Getting the Most out of Asterisk
I realized from a few emails that we webmasters have gotten over the last few days that I need to update the Asterisk howto for conferencing with some better information on how to make the audio quality the best possible for your call. I will be putting some more notes there over the next few days. But I wanted to take a little space here to let everyone who is trying out the Foundation’s new Asterisk server know the number one thing you can do to make sure the audio quality is good:
Everyone on the call needs to be using a headset or an external microphone!
That’s not necessarily an obvious fact, after all, most laptops and many integrated desktops include a built-in microphone and a nice set of speakers. The problem seems to be that the internal microphones pick up a lot of electronic noise from the hard drive and fan, overboost noise on the mic which sounds like white noise. You can even get audio echo and at worst feedback from the mic and speakers on the laptop. If you get a bunch of people on the call and even a couple are using built-in mics the quality of the whole call suffers.
As it turns out in our testing, even if you have a bunch of people in the room, just hook up a headset microphone, use the built-in speakers on the laptop, and the audio quality is good for everyone and it can pick up all the voices in the room (at least with our Logitech headsets).
So hopefully that will help some of you get a better experience with Asterisk. As always we’d like to know how it’s working for you.
Posted July 17th, 2007 by Denis Roy in category: Uncategorized
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3 Responses to “Getting the Most out of Asterisk”
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Ed Merks Says:
July 18th, 2007 at 9:50 am
We used it for the Modeling PMC call yesterday and it was so bad that I’m still suffering from partial hearing loss. I can imagine that using headsets would eliminate the feedback and horrible fading echoes. There are few things more disturbing than hearing our own voice talking back at you with a two second delay! Now we just need to get all our employers to fund the headsets we need, although I already have a Logitech one and it’s excellent and highly recommended.
Martin Oberhuber Says:
July 19th, 2007 at 8:48 am
We tried it twice for the DSDP-TM Committer meetings and also failed both times, but here are some hints to get things working better:
(1) Be sure to get Idefisk-1.37 and NOT Idefisk-2.0 — one of us reported his whole Windows installation corrupted badly by Idefisk-2.0, plus the 2.0 version definitely won’t work.
(2) In Idefisk-1.37, right-click on the background of the window, General Options > Disable Check for Updates (in order not to get the bad 2.0 version accidentally).
(3) In Idefisk-1.37, right-click on the background of the window, Filter Options. Enable at least Noise Reduction and Send CN frames when silence detected — this will reduce noise and improve quality for everyone.
(4) All participants should drag the little microphone gain slider in Idefisk to the left-hand-side (about 1/3 for most of us) or they will be very distorted. As an alternative, “Digital automatic gain detection” in the Filter Options dialog might also help.
But even with all of these little helpers, and even in a 1-on-1 conference, we still experienced problems like one person frequently dropping off (loss of UDP packets?) , others not being able to talk (microphone problem?), yet others not being able to connect (Ask your IT Personnel to disable firewall on UDP Port 4569 or it will not work!)
All these quirks let me wonder whether there are better alternatives to Idefisk? - With Skype, my call quality has always been surprisingly good even without any manual filter settings. I’d love to see progress from the Foundation, improving the conferencing solution.
Karl Matthias Says:
July 19th, 2007 at 6:30 pm
It seems that Idefisk is not necessarily the best client. With the settings Martin recommended (and I also had added to the wiki page) it is much better. Anyone who is having trouble with quality and who is NOT firewalled should try SJphone, a SIP client. It’s linked from the bottom of the wiki page now.
Today we had a test call with Martin and a few of us from the Foundation. The call sounded great by all accounts except for a jitter which affected people every few minutes. I think I tracked this down to a problem with the number of threads Asterisk was spawning to start with, and I’ve upped the total number. Hopefully we’ll see an improvement in jitter.
I really think that if everyone tweaks their client a bit with the settings now provided on the wiki it will really improve quality.
As for headsets… these things are super cheap these days. I know it’s a hassle to have to get them, but I have a mono $4 set that I used before the Logitechs and they sound at least as good as a phone. I think the Logitechs were like $16 + shipping.