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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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Echo and Sidetone

Echo and Sidetone


A telephone is a duplex device, meaning it is both transmitting and receiving on the same pair of wires. The phone network must ensure that not too much of the caller's voice is fed back into his or her receiver. This feature, is referred to as "sidetone," and is achieved by phasing the signal so that some cancellation occurs in the network before the signal is fed to the receiver. Callers who hear no sidetone may consider the phone "dead." Very little sidetone will convince callers that they're not being heard and cause them to shout. "Can you hear me now?!" Too much sidetone causes callers to lower their voices and not be heard well at the other end of the line. This also makes for a very unpleasant call.

A telephone on a short loop with no loop compensation will appear to have too much sidetone, and callers will lower their voices. In this case, the percentage of sidetone is the same, but as the overall decibel level is higher the sidetone level will also be higher.

With the advent of digital phones, and standalone voip phones, this concept has changed dramatically. Each manufacturer has their own theories on the best levels of sidetone. Some manufacturers will also eliminate sidetone entirely, but generally in order to work around another technical limitation of their firmware or hardware design. As latency is added, by a slow server or sporadically latent voip link, sidetone can turn into a delayed echo, which is an entirely different subject — Causes of Echo

See also:

Created by denon, Last modification by piotr on Thu 09 of Feb, 2006 [17:16 UTC]

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