VoIP communication does not always correspond to quality that we expect from it. This is due to many factors, causing delays in the network. In addition to delays, there is a possibility of loss of traffic packages.
In this case, data arrives damaged, and the sides cannot hear each other, mark interference during the conversation and do not make out some phrases. Loss of packages that exceeds more than 5% is already critical for VoIP networks.
One of the most important parameters affecting the voice quality is the so-called jitter. This is a "jitter" of the digital signal, or random delay in data transfer that occurs when there are uneven time slots for transfer. The parameter is calculated in milliseconds and is rarely seen by man. However, when it comes to the transmission of data in real time - for example, video or voice traffic, the jitter becomes very noticeable. In a telephone conversation the interlocutors begin to interrupt each other or there are breaks, and it seems that people are talking on the radio. In such circumstances, efficient GSM voice termination is hardly possible because subscribers will quickly finish the conversation, and the terminator will have more and more short (<30 seconds) and zero calls (failed, "successful" and so on). The allowable value jitter is 150 ms.
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Causes of Jitter
There are 3 main causes of jitter in VoIP networks:
- The low bandwidth connection.
- Incorrect operation of VoIP equipment.
- High delay of VoIP-signal.
Normal operation of GSM-gateways requires a minimum throughput of 42 kbps per channel. The higher the score, the lower the value of the jitter in the network is. It is also important to correctly set up the gsm gateway to have it operating properly. To do this, you should contact a qualified technician. To prevent delays in VoIP-signal, you can use a method to prioritize traffic, which is to build queues to transfer a particular type of data (for example, you can set the priority for voice traffic).
In addition to these preventive measures, the so-called "jitter buffer" is used to combat the jitter. This mechanism is implemented in the IP PBX Asterisk, and many VoIP devices. The jitter buffer compensates for the uneven speed of traffic entering the receiving side, by creating a temporary repository for data packages. Its goal is that all incoming packages are sorted in the order that corresponds to the time stamp and then issued the codec. In the case of loss of traffic, the appliance re-requests lost packages.
The VoIP equipment calculates a buffer size automatically, or the user should set it in the settings. It should not be too large to not increase the delay very. In this case, the jitter buffer size may be small; otherwise it will cause traffic package loss when the delay time in the VoIP network is changed.
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