How to keep good voice quality in poor network connectivity

For air traffic controllers and pilots of aircrafts, good voice quality is crucial. EUROCAE has set the ED-138 standard for network performance to ensure quality networks. However, it is a real challenge in some areas, or even not possible to reach this network performance.

This might lead to a potential issue with the sound quality. The cause of worse sound quality can be either jitter or packet loss. Jitter means that the sound packets sent at a regular interval are not received as sent due to congestions on the network. This can cause an irregular broadcasting at the radio site (or the other way around).

Packet loss is mostly caused by bandwidth issues or congestions in network equipment. Even the loss of 1 audio packet can have an impact on the Voice quality. Since the communication between the Air Traffic Controller and the pilot should always be with minimum delays, compensating for poor network quality is a challenge and network specific.

Different ways to compensate for poor network quality

To compensate for packet loss, you can retransmit lost packets on the network. This will keep the audio quality high. The TCP protocol for example has this feature integrated. However this means that you need to wait for all packets to arrive before you can start the radio transmission. This retransmission causes unacceptable delays in radio transmission.Compensating for jitter on the IP network comes with the same tradeoff as retransmitting lost packets. Buffering is required to ensure that the radio transmission is seamless without gaps in the audio. Therefor the UDP protocol is used, as lost packets or packets with too much jitter will be too late for transmission any way.Compensation always means that buffering the audio is required.

Audio buffering as a solution

As indicated, the solution lies in buffering audio on the side of the sender transmitting the radio waves. MEP implemented buffering in such a way that it can handle packets that have been overtaken by other packets, or even packets that have been accidentally duplicated on the network. MEP has full control over the amount of time the audio is buffered and is able to configure this in the optimal way given the real network performance.This ensures the minimal delay on the network and the best audio quality.

Solution for Climax and Simulcast

For Climax and Simulcast we take the slowest set-up point in the network as the benchmark and make sure it always has enough time to receive audio. For all other connections we ensure the least amount of buffering necessary.

Some examples of project implementations

MEP has implemented a Simulcast system on many offshore oil riggs, even when an ED-138 network is not available, allowing good quality Voice sound with helicopters and ships.MEP has implemented excellent sound quality between air traffic controllers located on several remote islands and high mountains where only low quality IP connections are available. These locations are required to achieve good radio coverage between the mountains and the islands.

Conclusion

The ED-138 standard network quality cannot always be achieved. We don't shy away from the challenge of a poor network connection. With our solution a poor network connection no longer means poor voice quality.

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