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AsteriskAndVoip

This version was saved 15 years, 11 months ago View current version     Page history
Saved by PBworks
on April 19, 2008 at 1:47:42 am
 

one of the founders of NetVoice

open source, Linux, and in particular Asterisk

also small telephone company

rack of Asterisk running at Bell Canada


 

"how I stopped worrying and learned to love the dialtone"

e164.org

 

  • stations -- end points, aka a phone
  • trunks -- tie you to another system
  • PBX -- sit between stations and trunks

could all have a phone tied to Telus, but expensive, and don't have features -- transfers, voicemail, etc. etc.

  • PBX == private branch exchange; this means you have a small telephone switch in your office, less trunks to the phone company

* FXO/FXS ports -- if you plug two phones into each other, nothing happens

* FXO port that you plug into the wall

* FXS is what you plug a station into (aka a phone)

* What is Asterisk? open source platform for telephony applications

* started in the US about 5 years ago by Mark Spencer

* vendor-neutral implementation -- works with lots of hardware

* lots of applications on top -- Asterisk works a bit like CGI-BIN, can communicate to other systems

* old fashioned way was to do processing directly on the hardware with DSPs; DSPs are expensive!

* as CPUs become more powerful, moved to processing directly on the processor, the interface port is nothing more than AD / DA converters, so they became much less expensive

* this breakthrough is what Asterisk exploited -- didn't need expensive hardware

* founder of Asterisk went to start a hardware company, Digium, which provided the now inexpensive hardware boards

* all successfull open source projects have had corporate sponsors like Digium to Asterisk; there are many other companies that provide the hardware boards

  • What can Asterisk do?

* scriptable telephony-to-anything

* MySQL connector

* your web app can make phone calls

  • Alphabet soup of protocols

* IAX, not a real standard -- a common standard in wide use, for tying Asterisk systems together

* SIP, H.323, MGCP, MEGACO, SCCP - JINGLE!

* SIP is new

* H.323 was the original, designed by telephony-heads, it's going to die

  • More on SIP

* standards based, so commodity hardware

* e.g. Sippura, now bought by LinkSys, less than $100 for a gateway (small box, size of a deck of cards)

* another open source system, SIP Express Router

* not all features of Asterisk, only handles SIP

* very efficient, can handle 50K phones off a single CPU -- only handles call setup

* problem with SIP is that it puts IP address in packets -- this means that there are NAT issues

* ways to compensate, but NAT is still problematic

  • system requirements for Asterisk

* 50 people, 23 trunks, 1 computer

* compression is what takes processing power

 

  • Freeswitch

* Asterisk has strong ties to one company, Digium

* Digium has a heavey hand

  • OpenPBX - fork of Asterisk
  • Jingle - SIP, h.323, and Jingle all do signalling

* RTP actually does transfer

 

Boris Notes

  • trixbox.org -- CentOS + Asterisk + flexible web admin tool
  • tallk to Jonathan
  • local Asterisk users group