Echo Elimination & DTMF Problems

The easiest echo to fix is:

  1. Echo that you or the person at the other end of the call always hears on a VoIP phone system when you’re talking on an analog line or trunk
  2. Echo that you or the person at the other end of the call always hears on a regular phone system connected to a VoIP phone line (adapter), and where you don’t hear the echo when you connect your butt-set directly on the line (with the phone system disconnected).

What to check

  • Use a butt-set or 2500 set and check for echo off of the analog pairs, if there is a problem using an analog set, then nothing in this article can solve it and you’ll need to contact 611 line repair. However, if this sounds clean, then it is likely that it will not be fixed by 611 line repair, unless detailed below.

Why is echo a problem on VoIP and not other technologies?

That hybrid is the source of most echo problems that you can fix. The hybrid is basically two transformers wound around a core, or an electronic version of that. Some hybrids are better than others. What makes them better? They’re more efficient, meaning that there is less “leakage” between transmit and receive, or reflected back out the two wire side. On a regular 2500 set, the hybrid in the network purposely sends some of the transmitted signal from the mic to the receiver, which is called sidetone (which basically permits you to hear a low volume copy of what you’re speaking. The hybrid used on a trunk card is much more efficient than a 2500 set because the sidetone isn’t needed on a phone system (it’s created at the station so the conversation is more natural to the user).

Even though the hybrid on a trunk card is efficient, it’s not 100% efficient. Some of the transmitted data “leaks” and gets mixed with the received data, or reflected back to the caller on the 2 wire side. It’s at such low volume that the leakage just doesn’t matter using an analog or regular digital phone system. This has never mattered until now because you can’t hear it on Analog or Digital trunks (it’s happening at almost the exact same time, so you don’t hear it as “echo”).

A digital phone system takes analog speech and runs it through a codec (Coder-Decoder) that digitizes or “undigitizes” the audio. The codecs on traditiona l digital TDM phone systems are so fast that there is very little delay, and even though some of the transmitted data is mixed with the received data in the hybrid, the delay is so small that we’ve never noticed it (but it’s there). You’d be hard pressed to hear any difference between a call on an analog phone system and a digital (TDM) phone system. VoIP is a different story…

When the Internet was created, it was determined that packets of data would be much more efficient traveling long distances than a single digitized stream of data (like TDM in a digital phone system or on a T1). Packetization means that the web page you’re looking at could have gotten to your computer through several paths across the country. Some of the packets may have gone through Florida, and some may have gone through California. The delays for each packet are normally much less than 100ms (a tenth of a second) per segment between routers on the Internet (it may take quite a few segments to get to you). Your computer puts the packets together in the correct order so you see everything in the right place on your screen. If it takes half a second longer to get the top part of the web page than the bottom part, nobody cares.

When people decided to put voice on the Internet, they had to divide the conversation up into packets. That’s the only way the public Internet can get data from one place to another. There have been many methods of digitizing voice through the years. Until the Internet it really didn’t matter how it was done – the results were essentially the same. Unfortunately, the only way to digitize voice and put it on the Internet is to packetize it for the Internet using IP (Internet Protocol). While these packets work great for data, which can arrive a little late or in the wrong order, the whole concept works badly for voice over the public Internet.

One of the troubles with creating packets for IP is that it takes a while for the codec to do its work – digitizing the voice and putting it in the the packets. It’s not a real long time, but it’s long enough for us to start hearing that inefficiency in the hybrid because there’s a delay. With analog or TDM, there was little or no delay and we just didn’t notice the inefficiency as echo, even though it was there.

The better the hybrid matches the phone line, the less you’ll notice echo on a VoIP phone system. If all phone lines were exactly the same, the hybrid would be as efficient as possible. Because all POTS lines are different in one way or another, the hybrid is never as efficient as it could be.

Echo doesn’t just effect voice. While it’s annoying to talk on a call with echo, there’s a good chance that an automated attendant or voice mail system won’t work correctly on calls with echo. The DTMF digits will look pretty strange to the device hearing them. A 60ms DTMF digit followed by 50ms of an echo of that digit can cause recognition problems that are tough to diagnose, especially if the echo is intermittent on incoming calls.


Before you begin, be sure that the problem is repeatable from multiple sources!

First, try the built in configurations settings of the Voice Gateway; be sure to test after each change:

  • Adjust the “Echo Canceller” on the SPA9000 (Admin>Advanced>Voice>Regional)
  • Adjust the volume of the phone line (db) in the SPA400 – be careful, because too much may make it difficult to hear.
  • Adjust the impedance of the SPA400 – alternating between 600 & 900 ohm;
  • Adjust the impedance of the SPA400 – using other impedances around 600 & 900 ohms (without capacitors mf, uf & pf)
  • Adjust the impedance of the SPA400 – using .47mf + resistance settings

Other more extreme measures:

  • There may be load coils in the line which can degrade the quality – while analog voice may not be problem, you may experince problems with modems, xDSL and VoIP. You can try to call 611 line repair and have them check for load coils in your line. If you can get them to dispatch, tell them to make sure the tech comes on site with a TDR (Time Domain Reflectometer) which will help them locate the load coils.
  • Check for high loop current – place a DMM (digital multi meter) in series with a butt-set — and make sure you’re set to DC ma (and there are typically different connectors on the DMM to measure DC current). When you’re on-hook (hung-up) you should read 0 to .01ma; but when you go off-hook, you should get something in the range of 23 to 35ma. Much higher and they’ll be problems, and you can call 611 line repair due to high loop current.

Source: Mike Sandman Enterprises, Inc.