Nowadays, telecommunications traffic mostly passes over a packet-switching network and very often over the public internet. Speech data is translated into packets for call participants and redirected 'toward' a recipient. Packet switching networks do not ensure that packets travel the same path to their destination.
Another common and equally frustrating annoyance that affects phone calls is the dreaded Jitter. A cousin to latency, Jitter is referred to as packet delay variation carrying voice or video data over a communications channel. Jitter happens when the RTP packet stream traverses the network (LAN, WAN, or Internet) because it has to share network capacity with other data.
Jitter can cause many participants to leave the phone call and either attempt to re-establish a connection or move to another form of communication. We often tend to ‘drop the call’ and quickly try to reconnect straight away. According to Cisco, the acceptable amount of Jitter time for a customer should be limited to 30ms. Jitter is more about pacing and a series of RTP packets that are coming in. When they appear in the right series in a good steady stream at regular intervals then you've got low jitter. Whether they come in interspersed intervals with holes, or if they appear out of order, then you've got a high jitter.
There are three types of Jitter:
- Random Jitter: Also referred to as “Gaussian jitter”, random Jitter is unpredictable electronic timing noise. Usually resulting from clock timing problems or random electronic timing noise.
- Deterministic Jitter: This type of jitter can be predicted or determined. Reproducible, limited and may be periodic. Deterministic jitter can either be correlated to the data stream (data-dependent jitter) or uncorrelated to the data stream (bounded uncorrelated jitter).
- Total Jitter: Calculated using a bit error ratio (BER), as well as a combination of random and deterministic jitter. The mathematical formula for calculating total jitter is Total jitter = Deterministic jitter+2*BER*Random jitter.
There are a number of causes for Jitter. Some of which are:
- Network Congestion: Jitter is generally caused by congestion in the IP network. The congestion can occur either at the router interfaces or in a provider or carrier network if the circuit has not been provisioned correctly. Having too many devices hooked up to the same system, all being used at the same time, will run out of bandwidth, slowing your connection to a crawl.
- Wireless Networks: While a wireless network allows mobility and eliminates the need for wires running through the office, there is a chance that you will experience a degraded network link. WiFi is not necessarily secure or stable enough for our mobile devices to depend on for our phone calls.
Other notable causes:
Hardware capacity on source/destination
Bad routing through network/internet (traffic routed in a suboptimal fashion introducing delays, etc)
While Jitter can be minimized and reduced, through the use of Jitter buffers, this is not a ‘magic solution’. Depending on the size of the jitter buffer, out of sequence packets can be rearranged before being delivered. However, this does introduce some delay in addition to the original latency on the network.
Increasing the buffer size directly increases the delay of actions to the receiving end. Likewise having a buffer be too small can lead to call degradation as an excessive number of packets may be discarded. So you need to be cautious in the size of the buffer. Generally, they are only effective for delay variations of less than 100 ms, and even then, deterioration in quality may be easily noticeable to users.
Another issue to be cautious of when using Jitter buffers is ‘buffer bloat’. Bufferbloat occurs when your router is unable to transmit all the packets required and builds up too large a queue (rather than dropping packets when the queue length starts making latency noticeable). This queuing causes large latency and bursts of jitter. The real-time nature of voice means that this is not helpful. Therefore organizations should try to identify the sources of jitter on their networks instead of relying on jitter buffers.
Network monitoring does measure a nice selection of audio quality factors including Jitter, but provides a view to the private network, while businesses use inbound services to engage with customers from outside of their private network. Number Testing provides a wider variety and end-to-end perspective. Through the use of proactive monitoring and number testing, Spearline is able to alert issues that may have underlying jitter causes, rather than be overly reliant on Jitter Buffers. Be sure to read our whitepaper today for a better in-depth understanding of network monitoring and number testing.
*For more content, be sure to connect with Josh O'Farrell on LinkedIn here.
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