=====datacomm Keyword 1=====
SIP - Session Initiation Protocol

====Definition====
SIP is a signalling protocol.  I would guess that that means it is a certain method of having machines communicate.  It is typically used for voice and video communication.  SIP can be used for two-party (unicast) and multiparty (multicast).

You can modify IPs and ports, invite more people, and add/delete media streams.

SIP can take care of 5 things:

  * User Location      - Figures out what system will be used for the call.
  * User Availability  - Determining the availability of the user.
  * User Capabilities  - Determining how the call will be made (voice, video, etc).
  * Session Setup      - The making of the session, making the ringing happen.
  * Session Management - Transferring, terminating, and modifying the call.

At first I had no idea what SIP was all about and had difficulty understanding it. Now I have a pretty good understanding, I think.  I'll probably ask about it next class to make sure I am right, but I feel like SIP is basically just a 'code' of how machines communicate. Just like how humans use English, or Spanish, or Arabic. Computers can use SIP.

On further research, the analogy is fairly sound, but needs a little bit more to it.  SIP is basically a specific format that machines can use to communicate.  It's a set of certain things that need to be done while communicating.  It's not like a language.... it's more like the RULES behind a language.
====References====
http://www.voipmechanic.com/sip-basics.htm \\
http://en.wikipedia.org/wiki/Communications_protocol

=====datacomm Keyword 1 Phase 2=====
asterisk

====Definition====
A programming language to communicate and execute the basic functions of telephony services.
====Demonstration====
Example of a configuration file written using asterisks language.  Given an example, it is pretty easy to figure out what certain things mean.  The 600 numbers are the numbers to call, and then it will Dial and use the given protocol to try to connect to the defined name. The 20 is how long it will ring for.
<code >
exten => s,1,Answer()
exten => s,n,Wait(2)
exten => s,n,SendDTMF(1)
exten => s,n,Dial(SIP/Saad&SIP/Andrew&SIP/LairPhone&SIP/Dataking&SIP/sipura&SIP/PA168,20)
exten => s,n,VoiceMail(601@vm,u)

exten => 601,1,Dial(SIP/Saad,20)
exten => 601,n,VoiceMail(601@vm,u)

exten => 602,1,Dial(SIP/Andrew,20)
exten => 602,n,VoiceMail(602@vm,u)

exten => 603,1,Dial(SIP/LairPhone,20)
exten => 603,n,VoiceMail(603@vm,u)

exten => 604,1,Dial(SIP/PA168,20)
exten => 604,n,VoiceMail(604@vm,u)

exten => 605,1,Dial(SIP/Dataking,20)
exten => 605,n,VoiceMail(605@vm,u)

exten => 606,1,Dial(SIP/sipura,20)
exten => 606,n,VoiceMail(606@vm,u)
</code>