|
|
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:
The Modem Passthrough over VoIP feature performs the following functions:
For further details, see the following functions of the Modem Passthrough over VoIP feature:
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. This 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 g711alaw or g711ulaw options of the codec command.
See also the "Configuration Tasks" section.
Controlled Redundancy
You can enable payload redundancy, so that the Modem Passthrough over VoIP switchover causes the gateway emit redundant packets.
Packet Size
When redundancy is enabled, 10 millisecond sample-sized packets are sent. When redundancy is disabled, 20 millisecond 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 milliseconds depth. This 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 back to dynamic jitter buffers.
Figure 1 illustrates the connection from the client modem to a MICA modem network access server (NAS).

The Modem Passthrough over VoIP feature offers the following benefits:
Cisco IOS Release 12.1(3)T is required to run on the gateways for the Modem Passthrough over VoIP feature to work.
The following documents provide additional platform-specific or hardware related information to help implement VoIP:
Cisco AS5300 Universal Access Server gateways
Standards
ITU-T G.711
MIBs
None
For descriptions of supported MIBs and how to use MIBs, see the Cisco MIB web site on CCO at http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
RFCs
By default, the Modem Passthrough over VoIP capability and redundancy are disabled.
![]() |
Tips You need to configure Modem Passthrough in both the originating gateway and the terminating gateway for the Modem Passthrough over VoIP to operate. If you configure only one of the gateways in a pair, the modem call will not connect successfully. |
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 configuration:
To configure Modem Passthrough over VoIP for all the connections of a Cisco AS5300 Universal Access Server gateway, use the following commands beginning in global configuration mode:
| Command | Purpose | |
|---|---|---|
Step 1 | Router(config)#voice service voip | Enters the voice-service configuration mode. Configures voice service for all the connections for the gateways. |
Step 2 | Router(conf-voi-serv)#modem passthrough {nse
[payload-type number] codec {g711ulaw | g711alaw}
[redundancy] [maximum-sessions value]}
| Configures Modem Passthrough over VoIP for the Cisco AS5300 Universal Access Server gateways. The default behavior 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 between 96 and 119. If you do not specify the payload-type number, it defaults to 100. When the payload-type is 100, and you use the show running-config command, the payload-type parameter does not appear. 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 packet redundancy for modem traffic. The maximum-sessions is an optional parameter for the modem passthrough command in the voice-service configuration mode. This parameter determines the maximum simultaneous Modem Passthrough sessions. The recommended value for the maximum-sessions is 16. The value can be set from 1 to 26. |
Step 3 | Router(conf-voi-serv)# | Exits from voice-service configuration mode. |
Step 4 | Router(config)# | Exits the global configuration mode. |
You can configure Modem Passthrough over VoIP on a specific dial peer by performing one of the following:
The default behavior for the voice-service configuration mode is no modem passthrough. This 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 the Modem Passthrough to operate, you must define VoIP dial peers on both gateways to match the call, for example, by using a destination-pattern. The Modem Passthrough parameters associated with those dial peers then will apply to the call.
![]() |
Note When Modem Passthrough is configured individually for a specific dial peer, that configuration for the specific dial peer takes precedence over the global configuration. |
To configure Modem Passthrough over VoIP for a specific dial peer, use the following commands beginning in global configuration mode:
| Command | Purpose | |
|---|---|---|
Step 1 | Router(config)#dial-peer voice number voip | Enters the dial-peer configuration mode. Configures a specific dial peer in dial-peer configuration mode. |
Step 2 | Router(config-dial-peer)#modem passthrough {nse
[payload-type number] codec {g711ulaw | g711alaw}
[redundancy]} | system}
| Configures Modem Passthrough over VoIP for a specific dial peer. The default behavior for the Modem Passthrough for VoIP in dial-peer configuration mode is modem passthrough system. As required, the gateway defaults to 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 between 96 and 119. If you do not specify the payload-type number, it defaults to 100. When the payload-type is 100, and you use the show running-config command, the payload-type parameter does not appear. 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 packet redundancy for modem traffic. When the system keyword is enabled, the following parameters are not available: nse, payload-type, codec, and redundancy. Instead the values from the global configuration are used. |
Step 3 | Router(config-dial-peer)#Exit | Exits from dial-peer configuration mode and returns to the global configuration mode. |
Step 4 | Router(config)# | Exits the global configuration mode. |
To verify that the Modem Passthrough over VoIP feature is enabled, follow these steps:
Step 2 Enter the show dial-peer voice command to verify that Modem Passthrough over VoIP is enabled.
To troubleshoot the Modem Passthrough over VoIP feature, perform the following steps:
Use the following commands to monitor and maintain the Modem Passthrough over VoIP feature:
| Command | Purpose |
|---|---|
Router#show call {active | history} voice [brief]
| Displays information for the active call table or displays the voice call history table. The brief option displays a truncated version of either option. |
Router#show dial-peer voice [number] [summary] | Displays configuration information for dial peers. The number argument specifies a specific dial peer from 1- 32767. The summary option displays a summary of all dial peers. |
See the following sample configuration for Modem Passthrough over VoIP for the Cisco AS5300 Universal Access Servers:
version 12.1 service timestamps debug uptime service timestamps log uptime no service password-encryption ! voice service voip modem passthrough nse codec g711ulaw redundancy maximum-session 5 ! ! resource-pool disable ! ! ! ! ! ip subnet-zero ip ftp source-interface Ethernet0 ip ftp username lab ip ftp password lab no ip domain-lookup ! isdn switch-type primary-5ess cns event-service server ! ! ! ! ! mta receive maximum-recipients 0 ! ! controller T1 0 framing esf clock source line primary linecode b8zs pri-group timeslots 1-24 ! controller T1 1 shutdown clock source line secondary 1 ! controller T1 2 shutdown ! controller T1 3 shutdown ! ! ! interface Ethernet0 ip address 1.1.2.2 255.0.0.0 no ip route-cache no ip mroute-cache ! interface Serial0:23 no ip address encapsulation ppp ip mroute-cache no logging event link-status isdn switch-type primary-5ess isdn incoming-voice modem no peer default ip address no fair-queue no cdp enable no ppp lcp fast-start ! interface FastEthernet0 ip address 26.0.0.1 255.0.0.0 no ip route-cache no ip mroute-cache load-interval 30 duplex full speed auto no cdp enable ! ip classless ip route 17.18.0.0 255.255.0.0 1.1.1.1 no ip http server ! ! ! ! voice-port 0:D ! dial-peer voice 1 pots incoming called-number 55511.. destination-pattern 020.. direct-inward-dial port 0:D prefix 020 ! dial-peer voice 2 voip incoming called-number 020.. destination-pattern 55511.. modem passthrough nse codec g711ulaw redundancy session target ipv4:26.0.0.2 ! ! line con 0 exec-timeout 0 0 transport input none line aux 0 line vty 0 4 login ! ! end
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 The modified commands are marked by asterisks. |
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}
Syntax Description
nse Named Signaling Events (NSE). payload-type (Optional) NSE payload-type. number The value of the payload-type (96 - 119). codec Codec selections for upspeed. g711ulaw Codec G.711 U-Law 64000 bps for T1. g711alaw Codec G.711 A-Law 64000 bps for E1. redundancy Packet redundancy for modem traffic. maximum-sessions (Optional) Maximum simultaneous Modem Passthrough sessions. value The number of simultaneous Modem Passthrough sessions. The minimum value is 1 and the maximum value is 26. The default is 16 calls.
Defaults
Disabled
Command Modes
Voice-service configuration mode
Command History
12.1(3)T This command was introduced for the Cisco AS5300 Universal Access Servers.
Release
Modification
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 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 between 96 and 119. If you do not specify the payload-type number, it 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 packet redundancy for modem traffic.
The maximum-sessions is an optional parameter for the modem passthrough command. This parameter determines the maximum 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 pass though configuration in voice service configuration mode:
Router(conf-voi-serv)# modem passthrough nse payload-type 101 codec g711ulaw redundancy maximum-session 1
Related Commands
voice service voip Enters the voice-service configuration mode and specifies the voice encapsulation type.
Command
Description
To configure Modem Passthrough over VoIP for a specific dial peer, use the modem passthrough dial-peer configuration mode 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 {nse [payload-type number] codec {g711ulaw | g711alaw} [redundancy]} | system}
Syntax Description
nse Named signaling event. payload-type NSE payload-type. number The value of the payload-type (96 - 119). codec Voice compression for speech or audio signals. Codec selections for upspeed. g711ulaw Codec G.711 U-Law 64000 bps for T1. g711alaw Codec G.711 A-Law 64000 bps for E1. redundancy Packet redundancy for modem traffic. system Defaults to the global configuration.
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 mode
Command History
12.1(3)T This command was introduced for the Cisco AS5300 Universal Access Servers.
Release
Modification
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 between 96 and 119. If you do not specify the payload-type number, it 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 packet redundancy for modem traffic.
When the system keyword is enabled, the following parameters are not available: nse, payload-type, codec, and redundancy. Instead 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 the dial-peer configuration mode:
Router(config-dial-peer)# modem passthrough nse ? Router(config-dial-peer)# modem passthrough nse codec g711ulaw redundancy 16
Related Commands
dial-peer voice Enters dial-peer configuration mode.
Command
Description
Syntax Description
voice Specifies that the active call table displays voice call information. brief (Optional) Displays a truncated version.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
11.3(1)T This command was introduced. 12.0(4)XJ This command was modified for Store and Forward Fax. 12.1(3)T This command was modified for Modem Passthrough over Voice over IP on the Cisco AS5300 Universal Access Servers.
Release
Modification
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
This is sample output from the show call active voice command updated with the Modem Passthrough output:
Router# show call active voice GENERIC: SetupTime=104443 ms Index=1 PeerAddress=50110 PeerSubAddress= PeerId=100 PeerIfIndex=104 LogicalIfIndex=10 ConnectTime=104964 CallDuration=00:02:43 CallState=4 CallOrigin=2 ChargedUnits=0 InfoType=2 TransmitPackets=15720 TransmitBytes=2362904 ReceivePackets=15670 ReceiveBytes=2737904 TELE: ConnectionId=[0x4B091A27 0x3EDD0003 0x0 0xFEFD4] TxDuration=155310 ms VoiceTxDuration=155310 ms FaxTxDuration=0 ms CoderTypeRate=g711ulaw NoiseLevel=-75 ACOMLevel=11 OutSignalLevel=-13 InSignalLevel=-22 InfoActivity=2 ERLLevel=27 SessionTarget= ImgPages=0 GENERIC: SetupTime=104648 ms Index=1 PeerAddress=55240 PeerSubAddress= PeerId=2 PeerIfIndex=105 LogicalIfIndex=0 ConnectTime=104964 CallDuration=00:02:47 CallState=4 CallOrigin=1 ChargedUnits=0 InfoType=2 TransmitPackets=16026 TransmitBytes=2608248 ReceivePackets=16075 ReceiveBytes=2609164 VOIP: ConnectionId[0x4B091A27 0x3EDD0003 0x0 0xFEFD4] RemoteIPAddress=1.14.82.14 RemoteUDPPort=18202 RoundTripDelay=2 ms SelectedQoS=best-effort tx_DtmfRelay=inband-voice FastConnect=TRUE SessionProtocol=cisco SessionTarget=ipv4:1.14.82.14 OnTimeRvPlayout=40 GapFillWithSilence=0 ms GapFillWithPrediction=0 ms GapFillWithInterpolation=0 ms GapFillWithRedundancy=0 ms HiWaterPlayoutDelay=67 ms LoWaterPlayoutDelay=67 ms ReceiveDelay=67 ms LostPackets=0 ms EarlyPackets=0 ms LatePackets=0 ms VAD = enabled CoderTypeRate=g729r8 CodecBytes=20 SignalingType=cas Modem passthrough signaling method is nse: Buffer Fill Events = 0 Buffer Drain Events = 0 Percent Packet Loss = 0 Consecutive-packets-lost Events = 0 Corrected packet-loss Events = 0 Last Buffer Drain/Fill Event = 157sec Time between Buffer Drain/Fills = Min 0sec Max 0sec Router# 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>s FR <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> dBm 3 : 104443hs.1 +521 pid:100 Answer 50110 active dur 00:03:28 tx:20151/3036404 rx:20102/3517936 Tele 0:D:1: tx:199630/199630/0ms g711ulaw noise:-75 acom:11 i/0:-22/-13 dBm 3 : 104648hs.1 +316 pid:2 Originate 55240 active dur 00:03:28 tx:20102/3276712 rx:20151/3277628 IP 1.14.82.14:18202 rtt:3ms pl:40/0ms lost:0/0/0 delay:67/67/67ms g729r8 MODEMPASS nse buf:0/0 loss 0% 0/0 last 195s dur:0/0s
Table 1 provides an alphabetical listing of the show call active command fields and a description of each field.
| Field | Description |
|---|---|
ACOM Level | Current ACOM level for this call. This value is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for this call. |
Buffer Drain Events | Total number of jitter buffer drain events. |
Buffer Fill Events | Total number of jitter buffer fill events. |
CallOrigin | Call origin: answer or originate. |
CallState | Current state of the call. |
CoderTypeRate | Negotiated coder transmit rate of voice/fax compression during this call. |
ConnectionId | Global call identifier for this gateway call. |
ConnectTime | Time when the 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. |
Dial-Peer | Tag of the dial peer sending this call. |
ERLLevel | Current Echo Return Loss (ERL) level for this call. |
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. |
GapFillWith Redundancy | 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. |
GapFillWithPrediction | Duration of the voice signal played out with 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. Examples of such pullout are frame-eraser or frame-concealment strategies in G.729 and G.723.1 compression algorithms. |
GapFillWithSilence | Duration of voice signal replaced with silence because voice data was lost or not received in time for this call. |
HiWaterPlayoutDelay | High water mark Voice Playout FIFO Delay during this call. |
Index | Dial-peer identification number. |
InfoActivity | Active information transfer activity state for this call. |
InfoType | Information type for this call. |
InSignalLevel | Active input signal level from the telephony interface used by this call. |
Last Buffer Drain/Fill Event | Time since the last jitter buffer drain/fill event, in seconds. |
LogicalIfIndex | Index number of the logical interface for this call. |
LoWaterPlayoutDelay | Low water mark Voice Playout FIFO Delay during this 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 2833are used for signaling codec upspeed. |
NoiseLevel | Active noise level for this call. |
OnTimeRvPlayout | Duration of voice playout from data received in time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values. |
OutSignalLevel | Active output signal level to telephony interface used by this call. |
PeerAddress | Destination pattern associated with this peer. |
PeerId | ID value of the peer table entry when this call was made. |
PeerIfIndex | Voice-port index number for this peer. |
PeerSubaddress | Subaddress when this call is connected. |
Percent Packet Loss | Total percent packet loss. |
ReceiveBytes | Number of bytes received by the peer during this call. |
ReceiveDelay | Average Playout FIFO Delay plus the Decoder Delay during this call. |
ReceivePackets | Number of packets received by this peer during this call. |
RemoteIPAddress | Remote system IP address for the VoIP call. |
RemoteUDPPort | Remote system UDP listener port when voice packets are sent. |
RoundTripDelay | Voice packet round trip delay between the local and remote system on the IP backbone during this call. |
SelectedQoS | Selected RSVP quality of service (QoS) for this call. |
SessionProtocol | Session protocol used for an Internet call between the local and remote router through the IP backbone. |
SessionTarget | Session target of the peer used for this call. |
SetupTime | Value of the system UpTime when the call associated with this entry was started. |
Time between Buffer Drain/Fills | Minimum and maximum durations between jitter buffer drain/fill events, in seconds. |
TransmitBytes | Number of bytes sent from this peer during this call. |
TransmitPackets | Number of packets sent from this peer during this call. |
TxDuration | Duration of transmit path open from this peer to the voice gateway for this call. |
VADEnable | Whether voice activation detection (VAD) was enabled for this call. |
VoiceTxDuration | Duration of voice transmission from this peer to the voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value. |
Related Commands
show call history voice Displays the Voice over IP call history table. show dial-peer voice Displays configuration information for dial peers. show num-exp Displays how the number expansions are configured in Voice over IP. show voice port Displays configuration information about a specific voice port.
Command
Description
Syntax Description
voice Specifies that the call history tables shows voice call information. last number (Optional) Displays the last calls connected, where the number of calls that appear is defined by the argument number. Valid values are from 1 to 2147483647. brief (Optional) Displays a truncated version of the call history table.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
11.3(1)T This command was introduced. 12.0(4)XJ This command was modified for Store and Forward Fax. 12.1(3)T This command was modified for Modem Passthrough over Voice over IP on the Cisco AS5300 Universal Access Servers.
Release
Modification
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 Voice over IP 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 keyword last, and define the number of calls you want to see with the argument number. To display a truncated version of the call history table, use the brief keyword.
Examples
This is sample output from the show call history command updated with the Modem Passthrough over Voice IP feature.
Router# show call history voice GENERIC: SetupTime=104648 ms Index=1 PeerAddress=55240 PeerSubAddress= PeerId=2 PeerIfIndex=105 LogicalIfIndex=0 DisconnectCause=10 DisconnectText=normal call clearing. ConnectTime=104964 DisconectTime=143329 CallDuration=00:06:23 CallOrigin=1 ChargedUnits=0 InfoType=speech TransmitPackets=37668 TransmitBytes=6157536 ReceivePackets=37717 ReceiveBytes=6158452 VOIP: ConnectionId[0x4B091A27 0x3EDD0003 0x0 0xFEFD4] RemoteIPAddress=1.14.82.14 RemoteUDPPort=18202 RoundTripDelay=2 ms SelectedQoS=best-effort tx_DtmfRelay=inband-voice FastConnect=TRUE SessionProtocol=cisco SessionTarget=ipv4:1.14.82.14 OnTimeRvPlayout=40 GapFillWithSilence=0 ms GapFillWithPrediction=0 ms GapFillWithInterpolation=0 ms GapFillWithRedundancy=0 ms HiWaterPlayoutDelay=67 ms LoWaterPlayoutDelay=67 ms ReceiveDelay=67 ms LostPackets=0 ms EarlyPackets=0 ms LatePackets=0 ms VAD = enabled CoderTypeRate=g729r8 CodecBytes=20 cvVoIPCallHistoryIcpif=0 SignalingType=cas Modem passthrough signaling method is nse Buffer Fill Events = 0 Buffer Drain Events = 0 Percent Packet Loss = 0 Consecutive-packets-lost Events = 0 Corrected packet-loss Events = 0 Last Buffer Drain/Fill Event = 373sec Time between Buffer Drain/Fills = Min 0sec Max 0sec GENERIC: SetupTime=104443 ms Index=2 PeerAddress=50110 PeerSubAddress= PeerId=100 PeerIfIndex=104 LogicalIfIndex=10 DisconnectCause=10 DisconnectText=normal call clearing. ConnectTime=104964 DisconectTime=143330 CallDuration=00:06:23 CallOrigin=2 ChargedUnits=0 InfoType=speech TransmitPackets=37717 TransmitBytes=5706436 ReceivePackets=37668 ReceiveBytes=6609552 TELE: ConnectionId=[0x4B091A27 0x3EDD0003 0x0 0xFEFD4] TxDuration=375300 ms VoiceTxDuration=375300 ms FaxTxDuration=0 ms CoderTypeRate=g711ulaw NoiseLevel=-75 ACOMLevel=11 SessionTarget= ImgPages=0 Router# 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>s FR <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>dBm
Table 2 provides an alphabetical listing of the fields for the show call history command and a description of each field.
| Field | Description |
|---|---|
ACOMLevel | Average ACOM level for this call. This value is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for a particular call. |
Buffer Drain Events | Total number of jitter buffer drain events. |
Buffer Fill Events | Total number of jitter buffer fill events. |
CallOrigin | Call origin: answer or originate. |
CoderTypeRate | Negotiated coder rate. This value specifies the transmit rate of voice/fax compression to its associated call leg for this call. |
ConnectionID | Global call identifier for the gateway call. |
ConnectTime | Time 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 this call was disconnected. |
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 that are 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 the 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 the voice call. |
Index | Dial peer identification number. |
InfoType | Information type for this call. |
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 the 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 2833are used for signaling codec upspeed. |
NoiseLevel | Average noise level for this call. |
OnTimeRvPlayout | Duration of voice playout from data received on time for this call. You can 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 | Index number of the logical interface through which this call was made. For ISDN media, this would be the index number of the B channel used for this call. |
PeerSubAddress | Subaddress where this call is connected. |
ReceiveBytes | Number of bytes received by the peer during this call. |
ReceiveDelay | Average Playout FIFO Delay plus the Decoder Delay during this voice call. |
ReceivePackets | Number of packets received by this peer during this call. |
RemoteIPAddress | Remote system IP address for this call. |
RemoteUDPPort | Remote system UDP listener port where voice packets are sent. |
RoundTripDelay | Voice packet round trip delay between the local and remote system on the IP backbone for this call. |
SelectedQoS | Selected RSVP QoS for this call. |
Session Protocol | Session protocol to be used for an Internet call between the local and remote router through the IP backbone. |
Session Target | Session target of the peer used for the third call. |
SetUpTime | Value of the system UpTime when the call associated with this entry was started. |
Time between Buffer Drain/Fills | Minimum and maximum durations between jitter buffer drain/fill events, in seconds. |
TransmitBytes | Number of bytes sent by this peer during this call. |
TransmitPackets | Number of packets sent by this peer during this call. |
TxDuration | Duration of the transmit path open from this peer to the voice gateway for this call. |
VADEnable | Whether voice activation detection (VAD) was enabled for this call. |
VoiceTxDuration | Duration of voice sent from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration by the TxDuration value. |
Related Commands
show call active Displays active call information for a call in progress.
Command
Description
To display configuration information for dial peers, use the show dial-peer voice command in privileged EXEC mode.
show dial-peer voice [number] [summary]
Syntax Description
number (Optional) A specific dial peer. By using this option, you can see configuration information for a single dial peer identified by the number argument. Valid entries are any integers that identify a specific dial peer from 1 to 32767. summary (Optional) Displays a summary of all voice dial peers.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
11.3(1)T This command was introduced. 11.3(1)MA The summary keyword was added for the Cisco MC3810. 12.0(3)XG This command was modified to support VoFR for the Cisco 2600 series and Cisco 3600 series routers. 12.0(4)T Support was added for VoFR for the Cisco 7200 series routers. 12.1(3)T This command was modified for Modem Passthrough over Voice over IP on the Cisco AS5300 Universal Access Servers.
Release
Modification
Usage Guidelines
Use the show dial-peer voice privileged EXEC command to display the configuration for all VoIP and POTS dial peers configured for the router. To show configuration information for only one specific dial peer, use the argument number to identify the dial peer.
Examples
This is sample output from the show dial-peer voice command for a POTS dial peer updated with the Modem Passthrough over Voice over IP feature:
Router# show dial-peer voice
VoiceEncapPeer100
information type = voice,
tag = 100, destination-pattern = \Q55250',
answer-address = \Q', preference=0,
numbering Type = \Qunknown'
group = 100, Admin state is up, Operation state is up,
incoming called-number = \Q', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
huntstop = disabled,
application associated:
type = pots, prefix = \Q55250',
forward-digits default
session-target = \Q', voice-port = \Q0:D',
direct-inward-dial = disabled,
digit_strip = enabled,
register E.164 number with GK = TRUE
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0,
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.
VoiceOverIpPeer2
information type = voice,
tag = 2, destination-pattern = \Q55240',
answer-address = \Q', preference=0,
numbering Type = \Qunknown'
group = 2, Admin state is up, Operation state is up,
incoming called-number = \Q', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
modem passthrough = nse, payload type = 117, codec = g711ulaw, redundancy,
huntstop = disabled,
application associated:
type = voip, session-target = \Qipv4:1.14.82.14',
technology prefix:
settle-call = disabled
ip precedence = 0, UDP checksum = disabled,
session-protocol = cisco, session-transport = udp, req-qos = best-effort,
acc-qos = best-effort,
fax-rate = disable, payload size = 20 bytes
codec = g729r8, payload size = 20 bytes,
Expect factor = 10, Icpif = 30,
Playout: Mode adaptive,
Expect factor = 10,
Max Redirects = 1, Icpif = 30,signaling-type = cas,
Table 3 explains the fields contained in both of these examples.
| Field | Description |
|---|---|
Accepted Calls | Number of calls from this peer accepted since system startup. |
acc-qos | Lowest acceptable quality of service configured for calls for this peer. |
Admin state | Administrative state of this peer. |
Charged Units | Total number of charging units applying to this peer since system startup. The unit of measure for this field is in hundredths of seconds. |
codec | Default voice coder rate of speech for this peer. |
Connect Time | Accumulated connect time to the peer since system startup for both incoming and outgoing calls. The unit of measure for this field is in hundredths of seconds. |
dest-pat | Destination pattern (telephone number) for this peer. |
Expect factor | User-requested Expectation Factor of voice quality for calls through this peer. |
Failed Calls | Number of failed call attempts to this peer since system startup. |
group | Group number associated with this peer. |
ICPIF | Configured Calculated Planning Impairment Factor (ICPIF) value for calls sent by a dial peer. |
incall-number | Full E.164 telephone number to be used to identify the dial peer. |
Last Disconnect Cause | Encoded network cause associated with the last call. This value will be updated whenever a call is started or cleared and depends on the interface type and session protocol being used on this interface. |
Last Disconnect Text | ASCII text describing the reason for the last call termination. |
Last Setup Time | Value of the System Up Time when the last call to this peer was started. |
Modem passthrough | Modem Passthrough signalling method is NSE. |
Operation state | Operational state of this peer. |
Payload type | NSE payload-type. |
Permission | Configured permission level for this peer. |
Poor QOV Trap | Whether Poor Quality of Voice trap messages have been enabled or disabled. |
Redundancy | Packet redundancy (RFC 2198) for modem traffic. |
Refused Calls | Number of calls from this peer refused since system startup. |
req-qos | Configured requested quality of service for calls for this dial peer. |
session-target | Session target of this peer. |
sess-proto | Session protocol to be used for Internet calls between local and remote router through the IP backbone. |
Successful Calls | Number of completed calls to this peer. |
tag | Unique dial peer ID number. |
VAD | Whether or not voice activation detection (VAD) is enabled for this dial peer. |
Related Commands
show call active voice Displays the Voice over IP active call table. show call history voice Displays the Voice over IP call history table. show num-exp Displays how the number expansions are configured in Voice over IP. show voice port Displays configuration information about a specific voice port.
Command
Description
To enter the voice service configuration mode and specify the voice encapsulation type, use the voice service global configuration command. To exit the voice service configuration mode, use the exit command.
voice service voip
Syntax Description
voip Specifies Voice over IP encapsulation.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
12.1(1)XA This command was introduced for VoATM on the Cisco MC3810 series. 12.1(2)T This command was implemented in Cisco IOS Release 12.1(2)T on the Cisco MC3810 series. 121(3)T This command was implemented in Cisco IOS Release 12.1(3)T for VoIP on the Cisco AS5300 Universal Access Servers.
Release
Modification
Usage Guidelines
Use the voice service command to switch to the voice-service configuration mode from the global configuration mode and to specify a voice encapsulation type. Use the exit command to exit the voice-service configuration mode and return to the global configuration mode.
Examples
The following example shows how to access the voice-service configuration mode and specify VoIP voice encapsulation, beginning in global configuration mode:
Router(config)# voice service voip Router(config-voice-service)#
Related Commands
modem passthrough Configures Modem Passthrough over VoIP.
Command
Description
DSPdigital signal processor
DSPWareThe firmware running on the DSP coprocessor
Modem PassthroughThe transport of modem signals through a packet network by using PCM encoded packets.
NASnetwork access server
NSENamed Signaling Event
PCMPulse Code Modulation
PSTNpublic switched telephone network
SAAService Assurance Agent
VADvoice activity detection
![]()
![]()
![]()
![]()
![]()
![]()
![]()
Posted: Tue Sep 19 17:48:59 PDT 2000
Copyright 1989-2000©Cisco Systems Inc.