What Is: Jitter

What is Jitter

What is Jitter?

Today, telecommunications traffic largely passes over a packet-switching network and very often over the public internet. Speech data translates into packets for call participants and redirects ‘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 jitter. A cousin to latency, jitter refers to 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, 30ms is the acceptable amount of jitter time for a customer. It 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.

What are the different types of jitter?

  1. Random: Also referred to as “Gaussian jitter”, this has unpredictable electronic timing noise; usually resulting from clock timing problems or random electronic timing noise.
  2. Deterministic: Predicted or determined, this is reproducible, limited, and it may be periodic. It either correlates to the data stream (data-dependent jitter) or uncorrelated to the data stream (bounded uncorrelated jitter).
  3. 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 = deterministic +2*BER*random.

What are the causes?

  • 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:

  • Misbehaving hardware

  • Hardware capacity on source/destination

  • Network capacity

  • Bad routing through network/internet (traffic routed in a suboptimal fashion introducing delays, etc)

How to reduce jitter

While minimizing jitter is possible through the use of buffers, this is not a ‘magic solution’. Depending on the size of the buffer, rearranging out of sequence packets is possible 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’. Buffer bloat 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 buffers. Be sure to read our whitepaper today for a better in-depth understanding of network monitoring and number testing.

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If you are new to Spearline and would like to find out more about how you can benefit from our platform, we would love to speak with you. Please send us a brief message, and we will be in contact with you shortly.

About Us

Spearline is a technology company that proactively tests toll and toll-free numbers for connectivity and audio quality globally. It enables organizations to provide uninterrupted services to customers around the world. For further information, or if you have any further questions please contact us.

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