 | This article needs additional citations for verification .(May 2024) |
In computer networking, packet delay variation (PDV) is the difference in end-to-end one-way delay between selected packets in a flow with any lost packets being ignored. [1] The effect is sometimes referred to as packet jitter, although the definition is an imprecise fit.
The term PDV is defined in ITU-T Recommendation Y.1540, Internet protocol data communication service - IP packet transfer and availability performance parameters, section 6.2.
In computer networking, although not in electronics, usage of the term jitter may cause confusion. From RFC 3393 (section 1.1):
The variation in packet delay is sometimes called "jitter". This term, however, causes confusion because it is used in different ways by different groups of people. ... In this document we will avoid the term "jitter" whenever possible and stick to delay variation which is more precise.
The means of packet selection for measurement is not specified in RFC 3393, but could, for example, be the packets that had the largest variation in delay in a selected time period.
The delay is specified from the start of the packet being transmitted at the source to the start of the packet being received at the destination. A component of the delay which does not vary from packet to packet can be ignored, hence if the packet sizes are the same and packets always take the same time to be processed at the destination then the packet arrival time at the destination could be used instead of the time the end of the packet is received.
Instantaneous packet delay variation is the difference between successive packets—here RFC 3393 does specify the selection criteria—and this is usually what is loosely termed "jitter", although jitter is also sometimes the term used for the variance of the packet delay. As an example, say packets are transmitted every 20 ms. If the second packet is received 30 ms after the first packet, IPDV = +10 ms. This is referred to as dispersion. If the second packet is received 10 ms after the first packet, IPDV = −10 ms. This is referred to as clumping.
It is also possible to visualize (I)PDV measurements, which makes interpreting and understanding the network easier, or (for bigger datasets) possible at all.
One possible diagram type are simple point cloud diagrams in which the x-axis represents the packet number and the y-axis contains the corresponding (I)PDV values, one dot for each measurement.
Another type are distribution histograms which are more useful for bigger datasets or even comparisons of different paths or technologies.
The effects of PDV in multimedia streams can be mitigated by a properly sized buffer at the receiver. As long as the bandwidth can support the stream, and the buffer size is sufficient, buffering only causes a detectable delay before the start of media playback.
However, for interactive real-time applications, e.g., voice over IP (VoIP), PDV can be a serious issue and hence VoIP transmissions may need quality-of-service–enabled networks to provide a high-quality channel.
Asynchronous Transfer Mode (ATM) is a telecommunications standard defined by the American National Standards Institute and ITU-T for digital transmission of multiple types of traffic. ATM was developed to meet the needs of the Broadband Integrated Services Digital Network as defined in the late 1980s, and designed to integrate telecommunication networks. It can handle both traditional high-throughput data traffic and real-time, low-latency content such as telephony (voice) and video. ATM provides functionality that uses features of circuit switching and packet switching networks by using asynchronous time-division multiplexing. ATM was seen in the 1990s as a competitor to Ethernet and networks carrying IP traffic as, unlike Ethernet, it was faster and designed with quality-of-service in mind, but it fell out of favor once Ethernet reached speeds of 1 gigabits per second.
The Internet Control Message Protocol (ICMP) is a supporting protocol in the Internet protocol suite. It is used by network devices, including routers, to send error messages and operational information indicating success or failure when communicating with another IP address. For example, an error is indicated when a requested service is not available or that a host or router could not be reached. ICMP differs from transport protocols such as TCP and UDP in that it is not typically used to exchange data between systems, nor is it regularly employed by end-user network applications.
In computer networking, the maximum transmission unit (MTU) is the size of the largest protocol data unit (PDU) that can be communicated in a single network layer transaction. The MTU relates to, but is not identical to the maximum frame size that can be transported on the data link layer, e.g., Ethernet frame.
Quality of service (QoS) is the description or measurement of the overall performance of a service, such as a telephony or computer network, or a cloud computing service, particularly the performance seen by the users of the network. To quantitatively measure quality of service, several related aspects of the network service are often considered, such as packet loss, bit rate, throughput, transmission delay, availability, jitter, etc.
The Transmission Control Protocol (TCP) is one of the main protocols of the Internet protocol suite. It originated in the initial network implementation in which it complemented the Internet Protocol (IP). Therefore, the entire suite is commonly referred to as TCP/IP. TCP provides reliable, ordered, and error-checked delivery of a stream of octets (bytes) between applications running on hosts communicating via an IP network. Major internet applications such as the World Wide Web, email, remote administration, and file transfer rely on TCP, which is part of the Transport layer of the TCP/IP suite. SSL/TLS often runs on top of TCP.
In computer networking, the User Datagram Protocol (UDP) is one of the core communication protocols of the Internet protocol suite used to send messages to other hosts on an Internet Protocol (IP) network. Within an IP network, UDP does not require prior communication to set up communication channels or data paths.
In electronics and telecommunications, jitter is the deviation from true periodicity of a presumably periodic signal, often in relation to a reference clock signal. In clock recovery applications it is called timing jitter. Jitter is a significant, and usually undesired, factor in the design of almost all communications links.
Network address translation (NAT) is a method of mapping an IP address space into another by modifying network address information in the IP header of packets while they are in transit across a traffic routing device. The technique was originally used to bypass the need to assign a new address to every host when a network was moved, or when the upstream Internet service provider was replaced, but could not route the network's address space. It has become a popular and essential tool in conserving global address space in the face of IPv4 address exhaustion. One Internet-routable IP address of a NAT gateway can be used for an entire private network.
Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for voice calls for the delivery of voice communication sessions over Internet Protocol (IP) networks, such as the Internet.
The leaky bucket is an algorithm based on an analogy of how a bucket with a constant leak will overflow if either the average rate at which water is poured in exceeds the rate at which the bucket leaks or if more water than the capacity of the bucket is poured in all at once. It can be used to determine whether some sequence of discrete events conforms to defined limits on their average and peak rates or frequencies, e.g. to limit the actions associated with these events to these rates or delay them until they do conform to the rates. It may also be used to check conformance or limit to an average rate alone, i.e. remove any variation from the average.
In computer networking and telecommunications, TDM over IP (TDMoIP) is the emulation of time-division multiplexing (TDM) over a packet-switched network (PSN). TDM refers to a T1, E1, T3 or E3 signal, while the PSN is based either on IP or MPLS or on raw Ethernet. A related technology is circuit emulation, which enables transport of TDM traffic over cell-based (ATM) networks.
Nagle's algorithm is a means of improving the efficiency of TCP/IP networks by reducing the number of packets that need to be sent over the network. It was defined by John Nagle while working for Ford Aerospace. It was published in 1984 as a Request for Comments (RFC) with title Congestion Control in IP/TCP Internetworks in RFC 896.
TCP tuning techniques adjust the network congestion avoidance parameters of Transmission Control Protocol (TCP) connections over high-bandwidth, high-latency networks. Well-tuned networks can perform up to 10 times faster in some cases. However, blindly following instructions without understanding their real consequences can hurt performance as well.
The Media Delivery Index (MDI) is a set of measures that can be used to monitor both the quality of a delivered video stream as well as to show system margin for IPTV systems by providing an accurate measurement of jitter and delay at network level (Internet Protocol, IP), which are the main causes for quality loss. Identifying and quantizing such problems in this kind of networks is key to maintaining high quality video delivery and providing indications that warn system operators with enough advance notice to allow corrective action.
T.38 is an ITU recommendation for allowing transmission of fax over IP networks (FoIP) in real time.
ITU-T Y.156sam Ethernet Service Activation Test Methodology is a draft recommendation under study by the ITU-T describing a new testing methodology adapted to the multiservice reality of packet-based networks.
Bufferbloat is a cause of high latency and jitter in packet-switched networks caused by excess buffering of packets. Bufferbloat can also cause packet delay variation, as well as reduce the overall network throughput. When a router or switch is configured to use excessively large buffers, even very high-speed networks can become practically unusable for many interactive applications like voice over IP (VoIP), audio streaming, online gaming, and even ordinary web browsing.
ITU-T Y.1564 is an Ethernet service activation test methodology, which is the new ITU-T standard for turning up, installing and troubleshooting Ethernet-based services. It is the only standard test methodology that allows for complete validation of Ethernet service-level agreements (SLAs) in a single test.
AES67 is a technical standard for audio over IP and audio over Ethernet (AoE) interoperability. The standard was developed by the Audio Engineering Society and first published in September 2013. It is a layer 3 protocol suite based on existing standards and is designed to allow interoperability between various IP-based audio networking systems such as RAVENNA, Wheatnet, Livewire, Q-LAN and Dante.
Deterministic Networking (DetNet) is an effort by the IETF DetNet Working Group to study implementation of deterministic data paths for real-time applications with extremely low data loss rates, packet delay variation (jitter), and bounded latency, such as audio and video streaming, industrial automation, and vehicle control.