It’s one of the most frustrating and complex problems that you’ll ever face deploying IP PBXs such as Asterisk, FreeSWITCH and 3CX when interconnecting with analog telephone (PSTN) lines – ECHO!
The usual first reaction is to add some hardware echo cancellation and hope for the best, but it usually is not the best solution. The echo is still there, you just have a powerful DSP hiding it for you. By solving the underlying problem(s), you’ll ensure that you have the best quality audio possible.
Choosing a telephony card or gateway with hardware-based echo cancellation is an excellent decision, and we strongly recommend their use, but they are an insurance policy, designed to remove echo caused in places you cannot fix, like at the far end of the call.
Today we’ll look at the common causes of echo in IP PBX systems with analog lines, and how best to fix each of them. Whether you’re using a Digium, Rhino or Sangoma card, or an external SIP gateway with FXO or FXS ports, this guide has some practical suggestions to help you on your way to echo-free calling.
My old phone system didn’t have echo!
This is a common and understandable complaint in new IP PBX installations with problematic analog telephone lines.
There are several reasons why an IP PBX (or another other sufficiently advanced PBX) can have serious echo problems even though your old mechanical key system or PBX worked just fine. The biggest reason is that VoIP introduces some additional latency that doesn’t exist on a simple old phone system which is just a big mechanical switch.
There are some other reasons too, like the fact you’re connecting a piece of grounded electrical equipment, which can amplify imbalances in your lines, but a lesson in electrical theory isn’t going to fix your echo problem.
Wiring mistakes
The number one cause of echo is the use of Cat5/Cat5e/Cat6 in place of proper Cat3 telephone cabling. You should never use anything other than Category 3 wiring for analog telephone wiring.
Why, you might ask? The signals for both the incoming and outgoing audio is sent simultaneously on the same copper pair. This requires that the circuit at both ends, yours and the CO, are “balanced”, so that the audio can be separated again, so that you do not hear echo. When you use non-Category 3 wiring, you alter the impedance (usually 600ohms in North America), creating an imbalance in the circuit, and the equipment on the line at both ends can now no longer properly separate the transmitted and received audio.
Also, telephone cabling should not be run in close proximity parallel to AC power cabling. This can further exacerbate imbalance problems, and also cause problems like audible humming.
Solution: Use only Category 3 cabling for connecting from the telephone company’s demarcation point to your equipment, and for connecting analog telephones. Do not use Cat5 or higher cable. Avoid cabling parallel to power cables and ballasts.
Wrong impedance settings
Your equipment, be it a card in your Asterisk system or a standalone gateway, should have a configurable setting for impedance matching. In North America, most telephone lines have an impedance of 600 ohms, but some, particularly on very long circuits or in rural areas, have an impedance of 900 ohms. Elsewhere in the world, a variety of different impedances are used, and you should select the right one for your location.
Even if you are certain of the impedance value that should be used for your line, if you are facing echo problems, you should try the different settings offered by your card or gateway to discover the one that best matches the properties of your line.
Solution: Check that your device is configured for the correct impedance, and try other impedance settings to see if they help.
Bridge Taps
A bridge tap is the situation where you have a pair of wires which “tap” somewhere along another pair of wires, creating a three way connection. Kind of like a “T-junction” for telephone wires. Such connections used to be quite common in the days of party lines.
These days, they are most likely to be found inside your home or office, rather than outside your control on the telephone network, unless you live in a very old or very rural area. Having extensions of a telephone line to multiple rooms in your house or office that have nothing connected to them will create a bridge tap.
Bridge taps cause a “reflection” of the signal, because the signal travels up the unused part of the “T”, and then reflects back down into the device that is connected to the line, causing unwanted echo. If the tap is long enough, it will also seriously degrade any DSL service you might have on that line.
Although far less common than some of the other causes, a bridge tap can cause a significant echo problem.
Solution: Disconnect any old, unused internal extension wiring at the source. Avoid the use of splitters or “Y” connectors.
Low Quality IP Phones
Some lower quality and “no-name” IP phones have very poor echo cancellation, especially when used in speakerphone mode. These can be the source of echo, by allowing the received audio to be transmitted again because of being too loud or poorly isolated, or they can make an existing echo problem even worse.
Solution: If you use your speakerphone a lot, make sure you are using quality phones from a manufacturer with quality speakerphone support, such as Aastra, Cisco or Polycom. If you’re stuck with the phones you’ve got, try turning the volume or gain settings down.
Other Reasons…
There are many other problems that can cause audio problems like echo and DTMF detection problems, such as:
- Induced sources of AC power on the line
- Loading coils
- Very high or very low loop current
These problems are all quite a bit trickier to diagnose and solve, and you should involve your PBX vendor/consultant, telephone company or a qualified technician for help in testing for and solving these kinds of problems. You can also contact Telephonyware for advice.
Sangoma has today announced the upcoming availability of their new D100 hardware transcoding card and software to enable high density server-based voice and data communications applications with up to 480 channels of voice transcoding per card.
