This document describes how to tune a Cisco voice network using IGX
8400 series switches and switch software releases 8.2.5x and later.
The reader should be familiar with configuring Cisco equipment and
basic concepts such as:
The information in this document is based on these software and
IGX 8400 series switch CVM and UVM cards in networks using switch
software 8.2.5x or later
MGX 8850 series edge switch VISM card release 1.5.04
FastPAD Multimedia device using software 8.0.1 or later
Voice Network Switching
The information in this document was created from the devices in a
specific lab environment. All of the devices used in this document started with
a cleared (default) configuration. If your network is live, make sure that you
understand the potential impact of any command.
For more information on document conventions, refer to the
Cisco Technical Tips
AAL1—ATM Adaptation Layer 1. AAL1 supports
connection-oriented Continuous Bit Rate (CBR) voice and video. AAL1 is commonly
used for circuit emulation service transmission over ATM networks.
AAL2—ATM Adaptation Layer 2. AAL2 supports
connection-oriented Variable Bit Rate (VBR) packetized voice and video. AAL2
does not have Convergence or Segmentation and Reassembly (SAR) sublayers.
AAL5—ATM Adaptation Layer 5. AAL5 is a Simple
Efficient Adaptation Layer (SEAL). The common part AAL5 supports both
connection-oriented and connectionless Variable Bit Rate (VBR) traffic.
ADPCM—Adaptive Differential Pulse Code Modulation.
ATM—Asynchronous Transfer Mode. A
connection-oriented protocol for the transmission of voice, data and video
using fixed-length cells of 53 octets. The rate of the cells is not periodic,
hence the protocol is asynchronous.
CAS—Channel Associated Signaling. A method of
signaling that allows PBXs or channel banks to communicate with each other. CAS
is implemented by transmitting the signaling required for a single channel of
traffic in the channel itself or in a signaling channel permanently associated
CCS—Common Channel Signaling. Like CAS, CCS is a
method of communicating between PBXs or channel banks. It is more sophisticated
than CAS in that it uses a feature rich High-level Data Link Control (HDLC)
framed protocol for signaling like Q.931, DPNSS, or QSIG. CCS is implemented by
transmitting the signaling information for multiple channels of traffic over
one or two timeslots (typically timeslot 16).
Call Agent External—call control element also known
as a Media Gateway Controller. Monitors resources of the overall system and
maintains control of all connections. The Cisco VSC 3000 is a Call Agent.
Class 5—Class 5 refers to a type of switch used in
the PSTN to provide local services to the end user. This switch connects
end-users to the PSTN and provides custom features such as call waiting and
three-way calling. Examples of these switches include the Lucent 5ESS and the
Convergence—The amount of time needed to make a
working mathematical model of an incoming speech sample.
CS-ACELP—Conjugate Structure Algebraic Code Excited
DASS2—Digital Access Signaling System Number 2.
dBm—Power level in decibels relative to 1 milliWatt.
DID—Direct Inward Dialing. Calls can be dialed from
a telephone connected to an extension on a PBX to the public network without
going through an operator.
DOD—Direct Outward Dialing. Calls can be dialed from
a telephone connected to the public network directly to extensions on a PBX
without going through an operator.
Double Talk—The situation where parties at both ends
of a conference are speaking simultaneously. A quality echo canceller will
provide a continuous speech path in both directions during double-talk.
DPNSS—Digital Private Network Signaling System
DS-0—Digital Signal level 0. Part of the North
American transmission hierarchy, transmitting at 64 kbps. A DS-0 is one DS-1
DTMF—Dual Tone Multifrequency. Generic name for
push-button telephone signaling that uses two tones to represent each digit on
the telephone keypad. The tones are in two groups within the speech band, a low
band and a high band. They are geometrically spaced to ensure that any two
frequencies of a valid combination are not harmonically related.
E&M—Ear and Mouth. A basic analog signaling
method. E&M lead signaling is a specific form of interface between a
switching system and a trunk in which the signaling information is transferred
across the interface via two-state voltage conditions on two leads, each with a
ground return, and separate from the leads used for message information.
End Path Delay—Used in this document to mean the
time it takes for a signal to pass from the echo canceller to the point of echo
ERL—Echo Return Loss. The difference in strength
between the original signal and the echo being returned, minus the loss
incurred when a signal traverses a hybrid. ERL is measured in dBm.
ERLE—The efficiency of the echo canceller, measured
in decibels. ERLE is the attenuation added to the ERL.
Front-end Clipping—Front-end clipping is defined as
the first part of words not being transmitted in the speech stream. Front-end
clipping occurs when the first part of syllables (talk spurts) are not
recognized by the speech detector.
Hybrid—A circuit that converts between the 4-wire
and the 2-wire local loop.
HNGTM—Hangover Time. The amount of time voice
activity detection (VAD) stays on after speech is no longer detected. A longer
hangover time will smooth out choppiness but consume more bandwidth. A shorter
hangover time will add choppiness and decrease bandwidth consumption. HNGTM
only applies to connections using VAD. Hangover time is hardcoded in the UVM at
ISDN—Integrated Services Digital Network.
LD-CELP—Low Delay Code Exited Linear Prediction.
MF—Multifrequency. Push-button telephone signaling
that uses two of six possible tones to encode ten digits and five special
MGCP—Media Gateway Control Protocol as specified in
Media Gateway—The Media Gateway performs the mapping
and translation function between the IP and telephony networks. It is also
responsible for support services and network usage. The gateway is defined in
and in IETF drafts.
OAM—Operations, Administration, and Maintenance.
Special purpose ATM cells perform fault management, continuity checking, and
performance measurement functions.
Off-hook—Lifting the handset off the cradle closes
the switch hook, and current flows through the phone. The circuit is said to
have gone off-hook when the central office is informed that a subscriber
requires service. Off-hook is the opposite of on-hook. The terms on-hook and
off-hook describe the state of the signaling equipment regardless of the type
of signaling used.
On-hook—Returning the handset to the cradle opens
the switch hook, the current ceases to flow. The handset is now on-hook.
On-hook is the opposite of off-hook.
QSIG—The QSIG protocol provides signaling for
private integrated services network exchange (PINX) devices. It is based on
Telephony Union (ITU) Recommendation Q.931
Sidetone—Sidetone is an intentional byproduct of the
hybrid in the phone (for example, sound is transmitted from microphone to
receiver). A portion of the speech is allowed to bleed over into the ear piece
so that speakers can judge how loudly they are talking. Two speakers may
therefore experience quite different sidetone conditions at their respective
Signaling—Signaling is the exchange of information
regarding the establishment and control of connections. For example, CAS
signaling typically uses 2 bits on a T1 or 4 bits on an E1 to indicate on-hook
and off-hook status.
Talkspurt—The term used when one party on a
telephone call is talking. It applies to VAD from the time when speech is first
detected to the end of the hangover time.
Tandem—Tandem is used in this document to mean a
voice connection that undergoes the demultiplex/multiplex, decompress/compress
cycle in an intermediate switch before being routed to the destination.
VAD—Voice Activity Detection. The process used on
CVM or UVM hardware to determine whether a party is talking on one end of a
phone. If no party is talking, usually no data is transmitted and a significant
bandwidth saving can be achieved.
VISM—Voice Interworking Service Module. The Cisco
Voice Interworking Service Module (VISM) Release 1.5 is a front and back card
set designed to operate on the Cisco MGX 8850 Wide Area Edge Switch platform.
The VISM supports VOIP Switching, VOIP Multiservice Access with Call Control,
and AAL2 trunking.
VOIP—Voice Over Internet Protocol. VOIP is used in
this document to mean the transmission of voice traffic in packet
VNS—Voice Network Switching. A product from Cisco
that intelligently routes voice calls across a Cisco WAN Switching network.
The following commands are referenced in this Tech Note. Commands that
have a different syntax for earlier releases of switch software are indicated
in parentheses (). Due to functional differences in the cards, the CVM command
structure differs from the UVM command structure as follows:
CVM commands are in a slot.channel or a
(for example, 4.1-24)
UVM commands are in a slot.line.channel or a
format (for example, 4.1.1-24)
VISM commands use a different syntax and are presented in the VISM
All commands that require SuperUser-level access are indicated by an
asterisk (*). All commands that require Service-level access are indicated by a
double asterisk (**).
and Service-level Commands
This section describes the procedure for tuning voice connections in
the IGX 8400 series switch using the CVM. It is assumed that the reader is
familiar with the addcon command required to create
a voice connection in an IGX 8400 network.
There are three models of the CVM card: Model A , Model B, and Model C.
The major difference between Model A and Model B is that Model B allows dynamic
noise (or pink noise) injection. Noise is played out in the ear piece while the
remote side is not talking, which gives the impression that the line is
present. With the Model B card, it is possible to dynamically match the noise
at the remote end of a telephone conversation without passing large numbers of
management packets. The noise level is measured at the remote end and a message
is passed back to the source. A representation of the level is then played into
the ear piece. This feature is enabled by using the
cnfvchparm command and setting Bkgnd
Noise to zero. The CVM Model C card is used for connecting
contiguous bundles of up to 24 timeslots across an IGX network. Model C is
primarily used for legacy data applications. The
Differences in Functionality Between CVM
Models on an IGX Switch Tech Note provides more information about CVM
It is assumed that the PBX configuration details are available. If they
are not, some guidelines are provided in the PBX
Specifics section of this document.
To configure an E1 circuit line to the PBX using the cnfln
<slot_number> command, you need to know the physical
connection, ground requirements, whether CRC is used, and the type of PBX
The physical E1 type connection to the PBX can be either BNC or DB15
and can be grounded or not grounded. The BNC connection is 75 Ohm unbalanced
and the DB15 connection is 120 Ohm balanced. If the physical connection to the
PBX requires that earth ground be disabled, do the following:
for the E1 BNC interface, remove the nuts from both the Tx and Rx
connections of the BNC sockets on the CVM E1 backcard. Then use the
cnfln <slot_number> command and select the
75 ohm no gnd option.
for the E1 DB15 interface, remove the nuts from both the TX and Rx
connections of the BNC sockets as there are no other earthing options on the
CVM E1 backcard. An incorrectly configured E1 DB15 interface will not function
If the physical connection to the PBX requires earth ground, leave both
TX and Rx nuts in place. The G.703 convention is to connect the TX nut to earth
ground, but this does not provide any operational benefits on the CVM E1
An incorrectly configured E1 BNC interface that requires no earth
ground exhibits framing errors in the dsplnerrs
<slot_number> display after the line has been activated
using the upln <slot_number> command. A
correctly configured E1 BNC interface exhibits no errors in the
dsplnerrs <slot_number> screen.
The next stage to consider is whether there is error detection on the
voice channels or timeslots. Protection of data on the timeslots is performed
by running a CRC (called CRC4) in Timeslot 0. An incorrectly configured E1 will
result in CRC Errs in the dsplnerrs
<slot_number> display on the IGX 8400 or in CRC errors on
The final configuration step is to determine whether the PBXs are using
Channel Associated Signaling (CAS) or Common Channel Signaling (CCS) and to
reflect the setting on the IGX 8400 using the cnfln
Following are the main differences between CAS and CCS:
E1 CAS signaling uses ABCD bits for each channel that are passed
constantly in Timeslot 16
E1 CCS signaling uses a framed protocol passed in Timeslot 16 that
sends indications such as off-hook only when a change
E1 CCS signaling is feature rich; for example, with Q.931 and DPNSS
there are many supplementary services such as camp on.
If CAS is selected, the IGX automatically routes the ABCD signaling
bits between the PBXs on the connection. No addcon
command for timeslot 16 is allowed. This works for point-to-point
configurations and the more complicated point-to-multipoint networks. Set the
cnfvchtp <channel_number> command to monitor
timeslot usage. When cnfvchtp is correctly
configured to match the PBX signaling, the dspconst
screen indicates the status of a connection (for example, on-hook, off-hook,
modem upgrade). To determine the PBX signaling, issue the dspsig
<channel_number> command to view snapshots of the signaling
states, which can then be configured in cnfvchtp.
If the PBX is pulse dialing on the signaling channel, the
cnfchdl command is used to configure out-of-band
signaling to pass the signaling without distortion. When there are dissimilar
signaling systems between the PBXs, use the dspsig
command to obtain the signaling states and the
cnfrcvsig and cnfxmtsig
command to manipulate the signaling bits. For example, to convert from T1
E&M to E1 SSDC5a signaling the following settings may be used:
The conditioning criteria to apply to the signaling bits when the
connection is derouted can be configured using the
cnfcond command. Configuring
cnfcond allows a defined pattern to appear on the
ABCD signaling bits when the connection fails. This command also allows timed
pulses to be applied to the signaling to ensure the PBX returns to a known
If CCS is used, a transparent connection must be added between two PBXs
even though the data is framed. The CAS multipoint feature is not supported
with CCS on the CVM. CCS is enabled using the command addcon
<slot.16 node slot.16 t> where slot
refers to the CVM card position and node refers to the
remote IGX 8400. In addition to the addcon command,
ensure that cnfvchtp <slot.16> is set to
No Sig at each end of the connection in the IGX 8400
network. If cnfln is incorrectly configured as CAS,
a CCS PBX will not work. A CAS PBX will work if
cnfln is incorrectly configured for CCS, but
bandwidth will be wasted because the ABCD bits will pass through continuously.
CCS circuits do not allow for individual timeslot or signaling state
monitoring using the dspconst or
dspsig commands on the IGX 8400.
The normal clocking condition to the PBX is
normal, which implies that the CVM times the TX data and
expects the Rx data frequency to match. This means that the CVM is providing
clock to the PBX and that the PBX is using the receive timing to clock transmit
data out to the CVM. To configure, set cnfln to
Loop clock: No on the IGX 8400 and PBX to loop clock. If
the PBX is connected to a digital ISDN service or a Building Integrated Timing
Supply (BITS), then it is acquiring a clock reference from another source. In
this case declare the PBX to be a clock source to IGX using the
cnfclksrc command. If the PBX is not connected to
ISDN, BITS, or another known clock source, do not declare it as a clock source.
To ensure that the PBX clocking is consistent with the configuration:
Refer to the dsplnerrs screen to ensure
the clocking is not causing frame slips. The cnfln
command may be required to adjust the clock configuration to Loop or Local.
Verify that the PBX is not detecting frame slips.
Use the cnflnalm command to make the
alarming of both circuit line and trunk alarms more sensitive so the operator
is made aware of any problems.
After the physical and protocol sides of the circuit line have been
configured, bring the circuit online using the upln
command. After a few seconds the dsplns display
should show Clear - OK. If there are minor or major
alarms, check the physical interface and the cnfln
Use the dsplnerrs command to determine if
the link is working correctly. The information provided by the
dsplnerrs command is summarized below.
Integrated ("hard") Alarms
Bipolar Errors - number of times two consecutive pulses have
the same polarity (T1 lines only).
Loss of Signal (RED) - signal level at receive input is below
Frame Slips - number of times a frame is inserted or deleted to
reestablish synchronization. This is commonly caused by a clock mismatch
between the PBX and the IGX 8400.
AIS (BLU) - a string of 2048 or more consecutive ones has been
detected. This is known as a 'keepalive' signal sent in the downstream
direction of a fault.
Out of Frames - number of times a loss of frame synchronization
is detected on this circuit line.
Out of Frame (RED) - frame synchronization loss.
Loss of Signal - number of times that the signal level at the
circuit line input went below the minimum acceptable level.
Remote Out of Frame (YEL) - far end receiver out of
Frame bit errors - number of times the frame bit failed to
alternate (E1 lines only).
CRC errors - number of times the generated CRC character did
not match the received CRC character. (CRC checking must be enabled on E1 lines
using the cnfln command.)
AIS-16 - number of times the Alarm Information Signal (Blue
alarm) was received (E1 lines only).
Out of Mframes - number of times a multiframe synchronization
error was detected (E1 lines only).
Modem traffic is different from voice traffic in that voice traffic
consists of peaks and troughs of volume and has a mathematically modeled
variance. The CVM voice compression algorithms do not work well with high speed
modems. To avoid negative impact to modem traffic, after the CVM detects a
modem (for V.25 modems this is typically a 2100 Hertz tone), the connection is
upgraded from the current configuration to a pulse code modulation (PCM) clear
channel for the duration of the modem call. If voice is detected on the next
call, the connection is then downgraded to the original configuration (for
example, c32) to re-enable bandwidth savings.
The IGX 8400 routinely polls all the CVMs and UVMs to monitor the
status of modem calls. The modem polling interval can be adjusted using the
cnfnodeparm command or modem polling can be disabled
using the off1 command.
For tuning modem calls, the cnfcdpparm
command must be configured at both ends of the connection to:
adjust the modem/fax detection silence maximum (MDM Detect
Silence Max.) from 0C Hex (one second) to 24 Hex (three seconds).
The modem/fax detection silence parameter defines the amount of time a channel
stays in a modem/fax detected state.
adjust the modem stationary coefficient (MDM Stationary
Coef.) from 14 Hex to 25 Hex. The modem stationary coefficient is
used to differentiate between slow modems (< 4800 baud) and fast modems
The bandwidth of the connection upgrade also affects modem performance.
The cnfvchparm command must be configured at both
ends of the connection to enable a compressed voice connection (for example,
c32) to be upgraded to:
A 64 kbps clear channel connection that will work with any type of
fax or modem.
A 32 kbps ADPCM connection that is optimized for faxes at 9600 bps.
To troubleshoot CAS connections using test calls, it is necessary to
identify the timeslot the PBX is using. PBXs typically seize 64-kbps trunks
randomly and do not select the same trunk for multiple calls. This dynamic
behavior can prolong testing. Some PBXs can be configured to seize only one
trunk during out-of-service testing, but if the PBX technician or a maintenance
window is not available, the following procedure can be used to troubleshoot
Dial a fax machine at the far end that has a handset and at the
same time watch the dspconst screen.
When the remote fax answers, an "M" appears in the timeslot the PBX
has selected. After the handset is taken off-hook from the remote fax, the CVM
downgrades the call and the M eventually goes away. After the M is gone from
the dspconst screen, a voice call has been
Continuously press the "#" key on the local telephone keypad. If
the telephone does not generate a continuous tone, find one that does and start
Use the dchst <slot.channel> <1>
command to display the power level received from the handset. The
receive level should be -13 dBm.
Calculate the number of dBs of gain/loss needed to have a receive
level of -13 dBm. If the level received is not within +/- 3dB of this figure,
adjust the output levels of the PBX so that the received level is -13 dBm.
If PBX gain adjustment is not possible, use the
cnfchgn <slot.channel> command to adjust the
receive level to -13dB by inserting loss or gain into the input of the CVM. Use
the dchst <slot.channel> <1> command to
confirm that the level is correct.
At the far end of the connection, use the same commands to
configure gain/loss in the CVM output to compensate for the loss/gain inserted
at the input. This is to ensure that the signal leaves the IGX 8400 at the same
level it came in. The IGX 8400 network must have a flat response and insert no
loss or gain.
Ensure the signal level is similar when calls are made from a
variety of handsets, or from offnet or remote locations.
If the levels vary widely, review the voice loss plan.
Repeat this process in the opposite direction.
Do not assume that the gain/loss values will be identical for both
directions. Many PBXs are configured differently even in the same network.
After the gain/loss is configured for transmit/receive paths at
each end of the voice connection, configure the remaining connections with the
Voice Activity Detection (VAD) is the most complicated algorithm
implemented on the CVM. The VAD function requires the CVM to constantly monitor
every voice channel to detect the presence of voice or modem activity.
Depending on the type of connection configured for the channel, VAD determines
whether to build and transmit fast packets for the connection. The suppression
of fast packet transmission in the IGX 8400 network results in bandwidth
The following are VAD connection types:
The VAD algorithm performs the opposite function of the Adaptive Voice
algorithm. Adaptive voice was useful for the CVM Model A with VAD connections
that originated in a combination of quiet and noisy rooms. In that environment,
the CVM VAD static background noise injection algorithm was suboptimal.
However, the CVM Model B uses a dynamic background noise matching algorithm
that greatly improves VAD performance. For all CVM Model B connections using
VAD, Adaptive Voice must be disabled using one of the following commands:
When a VAD connection is added, the default channel utilization is 60
percent. Percent utilization is used as a factor in building a load
model. Each IGX 8400 maintains a static load model of connection
bandwidth and resource requirements. Based on the static load model, decisions
regarding connection routing are made. If a trunk does not have the bandwidth
available to support a target connection, an alternate trunk must be found or
the connection will de-route and traffic will stop. If a network has a lot of
callers in noisy conditions, utilization on voice connections may exceed 60
percent. In this case, the percent utilization must be increased to reflect
actual use. If the load model does not reflect actual use, there may be voice
(VAD connections) or non-timestamped (non-VAD connections) packet drops on
network trunks resulting in poor voice quality. The command
cnfchutl is used to increase or decrease channel
The default VAD settings work well for most connections. For
environments that require additional connection tuning, follow these steps:
Locate a point with average background noise characteristics in the
location under test.
From the test point in the building with
average listening conditions, place a call to a similar
location. Identify the timeslot the PBX is using for the call. If necessary,
follow the steps listed in the Adjusting Gain / Loss on CAS
Connections section to determine the timeslot.
Issue the dchst <slot_number.connection_number>
1 command and watch Registers 1 and 2 to determine whether cells
are generated when the two parties speak. If either party stops speaking (for
example, leaves the handset close to their head without putting their hand over
the mouthpiece), confirm that cell generation stops.
If cells do not stop when the speaker is silent, adjust the VAD
threshold with the cnfchvad command. Adjust the
VAD Mid Power and VAD Low Power
parameters to raise the VAD threshold. Note that the lower the VAD threshold,
the more FastPackets generated, and the more trunk bandwidth that's required.
The increase in required bandwidth must be reflected by increasing connection
utilization using the cnfchutl command.
If the VAD threshold is too high, front-end clipping will be
experienced. Make test calls from different points in the building to ensure
that VAD operates satisfactorily. The efficiency gained by VAD varies among
calls and among handsets. It is most important to ensure that the average call
has good efficiency and that all callers experience good quality.
Details of the dchst
<slot_number.connection_number> 1 screen are shown below.
Note that Channelized Data Pad (CDP) and Channelized Voice Module (CVM) are
i3 TRM SuperUser IGX 8420 9.1.13 Mar. 21 2000 20:05 CST
Channelized Data Pad state display for channel 16.1 Snapshot
Transmit dBm0: -70.0 Level of signal transmitted to the CLN
Receive dBm0: -67.0 Level of signal received from the CLN
Register 0 = 2B2D TX PCM Value (MSB) | RX PCM Value (LS byte)
Register 1 = FFFF TX Packet count (# of packets transmitted to Cell Bus)
Register 2 = FFFF RX Packet count (# of packets received from Cell Bus)
Register 3 = 1583 DSP # to which the current connection is assigned
Register 4 = 0000 Lost packet count for G.729 (g729r8) and G.728 (l16) connections
Register 5 = 3601
Register 6 = 160C
Last Command: dchst 16.1 1
The default VAD settings are provided below. CDP and CVM are again
i3 TRM SuperUser IGX 8420 9.1.13 Mar. 21 2000 19:30 CST
CDP Models All
Sample Bkgnd Power Thresholds ZCR Stat. Hang Pri Detect
>From 16.1 Delay Noise HPF High Mid Low High Low Coef. over Float upgrade
16.1-9 A8 67 ON 3160 40 40 50 15 30 42 ON 64K
16.12-24 A8 67 ON 3160 40 40 50 15 30 42 ON 64K
Last Command: dspchvad 16.1
The following table lists the hexadecimal values for integer dBm0
values used for the following parameters:
VAD High Pwr Thrsh (cnfchvad)
VAD Mid Pwr Thrsh (cnfchvad)
VAD Low Pwr Thrsh (cnfchvad)
MDM Low Pwr Thrsh (cnfcdpparm)
The CVM does not directly support video teleconferencing. The CVM can
provide bandwidth savings for video teleconferencing connections routed over
the CVM to PBX connection. The steps to configure a connection to support video
teleconferencing are as follows:
Connect the number of channels needed to support video traffic and
configure the PBX to bar these channels for voice.
Disable the Echo Canceller on the connection using the
Add connections in the IGX network as ?v? type using the
Set the gain to zero using the cnfchgn
Set the delay to Hex 01 and disable the high pass filter using the
When no video is being transmitted from the PBX, VAD detects
silence and suppresses fast packet generation.
Echo cancellers are used to eliminate echo caused by 2-wire to 4-wire
converters or hybrids in a telecommunications network. An echo canceller
achieves this by:
Modeling the measured echo on individual voice channels.
Subtracting the measured echo (echo replica) from the reflected
Continuously adapting to the echo (convergence).
Recognizing the difference between echo and speech.
Disabling echo cancellation when modems are used.
The following diagram illustrates how an echo canceller functions. Note
that the algorithm is independently performed on each channel (DS-0) of a T1 or
E1 signal. Thus, the echo that was introduced in the analog portion of the
circuit is eliminated in the digital portion of the circuit.
The echo canceller is inserted between the circuit line termination of
the IGX and the connected PBX or channel bank. The echo canceller continuously
observes the signal (speech) going from the IGX to the PBX (transmit
direction). The echo canceller stores the transmitted signal and compares it to
the received signal. Choosing moments when there is no speech in the receive
direction, the echo canceller assumes that all the energy coming from that
direction is echo caused by reflections at the 2-wire termination on the tail
side of the call. Therefore, the signal should be a delayed, attenuated version
of the original signal that is already stored. The echo canceller uses the DSP
to calculate the delay and reduction in the original signal necessary to
completely cancel out the received signal. This process is called
convergence and is used to create a mathematical model of
the echo delay and amplitude of the echo in the tail circuit. The calculation
is then applied continuously to the call, reducing the reflected portion of the
received signal by at least 30 dBm.
The echo canceller at each end of a call reduces the echo in each tail
circuit so that the echo is imperceptible, even at the level of delay
introduced by the IGX 8400. If PBX hybrid balance is good, set the echo return
loss to low using the cnfchec and
cnfecparm commands to improve the convergence time
of the echo canceller. It is important for the signal level coming into the IGX
to be set correctly to get the best voice quality, best efficiency for the VAD
algorithm, and best performance for the echo cancellers. To set the correct
gain/loss in the IGX, you must make a test call and measure the signal strength
level using the dchst command.
Due to the differences in paths and terminations, the convergence
process must be repeated at the beginning of each call. The echo canceller uses
signaling information and speech energy to determine when a call is beginning.
While it is possible to configure some cancellers to converge based on changes
of the signaling bits, most cancellers attempt to converge continuously
whenever speech is present. When coupled with VAD, the echo canceller will
attempt to converge at the beginning of each talkspurt. In conditions of high
reflected signal (low echo return loss), this can result in the talker hearing
echo at the beginning of phrases.
For call paths with echo return loss (ERL) higher than 6 dBm, set
configurable echo cancellers to a value of 0. For call paths with low ERL (6-10
dBm), use the value of 6. The canceller can converge much faster if the ERL is
known. If the ERL diverges from the configured value, the canceller will have
great difficulty converging and bad echo will result. Convergence may take from
20 to 200 milliseconds.
Another difficult situation for the echo canceller is double-talk. It
is impossible to run the echo calculation when both callers are talking.
Therefore, the echo canceller must recognize double-talk and continue
cancellation based on the information before double-talk was detected. Echo
cancellation may be poor or other anomalies may occur if double-talk is
detected too late or not at all.
There is usually some form of residual suppression, center-clipper, or
nonlinear processing feature in echo cancellers. This feature recognizes that
signals with very low power are usually mixed up with noise. To guard against
some of this noise being echo, the canceller suppresses it all and transmits
idle code instead. This may give rise to clipping on quiet calls, particularly
when double-talk is present and the two directions of the call have very
different power levels.
An enhancement that center-clipper provides is noise matching. The
noise matching function recognizes that some calls may suffer from choppy voice
due to the level of background noise during speech being changed to silence
while the signal is clipped. The noise matching function continuously samples
the noise level before the echo cancellation in the receive direction, and
injects an appropriate level of noise after the clipper. The listener no longer
hears noise discontinuities due to the center-clipper. Noise matching should
generally be left enabled in the echo canceller, even on VAD connections. This
function prevents the remote listener from hearing background noise
discontinuities caused by the IGX building and transmitting packets during
silent periods, such as during VAD hangover time (HNGTM).
Echo cancellers have a tone detection feature to identify fax and fast
modem calls. Echo canceling is disabled when a 2100 Hz tone is detected and is
not re-enabled until the end of the call. The end of the call is identified by
signal power decreasing below a threshold. For normal applications, this
feature should be enabled.
The CVM supports an optional 24-channel or 32-channel integrated echo
canceller (IEC) which provides:
on tail circuits with:
The IEC performs the same function as an external echo canceller.
However, the integrated echo canceller is located between the gain/loss
insertion circuitry and the packet assembly and disassembly circuitry. Note
that the command dchst
<slot_number.channel_number> displays the input and output
dBm levels at the point between the gain insertion and the echo canceller.
The IEC provides basically the same configurable internal options as a
Tellabs echo canceller. The dspecparm and
cnfecparm commands are used to monitor and configure
the parameters of the optional IEC on the CVM. The configurable options are
referenced to the corresponding parameters of the
cnfecparm command described below.
i3 TRM SuperUser IGX 8420 9.1.13 Mar. 22 2000 16:41 CST
IEC Slot 16 Parameters
1 IEC Echo Return Loss High (.1 dBs) [ 60] (D)
2 IEC Echo Return Loss Low (.1 dBs) [ 30] (D)
3 IEC Tone Disabler Type [ G.164]
4 IEC Nonlinear Processing [Center Clipper]
5 IEC Nonlinear Processing Threshold [ 18] (D)
6 IEC Noise Injection [ Enabled]
7 IEC Voice Template [ USA]
Last Command: cnfecparm 16
Parameters 1 and 2 specify options which can be selected for each
channel of the circuit line in multiples of 0.1 dB. The echo return loss value
selected represents the minimum ERL required for the echo cancellation circuit
to be enabled. If the measured ERL is less than the value specified, the signal
is not considered echo and the echo convergence mechanism is frozen, even
though echo is still canceled based on the most recent echo model.
Parameter 3 allows selection of the G.164 or G.165 tone disabling
protocol to support high-speed modem transmission. The G.164 protocol is the
older mechanism originally developed for echo suppresser technology. It
requires the detection of a 2100 Hz sine wave to disable the echo canceller.
The G.165 protocol requires the detection of a 2100 Hz sine wave with phase
reversals every 450 milliseconds. Two of these phase reversals are required to
disable the echo canceller. Low speed modems generate the 2100 Hz sine wave and
high speed modems generate the 2100 Hz sine wave with phase reversals. This
implies that G.164 can work with both low and high speed modems. It is
recommended to use G.165 for most connections.
Parameter 4 allows selection of either the standard center clipper
technique or the newer multiplying technique. In the conventional center
clipper mechanism, any post-cancelled signal below the threshold specified in
Parameter 5 is eliminated using an abrupt transition. If it's enabled in
Parameter 6, it's replaced by low-level synthesized noise. Using the
multiplying technique, the transition from signal to injected noise is done
slowly, over a period of approximately one second. It is recommended to use the
center clipper technique for most connections.
Parameter 5 specifies, in dBm, the threshold at which the nonlinear
processor is activated. If the ERL of the tail circuit plus the ERLE provided
by the echo canceller exceeds this value, the remaining signal will be
Parameter 6 allows the user to enable or disable the noise injection
function. If this function is disabled, silence is sent when the nonlinear
processor is activated. When nonlinear processing is enabled in the center
clipper mode, noise injection is optionally enabled to fill the periods when
the echo signal is being cut off. The level of the noise injected is dynamic.
It is approximately equal to the background noise content of the current
In a VAD application noise injection should be disabled since the IEC
inserts its noise from the far end of the network. Packets would have to flow
to get the artificial noise to the listener. If you are going to use dynamic
noise insertion, disable the echo cancellers injected noise. In addition, using
the cnfvchparm command set the inserted noise level
to "0." This enables the dynamic noise insertion feature of the CVM Model B
card. (If you have model A cards, please contact your supplier. These cards
will not work with dynamic noise injection). Reset the card with the command
resetcd <slot_number> h to ensure the
parameters are downloaded to the card. The noise injection schemes used by the
Model B card and the echo canceller are different.
Parameter 7 allows the user to select the USA or UK echo cancellation
template. The UK template is provided solely for better performance in
environments using analog tail circuits, which are typical in the UK. This
provides a high-power input into the network. The USA setting should be
interpreted as a low power input.
The dspchec and
cnfchec commands allow the user to monitor and
specify the parameters that determine the operations of a single channel or a
range of channels of the optional IEC on the CVM. The
cnfchec command allows per-channel configuration to:
Enable or disable echo cancellation.
Select the high or low minimum ERL set with the
Enable or disable the disabling of the echo cancellation due to tone
Enable or disable the convergence function.
Enable or disable the nonlinear processor function.
Display the voice template selected for the line with the
i3 TRM SuperUser IGX 8420 9.1.13 Mar. 22 2000 17:04 CST
Echo Echo Return Tone Conver- Nonlinear Voice Bkgrnd
Channels Cancel Loss(.1 dBs) Disabler gence Processing Tmplt Filter
16.1 Enabled Low 30 Enabled Enabled Enabled USA -
16.2-24 Disabled High 60 Enabled Enabled Enabled USA -
Last Command: cnfchec 16.1
Echo Cancel [enable|disable]. Enable or disable IEC. The IEC is
bypassed by "disable".
Echo Return Loss [high|low]. Selects one of two options configured in
the cnfecparm command. The IEC will not converge
while the echo signal is within this amount of the speech signal. If this
setting is higher than the ERL, the IEC will not converge. Choose "high" only
if the ERL is at least 3 db better than the "high" setting configured using the
Tone Disabler [enable|disable]. The tone disabler feature allows the
IEC to detect a preamble tone associated with dial-up modems and disable itself
when a modem is detected. This is essential for high speed full-duplex modems.
Convergence [enable|disable]. Disabling the convergence function of a
channel has the effect of freezing the echo canceller in its current state,
preventing it from doing any further improvement or modification of the
normally adaptive echo cancellation process. This configuration is typically
used only for troubleshooting.
Nonlinear Processing [enable|disable] Since there is always a small
amount of echo that bypasses an echo canceller, it is sometimes desirable to
process this residual echo in a nonlinear fashion. If nonlinear processing is
enabled, the IEC stops sending all data when the echo signal is sufficiently
below the speech signal. There is a configurable threshold that defines how far
below the speech signal the echo signal must be before the nonlinear processing
Voice Template [USA|UK]. Each of these template choices represents a
set of internal IEC parameters which are not otherwise available to the user.
The USA template is optimized for voice levels approximately from -10 dBm0 to
-50 dBm0. The UK template is optimized for voice levels that reach above -10
dBm0 to +3 dBm0. When the UK template is chosen, IEC performance on voice
levels in the -10 dBm0 to -50 dBm0 is compromised. The UK template should only
be used when voice levels are extremely high.
Echo cancellers represent a relatively complex solution to a complex
problem. However, there are some straightforward measures which can improve
Verify that all echo cancellation parameters are correct.
Reduce, as much as possible, the level of echo (ERL) seen by the
echo canceller. Adding loss in the tail circuit is always helpful. Sometimes
it's possible to find the specific 2-wire termination causing echo. Echo can be
improved by line build-out or impedance options on trunk cards. Replace 2-wire
circuits with 4-wire circuits to eliminate hybrid echo.
Echo cancellers can usually accommodate up to 32 milliseconds of
tail circuit delay. If the delay is close to this limit, an extended version of
the echo canceller may be needed.
Echo cancellers have difficulty with double-talk when the signal
levels in each direction of a call differ by more than 10 dB. It may be
possible to change the network loss plan to allow for this signal level.
Delay introduced by the IGX 8400 may be reduced on VAD connections
by configuring Sample Input Delay from A8 to 50 if only on-net calls are made.
Reducing delay may improve echo canceller performance.
IGX 8400 network changes to route voice connections over the
smallest number of hops and to balance load evenly across trunks will reduce
delay and may improve echo canceller performance.
For troubleshooting purposes, test the problem connection with VAD
disabled and again as an uncompressed connection (p-type) to isolate the source
This section describes the procedure for tuning voice connections in
the IGX 8400 series switch using the UVM. It is assumed that the reader is
familiar with the addcon command required to create
a voice connection in an IGX 8400 network.
The UVM was introduced in switch software release 8.2.5x and is also
supported in releases 8.5, 9.1, 9.2, and later. The feature set of the UVM
varies depending on the release of the switch software. The features and
performance discussed in this section are with respect to switch software
release 9.1.13 and later using UVM firmware Model E version D (DED).
The UVM is the next generation voice card for the IGX. Most of the
commands used for the CVM apply to the UVM. For example, the dchst
<slot_number.line_number.channel_number> command is used to
monitor signal power levels. The sections relating to line features for the CVM
are the same and are not repeated here. UVM features include:
Channelized T1/E1/J1 interfaces
Voice coding types
ADPCM (G.726): 32 channels per UVM
LD-CELP (G.728): 16 channels per UVM
CS-ACELP (G.729): 16 channels per UVM
CS-ACELP (G.729A): 32 channels per UVM
Two line interface ports
Voice activity detection (new per channel
Integrated echo cancellation on back card using Mitel chipset.
Fax relay support for G.729 connections (new
Super-rate data connections
CAS switching only for VNS support (refer to the
VNS section of this document)
D-channel compression for CCS support using UVM Firmware DED and
D-channel compression for VNS signaling support (refer to the
VNS section of this document)
CAS or CCS signaling
Pass through (new cnflnpass command)
The UVM supports 16 channels configured for G729 voice compression. To
allow for full T1 or E1 capability, two UVM card sets must be chained together.
To chain UVM cards, add an external cable between two UVMs in the same chassis
and link them using the cnflnpass command. See the
"UVM Pass Through" diagram above.
To configure pass-through, separate channel numbers must be used to
denote the UVMs in separate slots. For example, to connect to a PBX with an E1
CCS interface using UVMs in slots 12 and 13, issue the following commands:
upln 12.1 – the Passing line
upln 12.2 – the Blocking line
upln 13.1 – the Inserting line
cnflnpass 12.1 13.1 – pass-through
requires the primary card to use line 1
Repeat commands 1 through 4 at the remote IGX 8400.
addcon 12.1.1-15 <remote_nodename> 12.1.1-15
addcon 12.1.16 <remote_nodename> 12.1.16
addcon 13.1.17-31 <remote_nodename> 13.1.17-31
Within the IGX network there are connections between cards 12 and 13,
but on the PBX side there are 30 connected timeslots with signaling to card 12.
The UVM supports fast modem detection and introduces a new feature, fax
UVM fast modem detection is available in all supported releases of
switch software. The UVM V.25 modem detector recognizes the steady 2100 Hz tone
output by V.25 fast modems (> 4800 baud) and fax machines to disable echo
cancellers at the beginning of transmission. This function is called the fast
modem capability and is the default configuration on UVM connections. The
monitoring of connections to detect fast modem calls is performed by the switch
software modem polling function. Modem polling can be
disabled using the off1 command or the frequency of
polls changed using the cnfnodeparm command. After a
fast modem has been detected, the connection can be upgraded from the current
compression level to 32 kbps or 64 kbps using the
cnfvchparm command. Modem polling is used to
determine when a modem call has been disconnected so that the connection can be
downgraded to the original bandwidth (usually less than 32 kbps or 64 kbps),
saving network resources.
To verify that a connection is using fast modem detection, (1) issue
the cnfchfax command and verify that the Fax field is disabled, and (2) issue
the cnfvchparm command and verify that the
V.25 Detect field is not disabled (either 64KB or 32KB).
Note that the V.25 Detect field is the only field
applicable to the UVM for the cnfvchparm command.
UVM fast modem upgrade to 32 kbps is not supported for g729ar8 and
g729ar8v connections. For g729ar8 and g729ar8v connections, configure the V.25
Detect field to 64KB.
For better fast modem performance, it may be necessary to change the
silence duration from 1600 to 5100 or higher. Issue the
cnfuvmchparm command and configure the SIL DUR
For networks with heavy fax use, the Line pct fast
modem parameter in the cnfln command may
need to be increased. This parameter is known as the percent fast modem
function and it is used by switch software to ensure that there is enough
CellBus bandwidth available to the UVM for the large number of FastPackets that
are generated for all the simultaneous fax calls. The Line pct fast
modem default is 20 percent, but this may need to be increased to 40
percent or higher to more closely represent fax usage over the UVM line. If the
Line pct fast modem parameter is too low, FastPackets will
be dropped. This will impact all voice connections on the card (that is, bad
voice quality) and cause faxes to be detected but unable to maintain
communication as observed using the dspconst screen
and noting a rapid change from M to +. The Line
pct fast modem parameter does not affect the load
Fax relay is a new feature. When a fax call is detected, the current
voice compression is disabled and replaced by a fax demodulation/modulation
algorithm. The algorithm manages fax negotiation and then transports the data
across the IGX 8400 network at 9.6 kbps or lower, as the fax bit stream
requires. The new cnfchfax command can be used to
enable or disable the feature on both sides of the connection.
Fax relay can be configured for G.729A connections but it is not
supported. This is due to the way that DSPs are allocated on the UVM card. Each
DSP will support two G.729A connections but only one fax relay call. If fax
relay is needed, use g729 or l16 which uses a whole DSP. This can be confusing
as it is possible to configure fax relay on a G.729A, however the firmware will
prevent the connection from upgrading.
Fax relay is delay sensitive and connections with long round trip
delays may not be able to support it. Long round trip delays are caused by:
The operation of UVM VAD is similar to VAD on the CVM. The
cnfuvmchparm command is used to tune VAD. Following
are VAD connection types:
The following default settings for VAD on the UVM displayed in the
cnfuvmchparm command may need to be tuned for better
voice performance. Changes to the cnfuvmchparm
settings must be made at both ends of a connection. Improving voice quality
typically comes at the cost of bandwidth savings.
NSE INJ (noise injection). Units are -10 dBm0.
Range is 1-15. Typically configured to 8 to represent -80 dBm.
VAD THLD (VAD threshold). Units are -1 dBm0.
Range is 1-255. Typically reduced from 40 (-40 dBm) to 45 or lower. Do not
configure noise injection and VAD threshold to be the same value.
MDM THLD (modem threshold). Units are -1 dBm0.
Range is 1-255. Below this threshold, the modem tone is ignored or not
detected. Typically configured at 40.
SIL DUR (silence detection window size). Units
are 20 milliseconds. Range is 1-255 (20 milliseconds - 5.1 seconds). Silence is
detected if the signal level stays below the silence detection threshold in
dBm0 for the duration specified by the silence detection window size in
milliseconds. Silence detection is used to determine when a fax/data modem call
SIL THLD (silence detection threshold). Units
are -1 dBm0. Range is 20-80.
7 (enable DC offset filter). For a PBX that
sends a non-standard idle code, a DC filter has been added during the
computation of bi-directional silence. This silence detection is used for
downgrading a connection in the V.25 modem state. The default value is 0
(disabled). To enable the filter, use 1.
8 (upper convergence speed threshold.) Used to
tune echo convergence threshold. Range is 12 (fastest convergence time) to 30
(slowest convergence time). Note that the fastest setting will increase the
instance of reconverge during normal pauses in speech such as the beginnings of
distinct sentences. Reconverge during speech may result in a brief period of
echo in the middle of a conversation.
9 (double talk detection threshold). 0 is equal
to the default value of 5 dB.Only configured to improve echo cancellation on
circuits with very poor ERL (<5 dB).
The PIU LVL setting is now hard-coded in the UVM
firmware. The PCM Interface Unit (PIU) value entered by the user must be zero.
The default value for this parameter is zero. The value is always displayed as
The echo canceller function is provided by the Mitel MT9122 chip
located on the UVM back card. It has the following features.
Adaptive filter for estimating the echo channel
Subtractor for canceling the echo
Double talk detector for disabling the filter adaptation during
periods of double-talk
Non Linear Processor for suppression of residual echo
Disable tone detector for detecting valid disable tones at the input
of receive and send paths
Narrow Band detector for preventing adaptive filter divergence caused
by narrow band signals. For example, if dual tone multifrequency (DTMF) tone is
present, this may cause the adaptive filter to diverge.
Offset Null filters for removing the DC component in PCM channels
12dB attenuator for signal attenuation
PCM encoder/decoder compatible with Mu/A
UVM echo cancellers are always active, but they need to reconverge
every time the ERL (or some other characteristic of the echo) changes. It is
possible that the ERL changes significantly when the called party goes from
ringing state to off-hook state, or goes from talking state to hold state. You
can determine the ERL for each of these states by sending DTMF tones and
watching Rx/TX levels on the dchst screen. A low ERL
during the ringing state or hold state would explain the echo heard during
A diagram of the Mitel MT9122 is shown here.
The Voice Interworking Service Module (VISM) uses high performance
digital signal processors and dual control processors with advanced software to
provide a fully nonblocking architecture that supports the following functions:
Eight T1 and E1 interfaces per service module, up to 24 service
modules per MGX 8800
Programmable echo cancellation up to 128 msec
ATM adaptation layer 1 (AAL1), AAL2, and AAL5 standards support
Primary Rate Interface (PRI) support
Fax and modem tone detection for compression and echo cancellation
Standards-based alarm and fault management
Simple Network Management Protocol (SNMP) configuration and access
Redundancy with standby switchover
PCM (G.711) support for a total of 192 DS0's per VISM
ADPCM (G.726) support for a total of 145 DS0's per VISM
CS-ACELP (G.729A/B) support for a total of 145 DS0's per VISM
No R1 and R2 signaling support at this time
CAS protocol support by mapping the CAS signals to Simple Gateway
Control Protocol (SGCP) events
For Release 1.5.04, the VISM supports 2 modes of operation:
VOIP Switching. In this mode, the VISM functions as a Media Gateway
to perform call control in conjunction with a Call Agent such as the Cisco
VSC3000 to provide voice service over existing packet networks.
AAL2 Trunking. In this mode, the VISM function is similar to the CESM
and also offers echo cancellation and G.711/G.726/G.729A/G.729B compression. No
Call Agent is required.
AAL2 Trunking and VOIP switching modes are incompatible and can not be
implemented on the same VISM. The default mode is VOIP Switching. If AAL2
Trunking mode is selected, the VISM will reset and any existing configuration
will be erased. A reset may take as long as five minutes. To view the existing
mode, issue the dspvismparam command.
mgx1.1.11.VISM8.s > dspvismparam
VISM mode: voipSwitching
CAC flag: enable
DS0s available: 240
Template number: 2
Percent of functional DSPs: 100
IP address: 0.0.0.0
Subnet mask 0.0.0.0
RTCP report interval: 1000
RTP receive timer: disable
Aal2 muxing status: disable
Tftp Server Dn TFTPDOMIAN
VISM firmware is bundled with MGX 8850 firmware. The VISM release train
is different from the MGX 8850 release train. This is reflected in the use of
unique identifiers in the firmware image filenames. Once a CCO user has
selected the MGX 8850 firmware bundle with the desired VISM release, the images
need to be downloaded and unbundled. All registered CCO users who are logged in
have access to the
Downloads - WAN
(registered customers only)
for firmware downloads.
Clocking is only supported on VISM line 1. The VISM uses line 1 to
derive clock used to transmit data for remaining seven T1 lines.
The VISM supports AAL2 Trunking through the VISM T1 back card or
through the SRM T3 interface. Use VoAAL2 as a solution for point-to-point
applications as well as for Integrated Voice/Data Access using other Cisco
multiservice access products such as the Cisco 2600 series, 3600 series, and
MC3810. An application for AAL2 Trunking is illustrated below.
The VISM supports VOIP Switching through the VISM T1 back card. VISM
operates in conjunction with a Call Agent, such as the Cisco VSC 3000, and two
networks are used. The Call Agent connects to the telephone SS7 network and
handles the call-control signaling. The VISM connects to an IP network (over an
AAL5 PVC) and handles the voice payload between the calling and called parties.
VISM and Call Agent communicate with each other and their activities
are coordinated through either SGCP or MGCP. In VOIP Switching mode, the VISM
also supports CAS backhaul through data networks. For CAS backhaul, the VISM
translates standard trunk signaling protocols to SGCP (MGCP) messages and
transmits the messages to the Call Agent. The trunk signaling translation
information is stored on the VISM in a file called a CAS variant. Supported CAS
The FastPAD Multimedia (FastPADmm) can no longer be ordered. Cisco will
support the FastPAD in customer networks until 2003. The following steps
describe how to adjust voice level settings at local and remote sites when
using the FastPAD Multimedia (FastPADmm).
Establish a connection with a person at the remote site.
From the Configure menu, select the expansion channel number where
the connection is established.
Using the <down arrow>, step through the configuration to the
point where the In and Out levels are set.
Speak to the person at the remote end and get feedback on how you
sound to them. If your voice sounds quiet, change the In at your end to be more
negative. If your voice sounds loud, change the In at your end to be less
negative. Continue adjustments until an appropriate level is reached.
Have the person on the other end speak, and adjust the Out
parameters at your end accordingly.
Repeat this process for each voice channel.
Save the settings to active and save the configuration.
The Cisco MC3810 is a compact, low-cost multiservice access
concentrator that integrates data, voice/fax, and video signals and connects
them to Asynchronous Transfer Mode (ATM), Frame Relay, or leased-line networks.
Seamlessly integrates data, voice, and video
Leased line, Frame Relay, and ATM service compatible
Cisco IOS® based multiprotocol routing, bridging and System Network
Two serial ports for packet data protocols, SNA
Six analog or 24/30 digital voice ports
Quality voice compression at 8 kbps (G.729, G.729A) or 32 kbps ADPCM
Per-call voice switching
Fax Relay to 9.6 kbps
Circuit emulation over ATM for video
Digital Access and Cross connect System (DACS) compatible drop/insert
structured trunk option
Seamless interoperability and management
The Cisco MC3810 supports pulse dialing with the following commands:
dial-type pulse, timing
pulse, and timing pulse-inter digit.
These commands are documented in the
MC3810 Multiservice Concentrator Configuration Guide and the
MC3810 Command Reference Guide.
Multi-Chassis Hunt Groups
Hunt group support has been enhanced to route an incoming call to
another configured outgoing trunk if it fails to terminate locally because all
ports are busy or if the intended outgoing trunk is down or congested. Using
multichassis hunt groups, the Cisco MC3810 can hunt between both local dial
peers and network dial peers on the terminating or tandem Cisco MC3810. The
system hunts among local dial peers first, and then hunts to the network dial
peers. A preference order defined with the preference command applies only
within the peer group, so all local peers will be searched first, even if a
network peer exists with a higher preference.
A/B Bit Conditioning
Cisco IOS Software Release 11.3(1)MA4 supports three new voice port
ignore, and define. These
commands allow the Cisco MC3810 to recognize and manipulate different ABCD bit
combinations such as on-hook and off-hook signals from the PBX.
In previous releases, the Cisco MC3810 that terminated a voice call
would only forward digits that exceeded the destination pattern. In Cisco IOS
Software Release 11.3(1)MA4, you can control the number of digits forwarded to
the telephony interface. This is critical in configuring a hierarchical dial
Voice Default Route
In previous releases, you could not set a voice default route using
all wildcards. In Cisco IOS Software Release 11.3(1)MA4, you can set a default
voice route for any fixed length dial string using all wildcards with the
Japan and Australia Call Progress Tones
Modifications have been made to support Japan and Australia call
Common Channel Signaling Support for International
Applications Including QSIG
In addition to the Voice switching capabilities currently available
in the MC3810 for FXS, FXO, and E&M, the MC3810 supports dynamic calling
using the ITU standard QSIG for Common Channel signaling. This implementation
supports full voice switching for both T1 (23B+1D) and E1 (30B+1D).
Transparent signaling for CCS PBXs is also supported providing
compressed voice over Frame Relay and ATM support for nearly any CCS-based PBX.
Bandwidth is allocated to voice calls dynamically using VAD.
These capabilities are backward compatible with systems using the
digital voice module currently shipped.
End-to-End Networking (IGX 8400 Internetworking)
The MC3810 can be used to extend high quality of service (QoS)
backbones using the Cisco IGX 8400, terminating voice and data on the MC3810 in
smaller branch offices. The IGX 8400 provides a scalable, robust head-end
solution to voice and data networking using the MC3810. Refer to
Cisco MC3810-IGX Interworking for additional information.
Full IOS Feature Set Support
The MC3810 offers the full range of Cisco IOS routing capabilities
including IP, X.25, AppleTalk, DecNET, Vines, and others.
Robust Applied Telephony Capabilities
The packet telephony capabilities of the MC3810 have been enhanced
to provide a more robust integrated Data/Voice/Video solution for branch
Call Detail Record (CDR)
The MC3810 supports collection and export of call records to a
central information base. Information included is time of call, originating
port, terminating port, and duration.
Multiflex Trunk module with integrated BRI
This module provides all the same functionality as the existing
Multiflex module but now provides an additional interface for BRI data backup.
The BRI module provides an S/T interface only, which is ideal for European
deployment. An inexpensive NT1 can be used to provide connectivity to ISDN
services in the US.
Facilities Data Link Capabilities on Multiflex Trunk
This feature provides carriers the full-featured remote management
capabilities that they demand from a manageable CSU.
This feature allows a port on the MC3810 to act like an
Off-Premise Extension to the PBX. When the PBX attempts to
make a connection to the remote voice port on an MC3810, OPX ring-through
allows the PBX to re-route the call if there is no answer.
Preference-Based Hunt Group
The multichassis hunt group is enhanced to allow use of the
preference command for selecting remote dial peers
before local dial peers using the priority values. This greatly extends the
capability of the product to support on-net to off-net call rerouting and
alternate call center applications.
This newly supported vocoder will be more reliable for digit
transport in networks with greater hop counts and support lower speed modems
(up to 9.6 kbps).
Multilength Dial Patterns
Dial strings of multiple lengths can now be supported in the same
network and on the same MC3810.
A large number of PBXs have been integrated with the MC3810.
Most PBX networks today use a tandem architecture
in which all calls are routed through one or more centrally located nodes
before reaching their destinations. This approach has several disadvantages:
It requires many E1/T1 or Fractional E1/T1 lines to support the trunk
groups that are needed for tandem connections. This type of trunking is very
bandwidth inefficient because traffic must be backhauled through intermediate
switching nodes and numerous small trunk groups carry less traffic than a
single large trunk group.
Multiple tandem hops affect voice quality. PBX tandem networks are
not amenable to voice compression because speech signals must go through
multiple compression/decompression cycles before reaching their final
destinations. The result is impaired voice quality, more voice compression
cards, more PBX trunk cards, and many signaling channels.
Separate trunk groups are needed for data and video communications
because compression can be applied to only voice connections. As noted,
multiple separate trunk groups are less efficient than a single large one.
PBX feature operation requires signaling channels on each trunk group
to carry feature messages between locations. These numerous signaling channels
each require 64 kbps of bandwidth.
Cisco Voice Network Switching (VNS) offers the solution to traditional
PBX networks. VNS works in conjunction with Cisco IGX 8400 series wide-area
switches to provide switched virtual circuits (SVCs) for voice and data
transmission over a Cisco wide-area Asynchronous Transfer Mode (ATM) or Frame
Relay network. Customers with tandem private branch exchange (PBX) networks
realize substantial savings on facility costs, simplified network topology, and
improved bandwidth efficiency with a VNS/IGX backbone. In addition, the VNS
system architecture is designed to provide scalability for small to very large
Voice Network Switching, in conjunction with standard QSIG and DPNSS
common channel signaling protocols, provides direct call-by-call routing for
PBX voice, data, and fax connections, enabling this information to be
transported across a Cisco WAN with efficiency and economy. It uses one hop
routing, which avoids multiple compression/decompression cycles, and removes
several shortcomings of existing PBX networks. VNS revolutionizes PBX
communications with advanced switching and signaling technology.
Industry standard signaling protocols for ISDN, Frame Relay, and ATM
are supported by VNS. The flexibility of the VNS platform also enables Cisco to
respond rapidly to changes in networking standards. Key standards include:
With VNS, voice calls are compressed once at the origination point and
decompressed once at the destination point. Voice quality is improved by
eliminating multiple compression/decompression cycles. Improved voice
transmission quality can increase the capacity of the network because more
aggressive voice compression can be used. For example, 16 kbps voice
compression can be used instead of 24 or 32 kbps compression. Network voice
capacity can be doubled without incurring additional bandwidth costs while
still maintaining acceptable quality. Another benefit of eliminating the
multiple compression/decompression cycles is that fewer voice compression
processor resources are required. These benefits are leveraged by the
comprehensive voice capabilities already available with Cisco IGX 8400
Voice Activity Detection (VAD) sends information only when speech
energy is present. When a person is not talking, no data is sent. Because each
individual speaks only 40 to 50 percent of the time during a typical
conversation, VAD can provide a 50 percent savings in bandwidth for voice
connections. When combined with ADPCM compression techniques, VNS provides
unparalleled efficiency and economy in transporting voice on ATM networks.
The UVM supports CAS for VNS by converting CAS signaling and DTMF tones
to CCS call-control messages.
The converted CCS messages for all channels on the line travel on a
regular t-type or a special td-type PVC connection from the UVM to another UVM
card. The VNS device can receive the CCS messages from t-type or td-type PVC
connections on the signaling channel of CAS-switching UVM cards in the network.
The UVM supports D-channel compression for VNS release 3.1 This feature
compresses the signaling traffic between the application UVM and the network
(VNS) UVM. D-channel compression reduces the consumed bandwidth from 64 kbps
per VNS signaling channel to 16 kbps or less. It applies to CCS lines or CAS
lines where the CAS-switching feature is operating. To enable D-channel
compression issue the following command:
Another advantage of VNS is that it enables users to migrate from a
tandem PBX backbone to a Cisco backbone without any change in network
VNS supports feature operation with the DPNSS and QSIG common channel
signaling protocols, which transmit feature messages as well as call setup and
tear-down messages. The Cisco IGX 8400 switches pass these messages
transparently to the destination PBX, operating as transit PBX nodes. This
transit node functionality provides a standards-based pass-through capability
that should support all PBXs that conform accurately to the DPNSS or QSIG
protocols. VNS simplifies PBX network design and management. Complex routing
tables, multiple trunk groups, and the complicated network topology normally
associated with PBX tandem networks are not required. Instead, the Cisco
backbone handles these tandem functions more efficiently by dynamically routing
all calls directly to their destinations based on a simple dialing plan.
VNS appears as QSIG "Transit Node"
Most PBX supplementary service messages are passed transparently
across an IGX network
Network dialing plan in VNS database (based on E.164)
E1 Primary Rate Interface connection to IGX CVM
Industry standard ISDN/QSIG signaling protocol stack:
Layer 2: Q.921
Layer 3: Q.931 call control
European Computer Manufacturers Association (ECMA) 143 basic call
ECMA 165 generic functional protocol
VNS appears as DPNSS "Transit Node"
PBX supplementary service messages passed transparently across IGX
Dialing plan in VNS database (based on E.164)
E1 Primary Rate Interface connection to IGX CVM
Industry standard DPNSS signaling protocol stack:
In some networks ATM trunks are used to transport voice traffic. To
ensure efficient use of ATM trunk bandwidth, IGX configuration changes may be
required. If the FastPacket to ATM cell combining values for voice connections
are less than those specified in the table below, trunk bandwidth may be
wasted. In the least efficient scenario, only one FastPacket (24 bytes) will be
transmitted as the payload of one ATM cell (53 bytes). The Service-level
command, cnfcmb, is used to change the FastPacket to
ATM cell combining value by adjusting the trunk wait time for a FastPacket. A
larger value indicates that the trunk card will wait a longer time for a second
FastPacket before sending out the ATM cell. In most cases, changing
cnfcmb significantly increases available trunk
bandwidth and minimally increases voice packet delay.
g729r8v or g729ar8v
g729r8 or g729ar8
This section explains how to configure a PBX to simplify the connection
to an IGX network. A common problem with new installations is improperly tuned
Not addressed in the following sections, but equally important is the
PBX clock configuration. The PBX clock source must be exactly the same as the
IGX clock source, or the clock sources used by the PBX and IGX must have the
same accuracy and stability. Equivalent clocks are required to keep the slip
rate between the PBX and IGX at an acceptable level.
Note: When you add connections between PBXs located in geographically
diverse locations, the cnfclnsigparm <slot>
command may be required to adjust for delay.
If voice connections are routed over tandem switches, the PBX network
must be tuned to avoid signal loss and degradation. Loss plans are covered in
detail in International Telephony Union (ITU) Recommendation G.171
ANSI TIA/EIA TSB 32
loss must be assessed at a number of locations to obtain the worst case
scenario when designing a network. The method for tuning PBXs described below
Follow the procedure in the following diagram.
The Integrated Services Digital Exchange (iSDX) is a family of digital
PBXs designed for business requirements of 30 to 3000 extensions. There are
four iSDX systems: iSDX-T, iSDX-L, iSDX-S, and iSDX Micro. All iSDX systems are
based on a common hardware platform and common software. Over 17,500 iSDX
systems have been sold in more than 40 countries worldwide, incorporating in
excess of 5.5 million lines.
The iSDX is at the forefront in the development of DPNSS and has the
highest level of compliance to this standard of any PBX. iSDX is the benchmark
with which all other PBXs must interwork.
A caveat with analog voice connections on the iSDX is that the default
connection type is SSDC5a, which is a variant of E&M Type V signaling. A
caveat with power and ground is that a nonstandard pin from the iSDX allows the
passing of ground between the PBX and the Cisco equipment. The iSDX does not
use the electrical earth that is supplied with the AC PSU. There is a specific
external earthing point for this purpose. There is no information about
problems with the DC version.
When attaching to a Northern Telecom Meridian with an analog trunk
configured for E&M TIE Line, one of the trunk parameters is CPAD. The
choices to configure this parameter are:
C OUT, which sets the trunk's input and output levels to 0 dBm. This
is the default setting.
C IN, which sets the trunk's input level to +7dB and output level to
The C OUT setting is preferred. The C IN setting results in low volume
from the PBX, which affects IGX VAD and onboard echo cancellation.
The Ericsson MD110 uses a proprietary protocol called System Link that
needs 256 kbps of bandwidth for remote PBX downloading. The 256 kbps of
bandwidth is distributed over timeslots 1, 3, 5, and 7. These timeslots must be
configured as transparent connections on the IGX (for example, no voice
compression can be used). There is no transparent connection requirement if
QSIG or another CCS-type protocol is used.
The table below shows a list of the clearing codes transmitted between
Clearing / Rejection Cause
Clearing / Rejection Cause
Number not obtainable
Message not understood
Signaling system incompatible (DPNSS) Reserved
Transferred (DPNSS) Reserved (DASS2)
Subscriber changed number
No reply from subscriber
Invalid request for supplementary service
Out of service
Subscriber out of service
Incoming calls barred
Outgoing calls barred
DTE controlled not ready
Remote procedure error
DCE controlled not ready
Subscriber call termination
Signal not understood
Invalid signal (DPNSS)
Service temporarily not available (DPNSS)
Local Procedure error
Facility not registered (DPNSS) Reserved (DASS2)