in BSD, Telephony

BSDCan 2006 Day 0xFFFE: VoIP and FreeBSD tutorial

I’m going to be posting a couple of journal entries where I will summarize the talks I attended at BSDCan 2006 last weekend. It might take me a little while to get through all the summaries, but eventually I’ll finish them all! I’m starting with Day -2 of BSDCan, i.e. 2 days before BSDCan really started I attended a tutorial on VoIP and FreeBSD.

VoIP and FreeBSD Tutorial

Presented by Massimiliano Stucchi

I attended this tutorial, presented by Massimiliano (Max), who works as a consultant for a major communications company in Italy. He mentioned that in his work he manages over 2000 channels — I took this to mean either simple FXO/FXS channels (single POTS lines) or PRIs. I took this to mean about 70 customers with PRIs since PRIs in Europe can carry up to 30 b-channels (whereas in North America, on the T-1 standard, one can carry 24 b-channels).

Primarily, VoIP in a free or open-source software environment means Asterisk, which is fine; that’s what I’m interested in. Max did briefly mention that there are other open-source PBXes out there such as VOCAL (which is probably not being actively maintained given that the "latest" release is for RedHat 7.3) and Bayonne, the GNU PBX which might show some promise. Still, when one thinks of open-source PBXes, one typically thinks of Asterisk.

Much of Max’s tutorial was a very basic introduction to Asterisk, which was fine although I felt like I didn’t learn too much more about the operation of the system. He did manage to clear up some terminology for me, though; in particular I think people use the word termination wrong. In the telephony world, termination is used to describe when calls leave your VoIP system and get into the PSTN. I think a lot of people, myself included, misuse the word termination to mean the completion of a call on a DID number, whereas this is actually termed origination. Confusing, yes. I guess it’s the same hurdle in understanding that you need to jump when thinking about what signalling to use for a FXO/FXS device: for an FXO device (which actually interfaces to an FXO, i.e. a CO) you need to use FXS signalling, because your FXO device is behaving as a "station" (like a regular analog phone is a station).

Another useful tip I got out of the tutorial was that faxing does not work well with Asterisk, and definitely not with pure IP. This is in part due to a number of factors: Asterisk does not have the capability to do the error control (ECM) necessary for faxing; although ECM is, strictly speaking, optional, most Class-1 fax devices expect to have it turned on. Moreover, the GSM codec does not handle faxing very well, therefore faxes will not work on IAX or SIP channels. Max felt that the T.38 protocol (faxing over UDP) is probably the way to go, but there is not yet wide support for this.

Max eventually got into some use cases of Asterisk, showing us parts of his dialplan, and discussing some of the syntax, but I felt that during this part of the talk he got a little distracted and started telling some stories & conversing semi-privately with some of the participants. To his credit, Max had some slides prepared, but he did not stick to them, particularly near the end of the tutorial. I think this would have been discouraging to any real newbie who came into the tutorial with absolutely no knowledge of Asterisk. At one point, a newbie asked him what it would take to implement a basic Asterisk setup just for testing — what hardware should be purchased? Max wasn’t able to answer this question directly, perhaps because he had never purchased small-scale hardware. I would have preferred to see an example of a starter setup, such as the use of cheap FXO clone cards to build an answering machine, for example.

Overall, though, I got some good tips from someone who has actually built a large-scale production Asterisk setup. I won’t try running a non-POTS fax line, that’s for sure.