Adding Voice Connections on IPX and IGX Switches

Document ID: 10720

Introduction
Prerequisites
      Requirements
      Components Used
      Conventions
How to Configure Voice Connections
Related Information

Introduction

Cisco INS® voice network switching hardware and software can be used in conjunction with Cisco IGX® and Cisco Internetwork Packet Exchange (IPX) switches equipped with channelized voice interface modules. This configuration provides switched virtual circuit capability over the ATM wide-area network for voice and data calls originating from private branch exchanges (PBXs). Voice network switching provides direct call-by-call routing for PBX voice, data, video, image, and fax connections across a Cisco StrataCom ATM WAN.

Voice network switching also supports circuit-based synchronous data transmission for video-conferencing and data connectivity. PBX data switching modules connect to video coder-decoders (codecs), and data terminals and controllers.

For more information, refer to the Voice Connections section of the Cisco WAN switch documentation.

Prerequisites

Requirements

There are no specific requirements for this document.

Components Used

This document is not restricted to specific software and hardware versions.

Conventions

For more information on document conventions, refer to Cisco Technical Tips Conventions.

How to Configure Voice Connections

Adding voice connections in IPX is not a simple matter of merely issuing an addcon command. The full procedure is described here:

Prepare the PBX port.

If either 120 ohm or 75 ohm without ground is required, you must remove the nuts from all of the BNC sockets on the E1 back card of the Channelized Data PAD (CDP) front card. If you do not, the interface will be unbalanced by the nuts earthing one connection to ground. This causes a 120 ohm connection to malfunction. It also causes the 75 ohm connection that needs to operate without ground framing errors to appear in the "dspclnerrs" display.

Once the circuit line is up, if the "dspclnerrs" screen is error-free, then the earthing arrangement is okay. Remember, for 75 ohm without ground, you must remove the nuts and select 75 ohm no ground in the "cnfcln" screen.
Configure the circuit line.
1. With the cnfcln command, select the correct type of PBX port: 75 ohm, with or without ground, or 120 ohm.
2. Select the option according to whether it uses channel associated signaling (CAS), or Common Channel signaling (CCS). A CCS PBX does not work if you have incorrectly selected CAS, but a CAS PBX still works even if you have incorrectly selected CCS. Using the CCS, however, wastes bandwidth, so it is worth getting it correct.
3. Also, indicate whether the PBX gives a cyclic redundancy check four counts in Timeslot 0. If you make the wrong selection, either the PBX or the IPX shows CRC errors as an alarm.
4. Select the clocking to the PBX. The standard condition is normal, which implies that the CDP times the Tx data, and expects the Rx data frequency to match.
Up the circuit line.

Issue the upcln command. After a few seconds the display "dspclns" should show clear. If not, check the physical interface and the cnfcln parameters.

Issue the dspclnerrs command to determine if the link is working error-free.
Configure the clock sources.
1. If the PBX is connected to a digital ISDN service, it acquires a clock reference from the carrier. In this case, declare the PBX to be a clock source to IPX with the cnfclksrcs command.
  
  If the PBX is not connected to ISDN, do not declare it as a clock source.
2. Ensure the PBX clocking is consistent with whatever you have selected.
3. Refer to the "dspclnerrs" screen to ensure the clocking is not causing any frame slips. Ensure the PBX is not detecting any frame slips.
4. Issue 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.
Adjust the CDP parameters.

With the cnfcdpparm command, adjust parameter 14 to set the modem or fax detect silent period to 24 Hex from 0C Hex. Failure to do so results in unreliable fax or modem detection.
Add the Signaling Channel.
- For CCS signaling (E1 systems) issue the addcon slot.16 node slot.16 command:
  This will add the signaling channel as a transparent 64Kbps connection.
- For CAS systems, no action is required.
Enable echo cancelers.

Issue the cnfchec command to enable echo cancelers for timeslots 1 to 15, and 17 to 31. Do not enable for Timeslot 16. If you do, the signaling channel does not work.

If the balance of the PBX hybrid is good, set the echo return loss to low to improve the convergence time of the echo canceler. It is best to determine which gives the best performance, through experimenting.

Leave the other echo canceler parameters as default.

Remember to issue this command at both ends of the link.
Configure the echo canceler parameters.

Issue the cnfecparm command to set the default parameters for the echo canceler. If you intend to use the dynamic noise insertion, then disable the echo canceler's injected noise. Reset the card with the resetcd (slot) h command to ensure the parameters are downloaded into the card.
Configure voice channel parameters.

With the cnfvchparm command set the inserted noise level to 0 (zero). This enables the dynamic noise insertion feature of the CDP Model B.

Note: If you do not have CDP Model B, contact Cisco for an upgrade from Model A to B. The Model B noise insertion firmware is a significant enhancement. Check your model CDP with the dspcds command. The first letter of the alpha code beside the CDP card in the display is the model number.

Decide if, on detection of faxes or modems, you want the CDP to promote the call to a clear 64 kbps which works with any kind of fax or modem, or if you want to promote it to a fax-optimized 32 kbps ADPCM that works with faxes at 9600bps. Select either 64 or 32 in the cnfvchparm command. Remember to issue this command at both ends of the link.
Add voice connections.

Issue the addcon command to add the connections.

If selecting 16 kbps ADPCM, always select =EBc16z=ED or =EBa16z=ED. This is better quality than the standard 16 kbps selection.
Configure CAS signaling.

If you are using CAS signaling, you must correctly set the cnfvchtp command if you want to get statistics on Timeslot usage. If this command is correctly set, the "dspconst" screen tells you when a connection is off-hook. If you do not know what signaling is being used by the PBX, the screen "dspsig" gives you snapshots of the PBX signaling state, which you can then relate to the options in cnfvchtp.

If the PBX is pulse-dialing on the signaling channel, you must set the cnfchdl command correctly to use outband signaling in order to pass the signaling undistorted.

If connecting between dissimilar signaling systems, issue the cnfrcvsig and cnfxmtsig commands to manipulate the signaling bits.

Issue the cnfvchtp command to let the software know what the off-hook condition is for this channel. Once this is set correctly, issuing the dspconst command shows you which connections are on-hook and which are off-hook.

You should also define what conditioning criteria you desire to apply to the signaling bits when the connection is unrouted. The cnfcond command allows you to define a pattern to appear on the ABCD signaling bits upon connection failure. This command also allows timed pulses to be applied to the signaling to ensure the PBX returns to a known state.

By this stage, your PBX should be synchronized and you can make calls over the IPX system. Make some test calls to ensure the signaling system is working reliably.
Configure the channel gain.

It is important for the signal level coming into the IPX to be set correctly to get best voice quality, to enable the echo cancelers to operate at peak performance, and to get best efficiency from the Voice Activity Detection.

In order to set the correct gain / loss in the IPX, you must make a test call and measure the signal strength level with the dchst command.

First, identify which timeslot the PBX has decided to use for your test call. This is easier said than done, as PBXs often use random timeslots. The easiest way to find which timeslot the PBX has selected is to call a fax machine (that has a handset) at the far end.
1. Dial the fax machine and watch the "dspconst slot 1" screen (at the same time).
2. When the remote fax answers, you should see an M appear in the timeslot position the PBX has selected.
3. Instruct your colleague to pick up the handset on the remote fax. This way, when it times out, the call is not cleared down. The M goes away after a time. This is normal behavior.
4. On the voice link to your colleague, press the digit # on your telephone keypad continuously. (If your telephone does not give a continuous tone, find one that does and start again).
5. Issue the dchst slot.channel 1 command.
  
  Look at the level received from the handset. Calculate how many dB's of gain / loss are needed to have a receive level of -13dBm.
  
  If the level received is not within +/- 3dB of this figure, adjust the output levels of the PBX so that the received level is -13dBm. This is not always possible, however, because customers are often reluctant to change PBX setups.
  
  Alternatively, issue the cnfchgn slot.channel command.
  
  Adjust the receive level to -13dB by inserting loss or gain into the input of the CDP.
  
  Confirm with the dchst command that it is correct.
6. At the far end of the connection, put in gain / loss in the CDP. This causes the signal to leave the IPX at the same level it came in, and there is a flat response across the IPX network.
7. Ensure the signal level is similar when calls are made from a variety of handsets or from off-net or remote locations. If the levels vary widely, it might be wise to talk to the client about their voice-loss plan.
8. Now set all the channels to have the same characteristics.
  
  Repeat this process, but instruct your colleague at the far end to hold down the digit #, so that you can set up the gain / loss in the opposite direction. Do not assume it to be the same as in the previous direction, since many PBX's are set up differently – even in the same network.
Once you have set up the gain / loss in both directions for transmit and receive paths for one voice channel, you can then set all the other voice channels to be the same.

Adjust the VAD threshold.

You must now determine if the VAD threshold is correct and giving good bandwidth efficiency.

From an average point in the building with average listening conditions, place a call to a similar location. If necessary, use the trick with the fax machine to determine which timeslot the PBX has selected.

Issue the dspchstats slot.channel 1 command:

Watch as cells are generated when the two parties speak. If either party stops speaking (leaving the handset close to their head, and without putting their hand over the mouthpiece), confirm that cell generation stops.

If cells do not stop under this condition, adjust the VAD threshold with the cnfcdpparm command:

Adjust the VAD Mid Power, and VAD Low Power parameters to raise the VAD threshold.

This table contains some example values:

VAD Mid power, Low Power	VAD Threshold
0020	-57dBm
0040	-54dBm Default
0080	-51dBm
0100	-48dBm
0200	-45dBm
0400	-42dBm
0800	-39dBm
1000	-36dBm

If the VAD threshold is too high, clipping is experienced.

Make more trial calls from different points and ensure that VAD operates satisfactorily. Do not expect every call to be exactly the same. The efficiency gained by VAD varies from call to call, and from handset to handset. It is most important to ensure that the average call has good efficiency, and that all callers have good quality.

Configure the channel utilization.

The IPX defaults a channel's utilization to 40 percent.

If your network has a lot of callers in noisy conditions, it may not be possible to achieve 40 percent average utilization. In this case, you should increase the utilization factor to whatever is being achieved, or to what you decide is your target utilization.

Use the cnfchutl command to change a channel's utilization. (This command has no effect on voice quality, but is used instead to determine how much reroute traffic can be supported down a trunk.)

Related Information

Updated: Apr 17, 2009

Document ID: 10720

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