Call admission control

Last updated

Call admission control (CAC) is a form of admission control that prevents or mitigates oversubscription of VoIP networks. CAC is used in the call set-up phase and applies to real-time media traffic as opposed to data traffic. CAC mechanisms complement and are distinct from the capabilities of quality of service tools to protect voice traffic from the negative effects of other voice traffic and to keep excess voice traffic off the network. Since it averts voice traffic congestion, it is a preventive Congestion Control Procedure. It ensures that there is enough bandwidth for authorized flows.

Integrated services with RSVP (which reserve resources for the flow of packets through the network) using controlled-load service ensures that a call cannot be set up if it cannot be supported. CAC rejects calls when either there is insufficient CPU processing power, the upstream and downstream traffic exceeds prespecified thresholds, or the number of calls being handled exceeds a specified limit. [1]

CAC can be used to prevent congestion in connection-oriented protocols such as ATM. In that context, there are several schemes available. [2] However, VoIP differs in that it uses RTP, UDP and IP, all of which are connectionless protocols.

Related Research Articles

<span class="mw-page-title-main">Asynchronous Transfer Mode</span> Digital telecommunications protocol for voice, video, and data

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.

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 Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE (VoLTE).

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.

<span class="mw-page-title-main">Frame Relay</span> Wide area network technology

Frame Relay is a standardized wide area network (WAN) technology that specifies the physical and data link layers of digital telecommunications channels using a packet switching methodology. Originally designed for transport across Integrated Services Digital Network (ISDN) infrastructure, it may be used today in the context of many other network interfaces.

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.

<span class="mw-page-title-main">Transport layer</span> Layer in the OSI and TCP/IP models providing host-to-host communication services for applications

In computer networking, the transport layer is a conceptual division of methods in the layered architecture of protocols in the network stack in the Internet protocol suite and the OSI model. The protocols of this layer provide end-to-end communication services for applications. It provides services such as connection-oriented communication, reliability, flow control, and multiplexing.

Differentiated services or DiffServ is a computer networking architecture that specifies a mechanism for classifying and managing network traffic and providing quality of service (QoS) on modern IP networks. DiffServ can, for example, be used to provide low-latency to critical network traffic such as voice or streaming media while providing best-effort service to non-critical services such as web traffic or file transfers.

An interexchange carrier (IXC), in U.S. legal and regulatory terminology, is a type of telecommunication company, commonly called a long-distance telephone company. It is defined as any carrier that provides services across multiple local access and transport areas (interLATA). Calls made on telephone circuits within the local geographic area covered by one local network are handled only by that intraLATA carrier, commonly called a local telephone exchange carrier. Local calls are usually defined by connections made without additional charge whether the connected call is in the same LATA or connects to another LATA with no charge. IntraLATA usually refers to rated or toll calls between LATA within state boundaries, as opposed to interstate, or calls between LATAs in different states.

Traffic shaping is a bandwidth management technique used on computer networks which delays some or all datagrams to bring them into compliance with a desired traffic profile. Traffic shaping is used to optimize or guarantee performance, improve latency, or increase usable bandwidth for some kinds of packets by delaying other kinds. It is often confused with traffic policing, the distinct but related practice of packet dropping and packet marking.

Network congestion in data networking and queueing theory is the reduced quality of service that occurs when a network node or link is carrying more data than it can handle. Typical effects include queueing delay, packet loss or the blocking of new connections. A consequence of congestion is that an incremental increase in offered load leads either only to a small increase or even a decrease in network throughput.

A session border controller (SBC) is a network element deployed to protect SIP based voice over Internet Protocol (VoIP) networks.

In communications, traffic policing is the process of monitoring network traffic for compliance with a traffic contract and taking steps to enforce that contract. Traffic sources which are aware of a traffic contract may apply traffic shaping to ensure their output stays within the contract and is thus not discarded. Traffic exceeding a traffic contract may be discarded immediately, marked as non-compliant, or left as-is, depending on administrative policy and the characteristics of the excess traffic.

The next-generation network (NGN) is a body of key architectural changes in telecommunication core and access networks. The general idea behind the NGN is that one network transports all information and services by encapsulating these into IP packets, similar to those used on the Internet. NGNs are commonly built around the Internet Protocol, and therefore the term all IP is also sometimes used to describe the transformation of formerly telephone-centric networks toward NGN.

<span class="mw-page-title-main">H.323</span> Audio-visual communication signaling protocol

H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences.

The Skype protocol is a proprietary Internet telephony network used by Skype. The protocol's specifications have not been made publicly available by Skype and official applications using the protocol are closed-source.

Voice over Internet Protocol (VoIP) recording is a subset of telephone recording or voice logging, first used by call centers and now being used by all types of businesses. There are many reasons for recording voice over IP call traffic such as: reducing company vulnerability to lawsuits by maintaining recorded evidence, complying with telephone call recording laws, increasing security, employee training and performance reviews, enhancing employee control and alignment, verifying data, sharing data as well as customer satisfaction and enhancing call center agent morale.

In computing, Microsoft's Windows Vista and Windows Server 2008 introduced in 2007/2008 a new networking stack named Next Generation TCP/IP stack, to improve on the previous stack in several ways. The stack includes native implementation of IPv6, as well as a complete overhaul of IPv4. The new TCP/IP stack uses a new method to store configuration settings that enables more dynamic control and does not require a computer restart after a change in settings. The new stack, implemented as a dual-stack model, depends on a strong host-model and features an infrastructure to enable more modular components that one can dynamically insert and remove.

Time-Sensitive Networking (TSN) is a set of standards under development by the Time-Sensitive Networking task group of the IEEE 802.1 working group. The TSN task group was formed in November 2012 by renaming the existing Audio Video Bridging Task Group and continuing its work. The name changed as a result of the extension of the working area of the standardization group. The standards define mechanisms for the time-sensitive transmission of data over deterministic Ethernet networks.

References

  1. "Call Admission Control" (PDF). Archived from the original (PDF) on 2005-05-19. Retrieved 2005-06-03.
  2. Call admission control schemes: a review. HG Perros, KM Elsayed, N Inc - IEEE Communications Magazine, 1996 - comsoc.org