When data is sent across the Internet, it is sent in “packets”. These small chunks of data are sent from one end of the connection to the other, and in both directions.
Occasionally, a connection will experience issues with the packets being transported. In the case of “high latency and high jitter”, the packets are taking longer than usual, and longer than preferred, to reach their destination. Here, the latency determines the time taken for the packets to reach their destination, while the jitter determines the fluctuation in latency; from low to high.
When packets reach their destination, they are queued in a buffer. This buffer stores packets until they are ready to be processed. Sometimes, these packets are buffered simply because there are packets missing and the existing packets are awaiting the arrival of these missing packets in order for them to be decoded.
Aircall calls that have both high latency and high jitter suggest that a proportion of this latency is caused by the jitter. When using a jitter buffer, the audio transcoder will wait for missing packets before passing the audio to the end-user. If the jitter is high enough, and the buffer large enough, then the transcoder may wait some time, leading to increased latency. A means to reduce latency can be to reduce the size of the jitter buffer and thus causing some audio to drop. While a preferred solution is to improve the quality of the connection itself.
Possible Causes of Jitter
Jitter can happen for many reasons, whether at the NAT (router), the transport route, or a physical layer, such as a VPN.
If the local network has heavy traffic, the NAT (Network Address Translator) may have issues prioritizing the audio packets. This may result in some packets leaving or entering the network quickly, while others are delayed. If your NAT supports it, utilize its QoS feature (Quality of Service) to prioritize VoIP audio packets.
Typically, packets traveling within the Internet from one location to another will visit numerous “hops” along the way. A poorly provisioned hop may cause issues to packets that traverse through it, leading to latency, packet loss, and jitter issues. This is otherwise known as congestion. Forcing an alternative route, via a DNS proxy or similar mechanism, may improve transport reliability.
A VPN (or Virtual Private Network) is a common tool used by businesses and corporate networks to secure Internet communications. However, using such services will often add latency to a network and may also cause network switching, whereby packets of data may be sent along more than one transport route to the destination. This increases the occurrence of jitter, as packets are no longer traveling in a predictable, sequential path.
An additional caveat of VPNs is that all traffic must first traverse to the DNS gateway before it is transported to and from its destination. This usually creates a longer physical path for data to traverse.
Depending on the properties of your VPN connectivity, it may be that disabling it specifically for VoIP calls can greatly improve your experience.
Still in need of assistance? Please feel free to reach out to the Support Team and we’ll be happy to assist!