TEL 500 Voice Communications Week 1 Write Up Session Initiation Protocol Lab Submitted To: Prof Ronny Bull By: Sai Sharan Korvi Date: 09/10/2014
ABSTRACT: Softphone is usually a software which can be used for making calls over internet. This software is installed on a personal computer from where it is connected to a server. We can establish a call between two different entities with the help of SIP (session initiation protocol), which performs even other tasks such as call termination. For a communication channel to be set up between two points, a software is used which is Asterisk. The primary goal of this lab is to setup a call between two different entities, namely a softphone and a hard phone. Putty is a text oriented console which is used to set up a softphone on personal computer workstation. The softphone is configured with the details of user such as IP address and MAC address followed by the hard phone setup. Now, a call can be established between these two phones by using the Session Initiation Protocol. PROCEDURE: PART 1: Firstly, we need to set up the machine of the asterisk. For this, we need to login to putty and connect to the asterisk. When we open the Putty console, we are asked to enter the Host Name to establish the connection with. Now, we need to login as root in putty using the password CaptainCrunch. Using the command cd /etc/asterisk, we can connect to asterisk.
Now, we need to create the directory files which is done by using the command touch. This command is generally used to create a new or empty file. There are different files which should be created such as: 1. 2. 3. 4. 5. touch logger.conf touch modules.conf touch sip.conf touch extensions.conf touch rtp.conf With the help of above commands, we have created files. Now, to check list of directory contents, we use the command ls.
Now, in order to edit the folders and add the code, we need to use the command vim. The commands used to open and edit the files are: 1. vim logger.conf The code for logger can be dumped here using the command mentioned above.
2. vim modules.conf To edit the modules file and add the code for it, the above command can be used. 3. vim sip.conf: This command can be used to open the SIP folder and add the code for it.
In this code, we can change the password and also edit the MAC address. 4. vim extensions.conf: This command can be used to open the file and edit it for dumping the code into it and making the changes to the extension numbers. 5 vim rtp.conf :
After editing the files and dumping the codes for all the files, we need to restart. The command used to restart is mentioned below: /etc/init.d/asterisk restart To set up a communication channel between two entities, we use the asterisk software. The command used to set up the channel is asterisk -rvvv. The two entities will be shown as offline, as we did not set up the soft phone yet. The status of both of them can be seen by using the command sip show peers.
When we use the command sip show peers command, we will be able to see the status of the two points between which we are about to set up a call now.
PART 2: Now, we have to set up a soft phone in the PC. This is done with the help of X-lite software which needs to be downloaded and installed. Once installed, we need to login with the User ID and the Account Name. Once we login with the details, our softphone is online. The status of the hard phone will now be offline because it has not been set up yet. However, as the soft phone has been set up, it will be shown as online. This can be checked with the help of a command sip show peers. This command shows the status of the entities, whether they are available or offline. For example, now as we have signed in soft phone, this will be in online mode. Whereas, the hard phone status will be offline because it has not been set up.
PART 3: Hard phone needs to be configured now after setting up the soft phone. In the admin settings of the hard phone, we get an option to add a new account. By using this, we can add our account to the hard phone with the help of MAC address. Here, we get an option to add a new SIP account. We need to select that option. One thing to be noticed here is, apart from just adding new accounts here, we can also delete the older SIP accounts.
After clicking on the SIP account, we have to give the details of the account which needs to be set. The server, user ID and the password the three details which are asked here. The setting up of the hard phone is done now and the only task left out is making the call.
PART 4: At last, we have set up the soft phone and the hard phone as well. If we now check the status of the peers, both the entities will be shown as online. Now, we can make a call from soft phone to the hard phone by dialing the extension number of the hard phone. We can see hard phone ringing when the call gets connected. Inversely, a call can now be made from the hard phone to the softphone. The extension number of the softphone is dialed to make a call to the softphone.
If we dial to hard phone, we need to dial the extension number of the hard phone.
CHALLENGES: In spite of being able to implement the entire process successfully, there are few situations where I had got few errors. They are: An incorrect password was given by me as the result of which the computer displayed as command incorrect. CONCLUSION: In this lab, establishing call between two different points has been successful. The major components used here are the workstation PC, soft phone and hard phone. The voice over internet protocol has been used in the process of setting up a communication channel between the two entities.