- VoIP Technology
- Define VoIP
- Network entities
- Telephone systems
- Company PBX
- VoIP telephone SET
- VoIP telephone adapters
- SIP Account
- SIP phone line
- SIP phone call
- ATA (FXS, FXO)
- Call Center Server
- Call Center Client
- Supported codecs
- SMS technology
- Ozeki VoIP SDK
- Developers Guide
SIP phone call
This article is a brief introduction about the phone calls that are made through the session initiation protocol in relation with Ozeki VoIP SIP SDK. After reading through this page you will be fully familiar with all the essential terms concerning SIP phone calls and what you will need for creating your own solution using Ozeki VoIP SIP SDK.
The SIP (Session Initiation Protocol) is a protocol to be used for Voice over IP communication. This communication was basically about voice, but today VoIP also covers video and data communication. A SIP phone call means the communication between two end points via VoIP using the SIP protocol for creating the phone line between the two parties (Figure 1).
The SIP is used for establishing the line between the communicating peers in VoIP. There is a need for a server called Private Branch Exchange (PBX) that registers the clients through a SIP account that contains user name, register name and registration password is there is a need for phone registration. When a phone (hard phone or softphone) has registered to the PBX it can start communication toward another registered phone.
The SIP call is started with a SIP INVITE message that is sent to the PBX. This message is sent from the PBX to the other party and an asynchronous 100 TRYING SIP message is sent back to the initiator party. If the remote party accepts the call, a 200 OK SIP message is sent from it to the PBX that transmits the message to the caller. The caller answers with an ACK SIP message and when the second client gets it, the PBX exits the call ant the two parties are communicating directly.
The SIP messaging and the PBX enters again to the communication line when one of the communicating clients want to end the call. At this point one of the peers sends a BYE SIP message to the other through the PBX and when an ACK is received for that the communication line is terminated.
According to this, the SIP phone call consists of a SIP establishment phase, the actual communication phase and the SIP termination phase. These three build up a whole communication between two VoIP clients.
This article introduced you the basic knowledge about the SIP phone call and showed how Ozeki VoIP SIP SDK can help you to fulfill your goals. If you have read through this page carefully, you already have all the knowledge you need to start on your own solution.
As you are now familiar with all the terms concerning this topic, now it is time to take a step further and explore what other extraordinary solution Ozeki VoIP SIP SDK can provide to you.
If you have any questions or need assistance, please contact us at firstname.lastname@example.org
You can select a suitable Ozeki VoIP SIP SDK license for your project on Pricing and licensing information page