Quality of Service (QoS) | Converged Networks

The Glossary of the Telecommunications Terms of the U.S. Federal Standard 10377 (available at www.its.bldrdoc.gov/fs-1037/fs-1037c.htm) defines quality of service (QoS) as:
1. The performance specification of a communications channel or system. . . . 2. A subjective rating of telephone communications quality in which listeners judge transmissions by qualifiers, such as excellent, good, fair, poor, or unsatisfactory.
This definition best expresses both the objective (that is, something based on a computable metrics) and subjective (that is, perception-based) aspects of the QoS concept. Three objectives that drive the need to integrate the Internet and the PSTN relate to QoS: (1) carrying voice across both the IP networks and the PSTN, (2) combining the PSTN transport and IP access to services, and (3) accessing the IP networks over the PSTN lines. The first item is associated with the most perceivable QoS requirements. Nevertheless, the IP network access aspect (and connected to it, issues of supporting VPNs) is equally important, as we will demonstrate.

To begin with, different applications (or, rather, their users) have different perceptions of what the QoS is. Using an application called telemedicine, a doctor may expect a copy of a brain tomogram taken in a remote laboratory. The transmission can be delayed for a few minutes; however, the doctor cannot afford to have any detail of the tomogram compromised (a missing detail could wrongly suggest a tumor or leave it undetected)—the chief QoS requirement here is that no data arrive in error. On the other hand, in an IP telephony application, an occasional error in the signal would cause no problem; however, a long delay (or a variable delay, called jitter) is likely to be unacceptable.
The model of routing for the PSTN is based on the concept of circuits, which are created end to end for the duration of the session. Circuits are mapped into fixed switched physical connections. Thus, any message between two end users in a session always traverses the same physical path for the duration of a session. [For conferencing, such circuits can be bridged (that is, joined) by switches or other devices that have switching fabric; thus, any multicast message will follow the same, predetermined set of physical circuits.] With this routing model, it is possible to determine whether a session that requires certain characteristics (such as bandwidth or loss tolerances) can be established. Once the session is established, it is relatively straightforward to guarantee that the requested characteristics will remain constant for the duration of the session.
One important factor in PSTN routing is the time that it takes to set up a circuit; the call setup time has traditionally been an essential QoS metric in the PSTN. Incidentally, this model, which naturally grew out of telephony, was applied by the ITU-T to the definition of virtual circuit for data communications standards. First, this concept was defined in X.25 and, subsequently, frame relay and asynchronous transfer mode (ATM) networks. ISDN access guarantees certain bandwidth (depending on a particular national standard) to the subscriber. Broadband ISDN (B-ISDN) access, in addition, specifies parameters that are needed by the ATM network. At this very moment, the mechanisms are being developed to back up B-ISDN by Intelligent Network, which could ensure policy-based networkwide QoS enforcement. Overall, in today’s PSTN, the main QoS metric is, as mentioned before, the call blocking rate.
The Internet routing model, on the other hand, has traditionally avoided stressing any built-in mechanism for creation and maintenance of virtual circuits. In the Internet, QoS issues (which also define their respective metrics) include bandwidth availability, latency (that is, end-to-end delay) control, jitter (that is, delay variation) control, and packet loss. Historically, the IP networks have been supporting what is called the best effort (but no guarantees) of packet transmission. In this system, no differentiation among different types of traffic is made, and neither the sequence of packet arrivals nor the arrival itself is guaranteed.
Whatever the end-to-end QoS requirements may be, at the network layer the packets travel (similarly to those of us who take airplanes) from hub to hub (that is, from router to router). Each router queues newly arriving packets for retransmission over the link to the most suitable (according to the routing table) router or destination host. Until very recently, most routers used a first-come, first-served queuing discipline, which is fair to all packets and, for this very reason, cannot make some packets more equal than others.
Overall, for applications such as voice or video over IP, a new network layer model was clearly needed, and such models have been researched and implemented since the 1990s. Two new approaches proposed mechanisms that are now called fair queuing and weighted fair queuing . With fair queuing and weighted fair queuing, routers are no longer required to treat packets equally. The incoming traffic is separated into well-defined flows. (A TCP connection is an example of a flow, although it may be difficult to detect by a router—all TCP connections between the same pair of hosts is a more realistic example; a voice session is another one.)

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