Set up your VOIP based SIP soft phone and know more about VOIP

VOIP (Voice over internet protocol)

       The  trending technology of present and near future. I have spent months of efforts on voip , its a huge topic so to minimize your time and also to refresh my voip skills , I have decided to write this blog.

Main Topics covered :-

SIP (session initiation protocol)
ASTERISK (open source voip server platform)

    SIP is a part of Application layer (fig1) in TCP/IP , its major function is to initiate/terminate a sip session and transfer it to TCP/UDP for transmission. In simple terms it holds the FROM and TO addresses in a post , the rest is carried using TCP/IP protocol. 

fig1 : SIP in TCP/IP

The basic operation of SIP is an INVITE and ACK via proxy, once the opposite Sip phone B accepts the call, a media session is established. Its simple as shown in fig 2. The proxy might be a VOIP services provided by certain websites such as  www.linphone.org, SIP2SIP.info etc or it might be your IP also.

fig 2 : Sip basic operation 

     Sip id : user@domain  or user@IP
           ex : hanu@linphone.org   or   hanu@192.168.1.13


SIP proxy is nothing but a VOIP server, You can deploy your own Voip Sip server using asterisk  www.asterisk.org. For example the your Skype and  even  facebook video calling services are nothing but a VOIP services and the two people calling using SKYPE are nothing but SIP endpoints. 

More features of SIP include video conferencing unit.
Example of a typical video conferencing unit using WEBRTC protcol - http://tresbu.com/trespond/

ASTERISK SERVER

  I am describing the possibilities of asterisk as a voip service. It has tremendous capabilities of using it in a call center, IVR etc. You can download any of the latest version of asterisk from this website http://www.asterisk.org/downloads. Further this blog guides you through the installation steps http://highsecurity.blogspot.in/2012/12/quick-and-dirty-asterisk-11-and-freepbx.html. The entire approach of asterisk is modular. You can add, modify the modules at your will. 

Here are the steps needed to configure asterisk as a voip service is as follows:- 
We have to make the following configuration :-

nano /etc/asterisk/sip.conf 
1) sip.conf

[1001]
type=friend
host=dynamic
context=default
username=1001
secret=1001
nat=yes
canreinvite=no
dtmfmode=rfc2833
qualify=yes
disallow=all
allow=ulaw

[1002]
type=friend
host=dynamic
context=default
username=1002
secret=1002
nat=yes
canreinvite=no
dtmfmode=rfc2833
qualify=yes
insecure=port,invite
disallow=all
allow=ulaw

2) nano /etc/asterisk/extensions.conf
Extensions.conf

[default]
exten => 1001,1,Answer()
exten => 1001,n,Dial(SIP/1001,20,tr)
exten => 1001,n,Hangup

exten => 1002,1,Answer()
exten => 1002,n,Dial(SIP/1002,20,tr)
exten => 1002,n,Hangup.


If you make the following changes you will be able to make calls using asterisk and any sip softphone.