1. Everything begins with the spoken word. A microphone on the handset records the sound pressure of the spoken word and converts it into an amplitude, i.e. an analogue signal. The amplitude represents, for example, the volume or frequency of the spoken word.
    
2. In the next step, this signal must be digitised. For this, the previous waveform or its amplitude must be "quantised". In this process, the state of the signal is defined with a unique, specific value. For example, with a 1 or a 0 depending on the amplitude level.
    
3. This resolves the previous wave into a raster with each "deflection" in the raster resembling the shape of the previous analogue amplitude.
    
4. In order to lose as little information as possible when digitising the signal, the sampling rate of the quantisation is crucial - i.e. how finely the raster samples the previous amplitude.
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5. But digitisation alone is not enough - the signal must now be processed, compressed and, above all, made ready for transmission. Of course, all this has to be done without interruption and in real time - and extremely efficiently.

Real-time communication

After the signal has been compressed, optimised and also divided into small data packets, the packets must now be sent to the receiver as efficiently and quickly as possible. For this purpose, various protocols are used that are directly integrated in the SIP protocol. The SIP network protocol takes over the control and the communication session between two or more participants. The SIP protocol not only negotiates the communication modalities, other protocols such as the (S)RTP or UDP protocol are also integrated into the protocol itself.

Communication in Internet telephony must take place in real time, i.e. the previously created data packets must arrive at the recipient quickly and, above all, at the right time, be unpacked and converted back. The so-called RTP (real-time transport protocol) has the task of preparing the packets for transmission. Similar to the postal service, this protocol ensures that the individual data packets can be sent as efficiently as possible and with all the necessary information that is later required for decoding the packets.

Once all packets are prepared, the so-called UDP protocol (User Datagram Protocol) takes over. UDP is a minimal connectionless network protocol that belongs to the transport layer of the Internet protocols. It enables the fast transmission of data packages via computer networks and the Internet.

However, the UDP protocol only works unilaterally, i.e. it sends data packets "come what may" and never knows whether the packets have arrived completely at the recipient. This is also the reason why the packets were packed efficiently and small in the previous steps, because if one of the packets cannot be sent in time or does not arrive completely at the recipient, it is considered lost.

If the data package has arrived completely at the receiver in time, the whole process starts again from the beginning, but in reverse order: The required data packages are combined and unpacked, the digital audio signal is converted to an analogue signal and output via the loudspeaker integrated in the handset.