Snom Audio

One of the central basic needs of human beings is communication.

This can be in writing, but also in pictures, films and music.

Direct interpersonal communication is particularly complex and is known to take place on different levels. Body language, gestures, facial expressions play just as important a role as intonation, pitch, tempo and volume as well as the content of what is said. All these components play an almost equal role. Since the visual components are missing in audio communication, it is all the more important when transmitting conversations to transmit them exactly as they originally sound - without delay, without distortions - just as if the speaker were sitting across from you.

And this is precisely where the first hurdle lies in telephony and even more so in digital. Because anything is possible here - for better or for worse!

For this reason, audio quality and its transmission with as little loss as possible is at the core of Snom's efforts. Offering the best "lifelike" telephony is what Snom is known and loved for.

On this page you will learn which challenges Snom masters and how telephony over IP (VoIP) works.

Have you ever wondered how it all works with (VoIP) telephony when you pick up the phone to make a call? 

In the following video, we would like to give you an understanding of the complicated processes of real-time communication.




Summary: How a VoIP call works

  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.
  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.

The codec

The codec not only digitises the signal, it also compresses it - in other words, it reduces the file size without degrading the signal too much. At the same time, the codec also decodes the signal at the remote station and converts it back to an analogue signal that we can understand.
Compression plays an important role in real-time communication. After all, the signal must arrive at the remote station in real time and without interruption. At the same time, compression always means a loss of quality - the original signal is cut.
The example of MP3 shows that compression does not always have to be a bad thing. With this type of compression, signals that are masked by other, louder signals are simply cut away. This means less information and thus smaller, more efficient files.
Codecs in IP telephony also work in a similar way to create data packets that are as small and efficient as possible. And this works as follows: The codec samples the audio signal every 125 microseconds at a sampling rate of 8,000 Hz and creates a sample. This sample is then compressed to 8bit by, for example, reducing the frequency range from the original 15,000 hearts to 3,000 - 3,400 hearts.

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.

Whitepaper Basics Audio Paper

Audio quality in
VoIP telephony

Get a good insight into VoIP audio technology in general
and technology in general and Snom in particular.

(German version)

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+49 30 - 39833-0
Office hours: Mo-Fr 9:00-17:00 (CET)


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phone +49 30 39833 0



Snom Technology GmbH
Aroser Allee 66
13407 Berlin

Phone: +49 30 39833-0
Fax: +49 30 39833-111


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