SIP trunking

Last updated

SIP trunking is a voice over Internet Protocol (VoIP) technology commonly referred to as 'elephant trunking'[ citation needed ] and a streaming media[ citation needed ] service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and unified communications facilities. [1] Most unified communications applications provide voice, video, and other streaming media applications such as desktop sharing, web conferencing, and shared whiteboard. [2]

Contents

Domains

The architecture of SIP trunking provides a partitioning of the unified communications network into two different domains of expertise: [3]

The interconnection between the two domains must occur through a SIP trunk.[ citation needed ] The interconnection between the two domains, created by transport via the Internet Protocol (IP), involves setting specific rules and regulations as well as the ability to handle some services and protocols that fall under the name of SIP trunking. [4]

The ITSP is responsible to the applicable regulatory authority regarding all the following law obligations of the public domain: [5]

The private domain instead, by nature, is not subject to particular constraints of law, and may be either the responsibility of the ITSP, the end user (enterprise), or of a third party who provides the voice services to the company [6] [7]

Architecture

Each domain has elements that perform the characteristic features requested of that domain, in particular the result (as part of any front-end network to the customer) is logically divided into two levels:

The private domain consists of three levels:

See also

Related Research Articles

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).

Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services over the Internet, rather than via the public switched telephone network (PSTN), also known as plain old telephone service (POTS).

<span class="mw-page-title-main">Media gateway</span>

A media gateway is a translation device or service that converts media streams between disparate telecommunications technologies such as POTS, SS7, Next Generation Networks or private branch exchange (PBX) systems. Media gateways enable multimedia communications across packet networks using transport protocols such as Asynchronous Transfer Mode (ATM) and Internet Protocol (IP).

Inter-Asterisk eXchange (IAX) is a communications protocol native to the Asterisk private branch exchange (PBX) software, and is supported by a few other softswitches, PBX systems, and softphones. It is used for transporting VoIP telephony sessions between servers and to terminal devices.

<span class="mw-page-title-main">Business telephone system</span> Multiline telephone system typically used in business environments

A business telephone system is a multiline telephone system typically used in business environments, encompassing systems ranging in technology from the key telephone system (KTS) to the private branch exchange (PBX).

The IP Multimedia Subsystem or IP Multimedia Core Network Subsystem (IMS) is a standardised architectural framework for delivering IP multimedia services. Historically, mobile phones have provided voice call services over a circuit-switched-style network, rather than strictly over an IP packet-switched network. Alternative methods of delivering voice (VoIP) or other multimedia services have become available on smartphones, but they have not become standardized across the industry. IMS is an architectural framework that provides such standardization.

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

An Internet telephony service provider (ITSP) offers digital telecommunications services based on Voice over Internet Protocol (VoIP) that are provisioned via the Internet.

<span class="mw-page-title-main">VoIP phone</span> Phone using one or more VoIP technologies

A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet. This is in contrast to a standard phone which uses the traditional public switched telephone network (PSTN).

<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.

<span class="mw-page-title-main">Acme Packet</span>

Acme Packet was a company based in Bedford, Massachusetts that sold session border controllers (SBCs), multiservice security gateways (MSGs), and session routing proxies (SRPs) to service providers and enterprises. It was a public company incorporated in Delaware. Acme Packet employs over 761 individuals in 31 countries.

UNIStim is a deprecated Telecommunications protocol developed by Nortel for IP Phone and IP PBX communications.

<span class="mw-page-title-main">Pbxnsip</span>

Pbxnsip is a software implementation of a telephone private branch exchange (PBX) produced by a company of the same name. Like any PBX, it allows attached telephones to make calls to one another, and to connect to other telephone services including the public switched telephone network (PSTN) and Voice over Internet Protocol (VoIP) services. Its name is a combination of the acronyms PBX and SIP.

<span class="mw-page-title-main">3CX Phone System</span> 3CX Phone System

The 3CXPhone System is the software-based private branch exchange (PBX) Phone system developed and marketed by the company, 3CX. The 3CX Phone System is based on the SIP standard and enables extensions to make calls via the public switched telephone network (PSTN) or via Voice over Internet Protocol (VoIP) services on premises, in the cloud, or via a cloud service owned and operated by the 3CX company. The 3CX Phone System is available for Windows, Linux, Raspberry Pi and supports standard SIP soft/hard phones, VoIP services, faxing, voice and web meetings, as well as traditional PSTN phone lines.

Cloud communications are Internet-based voice and data communications where telecommunications applications, switching and storage are hosted by a third-party outside of the organization using them, and they are accessed over the public Internet. Cloud services is a broad term, referring primarily to data-center-hosted services that are run and accessed over an Internet infrastructure. Until recently, these services have been data-centric, but with the evolution of VoIP, voice has become part of the cloud phenomenon. Cloud telephony refers specifically to voice services and more specifically the replacement of conventional business telephone equipment, such as a private branch exchange (PBX), with third-party VoIP service.

VaxTele SIP Server SDK is a complete development toolkit, which allows software vendors and Internet telephony service providers (ITSP) to develop SIP Server and (SIP) Session Initiation Protocol based VoIP systems for Microsoft Windows to install computer to computer voice chat, chat rooms, IVR systems, call center services, calling card services, dial/receive computer to PSTN and mobile phone calling services.

Federated VoIP is a form of packetized voice telephony that uses voice over IP between autonomous domains in the public Internet without the deployment of central virtual exchange points or switching centers for traffic routing. Federated VoIP uses decentralized addressing systems, such as ENUM, for location and identity information of participants and implements secure, trusted communications (TLS) for identify verification.

<span class="mw-page-title-main">Media gateway control protocol architecture</span>

The media gateway control protocol architecture is a methodology of providing telecommunication services using decomposed multimedia gateways for transmitting telephone calls between an Internet Protocol network and traditional analog facilities of the public switched telephone network (PSTN). The architecture was originally defined in RFC 2805 and has been used in several prominent voice over IP (VoIP) protocol implementations, such as the Media Gateway Control Protocol (MGCP) and Megaco (H.248), both successors to the obsolete Simple Gateway Control Protocol (SGCP).

Ingate Systems AB is a Swedish company that sells data network security and telecommunication equipment. The company primarily provides SIP Trunking of IP PBX:s on the US market. It is associated with sister company Intertex Data AB.

References

  1. "SIP trunking migration: Enterprise opportunities and challenges".
  2. "SIP Trunking Explained". Technology Convergence Group. Retrieved 8 September 2015.
  3. Gaboli, Ivan; Puglia, Virgilio (Jan 2011). "SIP Trunking the route to the new VoIP services". Kaleidoscope: Beyond the Internet? − Innovations for future networks and services, 2010 ITU-T, 13-15 Dec 2010. IEEE. ISBN   978-1-4244-8272-6.
  4. "SIP trunking explained". 2014-07-30.
  5. "Legal issues in different countries".
  6. "SIP trunking".
  7. "VoIP Office".
  8. "Role of Border Element". Cisco.
  9. "Acme Packet Net-Net session border controllers" (PDF). Acme Packet. Archived from the original (PDF) on 2011-07-17.
  10. "SIP Trunking Enterprise Solutions". Ingate Systems. Archived from the original on 2013-07-22.
  11. "Vitel Global Communications". Vitelglobal. Retrieved 2022-07-08.