| 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.
|
|
|
| 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 users
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)).
|
|
|