Thursday, July 19, 2007

Asterisk Improvements

The Modeling PMC had a call on our Asterisk server which did not go very well. I know a few others of you have tried it, some with luck, others not. I believe the biggest problems are all client-side, primarily poor audio quality due to a lack of headsets, and Idefisk not being well set up out of the box. I have updated the wiki page with instructions on how to configure Idefisk for best the audio quality. I recommended Idefisk initially for a few reasons:
  1. It's available on all major platforms.
  2. It's an IAX client which has major advantages for traversing firewalls.
  3. Setup out of the box is very easy
  4. It doesn't require registering an extension with the Asterisk server in order to place calls.
That may all be true, but the call quality appears not to be as good as that provided by SJphone a highly functional SIP client which is also available on all major platforms. If you're not screened by a firewall and can route traffic directly to the Internet I suggest trying that client instead. If you get no audio it's because the return RTP traffic is blocked and you're probably screened by a firewall or router. If anyone has success with other clients and would like to recommend them, please add them to the end of the wiki page.

Also, you really do need a headset. I know that's an unfortunate reality: hey I wish it were otherwise. But we've tried it in a number of configurations without a headset and the audio quality drops off massively for everyone on the call. I have an older mono headset that I got at the local computer store for about $4 and the quality is superior to a regular telephone, so we're not talking about major expenditure. So that's it for the client side.

I also have done some more testing and today with the help of Matt and Nathan from the Foundation and Martin Oberhuber was able to reproduce the jitter which some of you have mentioned. After debugging I tracked it down to a server problem where a lack of available threads for muxing/demuxing the audio and transcoding across the different codecs caused the muxer to pause. This would have been even worse for a larger conference. I have added a larger pool of threads on the server which should hopefully address this problem.

Everyone on the test call today agreed that the audio quality (without the jitter) was better than on a Skype conference. We're hoping to provide that level of quality going forward. As always please let me know if run into difficulty.

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