Improving Conference Call Quality: What You Can Do

Mozilla has a lot of conference calls. A lot. More time that is necessary is spent debugging audio issues and getting it so that everyone can just have a conversation. This post gives some tips on how you can make calls you are involved in a pleasant experience.

Firstly, if you are just listening, that’s fine – but mute yourself. All mobile phones and softphones, and some landline phones, have a mute button – use that for preference. If you are using a phone without a mute button, then “*1” mutes and unmutes you on the Mozilla system. The trouble with this is that a) people can hear the ‘beep’, and b) you can forget to unmute because there is no visual indicator. Prefer mute buttons with obvious visual indicators, and get into the habit of checking them before speaking.


VoIP can be much better quality than POTS, but sometimes has more lag, and requires a stable internet connection (hotel WiFi not good). Prefer VoIP clients (like Blink) which show latency and packet loss so you can tell easily when your connection is sucking. You can now get cheap international POTS calls from various services, but they normally keep the cost down by squeezing the bandwidth even more than a normal phone line – so much so that sometimes, even DTMF doesn’t work. So beware of that problem.

If you are planning to participate using VoIP, never do so just using the microphone and speakers built into your laptop. You’ll get poor sound quality, and everyone apart from you will get lots of echo too. (This is particularly bad, as you don’t notice the problem, but everyone else does.) At the very least, get a pair of headphones. Better, get a headset. Mozilla IT recommends the Plantronics .Audio 646 DSP USB headset. You can get them for £19 ($25; €17) from various branches of Amazon, and they work on Windows, Mac and Linux.

If you are a regular participant in Mozilla conference calls using VoIP, please, please get yourself one. If £19 is more than you can afford, let us know and we’ll get one for you. (The reason we don’t offer this to everyone straight off is that often the cost of postage is more then the cost of the headset.)

Consider making a test call to your provider’s echo service first to make sure everything is working fine.

Side Rant: VoIP clients

Why do all VoIP clients not have a built-in VU meter (with a green band for ‘OK’) and local echo test, so people can check their microphones without lots of “Can everyone hear me? Is this loud enough? Am I distorting?” at the start of every call?

Also, if you try and speak but you’ve muted yourself, it should tell you clearly and quickly! Either because the VU meter doesn’t move, or some other way. I’ve lost count of the times I’ve tried to join a conversation and thought that I was being ignored, but in fact I’d muted myself 5 minutes before.

I use Blink on Linux, and (as noted above) I love the fact that it shows latency and packet loss. But it doesn’t even have call history :-| Suggestions for better options welcome.

6 thoughts on “Improving Conference Call Quality: What You Can Do

  1. Is muting on the user’s end always enough? I have the vague recollection that if you’re on a poor/noisy connection, that noise can still get through and make listening hard.

    I’m paranoid and just mute both ways. :)

  2. Muting your phone only mutes the microphone in your telephone. So, if you’re on a POTS line and have line noise that will still go through. If you’re using VoIP it’s not really a problem.

    Now, if you control the call and have the capacity to mute people on your end, then that should take care of your problems.

  3. For the record, Google Hangout handles this nicely. If you mute yourself, and then try to talk, a big red warning at the top of the screen alerts you to this, and offers to unmute yourself.

    Really handy.

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