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Table Of Contents
Modem Passthrough over Voice over IP
Related Features and Technologies
Supported Standards, MIBs, and RFCs
Configuring Modem Passthrough over VoIP Globally
Configuring Modem Passthrough over VoIP for a Specific Dial Peer
Verifying Modem Passthrough over VoIP
Monitoring and Maintaining Modem Passthrough over VoIP
modem passthrough (voice-service)
Modem Passthrough over Voice over IP
This feature module describes the Modem Passthrough over Voice over IP (VoIP) feature on Cisco AS5300 universal access server gateways and includes information on the new feature in the following sections:
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Supported Standards, MIBs, and RFCs
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Feature Overview
The Modem Passthrough over VoIP feature provides the transport of modem signals through a packet network by using pulse code modulation (PCM) encoded packets. This feature is available on the Cisco AS5300 Universal Access Server for Cisco IOS Release 12.1(3)T.
The Modem Passthrough over VoIP feature performs the following functions:
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For further details, the functions of the Modem Passthrough over VoIP feature are described in the following sections.
Modem Tone Detection
The gateway is able to detect modems at speeds up to V.90.
Passthrough Switchover
When the gateway detects a data modem, both the originating gateway and the terminating gateway roll over to G.711. The roll over to G.711 disables the high-pass filter, disables echo cancellation, and disables VAD. At the end of the modem call, the voice ports revert to the prior configuration and the digital signal processor (DSP) goes back to the state before switchover. You can configure the codec by selecting the g711alaw or g711ulaw option of the codec command.
See also the "Configuration Tasks" section later in this document.
Controlled Redundancy
You can enable payload redundancy so that the Modem Passthrough over VoIP switchover causes the gateway to emit redundant packets.
Packet Size
When redundancy is enabled, 10-ms sample-sized packets are sent. When redundancy is disabled, 20-ms sample-sized packets are sent.
Clock Slip Buffer Management
When the gateway detects a data modem, both the originating gateway and the terminating gateway switch from dynamic jitter buffers to static jitter buffers of 200-ms depth. The switch from dynamic to static is to compensate for Public Switched Telephone Network (PSTN) clocking differences at the originating gateway and the terminating gateway. At the conclusion of the modem call, the voice ports revert to dynamic jitter buffers.
Figure 1 illustrates the connection from the client modem to a MICA technologies modem network access server (NAS).
Figure 1 Modem Passthrough Connection
Benefits
The Modem Passthrough over VoIP feature offers the following benefits:
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Restrictions
Cisco IOS Release 12.1(3)T must run on the gateways for the Modem Passthrough over VoIP feature to work.
Related Features and Technologies
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Related Documents
The following documents provide additional platform-specific or hardware-related information to help implement VoIP:
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Supported Platforms
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Supported Standards, MIBs, and RFCs
Standards
ITU-T G.711
MIBs
None
To obtain lists of MIBs supported by platform and Cisco IOS release and to download MIB modules, go to the Cisco MIB web site on Cisco Connection Online (CCO).
RFCs
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Prerequisites
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Configuration Tasks
By default, modem passthrough over VoIP capability and redundancy are disabled.
Tips
Redundancy can be enabled in one or both of the gateways. When only a single gateway is configured for redundancy, the other gateway receives the packets correctly, but does not produce redundant packets.
See the following sections for the Modem Passthrough over VoIP feature. The two configuration tasks can configure separately or together. If both are configured, the dial-peer configuration takes precedence over the global configuration. Consequently, a call matching a particular dial-peer will first try to apply the modem passthrough configuration on the dial-peer. Then, if a specific dial-peer is not configured, the router will use the global configuration:
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Configuring Modem Passthrough over VoIP Globally
For the Modem Passthrough over VoIP feature to operate, you need to configure modem passthrough in both the originating gateway and the terminating gateway so that the modem call matches a voip dial-peer on the gateway.
When using the voice service voip and modem passthrough nse commands on a terminating gateway to globally set up fax or modem passthrough with NSEs, you must also ensure that each incoming call will be associated with a VoIP dial peer to retrieve the global fax or modem configuration. You associate calls with dial peers by using the incoming called-number command to specify a sequence of digits that incoming calls can match. You can ensure that all calls will match at least one dial peer by using the following commands:
Router(config)# dial-peer voice tag voipRouter(config-dial-peer)# incoming called-number .To configure the Modem Passthrough over VoIP feature for all the connections of a Cisco AS5300 universal access server gateway, use the following commands beginning in global configuration mode:
Configuring Modem Passthrough over VoIP for a Specific Dial Peer
You can configure the Modem Passthrough over VoIP feature on a specific dial peer in two ways, as follows:
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The default behavior for the voice-service configuration mode is no modem passthrough. This default behavior implies that modem passthrough is disabled for all dial peers on the gateway by default.
To enable Modem Passthrough on the VoIP dial peers on both the originating and terminating gateway, configure modem passthrough globally or explicitly on the dial peer.
For modem passthrough to operate, you must define VoIP dial peers on both gateways to match the call, for example, by using a destination pattern or an incoming called number. The modem passthrough parameters associated with those dial peers then will apply to the call.
Note
To configure the Modem Passthrough over VoIP feature for a specific dial peer, use the following commands beginning in global configuration mode:
Verifying Modem Passthrough over VoIP
To verify that the Modem Passthrough over VoIP feature is enabled, perform the following steps:
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Troubleshooting Tips
To troubleshoot the Modem Passthrough over VoIP feature, perform the following steps:
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Monitoring and Maintaining Modem Passthrough over VoIP
To monitor and maintain the Modem Passthrough over VoIP feature, use the following commands in privileged EXEC mode:
Configuration Examples
The following is sample configuration for the Modem Passthrough over VoIP feature for the Cisco AS5300 universal access servers:
version 12.1service timestamps debug uptimeservice timestamps log uptimeno service password-encryption!voice service voipmodem passthrough nse codec g711ulaw redundancy maximum-session 5!!resource-pool disable!!!!!ip subnet-zeroip ftp source-interface Ethernet0ip ftp username labip ftp password labno ip domain-lookup!isdn switch-type primary-5esscns event-service server!!!!!mta receive maximum-recipients 0!!controller T1 0framing esfclock source line primarylinecode b8zspri-group timeslots 1-24!controller T1 1shutdownclock source line secondary 1!controller T1 2shutdown!controller T1 3shutdown!!!interface Ethernet0ip address 1.1.2.2 255.0.0.0no ip route-cacheno ip mroute-cache!interface Serial0:23no ip addressencapsulation pppip mroute-cacheno logging event link-statusisdn switch-type primary-5essisdn incoming-voice modemno peer default ip addressno fair-queueno cdp enableno ppp lcp fast-start!interface FastEthernet0ip address 26.0.0.1 255.0.0.0no ip route-cacheno ip mroute-cacheload-interval 30duplex fullspeed autono cdp enable!ip classlessip route 17.18.0.0 255.255.0.0 1.1.1.1no ip http server!!!!voice-port 0:D!dial-peer voice 1 potsincoming called-number 55511..destination-pattern 020..direct-inward-dialport 0:Dprefix 020!dial-peer voice 2 voipincoming called-number 020..destination-pattern 55511..modem passthrough nse codec g711ulaw redundancysession target ipv4:26.0.0.2!!line con 0exec-timeout 0 0transport input noneline aux 0line vty 0 4login!!endCommand Reference
This section documents new and modified commands. All other commands used with this feature are documented in the Cisco IOS Multiservice Applications Command Reference for Cisco IOS Release 12.1.
Note
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modem passthrough (dial-peer)
To configure Modem Passthrough over VoIP for a specific dial peer, use the modem passthrough dial-peer configuration command. To use the global default for a specific dial peer, use the modem passthrough system command. To disable modem passthrough for a specific dial peer, use the no modem passthrough command.
modem passthrough {system | nse [payload-type number] codec {g711ulaw | g711alaw} [redundancy]}
no modem passthrough
Syntax Description
Defaults
Defining system as the method in dial peer points to the voice service VoIP configuration default and is intended to simplify gateway configuration. The default is modem passthrough system. As required, the gateway defaults to no modem passthrough.
Command Modes
Dial-peer configuration
Command History
Usage Guidelines
Use the modem passthrough dial-peer configuration command to configure modem passthrough over VoIP for a specific dial peer. The payload type is an optional parameter for the nse keyword. Use the same payload-type number for both the originating gateway and the terminating gateway. The payload-type number can be set from 96 to 119. If you do not specify the payload-type number, the number defaults to 100.
Use the same codec type for both the originating gateway and the terminating gateway. g711ulaw codec is required for T1, and g711alaw codec is required for E1.
The redundancy keyword is an optional parameter for sending redundant packets for modem traffic.
When the system keyword is enabled, the following parameters are not available: nse, payload-type, codec, and redundancy. The system keyword overrides the configuration for the dial-peer and the values from the global configuration are used.
Examples
The following example shows how Modem Passthrough over VoIP is configured for a specific dial peer in dial-peer configuration mode:
Router(config-dial-peer)# modem passthrough nse codec g711ulaw redundancyRelated Commands
modem passthrough (voice-service)
To configure Modem Passthrough over VoIP for the Cisco AS5300 universal access server gateway, use the modem passthrough voice-service configuration command. To disable modem passthrough, use the no form of this command.
modem passthrough nse [payload-type number] codec {g711ulaw | g711alaw}
[redundancy] [maximum-sessions value]no modem passthrough
Syntax Description
Defaults
Disabled
Command Modes
Voice-service configuration mode
Command History
Usage Guidelines
Use the modem passthrough command to configure Modem Passthrough over VoIP for the Cisco AS5300 universal access server gateway. The default behavior for the voice service voip command is no modem passthrough.
The payload type is an optional parameter for the nse keyword. Use the same payload-type number for both the originating gateway and the terminating gateway. The payload-type number can be set from 96 to 119. If you do not specify the payload-type number, the number defaults to 100.
Use the same codec type for both the originating gateway and the terminating gateway. g711ulaw codec is required for T1, and g711alaw codec is required for E1.
The redundancy keyword is an optional parameter for sending redundant packets for modem traffic.
The maximum-sessions keyword is an optional parameter for the modem passthrough command. This parameter determines the maximum number of simultaneous modem passthrough sessions. The recommended value for the maximum-sessions is 16. The value can be set from 1 to 26.
Examples
The following example shows modem passthough configuration in voice-service configuration mode for NSE payload type 101 using codec G.711:
Router(conf-voi-serv)# modem passthrough nse payload-type 101 codec g711ulaw redundancy maximum-session 1Related Commands
voice service voip
Enters voice-service configuration mode and specifies the voice encapsulation type.
show call active
To show active call information for a call in progress, use the show call active privileged EXEC command.
show call active {voice | fax}[brief]
Syntax Description
voice
Specifies that the active call table displays voice call information.
fax
Specifies that the active call table displays fax call information.
brief
(Optional) Displays a truncated version.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Use the show call active privileged EXEC command to display the contents of the active call table. If you use the voice keyword, the active call table displays information about all the voice calls currently connected through the router or access server.
Examples
The following is sample output from the show call active voice command updated with the modem passthrough output:
Router# show call active voiceGENERIC:SetupTime=104443 msIndex=1PeerAddress=50110PeerSubAddress=PeerId=100PeerIfIndex=104LogicalIfIndex=10ConnectTime=104964CallDuration=00:02:43CallState=4CallOrigin=2ChargedUnits=0InfoType=2TransmitPackets=15720TransmitBytes=2362904ReceivePackets=15670ReceiveBytes=2737904TELE:ConnectionId=[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]TxDuration=155310 msVoiceTxDuration=155310 msFaxTxDuration=0 msCoderTypeRate=g711ulawNoiseLevel=-75ACOMLevel=11OutSignalLevel=-13InSignalLevel=-22InfoActivity=2ERLLevel=27SessionTarget=ImgPages=0GENERIC:SetupTime=104648 msIndex=1PeerAddress=55240PeerSubAddress=PeerId=2PeerIfIndex=105LogicalIfIndex=0ConnectTime=104964CallDuration=00:02:47CallState=4CallOrigin=1ChargedUnits=0InfoType=2TransmitPackets=16026TransmitBytes=2608248ReceivePackets=16075ReceiveBytes=2609164VOIP:ConnectionId[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]RemoteIPAddress=1.14.82.14RemoteUDPPort=18202RoundTripDelay=2 msSelectedQoS=best-efforttx_DtmfRelay=inband-voiceFastConnect=TRUESessionProtocol=ciscoSessionTarget=ipv4:1.14.82.14OnTimeRvPlayout=40GapFillWithSilence=0 msGapFillWithPrediction=0 msGapFillWithInterpolation=0 msGapFillWithRedundancy=0 msHiWaterPlayoutDelay=67 msLoWaterPlayoutDelay=67 msReceiveDelay=67 msLostPackets=0 msEarlyPackets=0 msLatePackets=0 msVAD = enabledCoderTypeRate=g729r8CodecBytes=20SignalingType=casModem passthrough signaling method is nse:Buffer Fill Events = 0Buffer Drain Events = 0Percent Packet Loss = 0Consecutive-packets-lost Events = 0Corrected packet-loss Events = 0Last Buffer Drain/Fill Event = 157secTime between Buffer Drain/Fills = Min 0sec Max 0secRouter# show call active voice brief<ID>: <start>hs.<index> +<connect> pid:<peer_id> <dir> <addr> <state>dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>delay:<last>/<min>/<max>ms <codec>MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>last <buf event time>s dur:<Min>/<Max>sFR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>sig:<on/off> <codec> (payload size)ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>sig:<on/off> <codec> (payload size)Tele <int>: tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm3 : 104443hs.1 +521 pid:100 Answer 50110 activedur 00:03:28 tx:20151/3036404 rx:20102/3517936Tele 0:D:1: tx:199630/199630/0ms g711ulaw noise:-75 acom:11 i/0:-22/-13 dBm3 : 104648hs.1 +316 pid:2 Originate 55240 activedur 00:03:28 tx:20102/3276712 rx:20151/3277628IP 1.14.82.14:18202 rtt:3ms pl:40/0ms lost:0/0/0 delay:67/67/67ms g729r8MODEMPASS nse buf:0/0 loss 0% 0/0 last 195s dur:0/0sTable 1 provides an alphabetical listing of the show call active command fields and a description of each field.
Related Commands
show call history
To display the fax call history table for a fax transmission, use the show call history privileged EXEC command.
show call history {voice | fax}[last number | brief]
Syntax Description
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Use the show call history voice privileged EXEC command to display the voice call history table. The call history table contains a listing of all calls connected through this router in descending time order since VoIP was enabled. You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the last keyword, and define the number of calls you want to see with the number argument. To display a truncated version of the call history table, use the brief keyword.
Examples
The following is sample output from the show call history command updated with the Modem Passthrough over Voice over IP feature:
Router# show call history voiceGENERIC:SetupTime=104648 msIndex=1PeerAddress=55240PeerSubAddress=PeerId=2PeerIfIndex=105LogicalIfIndex=0DisconnectCause=10DisconnectText=normal call clearing.ConnectTime=104964DisconectTime=143329CallDuration=00:06:23CallOrigin=1ChargedUnits=0InfoType=speechTransmitPackets=37668TransmitBytes=6157536ReceivePackets=37717ReceiveBytes=6158452VOIP:ConnectionId[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]RemoteIPAddress=1.14.82.14RemoteUDPPort=18202RoundTripDelay=2 msSelectedQoS=best-efforttx_DtmfRelay=inband-voiceFastConnect=TRUESessionProtocol=ciscoSessionTarget=ipv4:1.14.82.14OnTimeRvPlayout=40GapFillWithSilence=0 msGapFillWithPrediction=0 msGapFillWithInterpolation=0 msGapFillWithRedundancy=0 msHiWaterPlayoutDelay=67 msLoWaterPlayoutDelay=67 msReceiveDelay=67 msLostPackets=0 msEarlyPackets=0 msLatePackets=0 msVAD = enabledCoderTypeRate=g729r8CodecBytes=20cvVoIPCallHistoryIcpif=0SignalingType=casModem passthrough signaling method is nseBuffer Fill Events = 0Buffer Drain Events = 0Percent Packet Loss = 0Consecutive-packets-lost Events = 0Corrected packet-loss Events = 0Last Buffer Drain/Fill Event = 373secTime between Buffer Drain/Fills = Min 0sec Max 0secGENERIC:SetupTime=104443 msIndex=2PeerAddress=50110PeerSubAddress=PeerId=100PeerIfIndex=104LogicalIfIndex=10DisconnectCause=10DisconnectText=normal call clearing.ConnectTime=104964DisconectTime=143330CallDuration=00:06:23CallOrigin=2ChargedUnits=0InfoType=speechTransmitPackets=37717TransmitBytes=5706436ReceivePackets=37668ReceiveBytes=6609552TELE:ConnectionId=[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]TxDuration=375300 msVoiceTxDuration=375300 msFaxTxDuration=0 msCoderTypeRate=g711ulawNoiseLevel=-75ACOMLevel=11SessionTarget=ImgPages=0Router# show call history voice brief<ID>: <start>hs.<index> +<connect> +<disc> pid:<peer_id> <direction> <addr>dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes> <disc-cause>(<text>)IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>delay:<last>/<min>/<max>ms <codec>MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>last <buf event time>s dur:<Min>/<Max>sFR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>sig:<on/off> <codec> (payload size)ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>sig:<on/off> <codec> (payload size)Telephony <int>: tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBmTable 2 provides an alphabetical listing of the show call history command fields and a description of each field.
Table 2 show call history Field Descriptions
ACOMLevel
Current ACOM level for this call. ACOM is the combined loss achieved by the echo canceller, which is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call.
Buffer Drain Events
Total number of jitter buffer drain events.
Buffer Fill Events
Total number of jitter buffer fill events.
CallDuration
Length of the call in hours, minutes and seconds, hh:mm:ss.
CallOrigin
Call origin: answer or originate.
ChargedUnits
Total number of charging units applying to this peer since system startup. The unit of measure for this field is in hundredths of seconds.
CodecBytes
Payload size in bytes for the codec used.
CoderTypeRate
Negotiated coder rate. This value specifies the send rate of voice/fax compression to its associated call leg for this call.
ConnectionID
Global call identifier for the gateway call.
ConnectTime
Time when this call was connected.
Consecutive-packets-lost Events
Total number of consecutive (two-or-more) packet loss events.
Corrected packet-loss Events
Total number of packet loss events that were corrected using the RFC 2198 method.
DisconnectCause
Description explaining why this call was disconnected.
DisconnectText
Descriptive text explaining the disconnect reason.
DisconnectTime
Time when this call was disconnected.
FaxTxDuration
Duration of fax transmission from this peer to voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value.
GapFillWithInterpolation
Duration of the voice signal played out with a signal synthesized from parameters or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call.
GapFillWithRedundancy
Duration of the voice signal played out with a signal synthesized from redundancy parameters available because voice data was lost or not received in time from the voice gateway for this call.
GapFillWithSilence
Duration of a voice signal replaced with silence because voice data was lost or not received in time for this call.
GapFillWithPrediction
Duration of a voice signal played out with a signal synthesized from parameters, or samples of data preceding in time because voice data was lost or not received in time from the voice gateway for this call.
HiWaterPlayoutDelay
High water mark Voice Playout FIFO Delay during this voice call.
Index
Dial peer identification number.
InfoType
Information type for this call, for example, voice or fax.
Last Buffer Drain/Fill Event
Time since the last jitter buffer drain/fill event, in seconds.
LogicalIfIndex
Index number of the logical voice port for this call.
LoWaterPlayoutDelay
Low-water mark Voice Playout FIFO Delay during this voice call.
Modem passthrough signaling method is nse
Indicates that this is a modem passthrough call and named signaling events (NSEs)—also called telephone-events in RFC 2833—are used for signalling codec upspeed. The upspeed method is the method used to dynamically change the codec type and speed to meet network conditions. This means that you might move to a faster codec when you have both voice and data calls, and then slow down when there's only voice traffic.
NoiseLevel
Average noise level for this call.
OnTimeRvPlayout
Duration of voice playout from data received on time for this call. Derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.
Percent Packet Loss
Total percent packet loss.
PeerAddress
Destination pattern or number associated with this peer.
PeerId
ID value of the peer entry table to which this call was made.
PeerIfIndex
Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.
PeerSubAddress
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