VoIP and Asterisk8 May 2013, 7:30 p.m. MIT E51-145
The near-universal provision of voice services and their terminals (called "telephones") predates the Internet. While on some level, voice traffic via TCP/IP is just another protocol, there are challenges in making it "just work" like the traditional phones that we are all used to. There are the technical issues of the nature of the data, interfacing with the still robust telephone network, and of course the UI expectations and experience.
That means that the protocols involved - SIP and the related suite - were developed in the setting of a preexisting, mature, and complex switched network. I found that from the perspective of a systems administrator or network engineer there are complications, terminology, and conventions that aren't necessarily obvious.
This talk will provide insight into the these technologies from that perspective to allow you to grasp the protocols and the context in which they interoperate, using an example implementation of Asterisk.
The presentation will be available as a:
- Web-based presentation.
- Original Keynote format: presentation.key
- QuickTime Movie (presentation slides only, no narration): presentation.mov
- PDF: presentation.pdf
Configuration files: these are excerpted in the presentation. Note that these files contain some of the comments and guidance from the distributed samples - in general, other than the introductory information, most of the comments were deleted.
- SIP configuration file sip.conf which has entries for the devices and (fictional) SIP provider.
- Dialplan extensions.conf with the basic programming to handle incoming calls.