In the field of computer networking and other packet-switched telecommunication networks, the traffic engineering term quality of service (QoS) refers to resource reservation control mechanisms rather than the achieved service quality. Quality of service is the ability to provide different priority to different applications, users, or data flows, or to guarantee a certain level of performance to a data flow. For example, a required bit rate, delay, jitter, packet dropping probability and/or bit error rate may be guaranteed. Quality of service guarantees are important if the network capacity is insufficient, especially for real-time streaming multimedia applications such as voice over IP, online games and IP-TV, since these often require fixed bit rate and are delay sensitive, and in networks where the capacity is a limited resource, for example in cellular data communication. In the absence of network congestion, QoS mechanisms are not required.
A network or protocol that supports QoS may agree on a traffic contract with the application software and reserve capacity in the network nodes, for example during a session establishment phase. During the session it may monitor the achieved level of performance, for example the data rate and delay, and dynamically control scheduling priorities in the network nodes. It may release the reserved capacity during a tear down phase.
A best-effort network or service does not support quality of service. An alternative to complex QoS control mechanisms is to provide high quality communication over a best-effort network by over-provisioning the capacity so that it is sufficient for the expected peak traffic load.
In the field of telephony, quality of service was defined in the ITU standard X.902 as "A set of quality requirements on the collective behavior of one or more objects". Quality of Service comprises requirements on all the aspects of a connection, such as service response time, loss, signal-to-noise ratio, cross-talk, echo, interrupts, frequency response, loudness levels, and so on. A subset of telephony QoS is Grade of Service (GOS) requirements, which comprises aspects of a connection relating to capacity and coverage of a network, for example guaranteed maximum blocking probability and outage probability.
QoS is sometimes used as a quality measure, with many alternative definitions, rather than referring to the ability to reserve resources. Quality of service sometimes refers to the level of quality of service, i.e. the guaranteed service quality. High QoS is often confused with a high level of performance or achieved service quality, for example high bit rate, low latency and low bit error probability. See also Relation to subjective quality measures below.
Quality of Service Experience (QoSE) on the other hand, is the actual measure of user’s experience with an operator in terms of delivered quality with or without reference to what is being promised. This differs from QoS as the former is defined only in the context of user experience but not Quality of Experience (QoE) because it is not subjective.
When the Internet was first deployed many years ago, it lacked the ability to provide Quality of Service guarantees due to limits in router computing power. It therefore ran at default QoS level, or "best effort". There were four "Type of Service" bits and three "Precedence" bits provided in each message, but they were ignored. These bits were later re-defined as DiffServ Code Points (DSCP) and are largely honored in peered links on the modern Internet.
When looking at packet-switched networks, Quality of service is affected by various factors, which can be divided into "human" and "technical" factors. Human factors include: stability of service, availability of service, delays, user information. Technical factors include: reliability, scalability, effectiveness, maintainability, Grade of Service, etc.
Many things can happen to packets as they travel from origin to destination, resulting in the following problems as seen from the point of view of the sender and receiver: Dropped packets : The routers might fail to deliver (drop) some packets if they arrive when their buffers are already full. Some, none, or all of the packets might be dropped, depending on the state of the network, and it is impossible to determine what will happen in advance. The receiving application may ask for this information to be retransmitted, possibly causing severe delays in the overall transmission. Delay : It might take a long time for a packet to reach its destination, because it gets held up in long queues, or takes a less direct route to avoid congestion. In some cases, excessive delay can render an application, such as VoIP or online gaming unusable. Jitter : Packets from the source will reach the destination with different delays. A packet's delay varies with its position in the queues of the routers along the path between source and destination and this position can vary unpredictably. This variation in delay is known as jitter and can seriously affect the quality of streaming audio and/or video. Out-of-order delivery : When a collection of related packets is routed through the Internet, different packets may take different routes, each resulting in a different delay. The result is that the packets arrive in a different order than they were sent. This problem requires special additional protocols responsible for rearranging out-of-order packets to an isochronous state once they reach their destination. This is especially important for video and VoIP streams where quality is dramatically affected by both latency and lack of isochronicity. Error : Sometimes packets are misdirected, or combined together, or corrupted, while en route. The receiver has to detect this and, just as if the packet was dropped, ask the sender to repeat itself.
A defined Quality of Service may be required for certain types of network traffic, for example:
These types of service are called inelastic, meaning that they require a certain minimum level of bandwidth and a certain maximum latency to function.
By contrast, elastic applications can take advantage of however much or little bandwidth is available. Bulk file transfer applications that rely on TCP are generally elastic.
An alternative to complex QoS control mechanisms is to provide high quality communication by generously over-provisioning a network so that capacity is based on peak traffic load estimates. This approach is simple and economical for networks with predictable and light traffic loads. The performance is reasonable for many applications. This might include demanding applications that can compensate for variations in bandwidth and delay with large receive buffers, which is often possible for example in video streaming.
Commercial VoIP services are often competitive with traditional telephone service in terms of call quality even though QoS mechanisms are usually not in use on the user's connection to his ISP and the VoIP provider's connection to a different ISP. Under high load conditions, however, VoIP quality degrades to cell-phone quality or worse. The mathematics of packet traffic indicate that a network with QoS can handle four times as many calls with tight jitter requirements as one without QoS . The amount of over-provisioning in interior links required to replace QoS depends on the number of users and their traffic demands. As the Internet now services close to a billion users, there is little possibility that over-provisioning can eliminate the need for QoS when VoIP becomes more commonplace .
For narrowband networks more typical of enterprises and local governments, however, the costs of bandwidth can be substantial and over provisioning is hard to justify. In these situations, two distinctly different philosophies were developed to engineer preferential treatment for packets which require it.
Early work used the "IntServ" philosophy of reserving network resources. In this model, applications used the Resource reservation protocol (RSVP) to request and reserve resources through a network. While IntServ mechanisms do work, it was realized that in a broadband network typical of a larger service provider, Core routers would be required to accept, maintain, and tear down thousands or possibly tens of thousands of reservations. It was believed that this approach would not scale with the growth of the Internet, and in any event was antithetical to the notion of designing networks so that Core routers do little more than simply switch packets at the highest possible rates.
The second and currently accepted approach is "DiffServ" or differentiated services. In the DiffServ model, packets are marked according to the type of service they need. In response to these markings, routers and switches use various queuing strategies to tailor performance to requirements. (At the IP layer, differentiated services code point (DSCP) markings use the 6 bits in the IP packet header. At the MAC layer, VLAN IEEE 802.1Q and IEEE 802.1D can be used to carry essentially the same information)
Routers supporting DiffServ use multiple queues for packets awaiting transmission from bandwidth constrained (e.g., wide area) interfaces. Router vendors provide different capabilities for configuring this behavior, to include the number of queues supported, the relative priorities of queues, and bandwidth reserved for each queue.
In practice, when a packet must be forwarded from an interface with queuing, packets requiring low jitter (e.g., VoIP or VTC) are given priority over packets in other queues. Typically, some bandwidth is allocated by default to network control packets (e.g., ICMP and routing protocols), while best effort traffic might simply be given whatever bandwidth is left over.
Additional bandwidth management mechanisms may be used to further engineer performance, to include:
As mentioned, while DiffServ is used in many sophisticated enterprise networks, it has not been widely deployed in the Internet. Internet peering arrangements are already complex, and there appears to be no enthusiasm among providers for supporting QoS across peering connections, or agreement about what policies should be supported in order to do so.
One compelling example of the need for QoS on the Internet relates to this issue of congestion collapse. The Internet relies on congestion avoidance protocols, as built into TCP, to reduce traffic load under conditions that would otherwise lead to Internet Meltdown. QoS applications such as VoIP and IPTV, because they require largely constant bitrates and low latency cannot use TCP, and cannot otherwise reduce their traffic rate to help prevent meltdown either. QoS contracts limit traffic that can be offered to the Internet and thereby enforce traffic shaping that can prevent it from becoming overloaded, hence they're an indispensable part of the Internet's ability to handle a mix of real-time and non-real-time traffic without meltdown.
Asynchronous Transfer Mode (ATM) network protocol has an elaborate framework to plug in QoS mechanisms of choice. Shorter data units and built-in QoS were some of the unique selling points of ATM in the telecommunications applications such as video on demand, voice over IP.
|Priority Level||Traffic Type|
|3||Excellent Load (Business Critical)|
|4||Controlled Load (Streaming Multimedia)|
|5||Voice and Video
(Interactive Media and Voice)|
[Less than 100ms latency and jitter]
|6||Layer 3 Network Control Reserved Traffic |
[Less than 10ms latency and jitter]
|7||Layer 2 Network Control Reserved Traffic
[Lowest latency and jitter]
The research project MUSE defined a QoS concept in Phase I which was further worked out in another research project PLANETS The new idea of this solution is to agree on a discrete jitter value per QoS class which is imposed on network nodes. Including best effort, four QoS classes were defined, two elastic and two inelastic. The solution has several benefits:
The MUSE project finally elaborated its own QoS solution which is primarily based in: