Why is there an echo on my VoIP calls? How can I remove it?
Posted by Gabriel Yu on 15 April 2008 01:32 AM

Echo is not inherently a VoIP issue – the PSTN has been fighting echo for decades, especially with long-distance circuits. While the digital portion of VoIP cannot introduce echo, increased delay times due to codecs and buffering quickly makes even the slightest echo received very annoying. 

In PSTN the echo is most commonly generated where a 4-wire circuit is converted to a 2 wire circuit (note that telephone handsets are 2-wire devices); or by a poor quality handset.  These are fundamental analogue telephone design issues.

Firstly determine the characteristics of the echo:

  • Does the echo occur
    • on PSTN calls through the LINE port (Vigor 2100V, 2500V or 2700V (2S1L) models only) ?
    • on VoIP to VoIP calls ?
    • on VoIP to PSTN calls ?
  • Is it a 'talker' or 'listener' echo ?
    • Talker echo occurs when some proportion of the "talker's" voice is echoed back to the talker - i.e. the person speaking hears their own voice. A small amount of this is required, so that the talker doesn’t think that the line is dead.
    • Listener echo occurs when a second reflection causes some proportion of the Talker echo to go back to the listener.  This results in the listener hearing an echo of the talker's voice.
    • Convergence echo occurs at the start of a call, and automatically fixes itself.
  • Is the echo heard at your end, or the other end ?
    • If you hear echo, then the source of the echo is either the far end or somewhere in the PSTN network.  
    • If the far end hears echo then you generate the echo. 
Why does the echo occur ?
Echo is caused by:
  1. imperfect impedance matching in the “hybrid” (4-wire to 2-wire conversion point) in the analogue PSTN network.  Hybrid echo is an electrical signal reflection.
  2. acoustic feedback where the microphone of a telephone handset is picking up the sound from the speaker, possibly being carried by the materials of the handset itself. Can be caused by poorly designed echo cancelling circuitry in the telephone handset.
  3. Convergence echo occurs at the start of a call, and results from the time taken for the echo canceller to "converge".

As mentioned before, echo is present in all PSTN calls, but the short distance of local calls means that the echo is very quick, which means that we hardly notice it.  Long distance calls produce more delay, which creates a noticeable echo – so telephone companies have added expensive echo cancelling hardware to compensate.

Inherent in VoIP are several delays (time taken to do A-D encoding and decoding, jitter buffer to put packets back into order before playing).  When calling VoIP to VoIP on a fast line (with low ping times) the delay is not significant. However, when calling to PSTN the VoIP delays added to PSTN issues above are sometimes enough to cause an echo on local calls.  Curiously VoIP to long distance PSTN rarely experiences echo because the telcos’ echo cancellation filters are applied.  

What can be done to fix echo ?

There are a couple of mechanism to prevent echo; called ERL (Echo Return Loss) or ERLE (Echo Return Loss Enhance often named Echo Canceller).  Almost all Voice over IP devices and Gateways already incorporate an echo canceller to remove or reduce the echo level from analog loops. All DrayTek VoIP routers have G.168 Line Echo cancellation enabled by default.

Unfortunately there’s not much that the average end user can do.  Suggestions include:

  1. Try the call again, as it is possible that the echo canceller simply didn’t activate for some reason.
  2. Try a better quality telephone handset. Specifications of any echo cancellation circuitry is not often displayed on the box, so this may be very hit-or-miss.
  3. Use G.711 as the default code; since this provides no compression, and so reduces the delay added by VoIP.  If possible, please enable the single codec function, by telnet as shown here.
  4. Enable Voice Activity Detection function (if available).
  5. Try reducing the RTP packets size to 10ms.
  6. Enable the QoS function of your VoIP router. If your router doesn't support QoS, make sure you are using the latest firmware. 
  7. Check that your VoIP provider has G.168 echo cancellation active at their end of the PSTN connections – since their telco probably doesn’t provide echo cancellation for their local loop connections, due to the high cost.
  8. Reduce the transmit power level (gain), to minimise the volume of the echo – but don’t turn it down so low that the far end cannot hear you. This will probably also be hit-and-miss to find the ideal setting since gain on VoIP calls is usually much higher than PSTN calls.

Operators of VoIP gateways and PBX equipment (e.g. Asterisk) have more configuration options available - please check the manual and technical spport for your VoIP gateway/PBX. 

References:
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