BASICS OF IP TELEPHONY
Basic principles, terms and protocols
IP telephony (voip phone service) refers to voice communication, which is carried out over data networks, in particular over IP-networks (IP-Internet Protocol). Nowadays IP telephony is increasingly replacing traditional telephone networks due to the ease of deployment, low cost of calls, ease of configuration, high quality of communication and comparative connection security.
When making a call, the voice signals are converted into a compressed data packets that transferred over IP networks. When the receiver reaches the packets, they are decoded into the original voice signals. These processes are possible due to the large number of auxiliary protocols. In this context, the data transfer protocol is a language that allows to understand each other and ensure high-quality data transfer between two points.
Today Everyone has access to the global network, which makes it possible to cut connection costs or completely exclude them. VoIP Telephone servers are constantly being improved and their work algorithms become more resistant to delays or other problems of IP networks. New protocols and technologies are introduced every year to improve the quality of communication. For example, the problem of a busy line is very elegantly solved: redirection incomming calls to alternative destinations (including mobile phones, skype and other software) or IVR (Interactive voice response) or standby mode. Sip Systems is sip provider (using SIP and other related protocols) for VoIP communication. Find out more about basics of ip telephony service here!
IP telephony provides a way for computer networks and other devices to emulate traditional phones and phone lines. Most modern business PBX systems have migrated to VoIP technology already. In some circumstances, legacy phone lines (PSTN or POTS) are no longer available and VoIP is the only choice.
Businesses and consumers are already taking advantage of the cost savings and new features of making calls over a converged voice-data network, and the logical next step is to take those advantages to the wireless world. Early adopters are already touting the benefits of using WIFI to make inexpensive phone calls, but is this truly a technology that will take off any time soon? More information on Sip Systems blog – Wireless VoIP.
ASTERISK AND FREEPBX
Open Source Phone Systems
- The software or asterisk source code is completely free. You can obtain a downloadable version of asterisk/Linux or any other open source ISO from the leading players in the market such as trixbox, elastix, FreePBX, PBXinaflash, etc.
- Web GUI interface (like FreePBX solution) does not requires advanced skills on Linux systems and Asterisk engine code. So administrators don’t need to rely on asterisk command line to manage their phone system.
- Using IP based telephony, we can integrate other applications such as databases, CRM tools, email, click to dial and really any application or program you wish to integrate with Asterisk. It provides full customization for you or your business with unique or common tools.
- Source code of asterisk is always under heavy development, and does rely a lot on field developers and system administrators, it is constantly being updated with the latest and greatest features and bug fixes from sources that have fully tested the solution and are committed 100 percent.
- Common VoIP solutions:
The most popular free web-interface for telephone exchange based on asterisk is FreePBX. In general the advantage of this project are regular product updates, also easy management and flexibility. FreePBX is easy to customize and adapt to your changing needs. FreePBX is available as a free downloadable software package. It can be installed on your own server or virtual machine. FreePBX leverages standard computing servers and Linux OS. It is configured and managed using standard web browsers. It fully supports the ubiquitous SIP protocol and compatible with all softphones. FreePBX is the best platform for Advanced IP telephony service and business VoIP solutions.
FreePBX based on Asterisk provides extensive opportunities to provide conversations in IP-networks, allows you to more effectively manage the organization and save money on calls. But due to weak security or improper configuration VoIP-service can cause serious financial losses.
FreePBX VoIP server is a complex solution, and for its protection an integrated approach should be applied, blocking possible threats at all levels. Among other things, you should also worry about standard activities such as keeping the software version and server OS up-to-date, updating the firmware and configuring the security of the target hardware.
When it comes to call center software, most of the solutions you may be looking for can cost upwards to thousands of dollars. Above, Sip Systems offers popular and powerful call center programs (predictive dialer systems) that were tested by VoIP experts and guaranteed full free easy setup and usage.
Freeswitch is another open source project grown into millions of lines of code that hundreds of companies leverage to create successful businesses. If you’re already a software developer or network engineer with previous exposure to telephony concepts or already know you want to integrate intimately with call features, and/or you have big plans and need to look as far into the future as you can, you’ll want to go with FreeSWITCH.