In contrast to voice transmission, another protocol is used for IP-based telephony for setting up and clearing down VoIP connections, namely the Session Initiation Protocol (SIP). This network protocol ensures cross-manufacturer integration of VoIP components. With SIP-based systems, each participant has their own SIP address. A SIP address consists of two components. On the one hand, it contains the subscriber’s SIP user and on the other hand the domain name of the registrar server.
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With a virtual telephone system, you also enjoy maximum flexibility at the best. Because you can add or cancel phone numbers at any time yourself using an intuitive user interface. Location independence is another factor that entices many companies to switch to cloud telephony. In this way, branches abroad can even be linked to the cloud telephone system. Access to the telephone system is also possible from any device. If you want, you can even have your calls ringing on all end devices at the same time so that you never miss a call again.
How do the connection establishment and termination work with VoIP?
In order to be able to establish a connection, it is necessary that the caller (A-subscriber) knows the IP address of the recipient. To do this, the participants’ devices log on to a registrar server with their IP address, user name, and password. This enables users to make calls from anywhere, because they can log on to any SIP terminal worldwide, access all of their phone services, and keep their number. All you need is a highly supportive internet connection without buffering.
If a connection is to be established between two subscribers, the terminal of the subscriber. A sends a message with the subscriber B’s number to the server of his provider A. Thereupon, the server forwards the information to the server of provider B. So that the latter can use the terminal of the participant B can address. If this step worked without any problems, terminal B rings and sends a message back to terminal A. It recognizes that the other subscriber has been found and answers with a tone signal. When the connection is successfully established, communication then takes placing between the end devices and no longer via the SIP server.
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To end the call transfer, a terminal sends an appropriate SIP packet to its server. The server then notifies the other end devices of the disconnection and the call is ended.
What are the requirements for IP telephony?
To use Internet telephony, you only need three things –
- An internet connection
- A VoIP provider
- Suitable hardware
Since you are using an IP connection to make calls over the Internet, an Internet connection with sufficient bandwidth is essential to ensure excellent voice quality. Approximately 100 kbit/s of bandwidth is incurred per voice channel in the upload and download direction. You can find out how to calculate the required bandwidth and how to prioritize voice traffic in our VoIP bandwidth guide.
In addition to an Internet connection, a certain amount of hardware is required to use VoIP. However, it should be said in advance that it is not necessary to purchase additional hardware.