IT Solution offers VoIP (voice over IP), is the transmission of voice and multimedia content over Internet Protocol (IP) networks. VoIP historically referred to using IP to connect private branch exchanges (PBXs), but the term is now used interchangeably with IP telephony. VoIP is enabled by a group of technologies and methodologies used to deliver voice communications over the internet, enterprise local area networks or wide area networks. VoIP endpoints include dedicated desktop VoIP phones, softphone applications running on PCs and mobile devices, and WebRTC-enabled browsers.
There are three different "flavors" of VoIP service in common use today:
ATA -- The simplest and most common way is through the use of a device called an ATA (analog telephone adaptor). The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the analog signal from your traditional phone and converts it into digital data for transmission over the Internet. Providers like Vonage and AT&T CallVantage are bundling ATAs free with their service.
IP Phones -- These specialized phones look just like normal phones with a handset, cradle and buttons. But instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet connector. IP phones connect directly to your router and have all the hardware and software necessary right onboard to handle the IP call. Wi-Fi phones allow subscribing callers to make VoIP calls from any Wi-Fi hot spot.
Computer-to-computer -- This is certainly the easiest way to use VoIP. You don't even have to pay for long-distance calls. There are several companies offering free or very low-cost software that you can use for this type of VoIP. All you need is the software, a microphone, speakers, a sound card and an Internet connection, preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-computer calls, no matter the distance.
VoIP protocols and standards
VoIP endpoints typically use International Telecommunication Union (ITU) standard codecs, such as G.711, which is the standard for transmitting uncompressed packets, or G.729, which is the standard for compressed packets.
Many equipment vendors also use their own proprietary codecs.
Voice quality may suffer when compression is used, but compression reduces bandwidth requirements. VoIP typically supports non-voice communications via the ITU T.38 protocol to send faxes over a VoIP or IP network in real time.
Once voice is encapsulated onto IP, it is typically transmitted with the Real-Time Transport Protocol (RTP) or through its encrypted variant, the Secure Real-Time Transport protocol. The Session Initiation Protocol (SIP) is most often used to signal that it is necessary to create, maintain and end calls.
Within enterprise or private networks, quality of service (QoS) is typically used to prioritize voice traffic over non-latency-sensitive applications to ensure acceptable voice quality.
The two main types of VoIP telephones are hardware-based and software-based:
A hardware-based VoIP phone looks like a traditional hard-wired or cordless telephone and includes similar features, such as a speaker or microphone, a touchpad, and a caller ID display. VoIP phones can also provide voicemail, call conferencing and call transfer
Software-based IP phones, also known as softphones, are software clients installed on a computer or mobile device. The softphone user interface often looks like a telephone handset with a touchpad and caller ID display. A headset equipped with a microphone connects to the computer or mobile device to make calls. Users can also make calls via their computer or mobile device if they have a built-in microphone and speaker.