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Table of Contents

Modem Passthrough over Voice over IP

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:

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.

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).


Figure 1: Modem Passthrough Connection


Benefits

The Modem Passthrough over VoIP feature offers the following benefits:

Restrictions

Cisco IOS Release 12.1(3)T is required to run on the gateways for the Modem Passthrough over VoIP feature to work.

Related Features and Technologies

Related Documents

The following documents provide additional platform-specific or hardware related information to help implement VoIP:

Supported Platforms

Cisco AS5300 Universal Access Server gateways

Supported Standards, MIBs, and RFCs

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

Prerequisites

Configuration Tasks

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:

Configuring Modem Passthrough over VoIP Globally

For the Modem Passthrough over VoIP to operate, you need to configure Modem Passthrough over VoIP in both the originating gateway and the terminating gateway.

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)#exit

Exits from voice-service configuration mode.

Step 4 

Router(config)#exit

Exits the global configuration mode.

Configuring Modem Passthrough over VoIP for a Specific Dial Peer

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)#exit

Exits the global configuration mode.

Verifying Modem Passthrough over VoIP

To verify that the Modem Passthrough over VoIP feature is enabled, follow these steps:


Step 1   Enter the show run command to verify the configuration.

Step 2   Enter the show dial-peer voice command to verify that Modem Passthrough over VoIP is enabled.


Troubleshooting Tips

To troubleshoot the Modem Passthrough over VoIP feature, perform the following steps:

Monitoring and Maintaining Modem Passthrough over VoIP

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.

Configuration Examples

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
 

Command 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   The modified commands are marked by asterisks.

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

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
Release Modification

12.1(3)T

This command was introduced for the Cisco AS5300 Universal Access Servers.

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
Command Description

voice service voip

Enters the voice-service configuration mode and specifies the voice encapsulation type.

modem passthrough (dial-peer)

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}

no modem passthrough

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
Release Modification

12.1(3)T

This command was introduced for the Cisco AS5300 Universal Access Servers.

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
Command Description

dial-peer voice

Enters dial-peer configuration mode.

show call active

To show active call information for a call in progress, use the show call active command in privileged EXEC mode.

show call active {voice [brief]}

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
Release Modification

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.

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.


Table 1: show call active Command Field Descriptions
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 2833—are 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
Command Description

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.

show call history

To display the fax call history table for a fax transmission, use the show call history command in privileged EXEC mode.

show call history {voice [last number | brief]}

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
Release Modification

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.

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.


Table 2: show call history field Descriptions
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 2833—are 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
Command Description

show call active

Displays active call information for a call in progress.

show dial-peer voice

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
Release Modification

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.

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.


Table 3: show dial-peer voice Field Descriptions
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
Command Description

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.

voice service

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
Release Modification

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.

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
Command Description

modem passthrough

Configures Modem Passthrough over VoIP.

Glossary

DSP—digital signal processor

DSPWare—The firmware running on the DSP coprocessor

Modem Passthrough—The transport of modem signals through a packet network by using PCM encoded packets.

NAS—network access server

NSE—Named Signaling Event

PCM—Pulse Code Modulation

PSTN—public switched telephone network

SAA—Service Assurance Agent

VAD—voice activity detection


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Posted: Tue Sep 19 17:48:59 PDT 2000
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