I am not a audio person. I am more of a technical problem solver who is on an interesting project to provide remote interpretation to a business audience.

I am able to control what goes out (the interpretation) over wide band VoIP (G.722 codec at 7 kHz). and this is pretty good output for business voice calls.

The problem is the sound that the interpreters obtain. Often, clients user speakerphones/conference phones and these obviously deliver terrible sound, because speakers are often far away from the device, or make background noise, etc. The interpreters (who are at another location) must SIMULTANEOUSLY hear the (bad) sound AND interpret the content within a few moments. Bad sound is very detrimental to them and can even hurt their ears over time.

I was wondering if hiring a studio sound technician would be worthwhile to see if ON THE RECEIVING END of the conference call (what the interpreter listens to) the sound could be cleaned up FIRST and then rerouted to the interpreter?

Am I totally nuts, or would this be possible? and to what extent?


Sadly phones are not great sound reproduction devices. I've not see a business phone system (and I've seen many) where the sound can be changed in any way except volume.

Sometimes a problem is that speaker phones are turned up so loud that the phone itself resonates and adds lots of noise. In this case turning the volume down can actually make it easier to hear if it doesn't end up too quiet.

I think your best bet is to change the equipment. If the speaker phone being used is not a dedicated speaker unit (like a Polycom) then switching to one of those might help a lot. If it is, then you might be able to find a higher end model that will help.

You could also change the entire setup. Phone calls are very narrow bandwidth (300 - 3000 Hz in North America) which means a lot of fricatives and sibalents can get confused, among other problems. If there is enough network bandwidth available then changing to a network meeting system (Skype for business, GoToMeeting, etc) with good quality speakers and microphones might be the best available sound quality.

One more thing you could try, which is really outdated but can help, would be to set up a high quality speaker and microphone system and connect it to a hybrid, which is a box that connects audio equipment to an analog phone line. You probably want a pro for that kind of thing.


Sound can usually be improved on the receiving end. But there is no general advice, the approach depends on your specific VoIP termination, sound profile, site setup, etc. Sound engineering is also part science part voodoo, or "art" as some prefer, and requires some trial-and-error.

So answering your question requires the VoIP URI (for example the SIP address if you use SIP), and either an opportunity to join a call or a recording of a typical call. You probably don't want to post such information publicly, better send it to me by email.


Can you change the original signal, before it's sent? If you can, removing any unneeded frequencies might be a start to limit the frequencies that cause the distortion. Like high and low end. An expander or gate would also cut down on background noise. I don't really understand your setup though. Are the clients phones the receivers? I don't know.

Failing that, lower volume or better equipment.

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