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Glossary of Technical Terms

VoP | Protocol Stacks | Mobile Networks | Other Stacks | Support Acronyms

VoP

H.323
H.323 is an ITU standard for the usage of multimedia communication via packet-oriented networks that guarantees the interoperability between different equipment vendors. It defines how audiovisual conferencing data is transmitted across networks. The largest packet-oriented network is the Internet but also WAN, ISDN or dialup connections on which data is transported in packets (e.g. PPP) belong into this group. H.323 describes the general infrastructure and the utilization of different speech coders and protocol signaling stacks. The speech coders are defined in their respective sub standards, e.g. G.711 (Alaw and ulaw used in ISDN), G.722, G.723.1 and G.729.A for speech encoding. H.323 is definitely among the most widely deployed and mature standards.

IPv6 "Internet Protocol Version 6"( IPv6) is the "next generation" protocol designed by the IETF to replace the current version Internet Protocol, IP Version 4 ("IPv4"). IPv6 fixes a number of problems in IPv4, such as the limited number of available IPv4 addresses. It also adds many improvements to IPv4 in areas such as routing and network auto configuration.
IUA SIGTRAN ISDN User Adaptation (IUA) Layer provides functionality to transport ISDN signals from Switched Circuit Network (SCN) to IP network by back-hauling the Q.921 user messages over SCTP (Stream Control Transport Protocol) as per the IETF RFC 3057. IUA supports both ISDN Primary Rate access (PRA) as well as Basic Rate Access (BRA) including the support for both point-to-point and point-to-multi-point modes of communication

M2PA

MTP2 User Peer-to-Peer Adaptation Layer is a sigtran protocol for transporting SS7 MTP Level 2 user part signaling messages (i.e. MTP Level 3) over IP using the Stream Control Transmission Protocol (SCTP). Unlike M2UA, M2PA is used to support full MTP Level 3 message handling and network management between any two SS7 nodes communicating over an IP network. IP signaling points function as traditional SS7 nodes using the IP network instead of the SS7 network. Each switched circuit or IP signaling point has an SS7 point code. The M2PA protocol layer provides the same set of services as MTP Level 2 provides to MTP Level 3.
M2PA can be used between a signaling gateway and a media gateway controller, between a signaling gateway and an IP signaling point, and between two IP signaling points. Signaling points may use M2PA over IP or MTP Level 2 over standard SS7 links to send and receive MTP Level 3 messages.

M2UA

MTP2 User Adaptation Layer is a protocol defined by the IETF Sigtran Working Group for transporting SS7 MTP Level 2 user (i.e. MTP Level 3) signaling messages over IP using the Stream Control Transmission Protocol (SCTP). The M2UA protocol layer provides the equivalent set of services to its users as MTP Level 2 provides to MTP Level 3.
M2UA is used between the Signaling Gateway and Media Gateway Controller in VoIP networks. The signaling gateway receives SS7 messages over an MTP Level 1 and Level 2 interface from a signaling end point (SCP or SSP) or signal transfer point (STP) in the public switched telephone networks. The signaling gateway terminates the SS7 link at MTP Level 2 and transports MTP Level 3 and above to a Media Gateway Controller or other IP endpoint using M2UA over SCTP/IP. The signaling gateway maintains the Availability State of all media gateway controllers to manage signaling traffic flows across active SCTP associations.

M3UA M3UA is used for the transport of Signaling System 7 (SS7) MTP 3 user signaling messages over IP using the Stream Control Transmission Protocol (SCTP). Its uses include Voice over IP (VoIP) gateways and Third Generation wireless infrastructure (for the Universal Mobile Telecommunications System (UMTS). The M3UA protocol carries SS7 messages over IP, connecting call control and carrier services in the IP network to the existing SS7 network. Moreover, M3UA enables a single SS7/IP signaling gateway to converge multiple IP applications to appear as a single SS7 network element, thereby simplifying the network and ensuring the scalability, performance and reliability of the IP applications.

Media Gateway Controller Media gateway controllers (MGCs) or "softswitches" are the foundation for next-generation networks - offering the intelligence and reliability of the circuit-switched network with the speed and economy of the packet-switched network. As such, they must economically support both existing voice services like CLASS 4/5 features and evolving applications such as integrated access, VPN, Internet call waiting, click to dial, unified messaging, and enhanced roaming.

MEGACO MEGACO was developed by the ITU-T SG16 and IETF WG MEGACO and defines the control protocol between the media gateway controller (call agent) and the media gateway. The protocol provides - Control for various types of terminations, Support for negotiation of call capabilities, Multi user call scenarios, Rich termination dynamics, Quality of Service (QoS)and traffic measurement support, error reporting on protocol, call and capability and network failures.

Megaco based Media Gateway Toolkit A higher value solution is offered through the Media Gateway (MG) Toolkit that hides the syntactic and semantic details of protocol processing from the application. It provides the complete protocol processing function for different gateway control protocols and a generic interface to all other Media Gateway functions.

MGCP The Media Gateway Control Protocol (MGCP) is a complementary protocol to H.323 and SIP, designed for controlling media gateways from external call control elements in decomposed gateway architectures. Working in conjunction with the Gateway Location Protocol (GLP), it enables a caller with a PSTN phone number to locate the destination device and establish a session. It provides the gateway-to-gateway interface for the Session Initialization Protocol (SIP).
The MGCP is meant to simplify standards for the new Voice over Packet technology by eliminating the need for complex, processor-intense IP telephony devices, thus simplifying and lowering the cost of these terminals.

microSIP microSIP has been designed to meet the specific requirement of light weight protocol stack essential for user controlled terminals such as mobile phones, Personal Digital Assistants (PDAs) etc. The complete microSIP stack has a minimal memory footprint that can be further reduced depending on specific needs. At the same time, microSIP does not compromise on the features of an industry grade stack.

POTS Plain Old Telephone System is the collection of interconnected systems operated by the various telephone companies and administrations (telcos and PTTs) around the world. The PSTN started as human-operated analogue circuit switching systems (plugboards), progressed through electromechanical switches. By now this has almost completely been made digital, except for the final connection to the subscriber (the "last mile").

RTP/RTCP RTP/RTCP Stack complies with RFC 1889 and 1890. The stack provides application writers the flexibility to add new profiles to the stack. RTP/RTCP Stack can be used to build applications such as IP Phones, Gateways, Midcom Media Proxies and ALGs, Mixers, IVRs etc.

SCTP

The Stream Control Transmission Protocol is a reliable transport protocol operating on top of a connectionless packet network such as IP. SCTP is designed to transport PSTN signaling messages over IP networks, but is capable of broader applications.

It offers the following services to its users:

  • Acknowledged error-free non-duplicated transfer of user data.
  • Data fragmentation to conform to discovered path MTU size.
  • Sequenced delivery of user messages within multiple streams, with an option for order-of-arrival delivery of individual user messages.
  • Optional bundling of multiple-user messages into a single SCTP packet.
  • Network-level fault tolerance through supporting of multi-homing at either or both ends of an association.
SIGTRAN SIGTRAN was developed by the IETF working group SIGTRAN and defines the control protocol between the Signaling Gateway, Media Gateway Controllers and IP based Signaling Points. The Sigtran protocol consists of a modular, extensible structure with a common reliable transport protocol used for all signaling transport. The protocol transports message based signaling protocols messages, usually SS7, transparently over IP networks.

SIP An Internet Engineering Task Force (IETF) standard, SIP is an open, Internet-genuine protocol for establishing and managing multi-party, mixed-media sessions over converged networks. SIP enables the creation and deployment of feature-rich services that go far beyond simple VoIP calls.

SIP Server

The Session Initiation Protocol (SIP) is an application-layer control protocol. A signaling protocol for Internet telephony that allows for creation of innovative new services for IP Telephony networks. It allows packet-based networks to carry voice, video, and data. SIP can establish sessions for features such as audio/videoconferencing, interactive gaming, and call forwarding to be deployed over IP networks thus enabling service providers to integrate basic IP telephony services with Web, email, and chat services.

In addition to user authentication, redirect and registration services, the SIP Server supports traditional telephony features such as personal mobility, time-of-day routing and call forwarding based on the geographical location of the person being called.

SIP UA Toolkit A User Agent (UA) is a logical entity in a SIP network that initiates or responds to SIP requests. Almost every element of a SIP network exhibits a UA behavior either as a UA-Client by initiating SIP requests or as a UA-Server responding to requests or both. The UA thus forms the basic building block for all elements in a SIP network.
Softswitch A softswitch is a programmable network switch that processes signals for all kinds of packet protocols. Also known as a "Media Gateway Controller", "Call Server" or "Call Agent". It is a software that resides on fault-tolerant servers and performs call control functions such as protocol conversion, authorization, accounting and administration operations. Traditionally, voice-switching applications have been proprietary software running on Class 4 and Class 5 hardware. That has typically meant big waits for service providers that wanted new applications to differentiate their services. In a softswitch, solutions can come from multiple vendors, at all levels who supply open standards-based products and customers are free to choose best-in-class products to build their network. Open standards enable innovation and reduce costs.

SUA "SCCP User Adaptation" The Sigtran SUA layer defines the transport of SCCP-user messages (MAP & CAP over TCAP, RANAP, etc.) and new third generation network protocol messages over IP between two signaling endpoints using the Stream Control Transport Protocol (SCTP). Provision is made for protocol elements that enable a seamless, or as seamless as possible, operation of the SCCP User Protocol peers in the SS7 and IP domains. This protocol would also be used between any two signaling endpoints wholly contained within an IP network. It is the responsibility of SUA for triggering SCTP to initiate/shutdown an SCTP association. SUA also chooses the SCTP streams on which to send the data grams to the peer.

Protocol Stacks
SS7 Stacks Signaling System Number 7 (SS7) is the international digital signaling standard based on Common Channel Signaling (CCS) between Central Office switches worldwide.

Signaling System No. 7 (SS7) is a global standard for telecommunications defined by the International Telecommunication Union (ITU) Telecommunication Standardization Sector (ITU-T). The standard defines the procedures and protocol by which network elements in the public switched telephone network (PSTN) exchange information over a digital signaling network to effect wireless (cellular) and wireline call setup, routing and control. It is an industry proven protocol stack for building telecom equipment, network components as well as SS7 based applications.

The SS7 network and protocol are used for:

  • basic call setup, management, and tear down
  • wireless services such as personal communications services (PCS), wireless roaming, and mobile subscriber authentication
  • enhanced call features such as call forwarding, calling party name/number display, and three-way calling
  • efficient and secure worldwide telecommunications

Message Transfer Part (MTP)

The Message Transfer Part (MTP) is divided into three levels. The lowest level, MTP Level 1, is equivalent to the OSI Physical Layer. MTP Level 1 defines the physical, electrical, and functional characteristics of the digital signaling link. Physical interfaces defined include E-1 (2048 kb/s; 32 64 kb/s channels), DS-1 (1544 kb/s; 24 64kb/s channels), V.35 (64 kb/s), DS-0 (64 kb/s), and DS-0A (56 kb/s).

MTP Level 2 ensures accurate end-to-end transmission of a message across a signaling link. Level 2 implements flow control, message sequence validation, and error checking. When an error occurs on a signaling link, the message (or set of messages) is retransmitted. MTP Level 2 is equivalent to the OSI Data Link Layer.

MTP Level 3 provides message routing between signaling points in the SS7 network. MTP Level 3 re-routes traffic away from failed links and signaling points and controls traffic when congestion occurs. MTP Level 3 is equivalent to the OSI Network Layer.


ISDN User Part (ISUP)


The ISDN User Part (ISUP) defines the protocol used to set-up, manage, and release trunk circuits that carry voice and data between terminating line exchanges (e.g., between a calling party and a called party). ISUP is used for both ISDN and non-ISDN calls. However, calls that originate and terminate at the same switch do not use ISUP signaling.


Telephone User Part (TUP)


In some parts of the world (e.g., China, Brazil), the Telephone User Part (TUP) is used to support basic call setup and tear-down. TUP handles analog circuits only. In many countries, ISUP has replaced TUP for call management.


Signaling Connection Control Part (SCCP)


SCCP provides connectionless and connection-oriented network services and global title translation (GTT) capabilities above MTP Level 3. The primary difference between MTP and SCCP is in the addressing scheme and routing. SCCP is used as the transport layer for TCAP-based services.


Transaction Capabilities Applications Part (TCAP)


TCAP supports the exchange of non-circuit related data between applications across the SS7 network using the SCCP connectionless service. Queries and responses sent between SSPs and SCPs are carried in TCAP messages. For example, an SSP sends a TCAP query to determine the routing number associated with a dialed 800/888 number and to to check the personal identification number (PIN) of a calling card user. In mobile networks (IS-41 and GSM), TCAP carries Mobile Application Part (MAP) messages sent between mobile switches and databases to support user authentication, equipment identification, and roaming. CAP and INAP also use TCAP as the underlying layer.

MAP (Mobile Application Part)

The totality of a mobile application system (sometimes also called a wireless application) is often referred to as a Mobile Application Platform, and consists of two or more Mobile Application Parts (MAP). It enables signaling between different network entities. Some MAPs have specific data definition, transaction definition, overall output and input definition, and processing rules. The MAP stack is built on top of the Transaction Capabilities Application Part (TCAP) layer, which is the top most layer of the Signaling System Number 7 (SS7) stack.

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Mobile Networks
ALCAP The Access Link Control Application Part (Q.2630.1 / Q.2630.2) is the signaling protocol for control of the AAL2 (ATM Adaptation Layer 2) ATM (Asynchronous Transfer Mechanism) links between the 3G Radio Network Controller, 3G NodeB and 3G Mobile Switching Center.

CAP CAP is an important component of mobility networks used to implement CAMEL (Customized Applications for Mobile Network Enhanced Logic). CAMEL is an extension of Intelligent Network (IN) concept in GSM. However, an important difference between the two is that CAMEL can provide mechanisms to support services consistently independently of the serving network, thus enabling the user to experience uniform services even when she is outside the home network. Just like MAP, the CAP stack is built on top of the Transaction Capabilities Application Part (TCAP) layer.

Framing Protocols The Framing Protocols provide timing synchronization between different 3G Network Elements. The Framing Protocols are FP Iu-UP between the 3G Radio Network Controller and 3G Mobile Switching Center, FP Iub between the 3G Radio Network Controller and 3G Node B, and FP Iur between 3G Radio Network Controllers.

GTP-U The GPRS Tunneling Protocol – User Plane is used to transmit user data between the 3G Radio Network Controller and the 3G Serving GPRS Support Node.

GPRS (General Packet Radio Service) The General Packet Radio Service is a value-added service that allows both voice and high speed Internet data to be exchanged across a cellular network. GPRS enables instant high-speed data communication by which users can remain on-line and pay only for data actually exchanged. Hence GPRS facilitates the provisioning of customer services such as Internet browsing, mobile e-mail, multimedia communication and location-based services.

Gateway GPRS Support Node (GGSN)

GGSN is a GPRS support node which acts as a gateway between the GPRS network and packet switched public data network (PSPDN).

GPRS Tunnelling Protocol (GTP); Tunnel Protocol

GTP is a General Packet Radio Service (GPRS) protocol used in transmitting user data packets and signalling between GPRS support nodes (GSN) over the GPRS backbone network.

GSN(GPRS Support Node) A GPRS Support Node is an infrastructure node added to PLMNs to route and deliver packet data between mobile stations and external IP/X.25 packet data networks.
IPoA The Internet Protocol over ATM is the building block used to provide IP messaging over ATM interfaces in the 3G Radio Network Controller.

IuCS The circuit switched (CS) interface between the 3G Radio Network Controller and the 3G Mobile Switching Center. This consists of several protocols and underlying ATM hardware with an integrated management entity.

IuPS The packet switched (PS) interface between the 3G Radio Network Controller and the 3G Serving GPRS Support Node. This consists of several protocols and underlying ATM hardware with an integrated management entity.

Iub The interface between the 3G Radio Network Controller and the 3G NodeB. This consists of several protocols and underlying ATM hardware with an integrated management entity.
Iur The interface from between two 3G Radio Network Controllers. This consists of several protocols and underlying ATM hardware with an integrated management entity.

MAP The GSM / 3G PP compliant MAP Stack enables signaling between different network entities such as HLR, MSC/VLR, SGSN and GGSN. This stack is built on top of the Transaction Capabilities Application Part (TCAP) layer, which is the top most layer of the Signaling System Number 7 (SS7) stack.
NBAP The NodeB Application Protocol coordinates the functions of the NodeB between the 3G NodeB and the 3G Radio Network Controller.

PDCP The Packet Data Convergence Protocol enables the 3G radio interface to provide TCP/IP communication between the 3G Radio Network Controller and the 3G User Equipment.
PLMN(Public Land Mobile Network)

A Public Land Mobile Network is the full set of systems such as switches, transmission towers, etc. that enable a cellular network service to be provided over a given area.

RANAP The Radio Access Network Application Part manages the communication between the 3G Radio Network Controller, the 3G Serving GPRS Support Node and the 3G Mobile Switching Center.

RLC/MAC The Radio Link Control / Medium Access Control protocols manage and provide the data links for the radio interface between the 3G Radio Network Controller and the 3G NodeB.

RNSAP The Radio Network Service Application Part manages the handover of users’ calls between different 3G Radio Network Controllers, i.e. between different cell sites.

RRC The Radio Resource Control manages the control of the radio resources from the 3G Radio Resource Controller and the broadcast channel system information block from the 3G NodeB.
Radio Access Network Application Part (RANAP) A radio access network (RAN) signalling protocol that consists of mechanisms that handle all the procedures between the core network and radio access network.

SAAL-NNI The Signaling ATM Adaptation Layer – Network-Node Interface is used to provide reliable signaling between the ATM end points on the 3G Radio Network Controller, the 3G Serving GPRS Support Node and the 3G Mobile Switching Center, i.e. on network-side nodes.

SAAL-UNI The Signaling ATM Adaptation Layer – User-Network Interface is used to provide reliable signaling between the ATM end points on the 3G Radio Network Controller and the 3G NodeB, i.e. on user-side network nodes.

SCCP The Signaling Connection Control Part protocol provides connectionless and connection-oriented network services above MTP3 / MTP3-B, as used in the 3G Radio Network Controller, the 3G Serving GPRS Support Node and the 3G Mobile Switching Center.

Service Creation Environment (SCE) The Service Creation Environment is an integrated, graphical programming Environment that offers a suite of software tools to help a service designer plan, develop, test, deploy and maintain services in a fast, simple and flexible manner.

Serving GPRS Support Node (SGSN)

SGSN is a General Packet Radio Service Support (GPRS) support node that serves GPRS mobile by sending or receiving packets via a base station (BS) subsystem.

STC The Signaling Transport Converter (Q.2150.1 / Q.2150.2) is used to exchange ATM signaling messages with peer signaling entities and to receive info about the conditions of the signaling network.

Uu The interface between the 3G Radio Network Controller and the 3G User Equipment, i.e. the user’s mobile phone, PDA, laptop, etc. This consists of several protocols and underlying ATM hardware with an integrated management entity.

UMTS(Universal Mobile Telecommunication System) The Universal Mobile Telecommunication System is a value-added service that (in addition to voice) allows Internet data with highly flexible data rates and service qualities to be exchanged across cellular networks built using different technologies. UMTS enables a greater variety of cost-effective data services to be provided to users, and enables cellular networks based on different underlying technologies to be interoperable with each other.
Other Stacks

V5.2 LE/AN

The V5.2 Protocol defines the switching and signaling protocol between Access Network and the Local Exchange for the support of Analog Telephone Access, ISDN (BRI and PRI) Access and Other Digital or Analog access for Semi-permanent connections.

Defined by the European Telecommunications Standards Institute (ETSI), the recently adopted ETS V5.1/V5.2 standards define a set of universal signaling and switching characteristics for communications between a telecom Access Network (AN) and a digital Local Exchange (LE). ETS V5.1/V5.2 identifies a full range of procedural and protocol interface requirements, allowing vendors to develop non-proprietary, standards-based local access and local exchange equipment. The ultimate goal of ETS V5.1/V5.2 is to ensure connectivity and compatibility between competing vendors, giving operators greater choice when purchasing AN and LE equipment.

Broadband V5 (VB5)

VB5.x is a set of ETSI standards used to define the interconnection of local exchanges over broadband networks. This provides wireless, fiber, and copper access networks a common interface to the Access Node (AN) or Service Node (SN). VB5 Interface specifies the physical, procedural and protocol requirements for interfaces at the reference point between Access Network (AN) and Service Node (SN).

GR 303

GR-303 is an ISDN protocol based protocol for communication between the core and access network. It defines a set of requirements for Next Generation Integrated Digital Loop Carrier (NG-IDLC) systems that includes open interfaces for mix-and-match of Local Digital Switches (LDS) with Remote Digital Terminals (RDT).

GR-303 unbundles the interface between access equipment and Local Digital Switch (LDS) providers allowing Service Providers to mix and match equipment and facilities to meet their needs. GR-303's flexible concentration of bandwidth into the LDS for analog and ISDN services enables Service Providers to service additional customers with existing analog and digital media. GR-303 runs on dedicated redundant control channels between the Integrated Digital Terminal (IDT) within the LDS and the access equipment (Remote Digital Terminal (RDT)).




Last updated : February 2, 2004

 

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