VoIP
Step into the future of business communication with our advanced VoIP solutions. Hosted entirely in the cloud, our systems use Voice over Internet Protocol to deliver crystal-clear calls and a full suite of unified communications tools over your internet connection. This eliminates the need for expensive, on-premise hardware and provides a single, flexible platform for voice, video, and messaging. With HD call quality and seamless integration, our VoIP service is designed for the modern, agile workforce.
VoIP (Voice over Internet Protocol) is a suite of technologies and protocols that enables the transmission of voice communications and multimedia sessions over IP networks, such as the internet. Unlike the circuit-switched method of traditional PSTN (Public Switched Telephone Network) systems, VoIP uses packet-switching, which digitises and compresses voice signals into data packets for transmission.
The VoIP Process
The communication process is managed by several interconnected protocols working in tandem:
When a user speaks into a VoIP-enabled device (IP phone, softphone, or ATA), an ADC samples the analog voice signal thousands of times per second and converts it into a digital data stream.
A codec then compresses this digital data to reduce bandwidth requirements and packages it into small data packets. The choice of codec (e.g., G.711, Opus, or G.729) determines the balance between audio quality, compression level and bandwidth usage.
The Session Initiation Protocol (SIP) is the most common signaling protocol used to establish, modify, and terminate a call session. This “handshake” process negotiates session parameters, including the IP addresses, port numbers and codecs to be used.
The actual voice data packets are transported in real-time over the network using the Real-time Transport Protocol (RTP), which is typically encapsulated within the lighter User Datagram Protocol (UDP).
The companion RTP Control Protocol (RTCP) monitors the quality of the RTP session, providing feedback on transmission statistics, jitter, latency and packet loss.
At the receiving end, the device collects the incoming packets, uses the sequence numbers and timestamps to reassemble them correctly, and the codec decompresses the digital data. It is then converted back into an analog voice signal, allowing the recipient to hear the call.
Once the call is finished, SIP handles the coordinated closing of the session.