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This document explains how you can configure 1- and 2-port T1 and E1 Multiflex Voice/WAN interface cards (VWICs) on Cisco 2600 and 3600 routers and includes the following sections:
Multiflex VWICs support the following applications:
The following Multiflex VWICs are available:
Multiflex VWIC features include:
Table 1 shows the possible hardware configurations for the Multiflex VWICs.
| VWIC and Application | Cisco 2600 Series | Cisco 3620 and 3640 | Cisco 3660 |
|---|---|---|---|
1-port Data only | T1 or E1 VWIC in a chassis slot | T1 or E1 VWIC in a 1- or 2-port network module (NM-1E2W, NM-2E2W, NM-1E1R2W) | Planned for future availability |
2-port Data only | T1 or E1 VWIC in a chassis slot. Does not provide packet voice. Can provide two physical WAN connections with both ports supporting up to full T1/E1 speeds. | Planned for future availability | Planned for future availability |
2-port Drop-and-Insert | T1 or E1 D&I VWIC in a chassis slot. Does not provide packet voice. Provides WAN connections and digital cross-connect. | T1 D&I VWIC in a Digital T1 Packet Voice Trunk Network Module. Provides voice connections and digital cross-connect. Does not provide WAN connections. T1 or E1 D&I VWIC in a 1- or 2-port network module (NM-1E2W, NM-2E2W, NM-1E1R2W). Provides WAN connections and digital cross-connect. | T1 D&I VWIC in a Digital T1 Packet Voice Trunk Network Module. Provides voice connections and digital cross-connect. Does not provide WAN connections. |
1- or 2-port Voice only, no WAN connections | T1 VWIC in a Digital T1 Packet Voice Trunk Network Module | T1 VWIC in a Digital T1 Packet Voice Trunk Network Module | T1 VWIC in a Digital T1 Packet Voice Trunk Network Module |
T1/E1 Multiflex VWICs reduce networking life-cycle costs in the following ways:
T1/E1 Multiflex VWICs provide the following benefits of multifunction support for LAN-to-LAN routing, multiplexed voice and data, and voice:
The following restrictions apply to T1/E1 Multiflex VWIC configurations:
See Table 1 for summary information.
Digital T1 Packet Voice Trunk Network Modules requires 2-port T1 Multiflex VWIC for operation. For more information about these modules, see Configuring Digital T1 Packet Voice Trunk Network Modules on Cisco 2600 and 3600 Series Routers .
The following documents provide additional information about installing and configuring T1/E1 Multiflex VWICs:
The following Cisco IOS Release 12.0 documents provide information that can help you use T1/E1 Multiflex VWICs:
This feature is supported on the following platforms:
T1/E1 Multiflex VWICs support the standards, MIBs, and RFCs listed in this section.
RFC 1406
T1/E1 Multiflex VWICs require specific service, software, and hardware:
Voice, Video, and Home Applications Configuration Guide and Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0 provide information about setting up voice networks.
Configuring T1/E1 Multiflex VWICs includes the following tasks:
For detailed information about configuring a T1 Multiflex VWIC that is installed in a Digital T1 Packet Voice Trunk Network Module, see Configuring Digital T1 Packet Voice Trunk Network Modules on Cisco 2600 and 3600 Series Routers .
This section includes the following sections:
Follow the steps below if you are configuring T1 Multiflex VWICs installed in Digital T1 Packet Voice Trunk Network Modules for voice. Repeat Steps 2 and 3 for each voice card.
| Step | Command | Purpose | ||
|---|---|---|---|---|
| Router# configure terminal | Enter global configuration mode. | ||
| Router(config)# voice-card slot | Enter voice card interface configuration mode and specify the slot location by using a value from 0 to 5, depending upon your router. | ||
| Router(config-voice-ca)# codec complexity {high | | Specify 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 packet voice data modules (PVDMs) installed and the codec complexity. Here is a guideline:
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. See Step 7 in "Configuring Voice Dial Peers". 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. For more information about the ds0-group command, see Step 5. | ||
| Router(config)# controller T1 slot/port | Enter controller configuration mode for the VWIC. Valid values for slot are 0 through 5 and for port are 0 and 1. | ||
| Router(config-controller)# ds0-group ds0-group-no | (Voice only) This command defines the T1 channels for use by compressed voice calls as well as the signaling method the router uses to connect to the PBX or CO. Set up DS0 groups after you have specified codec complexity in voice-card configuration, as shown in Step 3. 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. ds0-group-no is a value from 0 to 23 that identifies the DS0 group. Note The ds0-group command automatically creates a logical voice port that is numbered as follows: slot/port:ds0-group-no. Although only one voice port is created, applicable calls are routed to any channel in the group.timeslot-list is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen to indicate a range of timeslots. For T1, allowable values are from 1 to 24. To map individual DS0 timeslots, define additional groups. The system maps additional voice ports for each defined group. See Step 2 of "Configuring Voice Ports". The signaling method selection for type depends on the connection that you are making:
|
Follow this procedure to configure T1 and E1 controllers. Skip Steps 1 and 2 if you are already in controller configuration mode.
Repeat the steps following Step 2 for each controller.
| Router# configure terminal | Skip this step if you are already in controller configuration mode. Enter global configuration mode. | ||
| Router(config)# controller {T1 | E1}slot/port
| Skip this step if you are already in controller configuration mode. Enter controller configuration mode for the T1 or E1 controller at the specified slot/port location. | ||
| Router(config-controller)# loopback {diagnostic | | (Optional, T1 only, testing) This command generates a local loopback test at the line or payload level or a remote loopback. For details, Configuring Digital T1 Packet Voice Trunk Network Modules on Cisco 2600 and 3600 Series Routers. | ||
| Router(config-controller)# clock source {line | 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:
| ||
| Router or
Router | Set the framing to SuperFrame (SF) or Extended SuperFrame (ESF) format, according to service provider requirements. Set 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. | ||
| Router | Set the line encoding according to your service provider's instructions. Bipolar-8 zero substitution (B8ZS), available only for T1 lines, encodes a sequence of eight zeros in a unique binary sequence to detect line coding violations. Alternate mark inversion (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), the default. | ||
| Router(config-controller)# line-termination {75-ohm |
| (E1 only) Enter 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 120-ohm to match the balanced 120-ohm interface. This is the default. 75-ohm is for an unbalanced BNC 75-ohm interface. | ||
| Router(config-if)# fdl {att | ansi | both}
| (T1 interfaces only) This command sets the Facility Data Link (FDL) exchange standard for the CSU controllers. The FDL is a 4-Kbps channel used with the Extended SuperFrame (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 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. att selects the AT&T TR54016 standard for extended superframe facilities data link exchange support. both enables both of the above standards. | ||
| Router(config-controller)# cablelength long {gain26 |
or cablelength short {133 | 266 | 399 | 533 | 655}
| (T1 interfaces only) 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:
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:
If you do not set the cable length, the system defaults to a setting of cablelength long gain26 0db. |
Perform the steps in this section if you are setting up Drop and Insert. If not, proceed to "Configuring Serial Interfaces".
| Router(config-controller)# tdm-group tdm-group-no | Enter this command to set up TDM channel groups for the Drop-and-Insert function with a 2-port Multiflex VWIC. tdm-group-no is a value from 0 to 23 for T1 and from 0 to 30 for E1; it identifies the group. timeslot-list is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen to indicate a range of timeslots. The valid range is from 1 to 24 for T1. For E1, the range is from 1 to 31. The signaling method selection for type depends on the connection that you are making. The fxs and fxo options allow you to specify a ground-start or loop-start line. The Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference includes additional information about these options. Note The group numbers for controller groups must be unique. For example, a TDM group should not have the same ID number as a DS0 group or channel group. | ||
| Router(config-controller)# channel-group | (Optional) Enter this command to set up channel groups for WAN data services with a 2-port Multiflex Drop-and-Insert VWIC. channel-group-no 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. timeslot-list is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen to indicate a range of timeslots. The valid range is from 1 to 24 for T1. For E1, the range is from 1 to 31. The optional speed setting defaults to 56 Kbps for T1 and 64 Kbps for E1. Note Although the CLI displays 48 as a speed option, it is not supported. | ||
| Router(config-controller)# no shutdown | Activate the controller. | ||
| Router(config-controller)# exit | Exit controller configuration mode. Skip the next step if you are not setting up Drop and Insert. | ||
| Router(config)# connect id {T1 | E1} slot/port-1 | This global configuration command sets up the connection between two T1 or E1 TDM groups of timeslots on the WVIC---for Drop and Insert. id is a name for the connection. Identify each controller by its slot/port location. tdm-group-no-1 and tdm-group-no-2 identify the TDM group numbers (from 0 to 23 or 30) on the specified controller. The groups were set up in Step 1. See the "Configuration Examples" section for sample Drop and Insert configurations. |
Step 1 Enter the show running-config command to display the current voice-card setting. If no codec complexity is shown, the default of medium complexity is set. The following example shows an excerpt from the command output:
Router# show running-config . . hostname router-alpha voice-card 1 codec complexity high . . .
Step 2 The privileged EXEC show controllers t1/e1 command displays the status of T1 or E1 controllers and displays information about clock sources and other settings for the ports:
Router# show controller T1 1/0
T1 1/0 is up.
Applique type is Channelized T1
Cablelength is short 133
Description: T1 WIC card Alpha
No alarms detected.
Framing is ESF, Line Code is B8ZS, Clock Source is Line Primary.
Data in current interval (1 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
Step 3 The privileged EXEC show connection all command displays the status of T1 or E1 TDM controller groups and how they are set up:
Router# show connection all ID Name Segment 1 Segment 2 State ======================================================================== 1 Test -T1 1/0 01 -T1 1/1 02 ADMIN UP
The way you set up serial and LAN interfaces depends on your application. This document does not explain all possible serial interface configuration options, nor does it show LAN interface configuration. For complete information, see the Cisco IOS Release 12.0 Cisco IOS Interface Configuration Guide and the Cisco IOS Interface Command Reference.
If you are not planning voice support, proceed to "Configuration Examples".
To configure VoIP, you must at least set up IP addresses for serial interfaces. When a user dials enough digits to match a configured destination pattern, the telephone number is mapped to an IP host through the dial plan mapper. The IP host has a direct connection to either the destination telephone number or a PBX that completes the call to the configured destination pattern.
To verify serial interface configuration, enter the privileged EXEC command show interfaces serial, which shows the status of all serial interfaces or of a specific serial interface, as in the following example. You can use this command to check the encapsulation, IP addressing, and other settings:
Router #show interface serial0/0:0
Serial0/0:0 is up, line protocol is up
Hardware is QUICC Serial
Internet address is 1.156.1.1/24
MTU 1500 bytes, BW 1536 Kbit, DLY 20000 usec,
reliability 255/255, txload 1/255, rxload 1/255
Encapsulation HDLC, loopback not set
Keepalive not set
Last input 00:00:00, output 00:00:00, output hang never
Last clearing of "show interface" counters never
Input queue: 0/75/0 (size/max/drops); Total output drops: 0
Queueing strategy: weighted fair
Output queue: 0/1000/64/0 (size/max total/threshold/drops)
Conversations 0/1/256 (active/max active/max total)
Reserved Conversations 0/0 (allocated/max allocated)
5 minute input rate 1000 bits/sec, 1 packets/sec
5 minute output rate 1000 bits/sec, 1 packets/sec
637 packets input, 64736 bytes, 0 no buffer
Received 181 broadcasts, 0 runts, 5 giants, 0 throttles
3617 input errors, 1506 CRC, 1646 frame, 0 overrun, 0 ignored, 0 abort
682 packets output, 67213 bytes, 0 underruns
0 output errors, 0 collisions, 1070 interface resets
0 output buffer failures, 0 output buffers swapped out
13 carrier transitions
Timeslot(s) Used:1-24, Transmitter delay is 0 flags
Follow these steps to set up voice ports to support the local and remote stations. This procedure applies only to T1 Multiflex VWICs installed in Digital T1 Packet Voice Trunk Network Modules when voice services are required.
This section does not show all the commands that you can use. To learn more, see Voice, Video, and Home Applications Configuration Guide and Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.
| Step | Command | Purpose | ||
|---|---|---|---|---|
| Router# configure terminal | Enter global configuration mode. | ||
| Router(config)# voice-port slot/port:ds0-group-no | Enter voice-port configuration mode. slot is the router location where the voice module is installed. Valid entries are from 0 to 3. port 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. For more information about DS0 groups, see Step 5 of "Configuring Voice Card and Controller Settings". Note This voice-port command syntax does not apply to analog voice network modules and voice interface cards. Specify voice interface cards by using slot/subunit/port, designating the router slot for the voice network module, the location of the voice interface card in the network module, and the port on the voice interface card. | ||
| Router(config-voice-port)# busyout monitor interface | (Optional) This command allows you to specify a LAN or WAN interface that will be monitored, and, when it is down, trigger a busyout (offhook) 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. | ||
| Router(config-voice-port)# comfort-noise | (Optional) 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. | ||
| Router(config-voice-port)# echo-cancel enable | (Optional) 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 25 milliseconds can cause problems. | ||
| Router(config-voice-port)# echo-cancel coverage {16 | | (Optional) This command adjusts the echo canceller by the specified number of milliseconds; the default is 16. | ||
| Router(config-voice-port)# connection {plar |trunk} | (Optional) This command sets up a connection mode for the voice port. plar specifies a private line automatic ring down (PLAR) connection, which rings a remote telephone when the dial peer goes off hook. trunk specifies a straight tie-line connection to a PBX. string specifies the remote telephone number or significant start digits of the number. See "Configuration Examples" for sample PLAR and trunk configurations. | ||
| Router(config-voice-port)# timeouts interdigit seconds | (Optional) This command 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 10 seconds, and the timeout can be set from 0 to 120 seconds. Note Changes to the default for this command normally are not required. Other timing settings may also be needed. For more information, see the Cisco IOS Release 12.0 Voice, Video, and Home Applications Configuration Guide. | ||
| Router(config-voice-port)# exit | Exit voice-port configuration mode. |
Follow this procedure to verify voice-port configuration. To learn more, see Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference.
Important command output is shown in bold.
To verify the voice-port configuration, enter the privileged EXEC show voice port slot/port:ds0-group command. The following sample output from the command shows explanatory information after the << characters:
cisco-router# show voice port 1/0:1 receEive and transMit Slot is 1, Sub-unit is 0, Port is 1 << voice-port 1/0:1 Type of VoicePort is E&M Operation State is DORMANT Administrative State is UP No Interface Down Failure Description is not set Noise Regeneration is enabled Non Linear Processing is enabled Music On Hold Threshold is Set to -38 dBm In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is enabled Echo Cancel Coverage is set to 8 ms Connection Mode is normal Connection Number is not set Initial Time Out is set to 10 s Interdigit Time Out is set to 10 s Region Tone is set for US
Follow these steps to set up voice dial peers to support the local and remote stations. This procedure applies only to T1 Multiflex VWICs installed in Digital T1 Packet Voice Trunk Network Modules when voice services are required.
This section does not show all the commands that you might need to enter. To learn more, see Voice, Video, and Home Applications Configuration Guide and Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.
| Step | Command | Purpose | ||
|---|---|---|---|---|
| Router# configure terminal | Enter global configuration mode. | ||
| Router(config)# dial-peer voice number pots | Enter dial-peer configuration mode and define a local dial peer that will connect to the plain old telephone service (POTS) network. number is one or more digits identifying the dial peer. Valid entries are from 1 to 2147483647. pots indicates a peer using basic telephone service. | ||
| Router | Configure the dial peer's destination pattern, so that the system can reconcile dialed digits with a telephone number. string is a series of digits that specify the E.164 or private dialing plan phone number. Valid entries are the digits 0 through 9 and the letters A through D. The plus symbol (+) is not valid. You can enter the following special characters:
When the timer (T) character is included at the end of the destination pattern, the system collects dialed digits as they are entered---until the interdigit timer expires (10 seconds, by default)---or the user dials the termination of end-of-dialing key (default is #). Note The timer character must be a capital T. | ||
| Router(config-dialpeer)# prefix string | (Optional) Include a dial-out prefix that the system enters automatically instead of people dialing it. string is a value from 0 to 9, and you can use a comma (,) to indicate a pause. Note There are other digit manipulation commands available to handle such situations as prefixes for special services, ignoring some digits, and dialing in to remote PBXs as though they are local. | ||
| Router(config-dialpeer)# port slot/port:ds0-group-no | This command associates the dial peer with a specific logical interface. slot is the router location where the voice module is installed. Valid entries are from 0 to 3. port indicates the voice interface card 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. | ||
| Router(config)# dial-peer voice number voip | Enter dial-peer configuration mode and define a remote VoIP dial peer. number is one or more digits identifying the dial peer. Valid entries are from 1 to 2147483647. voip indicates a VoIP peer using voice encapsulation on the IP network. | ||
| Router | The voice-card configuration codec complexity command sets the codec options that are available when you enter this command (see Step 3 of "Configuring Voice Card and Controller Settings"). If you do not set codec complexity, g729r8 with IETF bit-ordering is used. If you set codec complexity to high, the following options are available:
If you set codec complexity to medium, the following options are valid:
The optional bytes parameter sets the number of voice data bytes per frame. Acceptable values are from 10 to 240 in increments of 10 (for example, 10, 20, 30, and so on). Any other value is rounded down (for example, from 236 to 230). If you specify g729r8, then Internet Engineering Task Force (IETF) bit-ordering is used. For interoperability with a Cisco 2600, 3600, or AS5300 router running a Cisco IOS release earlier than Release 12.0(5)T or12.0(4)XH, you must specify the additional keyword pre-ietf after g729r8. | ||
| Router(config | (Optional) This setting is enabled by default and activates voice activity detection (VAD). VAD allows the system to reduce unnecessary voice transmissions caused by unfiltered background noise. | ||
| Router | (Optional) Dual-tone multifrequency (DTMF) describes the tone that sounds in response to a keypress on a touch-tone phone. DTMF tones are compressed at one end of a call and decompressed at the other end. If a low-bandwidth codec, such as a G.729 or G.723 is used, the tones can sound distorted. The dtmf-relay command transports DTMF tones generated after call establishment out-of-band by using a method that transmits with greater fidelity than is possible in-band for most low-bandwidth codecs. Without DTMF relay, calls established with low-bandwidth codecs can have trouble accessing automated phone menu systems, such as voice mail and interactive voice response (IVR) systems. A signaling method is supplied only if the remote end supports it, and the options are: Cisco proprietary (cisco-rtp), standard H.323 (h245-alphanumeric), and H.323 standard with signal duration (h245-signal). | ||
| Router | (Optional) Specify the transmission speed of a fax to be sent to this dial peer. disable turns off fax transmission capability, and voice specifies the highest possible fax speed supported by the voice rate. | ||
| Router | See Step 3 in this procedure. | ||
| Router | Configure the IP session target for the dial peer. ipv4:destination-address indicates IP address of the dial peer. dns:host-name indicates that the domain name server will resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device. There are also wildcards available for defining domain names with the keyword by using source, destination, and dialed information in the host name. For complete command syntax information, see Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0. |
Follow the procedure below to verify dial-peer configuration. To learn more about these commands, see Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.
Important command output is shown in bold.
Enter the privileged EXEC show dial-peer voice command. The following text is sample output from the command for a POTS dial peer:
router# show dial-peer voice 1
VoiceEncapPeer1
tag = 1, dest-pat = \Q+14085551000',
answer-address = \Q',
group = 0, Admin state is up, Operation state is down
Permission is Both,
type = pots, prefix = \Q',
session-target = \Q', voice-port =
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is "10"
Last Disconnect Text is ""
Last Setup Time = 0
The following text is sample output from the show dial-peer voice command for a VoIP dial peer:
Router# show dial-peer voice 10
VoiceOverIpPeer10
tag = 10, dest-pat = \Q',
incall-number = \Q+14087',
group = 0, Admin state is up, Operation state is down
Permission is Answer,
type = voip, session-target = \Q',
sess-proto = cisco, req-qos = bestEffort,
acc-qos = bestEffort,
fax-rate = voice, codec = g729r8,
Expect factor = 10,Icpif = 30, VAD = disabled, Poor QOV Trap = disabled,
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is "10"
Last Disconnect Text is ""
Last Setup Time = 0
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-Kb DS0 channels from one T1 or E1 and digitally cross-connect them to 64Kb DS0 channels on another T1 or E1.
Drop and Insert allows individual 64Kb 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:
This example in this section shows 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. WAN connections must be provided by separate links.
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's Multiflex VWIC ports: one from the PBX and one from the PSTN central office (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:
This example in this section 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's Multiflex VWIC ports: one from the PBX and one from the PSTN central office (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 T1 or 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 information (LMI) 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.
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 |
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 |
This configuration 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.
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, as well as VoIP traffic that originates on channels 13 through 24 from the PBX connected to the Digital T1 Packet Voice Trunk Network Module.
New or modified commands are included in the Command Reference section of Configuring Digital T1 Packet Voice Trunk Network Modules on Cisco 2600 and 3600 Series Routers. All other commands used with this feature are documented in the Cisco IOS Release 12.0 command references.
AMI---alternate mark inversion. Line-code type used on T1 and E1 circuits. In AMI, zeros are represented by 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, which is not maintained independent of the data stream. Sometimes called binary coded alternate mark inversion.
ATM---Asynchronous Transfer Mode. International standard for cell relay where multiple service types (such as voice, video, or data) are conveyed in fixed-length (53-byte) cells that allow cell processing to occur in hardware; thereby transit delays are reduced. ATM is designed to take advantage of high-speed transmission media such as E3, SONET, and T3.
B8ZS---binary 8-zero substitution. Line-code type, used on T1 and E1 circuits, in which a special code is substituted whenever 8 consecutive zeros are sent over the link. This code is then interpreted at the remote end of the connection. This technique guarantees that ones density is independent of the data stream.
CAS---channel-associated signaling. Trunk signaling (for example, in a T1 line) in which control signals, such as those for synchronizing and bounding frames, are carried in the same channel along with voice and data signals.
CCS---common channel signaling. Trunk signaling (for example, using Primary Rate Interface) in which a control channel carries signaling for separate voice and data channels.
CO---central office. Local telephone company office where all local loops in a given area connect and circuit switching of subscriber lines occurs.
codec---Coder-decoder. Device that typically uses pulse code modulation to transform analog signals into a digital bit stream and digital signals back into analog.
DTMF---Dual-tone multifrequency. Use of two simultaneous voice-band tones for dialing (such as touch tone).
Drop and Insert---(also called TDM Cross-Connect) Allows DSO channels from one T1 or E1 facility to be digitally cross-connected to DS0 channels on another T1 or E1. By using this method, channel traffic is sent between a PBX and CO PSTN switch or other telephony device, so that some PBX channels are directed 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 for external analog connectivity.
DSP---digital signal processor.
E1---European digital carrier facility used for transmitting data through the telephone hierarchy. The transmission rate for E1 is 2.048 megabits per second (Mbps).
E&M---rEceive and transMit, or Ear and Mouth. Type of signaling originally developed for analog two-state voltage telephony using the ear and mouth leads; in digital telephony, uses two bits.
ESF---Extended Superframe. Framing type used on T1 circuits that consists of 24 frames of 192 bits each with the 193rd bit providing timing and other functions. ESF is an enhanced version of SF format.
FDL---Facility Data Link. A 4-Kbps channel, provided by the Extended SuperFrame (ESF) T1 framing format. The FDL performs outside the payload capacity and allows a service provider to check error statistics on terminating equipment, without intrusion.
FXO---Foreign Exchange Office. A voice interface emulating a PBX trunk line to a switch or telephone equipment to a PBX extension interface.
FXS---Foreign Exchange Station. A voice interface for connecting telephone equipment; it emulates the extension interface of a PBX or the subscriber interface for a switch.
HBD3---High-Density Bipolar 3. Line code type used on E1 circuits.
IETF---Internet Engineering Task Force.
ISDN---Integrated Services Digital Network. Communication protocol offered by telephone companies. ISDN permits telephone networks to carry data, voice, and other source traffic.
packet---Logical grouping of information that includes a header containing control information and (usually) user data. Packets are most often used to refer to network layer units of data.
POTS---plain old telephone service.
PDVM---packet data voice module.
PSTN---Public Switched Telephone Network. General term referring to the variety of telephone networks and services in place worldwide.
QoS---quality of service. Measure of performance for a transmission system that reflects its transmission quality and service availability.
SF---Super Frame. Common framing type used on T1 circuits. SF consists of 12 frames of 192 bits each with the 193rd bit providing error checking and other functions. SF is superseded by ESF, but is still widely used. Also called D4 framing.
SNMP---Simple Network Management Protocol. Network management protocol used almost exclusively in TCP/IP networks. SNMP provides a means to monitor and control network devices, and to manage configurations, statistics collection, performance, and security.
T1---Digital WAN carrier facility. T1 transmits DS 1-formatted data at 1.544 Mbps through the telephone switching network by using alternate mark inversion or B8ZS coding.
T1 trunk---Digital WAN carrier facility. See T1.
TDM---time-division multiplexing.
Trunk---Physical and logical connection between two switches across which network traffic travels. A backbone is composed of a number of trunks.
UNI---User-Network Interface. ATM Forum specification that defines an interoperability standard for the interface between ATM-based products (a router or an ATM switch) located in a private network, and the ATM switches located within the public carrier networks. Also used to describe similar connections in Frame Relay networks.
VAD---voice activity detection.
Posted: Wed Mar 1 15:05:57 PST 2000
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