Table Of Contents
Digital E1 Packet Voice Trunk Network Module Interfaces
Supported Standards, MIBs, and RFCs
Configuring Voice Card and E1 Controller Settings
E1 Timing, Signaling, Framing, and Line Encoding
Verifying Voice Card and Controller Settings
Verifying Serial Interface Configuration
Monitoring and Maintaining E1 Digital Packet Voice Configuration
Digital E1 Packet Voice Trunk Network Module Interfaces
This document describes how to configure digital E1 packet voice trunk network module interfaces on Cisco 2600 and 3600 series routers and includes the following sections:
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Supported Standards, MIBs, and RFCs
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Feature Overview
Digital E1 packet voice trunk network modules for Cisco 2600 and 3600 series routers allow enterprises or service providers, using the equipped routers as customer premises equipment, to deploy digital voice and fax relay. These modules receive constant bit-rate telephony information over E1 interfaces and can convert that information to a compressed format, so that it can be transmitted as Voice over IP (VoIP), Voice over Frame Relay (VoFR), and Voice over ATM (VoATM).
Benefits
Digital E1 packet voice trunk network modules allow Cisco 2600 and 3600 series routers to provide E1 connectivity to private branch exchanges (PBXs) or to a central office (CO). With digital E1 connectivity, Cisco 2600 and 3600 series routers can provide greater voice density for enterprise and service provider VoIP networks. A digital E1 packet voice trunk network module is a complete solution, made up of a network module with installed packet voice data modules (PVDMs), and one E1 multiflex trunk voice/WAN interface card with either one or two E1 ports.
Restrictions
The following restrictions apply to digital E1 packet voice trunk network module configuration:
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Related Documents
The following documents can help you understand how to install Cisco 2600 and 3600 series routers:
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The following Cisco IOS Release 12.1 documents are also helpful:
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The following documents can help you troubleshoot ISDN, PRI, and BRI connections:
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For information about supported hardware on a Cisco 2600 or 3600 series router, go to the following URLs:
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http://www.cisco.com/univercd/cc/td/doc/product/access/acs_mod/cis3600/voice/4936vqsg.htmSupported Platforms
This feature is supported on the following platforms:
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Supported Standards, MIBs, and RFCs
MIBs
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Standards
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RFCs
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Prerequisites
Digital E1 packet voice capability requires specific service, software, and hardware:
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For high-volume applications, the memory required might be greater than these minimum values.
Support for digital E1 packet voice trunk network modules is included in Plus feature sets. The IP Plus feature set requires 16 MB of Flash memory.
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For Drop-and-Insert capability, you must install a two-port Drop-and-Insert E1 multiflex trunk voice/WAN interface card (VWIC-2MFT-E1-DI). To install a VWIC in a network module, see Cisco WAN Interface Cards Hardware Installation Guide.
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Cisco IOS Multiservice Applications Configuration Guide and Cisco IOS Multiservice Applications Command Reference for Cisco IOS Release 12.1 provide information about setting up voice networks.
Configuration Tasks
Complete the following tasks to configure a digital E1 packet voice trunk network module:
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Configuring Voice Card and E1 Controller Settings
The following steps specify codec settings for voice cards 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.
Step 1
Router# configure terminal
Enter global configuration mode.
Step 2
Router(config)# voice-card slot
Enter voice card configuration mode and specify the slot location by using a value from 0 to 5, depending on your router.
Step 3
Router(config-voice-ca)# codec complexity {high | medium}
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 PVDMs installed and the codec complexity. Here is a guideline:
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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" on page 19.
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Step 4
Router(config-voice-ca)# Exit
Exit the voice card configuration mode.
Step 5
Router(config)# controller E1 slot/port
Enter controller configuration mode for the E1 controller at the specified slot/port location. Valid values for slot and port are 0 and 1.
Step 6
Router(config-controller)# clock source {line [primary] | internal}
Configure 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:
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See "Verifying Voice Card and Controller Settings" on page 14, for more information about configurations for clocking.
Step 7
Router(config-controller)# framing crc4
Set the framing according to your service provider's instructions. Choose cyclic redundancy check 4 (CRC4) format.
Step 8
Router(config-controller)# linecode hdb3
Set the line encoding according to your service provider's instructions. E1 uses high density bipolar 3 (HDB3) encoding (similar to alternative mark inversion, or AMI).
Step 9
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cablelength short {133 | 266 | 399 | 533 | 655}(E1 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 E1 link, the command is rejected.
To set a cable length longer than 655 feet for an E1 link, use the cablelength long command. The keywords are as follows:
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To set a cable length 655 feet or less for an E1 link, use the cablelength short command. There is no default for cablelength short. The keywords are as follows:
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If you do not set the cable length, the system defaults to a setting of cablelength long gain26 0db.
Step 10
Enter a single timeslot number and a single range of values. For E1, the allowable values are from 1 to 31.
Step 11
Activate the controller.
Step 12
Exit controller configuration mode.
E1 Timing, Signaling, Framing, and Line Encoding
With the introduction of the digital E1 packet voice trunk network modules for the Cisco 2600 and 3600 series routers, you must set timing, signaling, framing, and line encoding. Digital E1 packet voice trunk network modules can connect to either a PBX (or similar telephony device) or to a central office (CO) to provide PSTN connectivity.
The differences that set E1 digital configuration apart from analog configuration are as follows:
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Timing
This section describes the five basic timing scenarios that can occur when a digital E1 packet voice trunk network module is connected to a PBX, CO, or both. In all of the examples below, the PSTN (or CO) and the PBX are interchangeable for purposes of providing or receiving clocking.
The digital E1 module has an onboard Phase-Lock Loop (PLL) chip that can either provide a clock source to both E1s or receive clocking that can drive the second E1 in the same digital E1 packet voice trunk network module. All timing commands are E1 controller configuration commands.
Single E1 Port Provides Clocking
In this scenario, the digital E1 module is the clock source for the connected device. The PLL generates the clock internally and drives the clocking on the E1 line.
Figure 1 Single E1 Port Providing Clock
The following configuration sets up this clocking method:
controller E1 1/0framing crc4linecoding hdb3clock source internalpri-group timeslots 1-31
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Single E1 Port Receiving Clock from the Line
In this scenario, the digital E1 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 E1 connection.
Figure 2 Single E1 Receiving Clock from Line
The following configuration sets up this clocking method:
controller E1 1/0framing crc4linecoding b8zsclock source linepri-group timeslots 1-31Dual E1s Receiving Clocking from the Line
In this scenario, the digital E1 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 pertain to the clocking method:
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The router can usually handle controlled slips because its single PLL architecture anticipates them.
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Figure 3 Dual E1s Receiving Line Clocking
In this scenario, the PLL derives clocking from the CO and puts the E1 port connected to the PBX into looped-time mode. This is usually the best method 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 E1 port.
The following configuration sets up this clocking method:
controller E1 1/0 << description - connected to the COframing crc4linecoding hdb3clock source line primarypri-group timeslots 1-31!controller E1 1/1 << description - connected to the PBXframing crc4linecoding hdb3clock source linepri-group timeslots 1-31The clock source line primary command tells the router to use this E1 port to drive the PLL. All other E1 ports configured as clock source line are then put into an implicit loop-timed mode. If the primary E1 port fails or goes down, the other E1 instead receives the clock that drives the PLL. In this configuration, E1 port 1/1 might see controlled slips, but these should not force it down. This method prevents the PBX from seeing slips.
Dual E1s—One Receives Clocking and One Provides Clocking
In this scenario, the digital E1 module receives clocking for the PLL from E1 0 and uses this clock as a reference to clock E1 1. If E1 0 fails, the PLL internally generates the clock reference to drive E1 1.
Figure 4 Dual E1s—One Receiving and One Providing Clocking
The following configuration sets up this clocking method:
controller E1 1/0framing crc4linecoding hdb3clock source linepri-group timeslots 1-31!controller E1 1/1framing crc4linecoding hdb3clock source internalpri-group timeslots 1-31Dual E1s—Both Clocks from Router
In this scenario, the router generates the clock for the PLL and therefore for both E1s.
Figure 5 Dual E1s—Both Clocks from Router
The following configuration sets up this clocking method:
controller E1 1/0framing crc4linecoding hdb3clock source internalpri-group timeslots 1-31!controller E1 1/1framing esflinecoding b8zsclock source internalpri-group timeslots 1-31Signaling
There are three types of signaling that you should consider for digital E1:
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controller E1 1/0ds0-group 1 timeslots 1-24 type e&m-wink-start
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controller E1 1/0ds0-group 1 timeslots 1-24 type fxo-ground-startor
controller E1 1/0ds0-group 1 timeslots 1-24 type fxs-loop-start
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Framing
Digital E1 packet voice trunk port adapters support two types of framing for E1 CAS: Extended Superframe (ESF) and Super Frame (SF), also called D4 framing. The framing type of the router and switch (CO or PBX) must match. The framing controller configuration command defines E1 framing, as in the following example:
controller E1 1/0framing esfor
controller E1 1/0framing sfLine Encoding
Digital E1 packet voice trunk port adapters support two types of line encoding for E1 CAS: B8ZS (bipolar-8 zero substitution) and AMI (alternate mark inversion). The line encoding of the router and switch (CO or PBX) must match. The linecoding controller configuration command defines E1 framing, as in the following example:
controller E1 1/0linecoding b8zsor
controller E1 1/0linecoding amiVerifying Voice Card and Controller Settings
To verify the configuration of voice card and controller settings, perform the following steps:
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Router# show running-config...hostname router-alphavoice-card 1codec complexity high...Step 2
Router# show controller E1 1/0E1 1/0 is up.Applique type is Channelized E1Cablelength is short 133Description: E1 WIC card AlphaNo alarms detected.Framing is CRC4, Line Code is HDB3, Clock Source is Line Primary.Data in current interval (1 seconds elapsed):0 Line Code Violations, 0 Path Code Violations0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
Configuring Serial Interfaces
The way you set up serial and LAN interfaces depends on your application. 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.
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 12.1 Cisco IOS Interface Configuration Guide and the Cisco IOS Interface Command Reference.
The "Configuration Example" section shows a sample configuration. For more information about setting up voice networks, see Cisco IOS Multiservice Applications Configuration Guide for Cisco IOS Release 12.1.
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Verifying Serial Interface Configuration
To verify serial interface configuration, enter the show interfaces serial privileged EXEC command, which displays the status of all serial interfaces or of a specific serial interface, as shown in the following example. You can use this command to check the encapsulation, IP addressing, and other settings:
Router# show interface serial0/0:0Serial0/0:0 is up, line protocol is upHardware is QUICC SerialInternet address is 1.156.1.1/24MTU 1500 bytes, BW 1536 Kbit, DLY 20000 usec,reliability 255/255, txload 1/255, rxload 1/255Encapsulation HDLC, loopback not setKeepalive not setLast input 00:00:00, output 00:00:00, output hang neverLast clearing of "show interface" counters neverInput queue: 0/75/0 (size/max/drops); Total output drops: 0Queueing strategy: weighted fairOutput 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/sec5 minute output rate 1000 bits/sec, 1 packets/sec637 packets input, 64736 bytes, 0 no bufferReceived 181 broadcasts, 0 runts, 5 giants, 0 throttles3617 input errors, 1506 CRC, 1646 frame, 0 overrun, 0 ignored, 0 abort682 packets output, 67213 bytes, 0 underruns0 output errors, 0 collisions, 1070 interface resets0 output buffer failures, 0 output buffers swapped out13 carrier transitionsTimeslot(s) Used:1-24, Transmitter delay is 0 flagsConfiguring Voice Ports
Follow these steps to set up voice ports to support the local and remote stations. Not all possible commands are shown here. To learn more, see Cisco IOS Multiservice Applications Configuration Guide and Cisco IOS Multiservice Applications Command Reference for Cisco IOS Release 12.1.
Verifying Voice Ports
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. Important command output is shown in bold.
Router# show voice port 1/0:1receive and transMit Slot is 1, Sub-unit is 0, Port is 1 << voice-port 1/0:1Type of VoicePort is E&MOperation State is DORMANTAdministrative State is UPNo Interface Down FailureDescription is not setNoise Regeneration is enabledNon Linear Processing is enabledMusic On Hold Threshold is Set to -38 dBmIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is enabledEcho Cancel Coverage is set to 8 msConnection Mode is normalConnection Number is not setInitial Time Out is set to 10 sInterdigit Time Out is set to 10 sRegion Tone is set for USConfiguring Voice Dial Peers
Follow these steps to set up voice dial peers to support the local and remote stations. Not all possible commands are shown here. To learn more, see Cisco IOS Multiservice Applications Configuration Guide and Cisco IOS Multiservice Applications Command Reference for Cisco IOS Release 12.1.
Step 1
Enter global configuration mode.
Step 2
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.
Step 3
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 telephone number. Valid entries are the digits 0 through 9 and the letters A through D. The plus symbol (+) is not valid. The following special characters can be entered:
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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 #).
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Step 4
(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.
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Step 5
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 E1 card.
Step 6
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.
Step 7
Router(config-dialpeer)# codec {g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g728 | g729r8 [pre-ietf] | g729br8} [bytes]The voice-card configuration codec complexity command sets the codec options that are available when you enter this command. See Step 3 of the "Configuring Voice Card and E1 Controller Settings" section.
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:
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If you set codec complexity to medium, the following options are valid:
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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 the IETF (Internet Engineering Task Force) bit-ordering is used. For interoperability with a Cisco 2600, 3600, or AS5300 router running a Cisco IOS release prior to Release 12.0(5)T or12.0(4)XH, you must specify the additional keyword pre-ietf after g729r8.
Step 8
(Optional) This setting is enabled by default. It activates voice activity detection (VAD). VAD allows the system to reduce unnecessary voice transmissions caused by unfiltered background noise.
Step 9
(Optional) Dual-tone multifrequency (DTMF) describes the tone that sounds in response to a keypress on a touch-tone telephone. 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 may have trouble accessing automated telephone menu systems, such as voicemail 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).
Step 10
(Optional) Specify the transmission speed of a fax to be sent to this dial peer. The disable keyword turns off fax transmission capability, and the voice keyword specifies the highest possible fax speed supported by the voice rate.
Step 11
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 Cisco IOS Multiservice Applications Command Reference for Cisco IOS Release 12.1.
Step 12
Configure the interface to forward digits for voice calls.
Step 13
(Optional) Disable hunting by the interface for dial peers.
Step 14
Exit interface configuration.
Verifying Voice Dial Peers
Enter the privileged EXEC show dial-peer voice command. The following text is sample output from the command for a POTS dial peer. Important command output is shown in bold.
Router# show dial-peer voice 1VoiceEncapPeer1tag = 1, dest-pat = \Q+14085551000',answer-address = \Q',group = 0, Admin state is up, Operation state is downPermission is Both,type = pots, prefix = \Q',session-target = \Q', voice-port =Connect Time = 0, Charged Units = 0Successful Calls = 0, Failed Calls = 0Accepted Calls = 0, Refused Calls = 0Last Disconnect Cause is "10"Last Disconnect Text is ""Last Setup Time = 0The following text is sample output from the show dial-peer voice command for a VoIP dial peer:
Router# show dial-peer voice 10VoiceOverIpPeer10tag = 10, dest-pat = \Q',incall-number = \Q+14087',group = 0, Admin state is up, Operation state is downPermission 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 = 0Successful Calls = 0, Failed Calls = 0Accepted Calls = 0, Refused Calls = 0Last Disconnect Cause is "10"Last Disconnect Text is ""Last Setup Time = 0Monitoring and Maintaining E1 Digital Packet Voice Configuration
This section presents some useful commands for understanding, maintaining, and troubleshooting your configuration. Table 1 lists the debug and show commands.
Table 1 Debug and Show Commands for Maintaining and Troubleshooting Your Configuration
Router# show dialplan number number
Shows which dial-peer is matched by a called number.
Router# show call active voice
Shows statistics for currently active voice calls.
Router# show call active fax
Shows statistics for currently active fax calls.
Router# show call history voice
Shows statistics on previous voice calls.
Router# show call history fax
Shows statistics on previous fax calls.
Router# show voice port
Shows the status of voice ports. See "Verifying Voice Ports" on page 18.
Router# show controller E1 slot/port
Shows the status of the E1 controller. See "Verifying Voice Card and Controller Settings" on page 14.
Router# show isdn status
Shows the status of an individual ISDN line.
Router# debug ccapi inout
Debugs the E1
Router# debug isdn q931
Debugs calls as they are set up and torn down on ISDN network connections (Layer 3) between the local router (user side) and the network.
Router# debug vpm all
Debugs the E1 signaling.
Router# debug vtsp all
Debugs the digits received and sent.
Router# debug voip ccapi inout
Debugs the call setup process.
Show Commands
This section illustrates some of the privileged EXEC show commands that are useful for analyzing your system. Note that important information appears in bold.
The show dialplan number command provides information about the dial peer associated with a specified dial-plan number. Notice that the dial peer is operational and that IP Precedence has been configured to the preferred setting of 5.
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Router# show dialplan number 75435Macro Exp.: ##75435VoiceOverIpPeer70000information type = voice,tag = 70000, destination-pattern = `##7....',answer-address = `', preference=0,group = 70000, Admin state is up, Operation state is up,incoming called-number = `', connections/maximum = 0/unlimited,DTMF Relay = disabled,application associated:type = voip, session-target = `ipv4:171.68.253.18',technology prefix:settlement: disabledip precedence = 5, UDP checksum = disabled,session-protocol = cisco, req-qos = best-effort,acc-qos = best-effort,fax-rate = 14400, payload size = 20 bytescodec = g729r8, payload size = 20 bytes,Expect factor = 10, Icpif = 30,signaling-type = cas,VAD = disabled, Poor QOV Trap = disabled,Connect Time = 0, Charged Units = 0,Successful Calls = 3, Failed Calls = 0,Accepted Calls = 3, Refused Calls = 0,Last Disconnect Cause is "10 ",Last Disconnect Text is "normal call clearing.",Last Setup Time = 344813.Matched: ##75435 Digits: 3Target: ipv4:171.68.253.18The show call active voice command displays information about a current call:
Router# show call active voiceGENERIC:SetupTime=94523746 msIndex=448PeerAddress=##73072PeerSubAddress=PeerId=70000PeerIfIndex=37LogicalIfIndex=0ConnectTime=94524043DisconectTime=94546241CallOrigin=1ChargedUnits=0InfoType=2TransmitPackets=6251TransmitBytes=125020ReceivePackets=3300ReceiveBytes=66000VOIP:ConnectionId[0x142E62FB 0x5C6705AF 0x0 0x385722B0]RemoteIPAddress=171.68.235.18RemoteUDPPort=16580RoundTripDelay=29 msSelectedQoS=best-efforttx_DtmfRelay=inband-voiceSessionProtocol=ciscoSessionTarget=ipv4:171.68.235.18OnTimeRvPlayout=63690GapFillWithSilence=0 msGapFillWithPrediction=180 msGapFillWithInterpolation=0 msGapFillWithRedundancy=0 msHiWaterPlayoutDelay=70 msLoWaterPlayoutDelay=30 msReceiveDelay=40 msLostPackets=0 msEarlyPackets=1 msLatePackets=18 msVAD = disabledCoderTypeRate=g729r8CodecBytes=20cvVoIPCallHistoryIcpif=0SignalingType=casThe show call history voice command shows statistics about previous calls:
Router# show call history voiceGENERIC:SetupTime=94893250 msIndex=450PeerAddress=##52258PeerSubAddress=PeerId=50000PeerIfIndex=35LogicalIfIndex=0DisconnectCause=10DisconnectText=normal call clearing.ConnectTime=94893780DisconectTime=95015500CallOrigin=1ChargedUnits=0InfoType=2TransmitPackets=32258TransmitBytes=645160ReceivePackets=20061ReceiveBytes=401220VOIP:ConnectionId[0x142E62FB 0x5C6705B3 0x0 0x388F851C]RemoteIPAddress=171.68.235.18RemoteUDPPort=16552RoundTripDelay=23 msSelectedQoS=best-efforttx_DtmfRelay=inband-voiceSessionProtocol=ciscoSessionTarget=ipv4:171.68.235.18OnTimeRvPlayout=398000GapFillWithSilence=0 msGapFillWithPrediction=1440 msGapFillWithInterpolation=0 msGapFillWithRedundancy=0 msHiWaterPlayoutDelay=97 msLoWaterPlayoutDelay=30 msReceiveDelay=49 msLostPackets=1 msEarlyPackets=1 msLatePackets=132 msVAD = disabledCoderTypeRate=g729r8CodecBytes=20cvVoIPCallHistoryIcpif=0SignalingType=casThe show isdn status command shows the status of ISDN calls:
Router# show isdn statusGlobal ISDN Switchtype = primary-qsigISDN Serial1/015 interface******* Network side configuration *******dsl 0, interface ISDN Switchtype = primary-qsig**** Master side configuration ****Layer 1 StatusACTIVELayer 2 StatusTEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHEDLayer 3 Status24 Active Layer 3 Call(s)Activated dsl 0 CCBs = 24CCBcallid=E3C, sapi=0, ces=0, B-chan=1, calltype=VOICECCBcallid=E3D, sapi=0, ces=0, B-chan=2, calltype=VOICECCBcallid=E3E, sapi=0, ces=0, B-chan=3, calltype=VOICECCBcallid=E3F, sapi=0, ces=0, B-chan=4, calltype=VOICECCBcallid=E40, sapi=0, ces=0, B-chan=5, calltype=VOICECCBcallid=E47, sapi=0, ces=0, B-chan=6, calltype=VOICECCBcallid=E48, sapi=0, ces=0, B-chan=7, calltype=VOICECCBcallid=E49, sapi=0, ces=0, B-chan=8, calltype=VOICECCBcallid=E50, sapi=0, ces=0, B-chan=9, calltype=VOICECCBcallid=E51, sapi=0, ces=0, B-chan=10, calltype=VOICECCBcallid=E52, sapi=0, ces=0, B-chan=11, calltype=VOICECCBcallid=E53, sapi=0, ces=0, B-chan=12, calltype=VOICECCBcallid=E54, sapi=0, ces=0, B-chan=13, calltype=VOICECCBcallid=E5B, sapi=0, ces=0, B-chan=14, calltype=VOICECCBcallid=E5C, sapi=0, ces=0, B-chan=15, calltype=VOICECCBcallid=E5D, sapi=0, ces=0, B-chan=17, calltype=VOICECCBcallid=E5E, sapi=0, ces=0, B-chan=18, calltype=VOICECCBcallid=E5F, sapi=0, ces=0, B-chan=19, calltype=VOICECCBcallid=E60, sapi=0, ces=0, B-chan=20, calltype=VOICECCBcallid=E61, sapi=0, ces=0, B-chan=21, calltype=VOICECCBcallid=E62, sapi=0, ces=0, B-chan=22, calltype=VOICECCBcallid=E63, sapi=0, ces=0, B-chan=23, calltype=VOICECCBcallid=E64, sapi=0, ces=0, B-chan=24, calltype=VOICECCBcallid=E6B, sapi=0, ces=0, B-chan=25, calltype=VOICEThe Free Channel Mask 0xFE000000Total Allocated ISDN CCBs = 24The show dial-peer voice summary command displays information about dial-peers that are active:
Router# show dial-peer voice summarydial-peer hunt 0TAG TYPE ADMIN OPER PREFIX DEST-PATTERN PREF SESS-TARGET PORT1 pots up up 3 0 1/015100 voip down down 1 0 ipv41.2.79.7200 voip down down 1 0 ipv41.2.79.31300 vofr up up 1 0 Serial0/0 990400 voip down down 1 0 ipv45.5.5.2The show voice call summary command displays a summary of all dial-peers that are active:
Router# show voice call summaryPORT CODEC VAD VTSP STATE VPM STATE========= ======== === ===================== ========================1/015.1 g729r8 y S_CONNECT S_TSP_CONNECT1/015.2 g729r8 y S_CONNECT S_TSP_CONNECT1/015.3 g729r8 y S_CONNECT S_TSP_CONNECT1/015.4 g729r8 y S_CONNECT S_TSP_CONNECT1/015.5 g729r8 y S_CONNECT S_TSP_CONNECT1/015.6 g729r8 y S_CONNECT S_TSP_CONNECT1/015.7 g729r8 y S_CONNECT S_TSP_CONNECT1/015.8 g729r8 y S_CONNECT S_TSP_CONNECT1/015.9 g729r8 y S_CONNECT S_TSP_CONNECT1/015.10 g729r8 y S_CONNECT S_TSP_CONNECT1/015.11 g729r8 y S_CONNECT S_TSP_CONNECT1/015.12 g729r8 y S_CONNECT S_TSP_CONNECT1/015.13 g729r8 y S_CONNECT S_TSP_CONNECT1/015.14 g729r8 y S_CONNECT S_TSP_CONNECT1/015.15 g729r8 y S_CONNECT S_TSP_CONNECT1/015.17 g729r8 y S_CONNECT S_TSP_CONNECT1/015.18 g729r8 y S_CONNECT S_TSP_CONNECT1/015.19 g729r8 y S_CONNECT S_TSP_CONNECT1/015.20 g729r8 y S_CONNECT S_TSP_CONNECT1/015.21 g729r8 y S_CONNECT S_TSP_CONNECT1/015.22 g729r8 y S_CONNECT S_TSP_CONNECT1/015.23 g729r8 y S_CONNECT S_TSP_CONNECT1/015.24 g729r8 y S_CONNECT S_TSP_CONNECT1/015.25 g729r8 y S_CONNECT S_TSP_CONNECTThe show voice dsp command displays current status of all DSP voice channels:
Router# show voice dspBOOT PAKTYPE DSP CH CODEC VERS STATE STATE RST AI PORT TS ABORT TX/RX-PAK-CNT==== === == ======== ==== ===== ======= === == ======= == ===== ===============C549 010 00 g729r8 3.3 busy idle 0 0 1/015 1 0 67400/8538401 g729r8 .8 busy idle 0 0 1/015 7 0 67566/8362302 g729r8 busy idle 0 0 1/015 13 0 65675/8185103 g729r8 busy idle 0 0 1/015 20 0 65530/83610C549 011 00 g729r8 3.3 busy idle 0 0 1/015 2 0 66820/8479901 g729r8 .8 busy idle 0 0 1/015 8 0 59028/6694602 g729r8 busy idle 0 0 1/015 14 0 65591/8108403 g729r8 busy idle 0 0 1/015 21 0 66336/82739C549 012 00 g729r8 3.3 busy idle 0 0 1/015 3 0 59036/6524501 g729r8 .8 busy idle 0 0 1/015 9 0 65826/8195002 g729r8 busy idle 0 0 1/015 15 0 65606/8073303 g729r8 busy idle 0 0 1/015 22 0 65577/83532C549 013 00 g729r8 3.3 busy idle 0 0 1/015 4 0 67655/8297401 g729r8 .8 busy idle 0 0 1/015 10 0 65647/8208802 g729r8 busy idle 0 0 1/015 17 0 66366/8089403 g729r8 busy idle 0 0 1/015 23 0 66339/82628C549 014 00 g729r8 3.3 busy idle 0 0 1/015 5 0 68439/8467701 g729r8 .8 busy idle 0 0 1/015 11 0 65664/8173702 g729r8 busy idle 0 0 1/015 18 0 65607/8182003 g729r8 busy idle 0 0 1/015 24 0 65589/83889C549 015 00 g729r8 3.3 busy idle 0 0 1/015 6 0 66889/8333101 g729r8 .8 busy idle 0 0 1/015 12 0 65690/8170002 g729r8 busy idle 0 0 1/015 19 0 66422/8209903 g729r8 busy idle 0 0 1/015 25 0 65566/83852The show voice trace command displays a trace of all active voice transitions:
Router# show voice trace1/015 1 State Transitions (state, event) -> (state, event) ...(S_NULL, E_TSP_INFO_IND) -> (S_SETUP_INDICATED, E_TSP_INFO_IND) ->(S_SETUP_INDICATED, E_TSP_INFO_IND) -> (S_SETUP_INDICATED, E_CC_PROCEEDING) ->(S_SETUP_INDICATED, E_CC_ALERT) -> (S_ALERTING, E_CC_BRIDGE) ->(S_ALERTING, E_CC_CONNECT) -> (S_CONNECT, E_CC_CAPS_IND) ->(S_CONNECT, E_CC_CAPS_ACK) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_TIMER) ->The show adapi command displays information about the call distribution application programming interface (CDAPI):
Router# show cdapiRegistered CDAPI Applications/Stacks====================================Application TSP CDAPI Application VoiceApplication Type(s) Voice Facility SignalingApplication Level TunnelApplication Mode EnblocSignaling Stack ISDNInterface Se1/015CDAPI Message Buffers=====================Used Msg Buffers 0, Free Msg Buffers 6400Used Raw Buffers 0, Free Raw Buffers 3200Used Large-Raw Buffers 0, Free Large-Raw Buffers 3202600-1#2600-1#2600-1#s vo call 1/015.11/015 1 vtsp level 0 state = S_CONNECTcallid 0x0EDE B01 state S_TSP_CONNECT clld 1 cllg 34565463472600-1# ***DSP VOICE VP_DELAY STATISTICS***Clk Offset(ms) -383401219, Rx Delay Est(ms) 61Rx Delay Lo Water Mark(ms) 61, Rx Delay Hi Water Mark(ms) 90***DSP VOICE VP_ERROR STATISTICS***Predict Conceal(ms) 0, Interpolate Conceal(ms) 0Silence Conceal(ms) 0, Retroact Mem Update(ms) 0Buf Overflow Discard(ms) 20, Talkspurt Endpoint Detect Err 0***DSP VOICE RX STATISTICS***Rx Vox/Fax Pkts 286, Rx Signal Pkts 0, Rx Comfort Pkts 0Rx Dur(ms) 24870, Rx Vox Dur(ms) 8510, Rx Fax Dur(ms) 0Rx Non-seq Pkts 0, Rx Bad Hdr Pkts 0Rx Early Pkts 0, Rx Late Pkts 0***DSP VOICE TX STATISTICS***Tx Vox/Fax Pkts 826, Tx Sig Pkts 0, Tx Comfort Pkts 0Tx Dur(ms) 24870, Tx Vox Dur(ms) 24790, Tx Fax Dur(ms) 0***DSP VOICE ERROR STATISTICS***Rx Pkt Drops(Invalid Header) 0, Tx Pkt Drops(HPI SAM Overflow) 0***DSP LEVELS***TDM Bus Levels(dBm0) Rx -12.5 from PBX/Phone, Tx -13.2 to PBX/PhoneTDM ACOM Levels(dBm0) +0.0, TDM ERL Level(dBm0) +23.5TDM Bgd Levels(dBm0) -12.1, with activity being voiceDebug Commands
This section illustrates some of the EXEC mode debug commands that are useful when analyzing and troubleshooting your system. Note that important information appears in bold.
The debug isdn q931 command displays information about call setup and teardown of ISDN network connections (Layer 3) between the local router (user side) and the network.
The debug voip ccapi inout EXEC command traces the execution path through the call control API, which serves as the interface between the call-session application and the underlying network-specific software.
During the capabilities exchange shown in the command output, both sides agree on what compression to use, and the debug voip ccapi inout output helps you determine what each side is negotiating.
This command shows how a call flows through the system. By using this debug level, you can see the call setup and teardown operations performed on both the telephony and network call legs:
Router# debug isdn q931Router# debug voip ccapi inout001041 ISDN Se1/015 RX <- SETUP pd = 8 callref = 0x1EC5 << the originating call001041 Sending Complete001041 Bearer Capability i = 0x8090A3001041 Channel ID i = 0xA98381001041 Calling Party Number i = 0x91, '0987654321'001041 Calling Party SubAddr i = 0x80, 'P123'001041 Called Party Number i = 0x91, '2312'001041 Called Party SubAddr i = 0x80, 'P321'001041 High Layer Compat i = 0x9181001041 Locking Shift to Codeset 5001041 Codeset 5 IE 0x31 i = 0x80001041 Codeset 5 IE 0x32 i = 0x800010180388626431 vtsp_tsp_call_setup_ind (sdb=0x81A57008, tdm_info=0x0,...0029107374182399 ISDN BR1/0 TX -> SETUP pd = 8 callref = 0x0001 << terminating call0029105245511244 Bearer Capability i = 0x8090A30029103079215104 Channel ID i = 0xA983810029103079215104 Calling Party Number i = 0x91, '0987654321'0029103079215104 Calling Party SubAddr i = 0x80, 'P123'0029103079215104 Called Party Number i = 0x91, '312'0029103079215104 Called Party SubAddr i = 0x80, 'P321'0029103079215104 Sending Complete0029103079215104 High Layer Compat i = 0x91810029103079215104 Locking Shift to Codeset 50029105245510852 Codeset 5 IE 0x31 i = 0x800029103079215104 Codeset 5 IE 0x32 i = 0x80002925 ISDN BR1/0 RX <- RELEASE_COMP pd = 8 callref = 0x8001002925 Cause i = 0x8096 - Number changed002925 Facility i = 0x91A4053132333435002925 User-User i = 0x08, 'USER', 0x20, 'INFORMATION'...003234359738368 Channel ID i = 0xA98381003234359738368 Calling Party Number i = 0x91, '0987654321'003234359738368 Calling Party SubAddr i = 0x80, 'P123'003234359738368 Called Party Number i = 0x91, '312'003234359738368 Called Party SubAddr i = 0x80, 'P321'003234359738368 Sending Complete003234359738368 High Layer Compat i = 0x9181003234359738368 Locking Shift to Codeset 5003236526034116 Codeset 5 IE 0x31 i = 0x80003234359738368 Codeset 5 IE 0x32 i = 0x80003209 ISDN BR1/0 RX <- CALL_PROC pd = 8 callref = 0x8003003209 Channel ID i = 0xA98381003224 ISDN BR1/0 RX <- PROGRESS pd = 8 callref = 0x8003003224 Progress Ind i = 0x8181 - Call not end-to-end ISDN, may havein-bandinfoTable 2 explains the codec negotiation values that appear—in hexadecimal format— during the capabilities exchange portion of the command output.
Reference Information
The information in this section helps you interpret the output from debug and show commands.
Table 3 shows Q.931 call disconnection causes. In the examples that follow, the disconnects are caused by normal call clearing.
These are codec capabilities bits that can appear in command output:
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These are fax capabilities bits that can appear in command output. The numbers following "FAX_" refer to the fax speed (for example, "144" means 14,400 bps):
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These are the VAD on and off capability bits:
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Configuration Example
This section displays the configuration example of a router running a digital E1 packet voice trunk network module interface:
Router#show running-configBuilding configuration...Current configuration!version 12.1service timestamps debug uptimeservice timestamps log uptimeno service password-encryption!hostname 2600-1!!!memory-size iomem 10voice-card 1!ip subnet-zerono ip domain-lookup!frame-relay switchingisdn switch-type primary-qsigisdn voice-call-failure 0voice hunt user-busy!!!controller E1 1/0pri-group timeslots 1-31!controller E1 1/1shutdown!!!interface Ethernet0/0ip address 1.2.79.1 255.255.0.0no ip directed-broadcastno cdp enable!interface Serial0/0no ip addressno ip directed-broadcastencapsulation frame-relayno ip mroute-cacheload-interval 30clockrate 800000frame-relay traffic-shapingframe-relay class voice-vcframe-relay interface-dlci 990vofr data 4 call-control 5frame-relay intf-type dce!interface Ethernet0/1no ip addressno ip directed-broadcastshutdownno cdp enable!interface Serial0/1ip address 5.5.5.1 255.0.0.0no ip directed-broadcastencapsulation frame-relayno ip mroute-cacheclockrate 800000frame-relay traffic-shapingframe-relay class voice-dataframe-relay interface-dlci 991frame-relay ip rtp header-compressionframe-relay intf-type dce!interface Serial1/015no ip addressno ip directed-broadcastip mroute-cacheno logging event link-statusisdn switch-type primary-qsigisdn overlap-receivingisdn protocol-emulate networkisdn incoming-voice voiceno isdn T309-enableisdn bchan-number-order ascendingfair-queue 64 256 0no cdp enable!router ripnetwork 172.28.0.0!router igrp 1redistribute connectednetwork 1.0.0.0!ip default-gateway 1.2.0.1ip classlessip route 223.255.254.254 255.255.255.255 1.2.0.1no ip http server!!map-class frame-relay voice-vcno frame-relay adaptive-shapingframe-relay cir 512000frame-relay bc 512000frame-relay fair-queueframe-relay voice bandwidth 512000frame-relay fragment 100!map-class frame-relay voice-datano frame-relay adaptive-shapingframe-relay cir 512000frame-relay bc 1000frame-relay fair-queueframe-relay fragment 200frame-relay ip rtp priority 2000 16383 500dialer-list 1 protocol ip permitdialer-list 1 protocol ipx permitno cdp run!voice-port 1/015compand-type a-law!dial-peer voice 1 potsdestination-pattern 3direct-inward-dialport 1/015forward-digits all!dial-peer voice 100 voipshutdowndestination-pattern 1session target ipv41.2.79.7!dial-peer voice 200 voipshutdowndestination-pattern 1session target ipv41.2.79.31!dial-peer voice 300 vofrdestination-pattern 1session target Serial0/0 990!dial-peer voice 400 voipshutdowndestination-pattern 1session target ipv45.5.5.2!!line con 0exec-timeout 0 0transport input noneline aux 0line vty 0 4password ardlogin!endCommand Reference
This section documents new or modified commands. All other commands used with this feature are documented in the Cisco IOS Release 12.1 command references.
pri-group
To specify a ISDN Primary Rate interface (PRI) on a channelized T1 or E1 controller, enter the pri-group controller configuration command. To remove the ISDN-PRI configuration, enter the no form of this command.
pri-group timeslots timeslot-range
no pri-group
Syntax Description
timeslot-range
Specifies a single range of values. For T1, the allowable range is from 1 to 23. For E1, the allowable values are from 1 to 15.
Defaults
There is no ISDN-PRI group configured.
Command Modes
Controller configuration
Command History
Usage Guidelines
The pri-group command applies to the configuration of Voice over Frame Relay and Voice over ATM on the Cisco MC3810 multiservice access concentrator and the Cisco 2600 and 3600 series routers.
Before you enter the pri-group command, you must specify an ISDN-PRI switch type and an E1 or T1 controller.
Note
Examples
The following example configures ISDN-PRI on all timeslots of controller E1 on a Cisco 2600 series router:
Router(config-controller)# pri-group timeslots 1-7, 16controller E1 4/0!controller E1 4/1pri-group timeslots 1-7,16!Related Commands
isdn switch-type
To configure the Cisco 2600 series router PRI interface to support QSIG signaling, enter this command.
Glossary
AAL—ATM Adaptation Layer. Service-dependent sublayer of the data link layer. The AAL accepts data from different applications and presents it to the ATM layer in the form of 48-byte ATM payload segments. AALs consist of two sublayers: convergence sublayer (CS) and segmentation and reassembly (SAR). AALs differ on the basis of the source-destination timing used, whether they use constant bit rate (CBR) or variable bit rate (VBR), and whether they are used for connection-oriented or connectionless mode data transfer. At present, the four types of AAL recommended by the ITU-T are AAL1, AAL2, AAL3/4, and AAL5.
AAL1—ATM Adaptation Layer 1. One of four AALs recommended by the ITU-T. AAL1 is used for connection-oriented, delay-sensitive services requiring constant bit rates, such as uncompressed video and other isochronous traffic.
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. Ones density is not maintained independent of the data stream. Sometimes called binary coded alternate mark inversion.
ATM—Asynchronous Transfer Mode. International standard for cell relay in which multiple service types (such as voice, video, or data) are conveyed in fixed-length (53-byte) cells. Fixed-length cells allow cell processing to occur in hardware, thereby reducing transit delays. 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 ones density 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.
CBR—constant bit rate. QoS class defined by the ATM Forum for ATM networks. CBR is used for connections that depend on precise clocking to ensure undistorted delivery.
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.
CES—circuit emulation service. Enables users to multiplex or concentrate multiple circuit emulation streams for voice and video with packet data on a single high-speed ATM link without a separate ATM access multiplexer.
CO—central office. Local telephone company office to which all local loops in a given area connect and in which 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 DS0 channels from one T1 or E1 facility to be digitally cross-connected to DS0 channels on another T1 or E1. 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, same as PVDM.
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.
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, emulates the extension interface of a PBX or the subscriber interface for a switch.
IETF—Internet Engineering Task Force
ISDN—Integrated Services Digital Network. Communication protocol, offered by telephone companies, that permits telephone networks to carry data, voice, and other source traffic.
IVR—interactive voice response. Term used to describe systems that provide information in the form of recorded messages over telephone lines in response to user input in the form of spoken words or more commonly DTMF signaling. Examples include banks that allow you to check your balance from any telephone and automated stock quote systems.
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
PVDM—packet voice data 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, 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.
Guest
