These days, most telecommunications traffic is routed through a packet-switching network, frequently routed over the public internet. For example, the network converts speech data into packets for call participants and is routed ‘towards’ a receiver. However, packet switching networks do not guarantee that packets will arrive on the same path as their destination.
The dreaded Jitter is another widespread and irritating nuisance that impacts your customer’s phone calls. Similar to latency, Jitter refers to the packet delay fluctuation when audio or video data travels across a communications connection. Because the RTP packet stream must share network capacity with other data, Jitter can occur.
Jitter can cause you to abandon a phone call. Leaving you to either attempt to reconnect or switch to another mode of communication. Frequently, we ‘drop the call’ and attempt to reconnect as soon as possible, hoping for a better connection. This issue is unacceptable as this can generate several problems for your business. Frustrated customers dealing with Jitter means a poor customer experience, loss in revenue and reputation.
According to Cisco, a customer’s tolerable Jitter time should be no more than 30ms. Jitter is more about timing and the receiving of RTP packets. When they appear in the correct series at regular intervals, you will have low Jitter. You will have high Jitter if they come in irregular intervals or appear out of order.
There are three types of Jitter:Â
- Random Jitter: Also referred to as Gaussian jitter. Random Jitter is unpredictable electronic timing noise resulting from clock timing problems or random electronic timing noise.Â
- Deterministic Jitter: This type of Jitter can be predicted, reproducible, limited and may be periodic. It can either be data-dependent Jitter correlated to the data stream or bounded Jitter uncorrelated to the data stream.Â
- Total Jitter: Calculated using a bit error ratio (BER) and a combination of random and deterministic Jitter. The mathematical formula for calculating total Jitter is –
Total Jitter = Deterministic Jitter + 2 * BER * Random Jitter.
What causes Jitter?
- Network Congestion –Â Congestion in the IP network is the most common cause of Jitter. If the circuit is not provided correctly, congestion might develop at:
– The router interfaces or
– The provider or carrier network.
If you connect too many devices to the same system, you will run out of bandwidth, slowing down your connection.
- Wireless Networks – While a wireless network provides mobility and removes the need for cables, there is a possibility of degraded network connectivity. Wi-Fi isn’t always safe or reliable enough for phone conversations on mobile devices.
- Poor and Misbehaving Hardware – Internet networks consist of many distinct hardware pieces, including a modem and a router, which often have switches. Lousy hardware, including an outdated modem, a broken Ethernet cable, or a malfunctioning router, contributes to issues with call quality.
While we can decrease and minimize Jitter by using buffers, this is not a “miracle solution.” We can adjust out-of-sequence packets before being transmitted, depending on the size of the Jitter buffer. However, this will add to the network’s initial latency and introduce some delays.
Increasing the buffer size delays activities at the receiving end. Similarly, having a too-small buffer size might cause call deterioration by discarding excessive packets. Therefore, you must consider the buffer size carefully. Furthermore, they are often only used for delay fluctuations of less than 100 ms, and even then, quality degradation may be evident.
Network monitoring measures a selection of audio quality factors, including Jitter. It provides a view of the private network while businesses use inbound services outside their private network with customers. Number Testing offers a wider variety and end-to-end perspective. Through proactive monitoring and number testing, Spearline can alert issues with underlying Jitter causes rather than rely on Jitter Buffers.
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