All of the devices used in this document started with a cleared default configuration. If your network is live, make sure that you understand the potential impact of any command. For more information on document conventions, refer to the Cisco Technical Tips Conventions. Jitter is defined as a variation in the delay of received packets. At the sending side, packets are sent in a continuous stream with the packets spaced evenly apart. Due to network congestion, improper queuing, or configuration errors, this steady stream can become lumpy, or the delay between each packet can vary instead of remaining constant.
The mechanism that handles this function is the playout delay buffer. The playout delay buffer must buffer these packets and then play them out in a steady stream to the digital signal processors DSPs to be converted back to an analog audio stream.
The playout delay buffer is also sometimes referred to as the de-jitter buffer. If the jitter is so large that it causes packets to be received out of the range of this buffer, the out-of-range packets are discarded and dropouts are heard in the audio.
For losses as small as one packet, the DSP interpolates what it thinks the audio should be and no problem is audible. When jitter exceeds what the DSP can do to make up for the missing packets, audio problems are heard. Enable Terminal Monitor in order to be able to see console messages through your Telnet session. Enter the show voice call summary command. Output similar to this appears:. The output that is given comes from the DSP that handles the call and is similar to this:.
Under this section, there are several parameters to look at. The main one is the number of Buf Overflow Discard ms that are seen. This counts the packets that are out of range for the playout delay buffer dropped. This may have some value in it, as long as it does not constantly increase. This number is a direct indication of excessive jitter. By default, this buffer runs in an adaptive mode where it dynamically adjusts to the amount of jitter present up to a point.
Configure the playout-delay command to change the defaults for the dynamic behavior of the de-jitter buffer. This buffer can also be set in fixed mode. This can fix some issues with jitter. 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.
The easiest and best place to start looking for jitter is at the router interfaces since you have direct control over this portion of the circuit. If the results of the test show a Jitter rating higher than 15—20ms, there is most likely an issue with the Internet Service Provider ISP. If unable to test plugged directly into the modem, the ISP should be able to run the test.
If they experience any jitter issues on their end, they should be able to resolve them. Test the network while the jitter is lower than ms to verify good audio. Jitter results can vary, making it very difficult to diagnose the problem. The fluctuation might have been high when the issue was occurring and corrected itself by the time the test completed.
Try running the test plugged directly into the router and work back until discovering what network device is causing jitter. Firewall Access Rules control the flow of inbound and outbound Internet traffic from the local network to the public Internet. Both routers and firewalls use access rules to control traffic and verify the source and destination addresses are permitted to send and receive traffic on the local network. Ensuring that the setting for SIP ALG is disabled is another frequent troubleshooting step to fix one-way audio issues.
While computer traffic will work fine on the network, the setting can affect the data stream for VoIP. Wireless jitter — One of the downsides of using a wireless network is a lower-quality network connection. Wired connections will help to ensure that voice and video call systems deliver a higher quality user experience.
Not implementing packet prioritization — For VoIP systems in particular, jitter occurs when audio data is not prioritized to be delivered before other types of traffic. QoS is the technology that manages data traffic in order to reduce jitter on your network and prevent or reduce the degradation of quality. QoS controls and manages network resources by setting priorities by which data is sent on the network.
Queuing - Enables you to prioritize or order packets so that delay-sensitive packets leave their queues more quickly than delay-insensitive packets. Link fragmentation and interleaving LFI - Routers do not pre-empt a packet that is currently being transmitted, so LFI reduces the sizes of larger packets into smaller fragments before sending them.
Compression - Payload or headers can be compressed, and this reduces the overall number of bits required to transmit the data. This requires less bandwidth, meaning queues shrink, which in turn reduces delay. Single Endpoint Where your network has control over just one of the endpoints aka single-ended , jitter is determined by measuring the mean round-trip time RTT , and the minimum RTT of a series of voice packets.
Double Endpoint In a double-ended path, the measurement used is the instantaneous jitter, or the variation between the intervals for transmitting and receiving a single packet. Jitter is the average difference between instantaneously measured jitter and the average instantaneous jitter throughout the transmission of a series of data packets.
Bandwidth Testing. Performing a bandwidth test can also determine the level of jitter. T roubleshooting network jitter can be tricky because of its unpredictability. Keeping jitter to a minimum begins by ensuring that your network is initially properly set up. Ensuring a quality network connection, enough bandwidth, and predictable latency can help reduce network jitter.
Jitter buffering - VoIP endpoints such as desk phones and ATAs usually include a jitter buffer to intentionally delay incoming data packets. A jitter buffer ensures that the receiving device can store a set number of packets and then realign them into the proper order, so that the receiver experiences minimum sound distortion. Jitter buffers are one way to address network jitter and latency but will not always work.
If a jitter buffer is too small then too many packets may be discarded, meaning bad call quality. If a jitter buffer is too large, then the additional delay can lead to conversational difficulty.
A typical jitter buffer configuration is 30ms to 50ms in size. You can increase buffer size to a point, but usually they are only effective for delay variations of less than ms. Perform a bandwidth test — Bandwidth testing sends files over a network to a specific computer, then measures the time required for the files to download at the destination.
This determines a theoretical data speed between the two points, measured in kilobits per second Kbps or megabits per second Mbps. Bandwidth tests can vary greatly. Factors that affect testing can be internet traffic, noise on data lines, file sizes, and load demand on the server at the time of testing.
Bandwidth testing should ideally be carried out several times to determine an average throughput. Improvements from within - Solving your VoIP network jitter problems may not be as challenging as you think.
Upgrade your ethernet cable - Outdated cables and switches can often cause high jitter issues. The latest cables are capable of transmitting data at MHz, as opposed to MHz, potentially solving ethernet jitter. Check your device frequency - A VoIP phone that operates at a higher frequency than a standard 2.
Some phones run at frequencies as high as 5. Reduce unnecessary bandwidth usage during work hours - Using large amounts of bandwidth for activities not related to work, like network gaming, or streaming video content can make jitter worse.
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