Client examples
Examples of SCCP client devices include the Cisco 7900 series of IP phones, Cisco IP Communicator softphone, and the 802.11b wireless Wireless IP Phone 7920, along with Cisco Unity voicemail server.
The Skinny Client Control Protocol (SCCP) is a proprietary network terminal control protocol originally developed by Selsius Systems, which was acquired by Cisco Systems in 1998.
SCCP is a lightweight IP-based protocol for session signaling with Cisco Unified Communications Manager, formerly named CallManager. [1] [2] The protocol architecture is similar to the media gateway control protocol architecture, in that is decomposes the function of media conversion in telecommunication for transmission via an Internet Protocol network into a relatively low-intelligence customer-premises device and a call agent implementation that controls the CPE via signaling commands. The call agent product is Cisco CallManager, which also performs as a signaling proxy for call events initiated over other common protocols such as H.323, and Session Initiation Protocol (SIP) for voice over IP, or ISDN for the public switched telephone network.
An SCCP client uses TCP/IP to communicate with one or more Call Manager applications in a cluster. It uses the Real-time Transport Protocol (RTP) over UDP-transport for the bearer traffic (real-time audio stream) with other Skinny clients or an H.323 terminal. SCCP is a stimulus-based protocol and is designed as a communications protocol for hardware endpoints and other embedded systems, with significant CPU and memory constraints.
Some Cisco analog media gateways, such as the VG248 gateway, register and communicate with Cisco Unified Communications Manager using SCCP.
Cisco acquired SCCP technology when it acquired Selsius Corporation in 1998. [3] For this reason the protocol is also referred to in Cisco documentation as the Selsius Skinny Station Protocol. Another remnant of the origin of the Cisco IP phones is the default device name format for registered Cisco phones with CallManager. It is SEP, as in Selsius Ethernet Phone, followed by the MAC address. Cisco also has marketed a Skinny-based softphone called Cisco IP Communicator.
Examples of SCCP client devices include the Cisco 7900 series of IP phones, Cisco IP Communicator softphone, and the 802.11b wireless Wireless IP Phone 7920, along with Cisco Unity voicemail server.
Other companies, such as Symbol Technologies, SocketIP, and Digium, have implemented the protocol in VoIP terminals and IP phones, media gateway controllers, and softswitches. An open source implementation of a call agent is available in the Asterisk and FreeSWITCH systems. [4] IPBlue provides a soft phone that emulates a Cisco 7960 telephone. [5] Twinlights Software distributes a soft phone implementation for Android-based devices. [6] The Cisco Unified Application Environment, the product acquired by Cisco when they purchased Metreos, supports using SCCP to emulate Cisco 7960 phones allowing applications to access all Cisco line-side features.
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).
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.
Systems Network Architecture (SNA) is IBM's proprietary networking architecture, created in 1974. It is a complete protocol stack for interconnecting computers and their resources. SNA describes formats and protocols but, in itself, is not a piece of software. The implementation of SNA takes the form of various communications packages, most notably Virtual Telecommunications Access Method (VTAM), the mainframe software package for SNA communications.
A softswitch is a call-switching node in a telecommunications network, based not on the specialized switching hardware of the traditional telephone exchange, but implemented in software running on a general-purpose computing platform. Like its traditional counterparts it connects telephone calls between subscribers or other switching systems across a telecommunication network. Often a softswitch is implemented to switch calls using voice over IP (VoIP) technologies, but hybrid systems exist.
An analog telephone adapter (ATA) or FXS gateway, is a device for connecting traditional analog telephones, fax machines, and similar customer-premises devices to a digital telephone system or a voice over IP telephony network.
Local Area Transport (LAT) is a non-routable networking technology developed by Digital Equipment Corporation to provide connection between the DECserver terminal servers and Digital's VAX and Alpha and MIPS host computers via Ethernet, giving communication between those hosts and serial devices such as video terminals and printers. The protocol itself was designed in such a manner as to maximize packet efficiency over Ethernet by bundling multiple characters from multiple ports into a single packet for Ethernet transport.
Skype for Business Server is real-time communications server software that provides the infrastructure for enterprise instant messaging, presence, VoIP, ad hoc and structured conferences and PSTN connectivity through a third-party gateway or SIP trunk. These features are available within an organization, between organizations and with external users on the public internet or standard phones.
A signaling protocol is a type of communications protocol for encapsulating the signaling between communication endpoints and switching systems to establish or terminate a connection and to identify the state of connection.
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).
Selsius Systems was a telecommunications company in Dallas, Texas that developed and marketed some of the first IP telephony gear. It was co-founded by Richard Platt and David Tucker. Selsius was incorporated in 1997 and acquired by Cisco Systems in November, 1998.
ZRTP is a cryptographic key-agreement protocol to negotiate the keys for encryption between two end points in a Voice over IP (VoIP) phone telephony call based on the Real-time Transport Protocol. It uses Diffie–Hellman key exchange and the Secure Real-time Transport Protocol (SRTP) for encryption. ZRTP was developed by Phil Zimmermann, with help from Bryce Wilcox-O'Hearn, Colin Plumb, Jon Callas and Alan Johnston and was submitted to the Internet Engineering Task Force (IETF) by Zimmermann, Callas and Johnston on March 5, 2006 and published on April 11, 2011 as RFC 6189.
This is a comparison of voice over IP (VoIP) software used to conduct telephone-like voice conversations across Internet Protocol (IP) based networks. For residential markets, voice over IP phone service is often cheaper than traditional public switched telephone network (PSTN) service and can remove geographic restrictions to telephone numbers, e.g., have a PSTN phone number in a New York area code ring in Tokyo.
Mobile VoIP or simply mVoIP is an extension of mobility to a voice over IP network. Two types of communication are generally supported: cordless telephones using DECT or PCS protocols for short range or campus communications where all base stations are linked into the same LAN, and wider area communications using 3G or 4G protocols.
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 Media Gateway Control Protocol (MGCP) is a telecommunication protocol for signaling and call control in hybrid voice over IP (VoIP) and traditional telecommunication systems. It implements the media gateway control protocol architecture for controlling media gateways connected to the public switched telephone network (PSTN). The media gateways provide conversion of traditional electronic media to the Internet Protocol (IP) network. The protocol is a successor to the Simple Gateway Control Protocol (SGCP), which was developed by Bellcore and Cisco, and the Internet Protocol Device Control (IPDC).
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.
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).