following benefits:
Busyout Monitor has the following restriction: Busyout Monitor monitors only locally connected LAN/WAN interfaces and does not monitor the status of remote devices. The feature cannot determine the status of the end-to-end path.
|
NoteIn some cases, for example, in a VoIP over Frame Relay environment, you can use the Frame Relay PVC end-to-end keepalive feature to track the end-to-end path and thereby busy out a port when its corresponding PVC is down. For more information about Frame Relay keepalive, refer to the Cisco IOS Wide-Area Networking Command Reference and the Cisco IOS Wide-Area Networking Configuration Guide publications. |
To configure Busyout Monitor, use the following commands beginning in privileged EXEC mode:
| Command | Purpose | |||
|---|---|---|---|---|
Step1 | Router# configure terminal | Enters global configuration mode. | ||
Step2 | Router(config)# voice-port slot/port:ds0-group-no | Enters voice-port configuration mode (Cisco2600/3600series). or Enters voice-port configuration mode (CiscoAS5300). or Enters voice-port configuration mode (CiscoAS5800). or Enters voice-port configuration mode (Cisco 7200 series).
| ||
Step3 | Router(config-voice-port)# busyout monitor interface interface number | (Optional) Allows you to specify a LAN or WAN interface that will be monitored, and, when it is down, triggers a busyout (off-hook) state on the voice port. This allows rerouting of calls. For example, if you specify Serial 1/0 as the interface and number, the voice port sends a busyout signal when that interface is down. You can issue the command repeatedly to specify as many interfaces, virtual interfaces, and subinterfaces as are required for a voice port. For example, if you issue the command three times so that three interfaces are monitored, the voice port only goes into busyout state when all three interfaces are down. When any one of the interfaces is operational, the busyout state is removed. | ||
Step4 | Router(config-voice-port)# exit | Exits voice-port configuration mode. |
|
NoteRepeat this procedure for each DS0 group that you create. |
To activate a voice port, use the following command in voice-port configuration mode:
| Command | Purpose |
|---|---|
Router(config-voiceport)#no shutdown | Activates the voice port. |
|
NoteIf you will not use a voice port, shut it down. |
This section contains the following configuration examples:
This section includes the following configuration examples:
These examples are not necessarily complete configurations. They are designed to illustrate specific tips and techniques, and only the relevant portions of the configurations are shown. Each configuration includes a brief introduction, side-by-side configurations for routers at either end, and explanations of key points.
Figure 20 shows how to set up a Cisco 2600 or 3600 router to collect digits from either a PBX/PSTN or a phone and route a VoIP call based on the digits received. The commands used in the configurations are explained inline. Only relevant sections of the configuration are shown. The example assumes that the IP portion of the network is already in place.
| Alpha Router | Beta Router |
hostname router-alpha ! voice-card 1 codec complexity high ! dial-peer voice 1 voip codec g723r53 fax-rate 14400 destination-pattern 5.... session target ipv4:192.168.100.1 ! dial-peer voice 2 pots destination-pattern 4.... prefix 4 port 1/0:1 ! controller T1 1/0 framing esf linecode b8zs clock source line ds0-group 1 timeslots 1-24 type e&m-wink ! interface serial 0/0 ip address 192.168.100.2 255.255.255.0 | hostname router-beta ! voice-card 1 codec complexity high ! dial-peer voice 1 voip codec g723r53 fax-rate 14400 destination-pattern 4.... session-target ipv4:192.168.100.2 ! dial-peer voice 2 pots destination-pattern 5.... prefix 5 port 1/0:1 ! controller T1 1/0 framing esf linecode b8zs clock source internal ds0-group 1 timeslot 1-24 type e&m-wink ! interface s0/0 ip address 192.168.100.1 255.255.255.0 |
In this configuration, the PBX seizes the T1 to the router, which expects to collect digits from the PBX. Upon collecting those digits, the router tries to match a dial peer to route the call. If the router receives the correct digits, it routes the call according to the configuration of the dial peer.
Here are some key points for consideration:
Figure 21 shows how to configure a Cisco 2600 or 3600 router to support FRF.12 fragmentation and queueing in a VoIP over Frame Relay network. FRF.12 is a Frame Relay Forum standard mechanism for fragmenting data packets. This fragmentation helps eliminate the delays that occur when sending voice and data over the same network---large data packets can delay smaller voice packets from being sent into the IP network. FRF.12 is also supported on the MC3810 and 7200 routers, which can be used as tandem nodes for VoIP networks.
|
NoteThis example shows VoIP over Frame Relay, which is not the same as VoFR. For more information about VoFR, see the Cisco IOS Release12.0(4)T feature module Voice over Frame Relay Using FRF.11 and FRF.12 . |
The following configuration fragments both the IP and IPX data traffic to 80 bytes, allowing the VoIP traffic to be only minimally delayed on the network. The FRF.12 setup also traffic-shapes the output traffic rate to match the provisioned CIR from the Frame Relay carrier. Matching the provisioned CIR from the Frame Relay carrier ensures that traffic is not dropped or delayed within the Frame Relay network.
Here are some key points for consideration:
| Alpha Router | Beta Router |
hostname router-alpha ! ipx routing ! voice-card 1 codec complexity high ! controller T1 1/0 framing esf linecode b8zs clock source line ds0-group 1 timeslot 1-24 type e&m-wink ! dial-peer voice 1 voip dtmf-relay h245-alpha codec g723r53 destination-pattern 5.... session target ipv4:192.168.100.2 ! dial-peer voice 2 pots destination-pattern 4.... prefix 4 port 1/0:1 ! interface serial 0/0 encapsulation frame-relay frame-relay traffic-shaping ! interface serial 0/0.1 point-to-point ip address 192.168.100.1 255.255.255.0 ipx network ABCD frame-relay interface-dlci 100 class cisco_frf12 ! map-class frame-relay cisco_frf12 frame-relay voice bandwidth 42000 frame-relay fragment 80 no frame-relay adaptive-shaping frame-relay cir 32000 frame-relay bc 1000 frame-relay mincir 64000 frame-relay fair-queue | hostname router-beta ! ipx routing ! voice-card 1 codec complexity high ! controller T1 1/0 framing esf linecode b8zs clock source line ds0-group 1 timeslot 1-24 type e&m-wink ! dial-peer voice 1 voip dtmf-relay h245-alpha codec g723r53 destination-pattern 4.... session target ipv4:192.168.100.2 ! dial-peer voice 2 pots destination-pattern 5.... prefix 5 port 1/0:1 ! interface serial 0/0 encapsulation frame-relay frame-relay traffic-shaping ! interface serial 0/0.1 point-to-point ip address 192.168.100.2 255.255.255.0 ipx network ABCD frame-relay interface-dlci 101 class cisco_frf12 ! map-class frame-relay cisco_frf12 frame-relay voice bandwidth 42000 frame-relay fragment 80 no frame-relay adaptive-shaping frame-relay cir 64000 frame-relay bc 1000 frame-relay mincir 64000 frame-relay fair-queue |
|
NoteWith the introduction of Cisco IOS Release 12.0(5)T and subsequent releases, Cisco VoIP gateways support H.323v2 (H.323 Version 2), which is backwards compatible with systems running H.323 Version 1. However, H.323 Version 2 features do not interoperate with H.323 Version 1 features in Cisco IOS releases prior to 11.3(9)NA or 12.0(3)T. Earlier Cisco IOS versions contain H.323 Version 1 software that does not support protocol messages with an H.323 Version 2 protocol identifier. All systems must be running either Cisco IOS Release 11.3(9)NA and later or Cisco IOS Release 12.0(3)T and later releases to interoperate with H.323 Version 2. Gateway Resource Availability Indication (RAI) messages are currently not supported on the Cisco 2600 and 3600 series. (These are messages that are sent to the Gatekeeper to inform it about the status of a Gateway DSP or DS0 availability.) |
Figure 22 shows how to configure a Cisco 2600 or 3600 series router to route VoIP calls through an H.323 gatekeeper. This setup shows calls being routed from a gateway in Zone-Alpha, through the gatekeeper, to a gateway in Zone-Beta.
| Gatekeeper | |
hostname router-gatekeeper ! gatekeeper zone local alpha alpha.com zone local beta beta.com no use-proxy alpha.com remote-zone beta.com no use-proxy beta.com remote-zone alpha.com zone prefix router-alpha 4.... zone prefix router-beta 5.... no shutdown ! interface ethernet 0/0 ip address 10.1.1.3 255.255.255.0 | |
| Alpha Router | Beta Router |
hostname router-alpha ! voice-card 1 ! controller T1 1/0 framing esf linecode b8zs clock source internal ds0-group 1 timeslot 1-24 type e&m-wink ! voice-port 1/0:1 ! dial-peer voice 1 voip dtmf-relay h245-alpha destination-pattern 5.... tech-prefix 1# session target ras ! dial-peer voice 2 pots destination-pattern 4.... prefix 4 port 1/0:1 ! gateway ! interface ethernet 0/0 ip address 10.1.1.1 255.255.255.0 h323-gateway voip interface h323-gateway voip id alpha ipaddr 10.1.1.3 1719 h323-gateway voip h323-id router-alpha@alpha.com h323-gateway voip tech-prefix 1# | hostname router-beta ! voice-card 1 ! controller T1 1/0 framing esf linecode b8zs clock source line ds0-group 1 timeslot 10-24 type e&m-wink ! voice-port 1/0:1 ! dial-peer voice 1 voip dtmf-relay h245-alpha destination-pattern 4.... tech-prefix 1# session target ras ! dial-peer voice 2 pots destination-pattern 5.... prefix 5 port 1/0:1 ! gateway ! interface ethernet 0/0 ip address 10.1.1.2 255.255.255.0 h323-gateway voip interface h323-gateway voip id beta ipaddr 10.1.1.3 1719 h323-gateway voip h323-id router-beta@beta.com h323-gateway voip tech-prefix 1# |
Here are some key points for consideration:
Figure 23 shows how to set up a Cisco 2600 or 3600 series router for a PLAR. PLAR is used to allow a station or DS0 to go off hook, and---without the user dialing digits---have a call completed to the far end. PLAR can also provide dial tone from a remote PBX for off-premises applications.
In this configuration, the phones off router Beta go off hook and receive dial tone from the PBX connected to router Alpha. From there, users can dial any digits in to the PBX as if their stations are directly connected to it.
Here are some key points for consideration:
| Alpha Router | Beta Router |
hostname router-alpha ! voice-card 1 ! ! controller T1 1/0 framing esf linecode b8zs ds0-group 1 timeslot 1 type fxo-loop ds0-group 2 timeslot 2 type fxo-loop ! dial-peer voice 1 voip dtmf-relay h245-alpha codec g729a destination-pattern 2.. session target ipv4:192.168.100.2 ! dial-peer voice 2 pots destination-pattern 101 port 1/0:1 ! dial-peer voice 3 pots destination-pattern 102 port 1/0:2 ! voice-port 1/0:1 connection plar 201 ! voice-port 1/0:2 connection plar 202 ! interface s0/0 ip address 192.168.100.1 255.255.255.0 | hostname router-beta ! dial-peer voice 1 voip destination-pattern 1.. dtmf-relay h245-alpha codec g729a session target ipv4:192.168.100.1 ! dial-peer voice 2 pots destination-pattern 201 port 1/1 ! ! dial-peer voice 3 pots destination-pattern 202 port 1/2 ! voice-port 1/1 ! ! voice-port 1 / 2 ! ! interface serial 0/0 ip address 192.168.100.2 255.255.255.0 |
Figure 24 shows how to configure a Cisco 2600 or 3600 router for a trunk connection. A trunk connection is like a "wire" between the two routers. It is a transparent connection, so it allows features such as hookflash (also called switchhook flash) or "hoot n' holler" (point-to-point) to pass. This type of trunk configuration can also be used for OPXs that require rollover to a centralized voice-mail system when the user does not answer.
A trunk connection can only be used between E&M ports or with FXO-to-FXS connections.
| Alpha Router | Beta Router |
hostname router-alpha ! voice-card 1 ! controller T1 1/0 framing esf linecode b8zs ds0-group 1 timeslot 1 type e&m-wink ds0-group 2 timeslot 2 type e&m-wink clock source line ! voice-port 1/0:1 connection trunk 1111 ! voice-port 1/0:2 connection trunk 1112 ! dial-peer voice 1 voip dtmf-relay h245-alpha codec g729a destination-pattern 111. session target ipv4:192.168.100.2 ! dial-peer voice 2 pots destination-pattern 2221 port 1/0:1 ! dial-peer voice 3 pots destination-pattern 2222 port 1/0:2 ! interface serial 0/0 ip address 192.168.100.1 255.255.255.0 | hostname router-beta ! voice-card 1 ! controller T1 1/0 framing esf linecode b8zs ds0-group 1 timeslot 1 type e&m-wink ds0-group 2 timeslot 2 type e&m-wink clock source line ! voice-port 1/0:1 connection trunk 2221 ! voice-port 1/0:2 connection trunk 2222 ! dial-peer voice 1 voip dtmf-relay h245-alpha codec g729a destination-pattern 222. session target ipv4:192.168.100.1 ! dial-peer voice 2 pots destination-pattern 1111 port 1/0:1 ! dial-peer voice 3 pots destination-pattern 1112 port 1/0:2 ! interface serial 0/0 ip address 192.168.100.2 255.255.255.0 |
In this configuration, a permanent and transparent path is set up between individual DS0s on each router. It passes dial tone from the remote PBX and passes DTMF digits out of band.
The connection trunk command establishes the permanent trunk connection between the routers. The digits after the command are passed internally within the router to match a dial peer so that the call can be set up.
Figure 25 shows an example of drop-and-insert. Drop-and-insert technology is one way to integrate old PBX technologies with VoIP. It allows you to take 64-Kbps DS0 channels from one T1 and digitally cross-connect them to 64-Kbps DS0 channels on another T1. Drop-and-insert is sometimes called TDM cross-connect.
Drop-and-insert allows individual 64-Kbps DS0 channels to be transparently passed, uncompressed, between T1 ports without passing through a DSP. Using this method, the channel traffic is sent between a PBX and CO switch (PSTN) or other telephony device, allowing the use, for example, of some PBX channels for long-distance service through the PSTN while the router compresses others for interoffice VoIP calls. In addition, drop-and-insert can cross-connect a telephony switch (from the CO or PSTN) to a channel bank to provide external analog connectivity.
Note the following design requirements:
The following configuration example shows how to configure drop-and-insert.
| Router RTR-A | Router RTR-B |
hostname RTR-A ! voice-card 1 codec complexity high ! controller T1 1/0 clock source line framing esf linecoding b8zs ds0-group 1 timeslots 1-12 type e&m-wink tdm-group 2 timeslots 13-24 type e&m ! controller T1 1/1 clock source line primary framing esf linecoding b8zs tdm-group 3 timeslots 13-24 type e&m ! voice-port 1/0:1 ! dial-peer voice 1 voip destination-pattern 4.... codec g723r63 dtmf-relay h245-alpha session target ipv4:192.168.100.2 ! dial-peer voice 2 pots destination-pattern 5.... prefix 5 port 1/0:1 ! interface serial 0/0 encapsulation ppp ip address 192.168.100.1 255.255.255.0 ! connect tdm1 T1 1/0 2 T1 1/1 3 | hostname RTR-B ! voice-card 1 codec complexity high ! controller T1 1/0 clock source line framing esf linecoding b8zs ds0-group 1 timeslots 1-12 type e&m-wink tdm-group 2 timeslots 13-24 type e&m ! controller T1 1/1 clock source line primary framing esf linecoding b8zs tdm-group 3 timeslots 13-24 type e&m ! voice-port 1/0:1 ! dial-peer voice 1 voip destination-pattern 5.... codec g723r63 dtmf-relay h245-alpha session target ipv4:192.168.100.1 ! dial-peer voice 2 pots destination-pattern 4.... prefix 4 port 1/0:1 ! interface serial 0/0 encapsulation ppp ip address 192.168.100.2 255.255.255.0 ! connect tdm1 T1 1/0 2 T1 1/1 3 |
Here are some key points for consideration:
This section includes three sample configurations to illustrate different scenarios:
Drop-and-insert technology is one way to integrate old PBX technologies with VoIP. It allows you to take 64-kbps DS0 channels from one T1 or E1 and digitally cross-connect them to 64-kbps DS0 channels on another T1 or E1.
Drop-and-insert allows individual 64-kbps DS0 channels to be transparently passed, uncompressed, between T1/E1 ports without DSP processing. Channel traffic is sent between a PBX and CO switch or other telephony device, allowing the use, for example, of some PBX channels for long-distance service through the PSTN while the router compresses others for interoffice VoIP calls. In addition, drop-and-insert can cross-connect a telephony switch (from the CO or PSTN) to a channel bank to provide external analog connectivity.
Keep the following considerations in mind:
Figure 26 shows drop-and-insert when a 2-port multiflex VWIC is installed in a digital T1 packet voice trunk network module VWIC slot and VoIP is configured. WAN connections must be provided by separate links.
The following configuration shows the configuration for drop-and-insert when a 2-port Multiflex VWIC is installed in a digital T1 packet voice trunk network module VWIC slot and VoIP is configured.
| Router RTR-A | Router RTR-B |
hostname RTR-A ! voice-card 1 codec complexity high ! controller T1 1/0 framing esf linecoding b8zs ds0-group 1 timeslots 1-12 type e&m-wink tdm-group 2 timeslots 13-24 type e&m ! controller T1 1/1 framing esf linecoding b8zs clock source line primary tdm-group 3 timeslots 13-24 type e&m ! voice-port 1/0:1 ! dial-peer voice 1 voip destination-pattern 4... codec g723r63 dtmf-relay h245-alpha session target ipv4:209.165.200.253 session target ipv4:209.165.200.252 ! dial-peer voice 2 pots destination-pattern 5... prefix 5 port 1/0:1 ! interface serial 0/0 encapsulation ppp ip address 209.165.200.252 255.255.255.224 ! connect tdm1 T1 1/0 2 T1 1/1 3 | hostname RTR-B ! voice-card 1 codec complexity high ! controller T1 1/0 framing esf linecoding b8zs ds0-group 1 timeslots 1-12 type e&m-wink tdm-group 2 timeslots 13-24 type e&m ! controller T1 1/1 framing esf linecoding b8zs clock source line primary tdm-group 3 timeslots 13-24 type e&m ! voice-port 1/0:1 ! dial-peer voice 1 voip destination-pattern 5. codec g723r63 dtmf-relay h245-alpha ! ! ! dial-peer voice 2 pots destination-pattern 4. prefix 4 port 1/0:1 ! interface serial 0/0 encapsulation ppp ip address 209.165.200.253 255.255.255.224 ! connect tdm1 T1 1/0 2 T1 1/1 3 |
In this example, two clock sources are available on each router multiflex VWIC ports: one from the PBX and one from the PSTN CO. However, the clock sources must be the same, so the system adjusts to this need.
The primary keyword of the clock source command, applied to T1 1/1, means that the PSTN is providing the clock source. The T1 1/0 port connected to the PBX is automatically put into looped-time mode, which means that the port takes the clocking received on its Rx (receive) pair and regenerates it back on its Tx (transmit) pair. While it is receiving clocking, it does not drive the on-board clock. It is "spoofing" the port so that the connected PBX does not detect clocking that is out of synchronization, which is indicated by slips. The router detects the slips as controlled and does not force the port to fail.
Here are some additional key points for consideration:
Figure 27 shows configuration for drop-and-insert when a 2-port Multiflex VWIC is installed in a Cisco 2600 series chassis slot or in a WIC slot of a Cisco 3600 series network module. Frame Relay data and PSTN voice calls travel between the PBXs, but no VoIP or VoIP over Frame Relay information is carried.
As in the previous example, two clock sources are available on each router multiflex VWIC ports: one from the PBX and one from the PSTN CO. However, the clock sources must be the same, so the system adjusts to this need.
The primary clock source is T1 or E1 1/0, connected to the PSTN, and its clock is a reference for T1or E1 1/1. If T1 1/0 fails, the clock source to drive T1 or E1 1/1 is generated from the line to the PBX.
The channel-group 0 command is configured in such a way that the service provider can send Frame Relay Link Management Interface (LMI) information on T1 channels 13 through 24 (17 through 31 on E1) for Frame Relay data services. This command automatically creates interface serial 1/0:0.
Interface serial 1/0:0 is where all WAN and Layer 3 protocol details are configured, for example, Frame Relay encapsulation or IP addresses.
| Router RTR-A | Router RTR-B |
hostname RTR-A ! controller T1 1/0 framing esf linecoding b8zs clock source line primary tdm-group 1 timeslots 1-12 channel-group 0 timeslots 13-24 ! controller T1 1/1 framing esf linecoding b8zs clock source line tdm-group 2 timeslots 1-12 ! interface serial 1/0:0 encapsulation frame-relay ! interface serial 1/0:1.1 ip address 209.165.200.252 255.255.255.224 frame-relay interface-dlci 100 br ! interface ethernet 0 ip address 209.165.200.250 255.255.255.224 ! router eigrp 1 network 209.165.200.224 ! connect tdm1 T1 1/0 1 T1 1/1 2 | hostname RTR-B ! controller T1 1/0 framing esf linecoding b8zs clock source line primary tdm-group 1 timeslots 1-12 channel-group 0 timeslots 13-24 ! controller T1 1/1 framing esf linecoding b8zs clock source line tdm-group 2 timeslots 1-12 ! interface serial 1/0:0 encapsulation frame-relay ! interface serial 1/0:1.1 ip address 209.165.200.253 255.255.255.224 frame-relay interface-dlci 100 br ! interface ethernet 0 ip address 209.165.201.1 255.255.255.224 ! router eigrp 1 network 209.165.200.224 network 209.165.201.0 ! connect tdm1 T1 1/0 1 T1 1/1 2 |
| Router RTR-A | Router RTR-B |
hostname RTR-A ! controller E1 1/0 framing crc4 linecoding hdb3 clock source line primary tdm-group 1 timeslots 1-15 channel-group 0 timeslots 17-31 ! controller E1 1/1 framing crc4 linecoding hdb3 clock source line tdm-group 2 timeslots 1-15 ! interface serial 1/0:0 encapsulation frame-relay ! interface serial 1/0:1.1 ip address 209.165.200.252 255.255.255.224 frame-relay interface-dlci 100 br ! interface ethernet 0 ip address 209.165.200.250 255.255.255.224 ! router eigrp 1 network 209.165.200.224 ! connect tdm1 T1 1/0 1 T1 1/1 2 | hostname RTR-B ! controller E1 1/0 framing crc4 linecoding hdb3 clock source line primary tdm-group 1 timeslots 1-15 channel-group 0 timeslots 17-31 ! controller E1 1/1 framing crc4 linecoding hdb3 clock source line tdm-group 2 timeslots 1-15 ! interface serial 1/0:0 encapsulation frame-relay ! interface serial 1/0:1.1 ip address 209.165.200.253 255.255.255.224 frame-relay interface-dlci 100 br ! interface ethernet 0 ip address 209.165.201.1 255.255.255.224 ! router eigrp 1 network 209.165.200.224 network 209.165.201.0 ! connect tdm1 T1 1/0 1 T1 1/1 2 |
Figure 28 shows how to use some T1 channels for passing voice from the PSTN to the PBX, and some channels for data services that also pass VoIP traffic. This setup requires both a digital T1 packet voice trunk network module with a multiflex VWIC installed and a separate multiflex VWIC.
The primary clock source is T1 1/0, and its clock is a reference for T1 1/1. If T1 1/0 fails, the clock source to drive T1 1/1 is generated internally.
| Router RTR-A | Router RTR-B |
hostname RTR-A ! controller T1 1/0 description - NM-HDV connected to PBX framing esf linecoding b8zs clock source internal tdm-group 1 timeslots 1-12 ds0-group 2 timeslots 13-24 type e&m-wink ! controller T1 1/1 description - xconnect to VWIC T1 framing esf linecoding b8zs clock source line tdm-group 2 timeslots 1-12 ! controller T1 2/0 description - connected to TELCO WAN framing esf linecoding b8zs channel-group 0 timeslots 13-24 tdm-group 3 timeslots 1-12 clock source line ! controller T1 2/1 description - xconnect to NM-HDV framing esf linecoding b8zs clock source internal tdm-group 4 timeslots 1-12 ! voice-port 1/0:2 ! interface serial 2/0:0 encapsulation frame-relay ! interface serial 1/0:0.1 ip address 209.165.200.252 255.255.255.224 frame-relay interface-dlci 100 br ! interface ethernet 0 ip address 209.165.200.250 255.255.255.224 ! router eigrp 1 network 209.165.200.224 ! dial-peer voice 1 voip destination-pattern 5... session target ipv4:209.165.200.253 ! dial-peer voice 2 pots destination-pattern 4... prefix 4 prefix 5 port 1/0:2 port 1/0:2 ! connect tdm1 T1 1/0 1 T1 1/1 2 connect tdm2 T1 2/0 3 T1 2/1 4 | hostname RTR-B ! controller T1 1/0 description - NM-HDV connected to PBX framing esf linecoding b8zs clock source internal tdm-group 1 timeslots 1-12 ! controller T1 1/1 description - xconnect to VWIC T1 framing esf linecoding b8zs clock source line tdm-group 2 timeslots 1-12 ! ! controller T1 2/0 description - connected to TELCO WAN framing esf linecoding b8zs channel-group 0 timeslots 13-24 tdm-group 3 timeslots 1-12 clock source line ! controller T1 2/1 description - xconnect NM-HDV framing esf linecoding b8zs clock source internal tdm-group 4 timeslots 1-12 ! voice-port 1/0:2 ! interface serial 2/0:0 encapsulation frame-relay ! interface serial 1/0:0.1 ip address 209.165.200.253 255.255.255.0 frame-relay interface-dlci 100 br ! interface ethernet 0 ip address 209.165.201.1 255.255.255.224 ! router eigrp 1 network 209.165.200.224 network 209.165.201.0 ! dial-peer voice 1 voip destination-pattern 4... session target ipv4:209.165.200.252 ! dial-peer voice 2 pots destination-pattern 5... ! connect tdm1 T1 1/0 1 T1 1/1 2 connect tdm2 T1 2/0 3 T1 2/1 4 |
The following connections are made by using channels 1 through 12 from the service provider:
Channels 13 through 24 pass Frame Relay LMI from the service provider for data services, and the channels terminate on the multiflex VWIC channel group. This serial interface is used for data traffic from the Ethernet, and VoIP traffic that originates on channels 13 through 24 from the PBX connected to the digital T1 packet voice trunk network module.
The configuration examples included in this section correspond to the topology shown in Figure 29. The routers each include a BRI VIC and a 2-slot voice network module, along with other VICs and modules that are included for the sake of completeness. Router A is connected to a PBX through the BRI VIC and is connected to Router B by a serial Ethernet interface. Router B includes a BRI VIC for connection to the PSTN, in order to process voice calls from off-premises terminal equipment.
The following example illustrates the configuration of a Cisco 3640 router for connection to a BRI VIC accessing a PBX:
vicbri_3640_s1#sh run Building configuration... Current configuration: ! version 12.0 service timestamps debug uptime service timestamps log uptime no service password-encryption ! hostname vicbri_3640_s1 ! logging buffered 200000 debugging ! ip subnet-zero ip host keyer 223.255.254.254 ! isdn switch-type basic-ni ! !
The following commands configure the ports on VICs. The last four specified ports are for FXO and E&M VICs.
voice-port 1/0/0 ! voice-port 1/0/1 ! voice-port 2/0/0 ! voice-port 2/0/1 ! voice-port 3/0/0 operation 4-wire type 2 ! voice-port 3/0/1 operation 4-wire type 2 ! voice-port 3/1/0 input gain 10 connection plar 39019 ! voice-port 3/1/1 input gain 10 connection plar 39020
The following commands configure dial peers to specify where incoming VoIP calls should be directed. In the first example, calls received with a starting digit of 5 are sent to the PBX via the BRIVIC.
dial-peer voice 10 pots destination-pattern 5..... port 1/1/0 !
This command sets up a local BRI connection:
dial-peer voice 11 pots destination-pattern 66002 port 1/0/0 !
In this example, calls with a starting digit of 9 are PSTN calls that are routed over IP:
dial-peer voice 13 voip destination-pattern 9....... session target ipv4:12.0.0.2 !
This command sets up an FXS connection over IP to the other router:
dial-peer voice 12 voip (calls to other router with FXS - go over IP) destination-pattern 7.... session target ipv4:12.0.0.2 !
The following global configuration commands define how to expand an extension number into a particular destination pattern:
num-exp 8 9529399 num-exp 1 550950 num-exp 2 76002
The following commands configure the Ethernet and serial interfaces:
interface Ethernet0/0 ip address 1.14.122.10 255.255.0.0 ip helper-address 223.255.254.254 no ip directed-broadcast ! interface Serial0/0 ip address 3.0.0.2 255.0.0.0 no ip directed-broadcast no ip mroute-cache no keepalive no fair-queue ! interface Ethernet0/1 ip address 11.0.0.1 255.0.0.0 no ip directed-broadcast ! interface Serial0/1 ip address 14.0.0.1 255.0.0.0 no ip directed-broadcast no keepalive shutdown no fair-queue clockrate 2000000
The following commands configure the BRI interfaces:
interface BRI1/0 no ip address no ip directed-broadcast isdn switch-type basic-ni1 isdn twait-disable isdn spid1 14085552121010 5552121 isdn spid2 14085552122010 5552122 isdn incoming-voice modem ! interface BRI1/1 no ip address no ip directed-broadcast isdn switch-type basic-ni1 isdn twait-disable isdn spid1 14085556362010 5556362 isdn spid2 14085556364010 5556364 isdn incoming-voice modem ! interface BRI2/0 no ip address no ip directed-broadcast isdn switch-type basic-ni1 isdn twait-disable isdn spid1 14085555711010 5555711 isdn spid2 14085555712010 5555712 isdn incoming-voice modem ! interface BRI2/1 no ip address no ip directed-broadcast isdn switch-type basic-ni1 isdn twait-disable isdn spid1 14085555162010 5555162 isdn spid2 14085555163010 5555163 isdn incoming-voice modem ! ip default-gateway 1.14.0.1 ip classless ip route 2.0.0.0 255.0.0.0 Ethernet0/1 ip route 2.0.0.0 255.0.0.0 Serial0/1 ip route 223.255.254.254 255.255.255.255 Ethernet0/0 ! ! ! line con 0 exec-timeout 0 0 transport input none line aux 0 line vty 0 4 login ! end vicbri_3640_s1#
The following example illustrates the configuration of a Cisco 2600 series router for connection to a BRI VIC accessing an ISDN telephone network:
vicbri_2600_s2#sh run Building configuration... Current configuration: ! version 12.0 service timestamps debug uptime service timestamps log uptime no service password-encryption ! hostname vicbri_2600_s2 ! logging buffered 200000 debugging ! ip subnet-zero ! isdn switch-type basic-ni ! !
The following commands configure the ports on VICs:
voice-port 1/0/0 ! voice-port 1/0/1 !
The following commands configure dial peers to specify where incoming VoIP calls should be directed. In the first example, a local FXS connection is made to Router A.
dial-peer voice 22 voip destination-pattern 6.... session target ipv4:12.0.0.1 !
This command sets up a connection to the PSTN via a BRI VIC:
dial-peer voice 23 pots destination-pattern 9.... port 1/1/0 !
This command sets up a local BRI connection:
dial-peer voice 24 pots destination-pattern 76003 port 1/0/0 !
This command sets up a connection to a PBX via Router A:
! dial-peer voice 26 voip destination-pattern 5.... session target ipv4:12.0.0.1 !
The following commands configure the Ethernet and serial interfaces:
interface Ethernet0/0 ip address 1.14.122.11 255.255.0.0 no ip directed-broadcast ! interface Serial0/0 ip address 2.0.0.1 255.0.0.0 no ip directed-broadcast no keepalive ! interface Ethernet0/1 ip address 11.0.0.2 255.0.0.0 no ip directed-broadcast ! interface Serial0/1 ip address 14.0.0.2 255.0.0.0 no ip directed-broadcast no keepalive no fair-queue
The following commands configure the BRI interfaces. Note that only one BRI VIC is installed in a VNM.
! interface BRI1/0 no ip address no ip directed-broadcast isdn switch-type basic-ni1 isdn twait-disable isdn spid1 14085551111 5551111 isdn spid2 14085551112 5551112 isdn incoming-voice modem interface BRI1/1 no ip address no ip directed-broadcast isdn switch-type basic-ni1 isdn twait-disable isdn spid1 14085552111 5552111 isdn spid2 14085552112 5552112 isdn incoming-voice modem ! ip classless ip route 3.0.0.0 255.0.0.0 Ethernet0/1 ip route 3.0.0.0 255.0.0.0 Serial0/1 ip route 223.255.254.0 255.255.255.0 Ethernet0/0 ! ! ! line con 0 exec-timeout 0 0 transport input none line aux 0 line vty 0 4 login ! end vicbri_2600_s2#
The following example configures R2 signalling and customizes R2 parameters on controller E1 2 of a CiscoAS5300. In most cases, the same R2 signalling type is configured on each E1 controller.
! Specify the E1 controller that you want to configure with R2 signalling. A controller ! informs the access server how to distribute or provision individual time slots for a ! connected channelized E1 line. You must configure one E1 controller for each E1 line. ! Configure channel associated signalling. The signalling type forwarded by the ! connecting telco switch must match the signalling configured on the CiscoAS5300. !The country code is ITU by default. ! controller E1 0 framing NO-CRC4 cas-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani cas-custom 0 ! controller E1 1 framing NO-CRC4 clock source line primary cas-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ! ! Customize some of the E1 R2 signalling parameters with the cas-custom channel ! controller configuration command. This example specifies the default R2 settings for ! Brazil. ! cas-custom 0 country brazil use-defaults metering category 2 answer-signal group-b 1 ! controller E1 2 ! controller E1 3 ! ! Configure voice port parameters. Be sure that the cptone command value is compatible ! with the country code defined by the cas-custom command. In this example, because ! ITU has no specific cptone value defined and uses aLaw E1 R2 signalling, the GB ! cptone command value is used. ! voice-port 0:0 cptone GB ! voice-port 1:0 cptone BR description Brasil Tone ! ! Define the parameters associated with the VoIP dial peer. ! dial-peer voice 101 voip destination-pattern +500.. session target ipv4:172.14.25.1 ! ! Define the parameters associated POTS dial peer. ! dial-peer voice 8221 pots destination-pattern 011822... direct-inward-dial port 0:0 ! ! Configure LAN interfaces. ! interface Ethernet0 ip address 172.13.103.33 255.255.0.0 no ip directed-broadcast no ip mroute-cache load-interval 30 no cdp enable ! interface FastEthernet0 ip address 173.14.25.100 255.255.0.0 no ip directed-broadcast bandwidth 1000000 load-interval 30 duplex full hold-queue 75 in ! no ip classless ip route 223.255.254.253 255.255.255.255 Ethernet0 ! ! line con 0 exec-timeout 0 0 logging synchronous level all transport input none escape-character BREAK line aux 0 rotary 1 transport preferred none transport input all flowcontrol hardware line vty 0 4 exec-timeout 60 0 password lab login ! end
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NoteWe strongly recommend that you specify your country default settings. To display a list of supported countries, enter the cas-custom country ? command. The default setting for all countries is ITU. |
The following example configures T1 CAS parameters on a CiscoAS5300:
! Enter global configuration mode. config terminal ! Enter controller configuration mode to configure your controller port. The controller ports are labeled 0 through 3 on the Quad T1/PRI and E1/PRI cards. controller t1 0 ! Enter your telco's framing type. framing esf ! Enter the clock source for the line. Configure other lines as clock source secondary ! or internal. Note that only one PRI can be clock source primary and one PRI can be ! clock source secondary clock source line primary ! Enter your telco's line code type. linecode b8zs ! Configure all channels for E&M, FXS, and SAS analog signalling. Enter 1-24 for T1. ! If E1, enter 1-31. ! Signalling types include e&m-fgb, e&m-fgd, e&m-immediate-start, fxs-ground-start, ! fxs-loop-start, sas-ground-start, and sas-loop-start. ! You must use the same type of signalling that your central office uses. ! For E1 using the Anadigicom converter, use cas e&m-fgb signalling. cas-group 1 timeslots 1-24 type e&m-fgb dtmf dnis ! Configure each additional controller (there are four). In this example, the ! controller number is 1, instead of 0. The clock source is secondary, instead of ! primary. The cas-group is 2, instead of 1 controller t1 1 framing esf linecode b8Zs clock source line secondary cas-group 2 timeslots 1-24 type e&m-fgb ! Configure each additional controller. controller T1 2 clock source internal cas-group 0 timeslots 1-24 type e&m-fgd mf ani-dnis controller T1 3 clock source internal ! Enter the dial peer configuration mode to configure a POTS peer. ! Specify destination pattern for this POTS peer. dial-peer voice 3070 pots destination-pattern +30... port 0:1 prefix 30 ! Specify destination pattern, and direct inward dial for each POTS peer. dial-peer voice 4080 pots destination-pattern +40... direct-inward-dial port 1:2 prefix 40 ! Specify the destination pattern and the direct inward dial for the dial peer. dial-peer voice 1050 pots destination-pattern +10... direct-inward-dial prefix 50 ! Specify the destination pattern and the direct inward dial for the dial peer. dial-peer voice 2060 pots destination-pattern +20... direct-inward-dial prefix 60 dial-peer voice 5050 voip answer-address 10... destination-pattern +50... end end
This section includes the following configuration examples:
These examples are not necessarily complete configurations. They are designed to illustrate specific tips and techniques, and only the relevant portions of the configurations are shown. Each configuration includes a brief introduction, side-by-side configurations for routers at either end, and explanations of key points.
Figure 30 shows how to set up a Cisco 7200 series router to collect digits from either a PBX/PSTN or a telephone and route a VoIP call based on the digits received. The commands used in the configurations are explained inline. Only relevant sections of the configuration are shown. The example assumes that the IP portion of the network is already in place.
| Alpha Router | Beta Router |
hostname router-alpha ! voice-card 1 codec high ! dial-peer voice 1 voip codec g723r53 fax-rate 14400 destination-pattern 5.... session target ipv4:192.168.100.1 ! dial-peer voice 2 pots destination-pattern 4.... prefix 4 port 1/0:1 ! controller T1 1/0 framing esf linecode b8zs clock source line ds0-group 1 timeslots 1-24 type e&m-wink ! interface serial 0/0 ip address 192.168.100.2 255.255.255.0 | hostname router-beta ! voice-card 1 codec high ! dial-peer voice 1 voip codec g723r53 fax-rate 14400 destination-pattern 4.... session-target ipv4:192.168.100.2 ! dial-peer voice 2 pots destination-pattern 5.... prefix 5 port 1/0:1 ! controller T1 1/0 framing esf linecode b8zs clock source internal ds0-group 1 timeslot 1-24 type e&m-wink ! interface s0/0 ip address 192.168.100.1 255.255.255.0 |
In this configuration, the PBX seizes the T1/E1 to the router, which expects to collect digits from the PBX. Upon collecting those digits, the router tries to match a dial peer to route the call. If the router receives the correct digits, it routes the call according to the configuration of the dial peer.
Here are some key points for consideration:
Figure 31 shows how to configure a Cisco 7200 series router to support FRF.12 fragmentation and queueing in a VoIP over Frame Relay network. FRF.12 is a Frame Relay Forum standard mechanism for fragmenting data packets. This fragmentation helps eliminate the delays that occur when sending voice and data over the same network---large data packets can delay smaller voice packets from being sent into the IP network. FRF.12 is also supported on the Cisco MC3810 and Cisco 7200 routers, which can be used as tandem nodes for VoIP networks.
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NoteThis example shows VoIP over Frame Relay, which is not the same as VoFR. For more information about VoFR, see the "Configuring Voice over Frame Relay." |
The following configuration fragments both the IP and IPX data traffic to 80 bytes, allowing the VoIP traffic to be only minimally delayed on the network. The FRF.12 setup also traffic-shapes the output traffic rate to match the provisioned CIR from the Frame Relay carrier. This ensures that traffic is not dropped or delayed within the Frame Relay network.
Here are some key points for consideration:
| Alpha Router | Beta Router |
hostname router-alpha ! ipx routing ! card type t1 1 ! dspint DSPfarm 1/0 codec high L30 ! controller T1 1/0 framing esf linecode b8zs clock source line ds0-group 1 timeslot 1-24 type e&m-wink ! dial-peer voice 1 voip dtmf-relay h245-alpha codec g723r53 destination-pattern 5.... session target ipv4:192.168.100.2 ! dial-peer voice 2 pots destination-pattern 4.... prefix 4 port 1/0:1 ! interface serial 0/0 encapsulation frame-relay frame-relay traffic-shaping ! interface serial 0/0.1 point-to-point ip address 192.168.100.1 255.255.255.0 ipx network ABCD frame-relay interface-dlci 100 class cisco_frf12 ! map-class frame-relay cisco_frf12 frame-relay voice bandwidth 42000 frame-relay fragment 80 no frame-relay adaptive-shaping frame-relay cir 32000 frame-relay bc 1000 frame-relay mincir 64000 frame-relay fair-queue | hostname router-beta ! ipx routing ! card type t1 1 codec high ! dspint DSPfarm 1/0 codec high L30 controller T1 1/0 framing esf linecode b8zs clock source line ds0-group 1 timeslot 1-24 type e&m-wink ! dial-peer voice 1 voip dtmf-relay h245-alpha codec g723r53 destination-pattern 4.... session target ipv4:192.168.100.2 ! dial-peer voice 2 pots destination-pattern 5.... prefix 5 port 1/0:1 ! interface serial 0/0 encapsulation frame-relay frame-relay traffic-shaping ! interface serial 0/0.1 point-to-point ip address 192.168.100.2 255.255.255.0 ipx network ABCD frame-relay interface-dlci 101 class cisco_frf12 ! map-class frame-relay cisco_frf12 frame-relay voice bandwidth 42000 frame-relay fragment 80 no frame-relay adaptive-shaping frame-relay cir 64000 frame-relay bc 1000 frame-relay mincir 64000 frame-relay fair-queue |
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NoteWith the introduction of Cisco IOS Release 12.0(5)T and subsequent releases, Cisco VoIP gateways support H.323v2 (H.323 Version 2), which is backwards compatible with systems running H.323 Version 1. However, H.323 Version 2 features do not interoperate with H.323 Version 1 features in Cisco IOS releases prior to 11.3(9)NA or 12.0(3)T. Earlier Cisco IOS versions contain H.323 Version 1 software that does not support protocol messages with an H.323 Version 2 protocol identifier. All systems must be running either Cisco IOS Release 11.3(9)NA and later or Cisco IOS Release 12.0(3)T and later releases to interoperate with H.323 Version 2. Gateway Resource Availability Indication (RAI) messages are currently not supported on the Cisco 7200 series. (These are messages that are sent to the Gatekeeper to inform it about the status of a Gateway DSP or DS0 availability.) |
Figure 32 shows how to configure a Cisco 7200 series router to route VoIP calls through an H.323 gatekeeper. This setup shows calls being routed from a gateway in Zone-Alpha, through the gatekeeper, to a gateway in Zone-Beta.
| Gatekeeper | |
hostname router-gatekeeper ! gatekeeper zone local alpha alpha.com zone local beta beta.com no use-proxy alpha.com remote-zone beta.com no use-proxy beta.com remote-zone alpha.com zone prefix router-alpha 4.... zone prefix router-beta 5.... no shutdown ! interface ethernet 0/0 ip address 10.1.1.3 255.255.255.0 | |
| Alpha Router | Beta Router |
hostname router-alpha ! card type t1 1 ! dspint DSPfarm 1/0 ! controller T1 1/0 framing esf linecode b8zs clock source internal ds0-group 1 timeslot 1-24 type e&m-wink ! voice-port 1/0:1 ! dial-peer voice 1 voip dtmf-relay h245-alpha destination-pattern 5.... tech-prefix 1# session target ras ! dial-peer voice 2 pots destination-pattern 4.... prefix 4 port 1/0:1 ! gateway ! interface ethernet 0/0 ip address 10.1.1.1 255.255.255.0 h323-gateway voip interface h323-gateway voip id alpha ipaddr 10.1.1.3 1719 h323-gateway voip h323-id router-alpha@alpha.com h323-gateway voip tech-prefix 1# | hostname router-beta ! card type t1 1 ! dspint DSPfarm 1/0 ! controller T1 1/0 framing esf linecode b8zs clock source line ds0-group 1 timeslot 10-24 type e&m-wink ! voice-port 1/0:1 ! dial-peer voice 1 voip dtmf-relay h245-alpha destination-pattern 4.... tech-prefix 1# session target ras ! dial-peer voice 2 pots destination-pattern 5.... prefix 5 port 1/0:1 ! gateway ! interface ethernet 0/0 ip address 10.1.1.2 255.255.255.0 h323-gateway voip interface h323-gateway voip id beta ipaddr 10.1.1.3 1719 h323-gateway voip h323-id router-beta@beta.com h323-gateway voip tech-prefix 1# |
Here are some key points for consideration:
Figure 33 shows how to set up a Cisco 7200 series router for a PLAR. PLAR is used to allow a station or DS0 to go off hook, and---without the user dialing digits---have a call completed to the far end. PLAR can also provide dial tone from a remote PBX for off-premises applications.
In this configuration, the phones off router Beta go off hook and receive dial tone from the PBX connected to router Alpha. From there, users can dial any digits in to the PBX as if their stations are directly connected to it.
Here are some key points for consideration:
| Alpha Router | Beta Router |
hostname router-alpha ! card type t1 1 ! dspint DSPfarm 1/0 ! controller T1 1/0 framing esf linecode b8zs ds0-group 1 timeslot 1 type fxo-loop ds0-group 2 timeslot 2 type fxo-loop ! dial-peer voice 1 voip dtmf-relay h245-alpha codec g729a destination-pattern 2.. session target ipv4:192.168.100.2 ! dial-peer voice 2 pots destination-pattern 101 port 1/0:1 ! dial-peer voice 3 pots destination-pattern 102 port 1/0:2 ! voice-port 1/0:1 connection plar 201 ! voice-port 1/0:2 connection plar 202 ! interface s0/0 ip address 192.168.100.1 255.255.255.0 | hostname router-beta ! dial-peer voice 1 voip destination-pattern 1.. dtmf-relay h245-alpha codec g729a session target ipv4:192.168.100.1 ! dial-peer voice 2 pots destination-pattern 201 port 1/1 ! ! dial-peer voice 3 pots destination-pattern 202 port 1/2 ! voice-port 1/1 ! ! voice-port 1 / 2 ! ! interface serial 0/0 ip address 192.168.100.2 255.255.255.0 |
Figure 34 shows how to configure a Cisco 7200 series router for a trunk connection. A trunk connection is like a "wire" between the two routers. It is a transparent connection, so it allows features such as hookflash (also called switchhook flash) or hoot `n' holler (point-to-point) to pass. This type of trunk configuration can also be used for OPXs that require rollover to a centralized voice-mail system when the user does not answer.
A trunk connection can only be used between E&M ports or with FXO-to-FXS connections.
| Alpha Router | Beta Router |
hostname router-alpha ! card type t1 1 ! dspint DSPfarm 1/0 ! controller T1 1/0 framing esf linecode b8zs ds0-group 1 timeslot 1 type e&m-wink ds0-group 2 timeslot 2 type e&m-wink clock source line ! voice-port 1/0:1 connection trunk 1111 ! voice-port 1/0:2 connection trunk 1112 ! dial-peer voice 1 voip dtmf-relay h245-alpha codec g729a destination-pattern 111. session target ipv4:192.168.100.2 ! dial-peer voice 2 pots destination-pattern 2221 port 1/0:1 ! dial-peer voice 3 pots destination-pattern 2222 port 1/0:2 ! interface serial 0/0 ip address 192.168.100.1 255.255.255.0 | hostname router-beta ! card type t1 1 ! dspint DSPfarm 1/0 ! controller T1 1/0 framing esf linecode b8zs ds0-group 1 timeslot 1 type e&m-wink ds0-group 2 timeslot 2 type e&m-wink clock source line ! voice-port 1/0:1 connection trunk 2221 ! voice-port 1/0:2 connection trunk 2222 ! dial-peer voice 1 voip dtmf-relay h245-alpha codec g729a destination-pattern 222. session target ipv4:192.168.100.1 ! dial-peer voice 2 pots destination-pattern 1111 port 1/0:1 ! dial-peer voice 3 pots destination-pattern 1112 port 1/0:2 ! interface serial 0/0 ip address 192.168.100.2 255.255.255.0 |
In this configuration, a permanent and transparent path is set up between individual DS0s on each router. It passes dial tone from the remote PBX and passes DTMF digits out of band.
The connection trunk command establishes the permanent trunk connection between the routers. The digits after the command are passed internally within the router to match a dial peer so that the call can be set up.
Figure 35 shows an example of drop-and-insert. Drop-and-insert technology is one way to integrate old PBX technologies with VoIP. It allows you to take 64-Kbps DS0 channels from one T1 and digitally cross-connect them to 64-Kbps DS0 channels on another T1. Drop-and-insert is sometimes called TDM cross-connect.
Drop-and-insert allows individual 64-Kbps DS0 channels to be transparently passed, uncompressed, between T1 ports without passing through a DSP. Using this method, the channel traffic is sent between a PBX and CO switch (PSTN) or other telephony device, allowing the use, for example, of some PBX channels for long-distance service through the PSTN while the router compresses others for interoffice VoIP calls. In addition, drop-and-insert can cross-connect a telephony switch (from the CO or PSTN) to a channel bank to provide external analog connectivity.
| Router RTR-A | Router RTR-B |
hostname RTR-A ! card type t1 1 ! codec high 1-30 ! dspint DSPfarm 1/0 ! codec high 1-30 controller T1 1/0 clock source line framing esf linecoding b8zs ds0-group 1 timeslots 1-12 type e&m-wink tdm-group 2 timeslots 13-24 type e&m ! controller T1 1/1 clock source line primary framing esf linecoding b8zs tdm-group 3 timeslots 13-24 type e&m ! voice-port 1/0:1 ! dial-peer voice 1 voip destination-pattern 4.... codec g723r63 dtmf-relay h245-alpha session target ipv4:192.168.100.2 ! dial-peer voice 2 pots destination-pattern 5.... prefix 5 port 1/0:1 ! interface serial 0/0 encapsulation ppp ip address 192.168.100.1 255.255.255.0 ! connect tdm1 T1 1/0 2 T1 1/1 3 | hostname RTR-B ! card type t1 1 ! codec high 1-30 ! dspint DSPfarm 1/0 ! codec high 1-30 ! controller T1 1/0 clock source line framing esf linecoding b8zs ds0-group 1 timeslots 1-12 type e&m-wink tdm-group 2 timeslots 13-24 type e&m ! controller T1 1/1 clock source line primary framing esf linecoding b8zs tdm-group 3 timeslots 13-24 type e&m ! voice-port 1/0:1 ! dial-peer voice 1 voip destination-pattern 5.... codec g723r63 dtmf-relay h245-alpha session target ipv4:192.168.100.1 ! dial-peer voice 2 pots destination-pattern 4.... prefix 4 port 1/0:1 ! interface serial 0/0 encapsulation ppp ip address 192.168.100.2 255.255.255.0 ! connect tdm1 T1 1/0 2 T1 1/1 3 |
Here are some key points for consideration:
This example, as depicted in Figure 36, shows the Busyout Monitor feature used on a digital voice interface. The feature instructs the voice gateway to busy out the voice port (all channels defined in the corresponding DS0 Group) if serial 2/1 fails. When the specified LAN/WAN interface becomes available again, the voice port is put back into service for handling VoIP calls.
hostname RTR-A ! voice-card 1 ! controller T1 1/0 framing esf linecoding b8zs clock source line ds0-group 1 timeslots 1-24 type e&m-wink ! voice-port 1/0:1 busyout monitor interface serial 2/1 ! interface serial 2/1 encapsulation ppp bandwidth 1544 ip address 10.168.100.1 255.255.255.0 ! interface ethernet 0/0 ip address 10.168.102.1 255.255.255.0 ! dial-peer voice 1 voip destination-pattern 5.... codec g711u dtmf-relay h245-alphanumeric session target ipv4:10.168.100.2 ! dial-peer voice 2 pots destination-pattern 4.... prefix 4 port 1/0:1 hostname RTR-B ! voice-card 1 ! controller T1 1/0 framing esf linecoding b8zs clock source line ds0-group 1 timeslots 1-24 type e&m-wink ! voice-port 1/0:1 busyout monitor interface serial 2/0 ! interface serial 2/0 encapsulation ppp bandwidth 1544 ip address 10.168.100.2 255.255.255.0 ! interface ethernet 0/0 ip address 10.168.101.1 255.255.255.0 ! dial-peer voice 1 voip destination-pattern 4.... codec g711u dtmf-relay h245-alphanumeric session target ipv4:10.168.100.1 ! dial-peer voice 2 pots destination-pattern 5.... prefix 5 port 1/0:1
Posted: Wed Jul 26 23:57:20 PDT 2000
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