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N g n study guide
NEXT GENERATION NETWORK

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Softswitch
A softswitch is a central device in a telephone network which connects calls from one phone line to another, entirely by means of software running on a computer system. This work was formerly carried out by hardware, with physical switchboards to route the calls. A softswitch is typically used to control connections at the junction point between circuit and packet networks. A single device containing both the switching logic and the switching fabric can be used for this purpose; however, modern technology has led to a preference for decomposing this device into a Call Agent and a Media Gateway. The Call Agent takes care of functions like billing, call routing, signalling, call services and so on and is the 'brains' of the outfit. A Call Agent may control several different Media Gateways in geographically dispersed areas over a TCP/IP link. The Media Gateway connects different types of digital media stream together to create an end-to-end path for the media (voice and data) in the call. It may have interfaces to connect to traditional PSTN networks like DS1 or DS3 ports (E1 or STM1 in the case of non-US networks), it may have interfaces to connect to ATM and IP networks and in the modern system will have Ethernet interfaces to connect VoIP calls. The call agent will instruct the media gateway to connect media streams between these interfaces to connect the call - all transparently to the end-users. The softswitch generally resides in a building owned by the telephone company called a central office. The central office will have telephone trunks to carry calls to other offices owned by the telephone company and to other telephone companies (aka the Public Switched Telephone Network or PSTN). Looking towards the end users from the switch, the Media Gateway may be connected to several access devices. These access devices can range from small Analog Telephone Adaptors (ATA) which provide just one RJ11 telephone jack to an Integrated Access Device (IAD) or PBX which may provide several hundred telephone connections.

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Typically the larger access devices will be located in a building owned by the telephone company near to the customers they serve. Each end user can be connected to the IAD by a simple pair of copper wires. The medium sized devices and PBXs will typically be used in a business premises and the single line devices would probably be found in residential premises. In more recent times (i.e., the IP Multimedia Subsystem or IMS), the Softswitch element is represented by the Media Gateway Controller (MGC) element, and the term "Softswitch" is rarely used in the IMS context.

Feature server as a part of softswitch
The feature server, often built into a call agent/softswitch, is the functional component that provides call-related features. Capabilities such as call forwarding, call waiting, and last call return, if implemented in the network, are implemented in the feature server. The feature server works closely with the call agent, and may call upon the media server to provide these services. These features do not require the subscriber to explicitly request them but tend to be triggered within the call handling logic. An example of a feature service is last call return, in which the user picks up the phone, dials *69, and hears, “The number that last called you was xxxxxx-xxxx. Press 1 to return this call.” When the call agent sees the dial string *69, it triggers an invocation of the feature server function. The feature server examines its database, finds the user and the caller identification of the last call, then asks the media server to play the announcement and collect a digit. When the media server returns a “1”, the feature server instructs the call agent to establish a call between the user and the party that last called that user

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Media Gateway Control Protocol
In computing, Media Gateway Control Protocol (MGCP) is a signaling and call control protocol used within a distributed Voice over IP system. It superseded the Simple Gateway Control Protocol (SGCP). Another protocol for the same purpose is Megaco, a co-production of IETF and ITU (Recommendation H.248-1). Both protocols follow the guidelines of the API Media Gateway Control Protocol Architecture and

Architecture
The distributed system is composed of a Call Agent (or Media Gateway Controller), at least one Media Gateway (MG) that performs the conversion of media signals between circuits and packets, and at least one Signaling gateway (SG) when connected to the PSTN. The Call Agent uses MGCP to tell the Media Gateway: “what events should be reported to the Call Agent how endpoints should be connected together what signals should be played on endpoints.” MGCP also allows the Call Agent to audit the current state of endpoints on a Media Gateway. The Media Gateway uses MGCP to report events (such as off-hook, or dialed digits) to the Call Agent. (While any Signaling Gateway is usually on the same physical switch as a Media Gateway, this needn't be so. The Call Agent does not use MGCP to control the Signaling Gateway; rather, SIGTRAN protocols are used to backhaul signaling between the Signaling Gateway and Call Agent). In MGCP, every command has a transaction ID and receives a response.

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Typically, a Media Gateway is configured with a list of Call Agents from which it may accept programming (where that list normally comprises only one or two Call Agents). In principle, event notifications may be sent to different Call Agents for each endpoint on the gateway (as programmed by the Call Agents, by setting the Notified Entity parameter). In practice however, it is usually desirable that at any given moment all endpoints on a gateway should be controlled by the same Call Agent; other Call Agents are available only to provide redundancy in the event that the primary Call Agent fails, or loses contact with the Media Gateway. In the event of such a failure it is the backup Call Agent's responsibility to reprogram the MG so that the gateway comes under the control of the backup Call Agent. Care is needed in such cases; two Call Agents may know that they have lost contact with one another, but this does not guarantee that they are not both attempting to control the same gateway. The ability to audit the gateway to determine which Call Agent is currently controlling can be used to resolve such conflicts. MGCP assumes that the multiple Call Agents will maintain knowledge of device state among themselves (presumably with an unspecified protocol) or rebuild it if necessary (in the face of catastrophic failure). Its failover features take into account both planned and unplanned outages.

Protocol Overview
MGCP packets are unlike what you find in many other protocols. Usually wrapped in UDP port 2427, the MGCP datagrams are formatted with whitespace, much like you would expect to find in TCP protocols. An MGCP packet is either a command or a response. Commands begin with a four-letter verb. Responses begin with a three number response code. There are eight (8) command verbs:
AUEP, AUCX, CRCX, DLCX, MDCX, NTFY, RQNT, RSIP

Two verbs are used by a Call Agent to query (the state of) a Media Gateway:
AUEP - Audit Endpoint

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AUCX - Audit Connection

Three verbs are used by a Call Agent to manage an RTP connection on a Media Gateway (a Media Gateway can also send a DLCX when it needs to delete a connection for its self-management):
CRCX - Create Connection DLCX - Delete Connection MDCX - Modify Connection

One verb is used by a Call Agent to request notification of events on the Media Gateway, and to request a Media Gateway to apply signals:
RQNT - Request for Notification

One verb is used by a Media Gateway to indicate to the Call Agent that it has detected an event for which the Call Agent had previously requested notification of (via the RQNT command verb):
NTFY - Notify

One verb is used by a Media Gateway to indicate to the Call Agent that it is in the process of restarting:
RSIP - Restart In Progress

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H.323
H.323 is an umbrella Recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audiovisual communication sessions on any packet network. It is widely implemented by voice and videoconferencing equipment manufacturers, is used within various Internet real-time applications such as GnuGK, NetMeeting and X-Meeting, and is widely deployed worldwide by service providers and enterprises for both voice and video services over Internet Protocol (IP) networks. It is a part of the ITU-T H.32x series of protocols, which also address multimedia communications over Integrated Services Digital Network (ISDN), Public Switched Telephone Network (PSTN) or Signaling System 7 (SS7), and 3G mobile networks. H.323 Call Signaling is based on the ITU-T Recommendation Q.931 protocol and is suited for transmitting calls across networks using a mixture of IP, PSTN, ISDN, and QSIG over ISDN. A call model, similar to the ISDN call model, eases the introduction of IP telephony into existing networks of ISDNbased PBX systems, including transitions to IP-based Private Branch eXchanges (PBXs). Within the context of H.323, an IP-based PBX might be an H.323 Gatekeeper or other call control element that provides service to telephones or videophones. Such a device may provide or facilitate both basic services and supplementary services, such as call transfer, park, pick-up, and hold. While H.323 excels at providing basic telephony functionality and interoperability, H.323’s strength lies in multimedia communication functionality designed specifically for IP networks.

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Contents 1 History 2 Protocols 3 Codecs 4 H.323 Architecture 4.1 H.323 Network Elements 4.1.1 Terminals 4.1.2 Multipoint Control Units 4.1.3 Gateways 4.1.4 Gatekeepers 4.1.5 Border Elements and Peer Elements 4.2 H.323 Network Signaling 4.2.1 H.225.0 Call Signaling 4.2.2 RAS Signaling 4.2.3 H.245 Call Control 4.2.3.1 Capability Negotiation 4.2.3.2 Master/Slave Determination 4.2.3.3 Logical Channel Signaling 4.2.3.4 Fast Connect 5 Use cases 5.1 H.323 and Voice over IP services 5.2 H.323 and Videoconference services

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History
The first version of H.323 was published by the ITU in November 1996 with an emphasis of enabling videoconferencing capabilities over a Local Area Network (LAN), but was quickly adopted by the industry as a means of transmitting voice communication over a variety of IP networks, including WANs and the Internet (see VoIP). Over the years, H.323 has been revised and re-published with enhancements necessary to better-enable both voice and video functionality over Packetswitched networks, with each version being backward-compatible with the previous version. Recognizing that H.323 was being used for communication, not only on LANs, but over WANs and within large carrier networks, the title of H.323 was changed when published in 1998. The title, which has since remained unchanged, is "Packet-Based Multimedia Communications Systems." The current version of H.323, commonly referred to as "H.323v6", was published in 2006. One strength of H.323 was the relatively early availability of a set of standards, not only defining the basic call model, but also the supplementary services needed to address business communication expectations. H.323 was the first VoIP standard to adopt the Internet Engineering Task Force (IETF) standard Real-time Transport Protocol (RTP) to transport audio and video over IP networks.

Protocols
H.323 is a system specification that describes the use of several ITU-T and IETF protocols. The protocols that comprise the core of almost any H.323 system are: H.225.0 Registration, Admission and Status (RAS), which is used between an H.323 endpoint and a Gatekeeper to provide address resolution and admission control services. H.225.0 Call Signaling, which is used between any two H.323 entities in order to establish communication.

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H.245 control protocol for multimedia communication, which describes the messages and procedures used for capability exchange, opening and closing logical channels for audio, video and data, control and indications. Real-time Transport Protocol (RTP), which is used for sending or receive multimedia information (voice, video, or text) between any two entities. Many H.323 systems also implement other protocols that are defined in various ITU-T Recommendations in order to provide supplementary services support or deliver other functionality to the user. Some of those Recommendations are: H.235 series describes security within H.323, including security for both signaling and media. H.239 describes dual stream use in videoconferencing, usually one for live video, the other for still images. H.450 series describes various supplementary services. H.460 series defines optional extensions that might be implemented by an endpoint or a Gatekeeper, including ITU-T Recommendations H.460.17, H.460.18, and H.460.19 for Network address translation (NAT) / Firewall (FW) traversal. In addition to those ITU-T Recommendations, H.323 utilizes various IETF Request for Comments for media transport and media packetization, including Real-time Transport Protocol (RTP).

Codecs
H.323 utilizes both ITU-defined codecs and codecs defined outside the ITU. Codecs that are widely implemented by H.323 equipment include: Video codecs: H.261, H.263, H.264 Audio codecs: G.711, G.729 (including G.729a), G.723.1, G.726 Text codecs: T.140

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H.323 Architecture
The H.323 system defines several network elements that work together in order to deliver rich multimedia communication capabilities. Those elements are Terminals, Multipoint Control Units (MCUs), Gateways, Gatekeepers, and Border Elements. Collectively, terminals, multipoint control units and gateways are often referred to as endpoints. While not all elements are required, at least two terminals are required in order to enable communication between two people. In most H.323 deployments, a gatekeeper is employed in order to, among other things, facilitate address resolution.

H.323 Network Elements
Terminals
Figure 1 - A complete, sophisticated protocol stack

Terminals in an H.323 network are the most fundamental elements in any H.323 system, as those are the devices that users would normally encounter. They might exist in the form of a simple IP phone or a powerful high-definition videoconferencing system.

Inside an H.323 terminal is something referred to as a "protocol stack," which implements the functionality defined by the H.323 system. The protocol stack would include an implementation of the basic protocol defined in ITU-T Recommendation H.225.0 and H.245, as well as RTP or other protocols described above.

The diagram, figure 1, depicts a complete, sophisticated stack that provides support for voice, video, and various forms of data communication. In reality,

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most H.323 systems do not implement such a wide array of capabilities, but the logical arrangement is useful in understanding the relationships.

Multipoint Control Units
A Multipoint Control Unit (MCU) is responsible for managing multipoint conferences and is comprised of two logical entities referred to as the Multipoint Controller (MC) and the Multipoint Processor (MP). In more practical terms, an MCU is a conference bridge not unlike the conference bridges used in the PSTN today. The most significant difference, however, is that H.323 MCUs might be capable of mixing or switching video, in addition to the normal audio mixing done by a traditional conference bridge. Some MCUs also provide multipoint data collaboration capabilities. What this means to the end user is that, by placing a video call into an H.323 MCU, the user might be able to see all of the other participants in the conference, not only hear their voices.

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Gateways
Gateways are devices that enable communication between H.323 networks and other networks, such as PSTN or ISDN networks. If one party in a conversation is utilizing a terminal that is not an H.323 terminal, then the call must pass through a gateway in order to enable both parties to communicate. Gateways are widely used today in order to enable the legacy PSTN phones to interconnect with the large, international H.323 networks that are presently deployed by services providers. Gateways are also used within the enterprise in order to enable enterprise IP phones to communicate through the service provider to users on the PSTN. Gateways are also used in order to enable videoconferencing devices based on H.320 and H.324 to communicate with H.323 systems. Most of the third generation (3G) mobile networks deployed today utilize the H.324 protocol and are able to communicate with H.323-based terminals in corporate networks through such gateway devices.

Gatekeepers
A Gatekeeper is an optional component in the H.323 network that provides a number of services to terminals, gateways, and MCU devices. Those services include endpoint registration, address resolution, admission control, user authentication, and so forth. Of the various functions performed by the gatekeeper, address resolution is the most important as it enables two endpoints to contact each other without either endpoint having to know the IP address of the other endpoint on. Gatekeepers may be designed to operate in one of two signaling modes, namely "direct routed" and "gatekeeper routed" mode. Direct routed mode is the most efficient and most widely deployed mode. In this mode, endpoints utilize the RAS protocol in order to learn the IP address of the remote endpoint and a call is established directly with the remote device. In the gatekeeper routed mode, call signaling always passes through the gatekeeper. While the latter requires the gatekeeper to have more processing power, it also gives the gatekeeper complete control over the call and the ability to provide supplementary services on behalf of the endpoints.

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H.323 endpoints use the RAS protocol to communicate with a gatekeeper. Likewise, gatekeepers use RAS to communicate with other gatekeepers. A collection of endpoints that are registered to a single Gatekeeper in H.323 is referred to as a “zone”. This collection of devices does not necessarily have to have an associated physical topology. Rather, a zone may be entirely logical and is arbitrarily defined by the network administrator. Gatekeepers have the ability to neighbor together so that call resolution can happen between zones. Neighboring facilitates the use of dial plans such as the Global Dialing Scheme. Dial plans facilitate “inter-zone” dialing so that two endpoints in separate zones can still communicate with each other.

Border Elements and Peer Elements
Figure 2 - An illustration of an administrative domain with border elements, peer elements, and gatekeepers

Border Elements and Peer Elements are optional entities similar to a Gatekeeper, but that do not manage endpoints directly and provide some services that are not described in the RAS protocol. The role of a border or peer element is understood via the definition of an "administrative domain". An administrative domain is the collection of all zones that are under the control of a single person or organization, such as a service provider. Within a service provider network there may be hundreds or thousands of gateway devices, telephones, video terminals, or other H.323 network elements. The service provider might arrange devices into "zones" that enable the service provider to best manage all of the devices under its control, such as logical arrangement by city. Taken together, all of the zones within the service provider network would appear to another service provider as an "administrative domain". The border element is a signaling entity that generally sits at the edge of the administrative domain and communicates with another administrative domain. This communication might include such things as access authorization information; call pricing information; or other important data necessary to enable communication between the two administrative domains. Peer elements are entities with the administrative domain that, more or less, help to propagate information learned from the border elements throughout

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the administrative domain. Such architecture is intended to enable large-scale deployments within carrier networks and to enable services such as clearinghouses. The diagram, figure 2, provides an illustration of an administrative domain with border elements, peer elements, and gatekeepers.

H.323 Network Signaling
H.323 is defined as a binary protocol, which allows for efficient message processing in network elements. The syntax of the protocol is defined in ASN.1 and uses the Packed Encoding Rules (PER) form of message encoding for efficient message encoding on the wire. Below is an overview of the various communication flows in H.323 systems.

H.225.0 Call Signaling
Once the address of the remote endpoint is resolved, the endpoint will use H.225.0 Call Signaling in order to establish communication with the remote entity. H.225.0 messages are: Setup and Setup acknowledge Call Proceeding Connect

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Alerting Information Release Complete Facility Progress Status and Status Inquiry Notify
Figure 3 - Establishment of an H.323 call

In the simplest form, an H.323 call may be established as follows (figure 3): In this example, the endpoint (EP) on the left initiated communication with the gateway on the right and the gateway connect the call with the called party. In reality, call flows are often more complex than the one shown, but most calls that utilize the Fast Connect procedures defined within H.323 can be established with as few as 2 or 3 messages. Endpoints must notify their gatekeeper (if gatekeepers are used) that they are in a call. Once a call has concluded, a device will send a Release Complete message. Endpoints are then required to notify their gatekeeper (if gatekeepers are used) that the call has ended.

RAS Signaling
Endpoints use the RAS protocol in order to communicate with a gatekeeper. Likewise, gatekeepers use RAS to communicate with peer gatekeepers. RAS is a fairly simple protocol comprised of just a few messages. Namely: Gatekeeper request, reject, and confirm messages (GRx) Registration request, reject, and confirm messages (RRx) Unregister request, reject, and confirm messages (URx) Admission request, reject, and confirm messages (ARx)

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Bandwidth request, reject, and confirm message (BRx) Disengage request, reject, and confirm (DRx) Location request, reject, and confirm messages (LRx) Info request, ack, nack, and response (IRx) Nonstandard message Unknown message response Request in progress (RIP) Resource availability indication and confirm (RAx) Service control indication and response (SCx) Admission confirm sequence (ACS)
Figure 4 - A high-level communication exchange between two endpoints (EP) and two gatekeepers (GK)

When an endpoint is powered on, it will generally send either a gatekeeper request (GRQ) message to "discover" gatekeepers that are willing to provide service or will send a registration request (RRQ) to a gatekeeper that is predefined in the system’s administrative setup. Gatekeepers will then respond with a gatekeeper confirm (GCF). If a GRQ has been sent the endpoint will then select a gatekeeper with which to register by sending a registration request (RRQ), to which the gatekeeper responds with a registration confirm (RCF). At this point, the endpoint is known to the network and can make and place calls. When an endpoint wishes to place a call, it will send an admission request (ARQ) to the gatekeeper. The gatekeeper will then resolve the address (either
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locally, by consulting another gatekeeper, or by querying some other network service) and return the address of the remote endpoint in the admission confirm message (ACF). The endpoint can then place the call. Upon receiving a call, a remote endpoint will also send an ARQ and receive an ACF in order to get permission to accept the incoming call. This is necessary, for example, to authenticate the calling device or to ensure that there is available bandwidth for the call. Figure 4 depicts a high-level communication exchange between two endpoints (EP) and two gatekeepers (GK).

H.245 Call Control
Once a call has initiated (but not necessarily fully connected) endpoints may initiate H.245 call control signaling in order to provide more extensive control over the conference. H.245 is a rather voluminous specification with many procedures that fully enable multipoint communication, though in practice most implementations only implement the minimum necessary in order to enable point-to-point voice and video communication. H.245 provides capabilities such as capability negotiation, master/slave determination, opening and closing of "logical channels" (i.e., audio and video flows), flow control, and conference control. It has support for both unicast and multicast communication, allowing the size of a conference to theoretically grow without bound. Capability Negotiation Of the functionality provided by H.245, capability negotiation is arguably the most important, as it enables devices to communicate without having prior knowledge of the capabilities of the remote entity. H.245 enables rich multimedia capabilities, including audio, video, text, and data communication. For transmission of audio, video, or text, H.323 devices utilize both ITUdefined codecs and codecs defined outside the ITU. Codecs that are widely implemented by H.323 equipment include: Video Codecs: H.261, H.263, H.264 Audio Codecs: G.711, G.729, G.729a, G.723.1, G.726 Text Codecs: T.140

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H.245 also enables real-time data conferencing capability through protocols like T.120. T.120-based applications generally operate in parallel with the H.323 system, but are integrated to provide the user with a seamless multimedia experience. T.120 provides such capabilities as application sharing T.128, electronic whiteboard T.126, file transfer T.127, and text chat T.134 within the context of the conference. When an H.323 device initiates communication with a remote H.323 device and when H.245 communication is established between the two entities, the Terminal Capability Set (TCS) message is the first message transmitted to the other side. Master/Slave Determination After sending a TCS message, H.323 entities (through H.245 exchanges) will attempt to determine which device is the "master" and which is the "slave." This process, referred to as master/slave determination, is important, as the master in a call settles all "disputes" between the two devices. For example, if both endpoints attempt to open incompatible media flows, it is the master who takes the action to reject the incompatible flow. Logical Channel Signaling Once capabilities are exchanged and master/slave determination steps have completed, devices may then open "logical channels" or media flows. This is done by simply sending an Open Logical Channel (OLC) message and receiving an acknowledgement message. Upon receipt of the acknowledgement message, an endpoint may then transmit audio or video to the remote endpoint.

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Fast Connect
Figure 5 - A typical H.245 exchange

A typical H.245 exchange looks similar to figure 5: After this exchange of messages, the two endpoints (EP) in this figure would be transmitting audio in each direction. The number of message exchanges is numerous, each has an important purpose, but nonetheless takes time. For this reason, H.323 version 2 (published in 1998) introduced a concept called Fast Connect, which enables a device to establish bi-directional media flows as part of the H.225.0 call establishment procedures. With Fast Connect, it is possible to establish a call with bi-directional media flowing with no more than two messages, like in figure 3. Fast Connect is widely supported in the industry. Even so, most devices still implement the complete H.245 exchange as shown above and performs that message exchange in parallel to other activities, so there is no noticeable delay to the calling or called party.

Use cases
H.323 and Voice over IP services
Voice over Internet Protocol (VoIP) describes the transmission of voice using the Internet or other packet switched networks. ITU-T Recommendation H.323 is one of the standards used in VoIP. VoIP requires a connection to the Internet or another packet switched network, a subscription to a VoIP service provider and a client (an analogue telephone adapter (ATA), VoIP Phone or "soft phone"). The service provider offers the connection to other VoIP

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services or to the PSTN. Most service providers charge a monthly fee, then additional costs when calls are made.[1] Using VoIP between two enterprise locations would not necessarily require a VoIP service provider, for example. H.323 has been widely deployed by companies who wish to interconnect remote locations over IP using a number of various wired and wireless technologies.

H.323 and Videoconference services
A videoconference, or video teleconference (VTC), is a set of telecommunication technologies allowing two or more locations to interact via two-way video and audio transmissions simultaneously. There are basically two types of videoconferencing; dedicated VTC systems have all required components packaged into a single piece of equipment while desktop VTC systems are add-ons to normal PC's, transforming them into VTC devices. Simultaneous videoconferencing among three or more remote points is possible by means of a Multipoint Control Unit (MCU). There are MCU bridges for IP and ISDN-based videoconferencing. Due to the price point and proliferation of the Internet, and broadband in particular, there has been a strong spurt of growth and use of H.323-based IP videoconferencing. H.323 is accessible to anyone with a high speed Internet connection, such as DSL. Videoconferencing is utilized in various situations, for example; distance education, telemedicine and business.[2]

International Conferences
H.323 has been used in the industry to enable large-scale international video conferences that are significantly larger than the typical video conference. One of the most widely attended is an annual event called “Megaconference”. The Mega conferences are special non-profit world-wide events which use the H.323 protocol to create a virtual conference involving hundreds of locations and thousands of people. Everyone in the world with H.323 equipment is invited to participate. They are the world’s largest video conferences. The first Mega conference was held in 1999, and it has been held annually ever since. The Mega conferences are run as professional conferences, with no central location. There are presentations (called Interactions) by users of H.323 technology, vendor presentations, roll calls, musical events and open periods called mega conference Cafes where anyone can talk to anyone.A particularly
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popular portion is the Roll Calls, where all registrants are given a moment to say hello to the world; they can say whetever they wish, sing a song, play a video or whatever. A network of 30 or so MCUs is created for the event, all cascaded together. Background chats are run for the presenters, the MCU managers and the audience, to coordinate the event in real-time. The event is also streamed out to the world, and is recorded for later distribution on DVDs.[3] There have been a number of spinoffs of the Mega conference, beginning with Mega conference Jr, which started in 2002. That event is intended for students of all ages, and students make all the presentations.[4] The Mega conferences and their spin-offs received the first-ever Internet2 Driving Exemplary Applications award in 2006

SIGTRAN
SIGTRAN is the name given to an Internet Engineering Task Force (IETF) working group that produced specifications for a family of protocols that provide reliable datagram service and user layer adaptations for SS7 and ISDN communications protocols. SIGTRAN is logically an extension of the SS7 protocol family. It supports the same application and call management paradigms as SS7 but uses an IP transport called Stream Control Transmission Protocol (SCTP) as its underlying transport vehicle. Indeed, the most significant protocol defined by the SIGTRAN group was SCTP, which is used to carry PSTN signaling over IP. The SIGTRAN group was significantly influenced by telecommunications engineers intent on using the new protocols for adapting VoIP networks to the PSTN with special regard to signaling applications. Recently, SCTP is finding applications beyond its original purpose wherever reliable datagram service is desired. The SIGTRAN family of protocols includes: ISDN User Adaptation (IUA) MTP2 User Peer-to-Peer Adaptation Layer (M2PA) MTP2 User Adaptation Layer (M2UA)

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MTP3 User Adaptation Layer (M3UA) Stream Control Transmission Protocol (SCTP) SCCP User Adaptation (SUA) V5 User Adaptation (V5UA)
ISDN ELEMENTS Integrated Services Digital Network (ISDN), originally "Integriertes Sprachund Datennetz" (German for "Integrated Speech and Data Net"), is a circuitswitched telephone network system, designed to allow digital transmission of voice and data over ordinary telephone copper wires, resulting in better voice quality than an analog phone. It offers circuit-switched connections (for either voice or data) in increments of 64 kbit/s. One of the major use cases is Internet access, where ISDN typically provides a maximum of 128 kbit/s. More broadly, ISDN is a set of protocols for establishing and breaking circuit switched connections, and for advanced call features for the user. It was introduced in the late 1980's.[1] In a videoconference, ISDN provides simultaneous voice, video, and text transmission between individual desktop videoconferencing systems and group (room) videoconferencing systems. ISDN elements SIGTRAN Integrated Services refers to ISDN's ability to deliver at minimum two simultaneous connections, in any combination of data, voice, video, and fax, over a single line. Multiple devices can be attached to the line, and used as needed. That means an ISDN line can take care of most people's complete communications needs at a much higher transmission rate, without forcing the purchase of multiple analog phone lines. Digital refers to its purely digital transmission, as opposed to the analog transmission of plain old telephone service (POTS). Use of an analog telephone modem for Internet access requires that the Internet service provider's (ISP) modem converts the digital content to analog signals before sending it and the user's modem then converts those signals back to digital when receiving. When connecting with ISDN there is no analog conversion.

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Network refers to the fact that ISDN is not simply a point-to-point solution like a leased line. ISDN networks extend from the local telephone exchange to the remote user and includes all of the telecommunications and switching equipment in between. The purpose of the ISDN is to provide fully integrated digital services to the users. These services fall under three categories: bearer services, supplementary services and teleservices.

The five-layer TCP/IP model 5. Application layer

DHCP · DNS · FTP · Gopher · HTTP · IMAP4 · IRC · NNTP · XMPP · POP3 · RTP · SIP · SMTP · SNMP · SSH · TELNET · RPC · RTCP · RTSP · TLS (and SSL) · SDP · SOAP · GTP · STUN · NTP · (more) 4. Transport layer TCP · UDP · DCCP · SCTP · RSVP · ECN · (more) 3. Network/internet layer IP (IPv4 · IPv6) · OSPF · IS-IS · BGP · IPsec · ARP · RARP · RIP · ICMP · ICMPv6 · IGMP · (more) 2. Data link layer 802.11 (WLAN) · 802.16 · Wi-Fi · WiMAX · ATM · DTM · Token ring · Ethernet · FDDI · Frame Relay · GPRS · EVDO · HSPA · HDLC · PPP · PPTP · L2TP · ISDN · ARCnet · LLTD · (more) 1. Physical layer Ethernet physical layer · Modems · PLC · SONET/SDH · G.709 · Optical fiber · Coaxial cable · Twisted pair · (mo

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The Message Transfer Part (MTP) is part of the Signaling System 7 (SS7) used for communication in Public Switched Telephone Networks. MTP is responsible for reliable, unduplicated and in-sequence transport of SS7 messages between communication partners. MTP is made up of three levels, corresponding to layers in the OSI model: MTP Level 1 corresponds to OSI Layer 1 (the physical layer), MTP Level 2 to OSI Layer 2 (the data link layer), and MTP Level 3 to OSI Layer 3 (the network layer). MTP Level 3 is usually abbreviated as MTP3. Likewise MTP Level 2 and MTP Level 1 are abbreviated as MTP2 and MTP1. MTP1 represents the physical layer. That is, the layer that is responsible for the connection of SS7 Signaling Points into the transmission network over which they communicate with each other. Primarily, this involves the conversion of messaging into electrical signal and the maintenance of the physical links through which these pass. In this way, it is analogous to the Layer 1 of ISDN or other, perhaps more familiar, protocols. MTP1 normally uses a timeslot in an E-carrier or T-carrier. MTP2 provides error detection and sequence checking, and retransmits unacknowledged messages. MTP2 uses packets called signal units to transmit SS7 messages. There are three types of signal units: Fill-in Signal Unit (FISU), Link Status Signal Unit (LSSU), Message Signal Unit (MSU). MTP3 provides routing functionality to transport signaling messages through the SS7 network to the requested endpoint. Each network element in the SS7 network has a unique address, the Point Code (PC). Message routing is performed according to this address. A distinction is made between a Signaling Transfer Point (STP) which only performs MTP message routing functionalities and a Signaling End Point (SEP) which uses MTP to communicate with other SEPs (that is, telecom switches). MTP3 is also responsible for network management; when the availability of MTP2 data links changes, MTP3 establishes alternative links as required and propagates information about route availability through the network. MTP is formally defined in ITU-T recommendations Q.701-Q.705. Tests for the MTP are specified in the ITU-T recommendations Q.781 for MTP2 and in Q.782

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for MTP3. These tests are used to validate the correct implementation of the MTP protocol. Different countries use different variants of the MTP protocols. In North America, the formal standard followed is the Telcordia Technologies (formerly Bellcore) document GR-246-CORE
SS7 protocol suite Layer Application Protocols INAP, MAP, IS-41... TCAP, CAP, ISUP, ... Transport Network Data link Physical SCCP MTP Level 3 MTP Level 2 MTP Level 1 ...

The Message Transfer Part (MTP) is part of the Signaling System 7 (SS7) used for communication in Public Switched Telephone Networks. MTP is responsible for reliable, unduplicated and in-sequence transport of SS7 messages between communication partners. MTP is made up of three levels, corresponding to layers in the OSI model: MTP Level 1 corresponds to OSI Layer 1 (the physical layer), MTP Level 2 to OSI Layer 2 (the data link layer), and MTP Level 3 to OSI Layer 3 (the network layer). MTP Level 3 is usually abbreviated as MTP3. Likewise MTP Level 2 and MTP Level 1 are abbreviated as MTP2 and MTP1. MTP1 represents the physical layer. That is, the layer that is responsible for the connection of SS7 Signaling Points into the transmission network over which they communicate with each other. Primarily, this involves the conversion of messaging into electrical signal and the maintenance of the physical links through which these pass. In this way, it is analogous to the Layer 1 of ISDN or other, perhaps more familiar, protocols. MTP1 normally uses a timeslot in an E-carrier or T-carrier.

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MTP2 provides error detection and sequence checking, and retransmits unacknowledged messages. MTP2 uses packets called signal units to transmit SS7 messages. There are three types of signal units: Fill-in Signal Unit (FISU), Link Status Signal Unit (LSSU), Message Signal Unit (MSU). MTP3 provides routing functionality to transport signaling messages through the SS7 network to the requested endpoint. Each network element in the SS7 network has a unique address, the Point Code (PC). Message routing is performed according to this address. A distinction is made between a Signaling Transfer Point (STP) which only performs MTP message routing functionalities and a Signaling End Point (SEP) which uses MTP to communicate with other SEPs (that is, telecom switches). MTP3 is also responsible for network management; when the availability of MTP2 data links changes, MTP3 establishes alternative links as required and propagates information about route availability through the network. MTP is formally defined in ITU-T recommendations Q.701-Q.705. Tests for the MTP are specified in the ITU-T recommendations Q.781 for MTP2 and in Q.782 for MTP3. These tests are used to validate the correct implementation of the MTP protocol. Different countries use different variants of the MTP protocols. In North America, the formal standard followed is the Telcordia Technologies (formerly Bellcore) document GR-246-CORE. M3UA

The introduction to this article provides insufficient context for those unfamiliar with the subject. Please help improve the article with a good introductory style. SS7 protocol suite Layer Application Protocols INAP, MAP, IS-41... TCAP, CAP, ISUP, ... Transport Network Data link SCCP MTP Level 3 MTP Level 2 ...

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Physical MTP Level 1 ...

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MTP-3 M3UA stands for MTP Level 3 (MTP3) User Adaptation Layer as defined by the IETF SIGTRAN working group. M3UA enables the SS7 protocol's User Parts (e.g. ISUP, SCCP and TUP) to run over IP instead of telephony equipment like ISDN and PSTN. It is recommended to use the services of SCTP to transmit M3UA. An open implementation of the M3UA standard can be found at OpenSS7's web site.

The five-layer TCP/IP model 5. Application layer

DHCP · DNS · FTP · Gopher · HTTP · IMAP4 · IRC · NNTP · XMPP · POP3 · RTP · SIP · SMTP · SNMP · SSH · TELNET · RPC · RTCP · RTSP · TLS (and SSL) · SDP · SOAP · GTP · STUN · NTP · (more) 4. Transport layer TCP · UDP · DCCP · SCTP · RSVP · ECN · (more) 3. Network/internet layer IP (IPv4 · IPv6) · OSPF · IS-IS · BGP · IPsec · ARP · RARP · RIP · ICMP · ICMPv6 · IGMP · (more) 2. Data link layer 802.11 (WLAN) · 802.16 · Wi-Fi · WiMAX · ATM · DTM · Token ring · Ethernet · FDDI · Frame Relay · GPRS · EVDO · HSPA · HDLC · PPP · PPTP · L2TP · ISDN · ARCnet · LLTD · (more) 1. Physical layer Ethernet physical layer · Modems · PLC · SONET/SDH · G.709 · Optical fiber · Coaxial cable · Twisted pair · (more)

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In the field of computer networking, the IETF Signaling Transport (SIGTRAN) working group defined the Stream Control Transmission Protocol (SCTP) as a transport layer protocol in 2000. As a transport protocol, SCTP operates analogously to TCP or UDP. Indeed it provides some similar services as TCP—ensuring reliable, in-sequence transport of messages with congestion control. (In the absence of native SCTP support, it may sometimes be desirable to tunnel SCTP over UDP.)
Contents 1 Message-based multi-streaming 2 Benefits 3 Motivations 4 Comparison between transport layers 5 Implementations 6 Packet structure 7 See also 8 External links

Message-based multi-streaming Whereas TCP transports a byte-stream, SCTP can transport multiple messagestreams. All bytes sent in a TCP connection must be delivered in that order, which requires that a byte transmitted first must safely arrive at the destination before a second byte can be processed even if the second byte manages to arrive first. If an arbitrary number of bytes are sent in one step and later some more bytes are sent, these bytes will be received in order, but the receiver can not distinguish which bytes were sent in which step. SCTP in contrast, conserves message boundaries by operating on whole messages instead of single bytes. That means if one message of several related bytes of information is sent in one step, exactly that message is received in one step. The term "multi-streaming" refers to the capability of SCTP to transmit several independent streams of messages in parallel. For example, transmitting two images in an HTTP application in parallel over the same SCTP association. You

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might think of multi-streaming as bundling several TCP-connections in one SCTP-association operating with messages instead of bytes. TCP ensures the correct order of bytes in the stream by conceptually assigning a sequence number to each byte sent and ordering these bytes based on that sequence number when they arrive. SCTP, on the other hand, assigns different sequence numbers to messages sent in a stream. This allows independent ordering of messages in different streams. However, message ordering is optional in SCTP. If the user application so desires, messages will be processed in the order they are received instead of the order they were sent, should these differ. Signaling in Public Switched Telephone Networks requires message-based delivery. Multi-Streaming also provides an advantage when used to transport PSTN services. If an SCTP connection is set up to carry, say, ten phone calls with one call per stream, then if a single message is lost in only one phone call, the other nine calls will not be affected. To handle ten phone calls in TCP, some form of multiplexing would be required to put all ten phone calls into a single byte-stream. If a single packet for phone call #3 is lost then all packets after that could not be processed until the missing bytes are retransmitted, thus causing unnecessary delays in the other calls. Benefits Benefits of SCTP include: Multihoming support, where one (or both) endpoints of a connection can consist of more than one IP address, enabling transparent fail-over between redundant network paths. Delivery of data in chunks within independent streams - this eliminates unnecessary head-of-line blocking, as opposed to TCP byte-stream delivery. Path Selection and Monitoring - Selects a "primary" data transmission path and tests the connectivity of the transmission path. Validation and Acknowledgment mechanisms - Protects against flooding attacks and provides notification of duplicated or missing data chunks. Improved error detection suitable for jumbo Ethernet frames. The designers of SCTP originally intended it for the transport of telephony (SS7) protocols over IP, with the goal of duplicating some of the reliability attributes of
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the SS7 signaling network in IP. This IETF effort is known as SIGTRAN. In the meantime, other uses have been proposed, for example the Diameter protocol and Reliable server pooling ("RSerPool"). Motivations Transmission Control Protocol (TCP) has provided the primary means to transfer data across the Internet in a reliable way. However, TCP has imposed limitations on several applications. From TCP provides both reliable data transfer and strict order-of- transmission delivery of data. Some applications need reliable transfer without sequence maintenance, while others would be satisfied with partial ordering of the data. In both of these cases, the head-of-line blocking offered by TCP causes unnecessary delay. The stream-oriented nature of TCP is often an inconvenience. Applications must add their own record marking to delineate their messages, and must make explicit use of the push facility to ensure that a complete message is transferred in a reasonable time. The limited scope of TCP sockets complicates the task of providing highlyavailable data transfer capability using multi-homed hosts. TCP is relatively vulnerable to denial-of-service attacks, such as SYN attacks. All these limitations affect the performance of IP over public switched telephone networks. Comparison between transport layers
Feature Name UDP TCP SCTP

Connection oriented

No

Yes Yes

Reliable transport

No

Yes Yes

Preserve message boundary

Yes No

Yes

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Ordered delivery

No

Yes Yes

Unordered delivery

Yes No

Yes

Data checksum

Yes Yes Yes

Checksum size (bits)

16

16

32

Path MTU

No

Yes Yes

Congestion control

No

Yes Yes

Multiple streams

No

No

Yes

Multi-homing support

No

No

Yes

Bundling

No

No

Yes

Signaling Connection and Control Part
SS7 protocol suite Layer Application Protocols INAP, MAP, IS-41... TCAP, CAP, ISUP, ... Transport Network Data link Physical SCCP MTP Level 3 MTP Level 2 MTP Level 1 ... ...

The Signaling Connection and Control Part (SCCP) is a transport layer protocol which provides extended routing, flow control, segmentation, connection-orientation, and error correction facilities in Signaling System 7

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telecommunications networks. SCCP relies on the services of MTP for basic routing and error detection.
Contents 1 Published specification 2 Routing facilities beyond MTP-3 3 Classes of service 3.1 Class 0: Basic connectionless 3.2 Class 1: Sequenced connectionless 3.3 Class 2: Basic connectionoriented 3.4 Class 3: Flow control connection oriented 4 Transport over IP Networks

Published specification The base SCCP specification is defined by the ITU-T, in recommendations Q.711 to Q.714, with additional information to implementors provided by Q.715 and Q.716. There are, however, regional variations defined by local standards bodies. In the United States, ANSI publishes its modifications to Q.713 as ANSI T1.112 or JT-Q.711 to JT-Q.714, whilst in Europe ETSI publishes ETSI EN 300 009, which documents its modifications to the ITU-T specification. Routing facilities beyond MTP-3 Although MTP-3 provides routing capabilities based upon the Point Code, SCCP allows routing using a Point Code and Subsystem number or a Global Title. A Point Code is used to address a particular node on the network, whilst a Subsystem number addresses a specific application available on that node. SCCP employs a process called Global Title Translation (which is similar to DNS resolution in IP networks) in order to determine Point Codes from Global Titles so as to instruct MTP-3 on where to route messages.

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SCCP messages contain parameters which describe the type of addressing used, and how the message should be routed:

Address Indicator Subsystem indicator: The address includes a Subsystem Number Point Code indicator: The address includes a Point Code Global title indicator No Global Title Global Title includes Translation Type (TT), Numbering Plan Indiciator (NPI) and Type of Number (TON) Global Title includes Translation Type only Routing indicator Route using Global Title only Route using Point Code/Subsystem number Address Indicator Coding Address Indicator coded as national (the Address Indicator is treated as international if not specified) Classes of service SCCP provides 5 classes of service to its applications: Class 0: Basic connectionless Class 1: Sequenced connectionless Class 2: Basic connection-oriented Class 3: Flow control connection oriented Class 4: Error recovery and flow control connection oriented The connectionless protocol classes provide the capabilities needed to transfer one Network Service Data Unit (NSDU) in the "data" field of an XUDT, LUDT or UDT message. When one connectionless message is not sufficient to convey the user data contained in one NSDU, a segmenting/reassembly function for protocol classes 0 and 1 is provided. In this case, the SCCP at the originating node or in a relay node provides segmentation of the information into multiple segments prior

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to transfer in the "data" field of XUDT (or as a network option LUDT) messages. At the destination node, the NSDU is reassembled. The connection-oriented protocol classes (protocol classes 2 and 3) provide the means to set up signalling connections in order to exchange a number of related NSDUs. The connection-oriented protocol classes also provide a segmenting and reassembling capability. If an NSDU is longer than 255 octets, it is split into multiple segments at the originating node, prior to transfer in the "data" field of DT messages. Each segment is less than or equal to 255 octets. At the destination node, the NSDU is reassembled.[1] Class 0: Basic connectionless The SCCP Class 0 service is the most basic of SCCP transports. Network Service Data Units passed by higher layers to the SCCP in the originating node are delivered by the SCCP to higher layers in the destination node. They are transferred independently of each other. Therefore, they may be delivered to the SCCP user out-of-sequence. Thus, this protocol class corresponds to a pure connectionless network service. As a connectionless protocol, no transport-level dialog is established between the sender and the receiver. Class 1: Sequenced connectionless SCCP Class 1 builds on the capabilities of Class 0, with the addition of a sequence control parameter in the NSDU which allows the SCCP User to instruct the SCCP that a given stream of messages should be delivered in sequence. Therefore, Protocol Class 1 corresponds to an enhanced connectionless service with in-sequence delivery. Class 2: Basic connection-oriented SCCP Class 2 provides the facilities of Class 1, but also allows for an entity to establish a two-way dialog with another entity using SCCP. Class 3: Flow control connection oriented Class 3 service builds upon Class 2, but also allows for expedited (urgent) messages to be sent and received, and for errors in sequencing (segment reassembly) to be detected and for SCCP to restart a connection should this occur.

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In the SIGTRAN suite of protocols, there are two primary methods of transporting SCCP applications across Internet Protocol networks: SCCP can be transported directly using the MTP level 3 User Adaptation protocol (M3UA), a protocol which provides support for users of MTP-3—including SCCP. Alternatively, SCCP applications can operate over the SCCP User Adaptation protocol (SUA) which is a form of modified SCCP designed specifically for use in IP networking.

V5 interface The introduction to this article provides insufficient context for those unfamiliar with the subject. Please help improve the article with a good introductory style. V5 is a set of telephone network protocols defined by ETSI by which a multiplexer in the access network of the PSTN can communicate with a telephone exchange. The protocols are designed to handle both POTS and ISDN traffic. They are based on the principle of common channel signalling where message-based signalling for all subscribers uses the same signalling channel(s) rather than separate channels existing for different subscribers. V5 comes in two forms: V5.1 (ETS 300 324) in which there is a 1 to 1 correspondence between subscriber lines and bearer channels in the aggregate link to the exchange. A V5.1 interface relates to a single aggregate E1 (2 Mbit/s) link between a multiplexer and an exchange. V5.2 (ETS 300 347) which provides for concentration where there are not enough bearer channels in the aggregate link(s) to accommodate all subscribers at the same time. A single V5.2 interface can control up to 16 E1 links at once and can include protection of the signalling channels. The Layer 3 protocols

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Control protocol - for setting up a V5 connection between an Access Network and a Local Exchange. PSTN protocol - For call setup messages to control POTS (Like Of Hook and Digit Messages) BCC protocol - Bearer Control allocates a 64K timeslot to a call. Only V5.2 supports it. Link control protocol - For managing up to 16 E1 links. Protection protocol - Allows the V5 protocol is duplicated in two or more links. V5.1 only supports the Control, PSTN and ISDN protocols. V5.2 also supports BCC, Link Control and Protection protocols. V5 Layer 3 protocols are transported on a Layer 2 protocol called LAPV5, a variation of the LAP-D or Link Access Procedures, D channel ISDN transport layer. As standard protocols, V5 allows the interoperability of Access Networks with Exchanges from different vendors. V5 is a circuit-switched protocol stack. see packet switched VoIP.

Softswitch A softswitch is a central device in a telephone network which connects calls from one phone line to another, entirely by means of software running on a computer system. This work was formerly carried out by hardware, with physical switchboards to route the calls. A softswitch is typically used to control connections at the junction point between circuit and packet networks. A single device containing both the switching logic and the switching fabric can be used for this purpose; however, modern technology has led to a preference for decomposing this device into a Call Agent and a Media Gateway. The Call Agent takes care of functions like billing, call routing, signalling, call services and so on and is the 'brains' of the outfit. A Call Agent may control several different Media Gateways in geographically dispersed areas over a TCP/IP link.

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The Media Gateway connects different types of digital media stream together to create an end-to-end path for the media (voice and data) in the call. It may have interfaces to connect to traditional PSTN networks like DS1 or DS3 ports (E1 or STM1 in the case of non-US networks), it may have interfaces to connect to ATM and IP networks and in the modern system will have Ethernet interfaces to connect VoIP calls. The call agent will instruct the media gateway to connect media streams between these interfaces to connect the call - all transparently to the end-users. The softswitch generally resides in a building owned by the telephone company called a central office. The central office will have telephone trunks to carry calls to other offices owned by the telephone company and to other telephone companies (aka the Public Switched Telephone Network or PSTN). Looking towards the end users from the switch, the Media Gateway may be connected to several access devices. These access devices can range from small Analog Telephone Adaptors (ATA) which provide just one RJ11 telephone jack to an Integrated Access Device (IAD) or PBX which may provide several hundred telephone connections. Typically the larger access devices will be located in a building owned by the telephone company near to the customers they serve. Each end user can be connected to the IAD by a simple pair of copper wires. The medium sized devices and PBXs will typically be used in a business premises and the single line devices would probably be found in residential premises. In more recent times (i.e., the IP Multimedia Subsystem or IMS), the Softswitch element is represented by the Media Gateway Controller (MGC) element, and the term "Softswitch" is rarely used in the IMS context. Feature server as a part of soft switch The feature server, often built into a call agent/softswitch, is the functional component that provides call-related features. Capabilities such as call forwarding, call waiting, and last call return, if implemented in the network, are implemented in the feature server. The feature server works closely with the call agent, and may call upon the media server to provide these services. These

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features do not require the subscriber to explicitly request them but tend to be triggered within the call handling logic. An example of a feature service is last call return, in which the user picks up the phone, dials *69, and hears, “The number that last called you was xxx-xxx-xxxx. Press 1 to return this call.” When the call agent sees the dial string *69, it triggers an invocation of the feature server function. The feature server examines its database, finds the user and the caller identification of the last call, then asks the media server to play the announcement and collect a digit. When the media server returns a “1”, the feature server instructs the call agent to establish a call between the user and the party that last called that user.

RTP PROTOCOL
The Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 which was made obsolete in 2003.Real time transport protocol can also be used in conjunction with RTSP protocol which enhances the field of multimedia applications. RTP does not have a standard TCP or UDP port on which it communicates. The only standard that it obeys is that UDP communications are done via an even port and the next higher odd port is used for RTP Control Protocol (RTCP) communications. Although there are no standards assigned, RTP is generally configured to use ports 16384-32767. RTP can carry any data with real-time characteristics, such as interactive audio and video. Call setup and tear-down for VoIP applications is usually performed by either SIP or H.323 protocols. The fact that RTP uses a dynamic port range makes it difficult for it to traverse firewalls. In order to get around this problem, it is often necessary to set up a STUN server. It was originally designed as a multicast protocol, but has since been applied in many unicast applications. It is frequently used in streaming media systems (in conjunction with RTSP) as well as videoconferencing and push to talk systems (in conjunction with H.323 or SIP), making it the technical foundation of the Voice over IP industry. It goes along with the RTCP and is built on top

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of the User Datagram Protocol (UDP). Applications using RTP are less sensitive to packet loss, but typically very sensitive to delays, so UDP is a better choice than TCP for such applications. The services provided by RTP include: Payload-type identification - Indication of what kind of content is being carried Sequence numbering - PDU sequence number Time stamping - allow synchronization and jitter calculations Delivery monitoring The protocols themselves do not provide mechanisms to ensure timely delivery. They also do not give any Quality of Service (QoS) guarantees. These things have to be provided by some other mechanism. Also, out of order delivery is still possible, and flow and congestion control are not supported directly. However, the protocols do deliver the necessary data to the application to make sure it can put the received packets in the correct order. Also, RTCP provides information about reception quality which the application can use to make local adjustments. For example if a congestion is forming, the application could decide to lower the data rate. RTP was also published by the ITU-T as H.225.0, but later removed once the RTP Profile for Audio and Video Conferences) which can be used (optionally) to provide confidentiality, message authentication, and replay protection for audio and video streams being delivered.

Contents
[hide]

1 Packet structure 2 Potential further development of RTP & RTCP 3 Mathematical background 4 Structure of RTP/RTCP applications

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+ Bits 0 32 64 96 96+(CC×32) 96+(CC×32) + (X×((EHL+1)×32))

0-1 2 3 4-7 8

915

16-31 Sequence Number

Ver. P X CC M PT

Timestamp SSRC identifier ... CSRC identifiers ... Extension header (optional). Data

Packet structure
The RTP header size is 12 bytes. Ver. (2 bits) Indicates the version of the protocol. Current version is 2. P (1 bit) Used to indicate if there are extra padding bytes at the end of the RTP packet. X (1 bit) Indicates if the extensions to the protocol are being used in the packet. CC (4 bits) Contains the number of CSRC identifiers that follow the fixed header. M (1 bit) Used at the application level and is defined by a profile. If it is set, it means that the current data has some special relevance for the application. PT (7 bits) Indicates the format of the payload and determines its interpretation by the application. SSRC Indicates the synchronization source.
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CSRC Contributing source ID. Extension header Indicates the length of the extension (EHL=extension header length) in 32bit units, excluding the 32bits of the extension header.

Potential further development of RTP & RTCP
The Real-time Transport Protocol (RTP) and the Real-time Transport Control Protocol (RTCP) are commonly used together. RTP is used to transmit data (e.g. audio and video) and RTCP is used to monitor QoS. The monitoring of quality of service is very important for modern applications. In large scale applications (e.g. IPTV), there is an unacceptable delay between RTCP reports, which can cause quality of service related problems. To reduce the size of the IP, UDP and RTP headers, Compressed RTP (CRTP) was developed. It is primarily used for reliable and fast point-to-point links, but it can be problematic in other applications. Therefore, Enhanced CRTP (ECRTP) was defined. Especially in VoIP over wireless applications, headers are significantly larger than the payload. The RObust Header Compression (ROHC) seems to be an increasingly deployed method for better efficiency

Real-time Transport Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP). RTCP provides out-of-band control information for an RTP flow. It partners RTP in the delivery and packaging of multimedia data, but does not transport any data itself. It is used periodically to transmit control packets to participants in a streaming multimedia session. The primary function of RTCP is to provide feedback on the quality of service being provided by RTP. RTCP gathers statistics on a media connection and information such as bytes sent, packets sent, lost packets, jitter, feedback and round trip delay. An application may use this information to increase the quality of service, perhaps by limiting flow or using a different codec.
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There are several type of RTCP packets: Sender report packet, Receiver report packet, Source Description RTCP Packet, Goodbye RTCP Packet and Application Specific RTCP packets. RTCP itself does not provide any flow encryption or authentication means. SRTCP protocol can be used for that purpose.

Problems and potential further development of RTCP
The Real-time Transport Control Protocol (RTCP) has some issues with deployment on large scale applications of types that could inflict very long delay between RTCP reports (such as IPTV). This could make the receiver's reporting messages and its evaluation by sender inaccurate relative to the real state of the session. Due to this there are some methods to deal with this issue: these are filtering, biasing and hierarchical aggregation.

Session Initiation Protocol (SIP-Protocol)
The Session Initiation Protocol (SIP) is a signalling protocol, widely used for setting up and tearing down multimedia communication sessions such as voice and video calls over the Internet. Other feasible application examples include video conferencing, streaming multimedia distribution, instant messaging, presence information and online games. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture for IP based streaming multimedia services in cellular systems.

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The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, adding or deleting media streams, etc. The SIP protocol is situated at the session layer in the OSI model, and at the application layer in the TCP/IP model. SIP is designed to be independent of the underlying transport layer; it can run on TCP, UDP, or SCTP. It was originally designed by Henning Schulzrinne (Columbia University) and Mark Handley (UCL) starting in 1996. SIP has the following characteristics: Transport-independent, because SIP can be used with UDP, TCP, SCTP, etc. Text-based, allowing for humans to read and analyze SIP messages.
Contents 1 Protocol design 2 SIP network elements 3 Instant messaging (IM) and presence 4 Conformance testing 5 Commercial applications

Protocol design
SIP clients typically use TCP or UDP (typically on port 5060) to connect to SIP servers and other SIP endpoints. SIP is primarily used in setting up and tearing down voice or video calls. However, it can be used in any application where session initiation is a requirement. These include Event Subscription and Notification, Terminal mobility and so on. There are a large number of. All voice/video communications are done over separate session protocols, typically RTP. A motivating goal for SIP was to provide a signaling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network

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(PSTN). SIP by itself does not define these features; rather, its focus is callsetup and signaling. However, it has been designed to enable the building of such features in network elements known as Proxy Servers and User Agents. These are features that permit familiar telephone-like operations: dialing a number, causing a phone to ring, hearing ringback tones or a busy signal. Implementation and terminology are different in the SIP world but to the enduser, the behavior is similar. SIP-enabled telephony networks can also implement many of the more advanced call processing features present in Signaling System 7 (SS7), though the two protocols themselves are very different. SS7 is a centralized protocol, characterized by a complex central network architecture and dumb endpoints (traditional telephone handsets). SIP is a peer-to-peer protocol. As such it requires only a simple (and thus scalable) core network with intelligence distributed to the network edge, embedded in endpoints (terminating devices built in either hardware or software). SIP features are implemented in the communicating endpoints (i.e. at the edge of the network) as opposed to traditional SS7 features, which are implemented in the network. Although many other VoIP signaling protocols exist, SIP is characterized by its proponents as having roots in the IP community rather than the telecom industry. SIP has been standardized and governed primarily by the IETF while the H.323 VoIP protocol has been traditionally more associated with the ITU. However, the two organizations have endorsed both protocols in some fashion. SIP works in concert with several other protocols and is only involved in the signaling portion of a communication session. SIP acts as a carrier for the Session Description Protocol (SDP), which describes the media content of the session, e.g. what IP ports to use, the codec being used etc. In typical use, SIP "sessions" are simply packet streams of the Real-time Transport Protocol (RTP). RTP is the carrier for the actual voice or video content itself. SIP is similar to HTTP and shares some of its design principles: It is human readable and request-response structured. SIP shares many HTTP status codes, such as the familiar '404 not found'. SIP proponents also claim it to be simpler than H.323. However, some would counter that while SIP originally

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had a goal of simplicity, in its current state it has become as complex as H.323. Others would argue that SIP is a stateless protocol, hence making it possible to easily implement failover and other features that are difficult in stateful protocols such as H.323. SIP and H.323 are not limited to voice communication but can mediate any kind of communication session from voice to video or future, unrealized applications.

SIP network elements
Hardware endpoints — devices with the look, feel, and shape of a traditional telephone, but that use SIP and RTP for communication — are commercially available from several vendors. Some of these can use Electronic Numbering (ENUM) or DUNDi to translate existing phone numbers to SIP addresses, so calls to other SIP users can bypass the telephone network, even though your service provider might normally act as a gateway to the PSTN network for traditional phone numbers (and charge you for it). Today, software SIP endpoints are common.

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SIP also requires proxy and registrar network elements to work as a practical service. Although two SIP endpoints can communicate without any intervening SIP infrastructure, which is why the protocol is described as peerto-peer, this approach is impractical for a public service. There are various implementations that can act as proxy and registrar. "SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users." "SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. " "Since registrations play an important role in SIP, a User Agent Server that handles a REGISTER is given the special name registrar." "It is an important concept that the distinction between types of SIP servers is logical, not physical."

Instant messaging (IM) and presence
A standard instant messaging protocol based on SIP, called SIMPLE, has been proposed and is under development. SIMPLE can also carry presence information, conveying a person's willingness and ability to engage in communications. Presence information is most recognizable today as buddy status in IM clients. Some efforts have been made to integrate SIP-based VoIP with the XMPP specification used by Jabber. Most notably Google Talk, which extends XMPP to support voice, plans to integrate SIP. Google's XMPP extension is called Jingle and, like SIP, it acts as a Session Description Protocol carrier. SIP itself defines a method of passing instant messages between endpoints, similar to SMS messages. This is not generally supported by commercial operators.

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Commercial applications
Firewalls typically block media packet types such as UDP, though one way around this is to use TCP tunnelling and relays for media in order to provide NAT and firewall traversal. One solution involves tunnelling the media packets within TCP or HTTP packets to a relay. This solution uses additional functionality in conjunction with SIP, and packages the media packets into a TCP stream which is then sent to the relay. The relay then extracts the packets and sends them on to the other endpoint. If the other endpoint is behind a symmetrical NAT, or corporate firewall that does not allow VoIP traffic, the relay would transfer the packets to another tunnel. One disadvantage of this approach is that TCP was not designed for real time traffic such as voice, so an optimized form of the protocol is sometimes used. As envisioned by its originators, SIP's peer-to-peer nature does not enable network-provided services. For example, the network can not easily support legal interception of calls (referred to in the United States by the law governing wiretaps, CALEA). Emergency calls (calls to E911 in the USA) are difficult to route. It is difficult to identify the proper Public Service Answering Point, PSAP because of the inherent mobility of IP end points and the lack of any network location capability. Many VoIP phone companies allow customers to bring their own SIP devices, as SIP-capable telephone sets, or softphones. The new market for consumer SIP devices continues to expand.

Session Description Protocol
Session Description Protocol (SDP) is a format for describing streaming media initialization parameters. It has been published by the IETF. SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. SDP does not provide the content of the media form itself but simply provides a negotiation between two end points to allow them to agree on a media type and format. This allows SDP to support upcoming media types and formats, enabling systems based on this technology to be forward compatible.

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SDP started off as a component of the Session Announcement Protocol (SAP), but found other uses in conjunction with RTP, RTSP, SIP and just as a standalone format for describing multicast sessions. There are five terms related to SDP: 1. Conference: It is a set of two or more communicating users along with the software they are using. 2. Session : Session is the multimedia sender and receiver and the flowing stream of data. 3. Session Announcement: A session announcement is a mechanism by which a session description is conveyed to users in a proactive fashion, i.e., the session description was not explicitly requested by the user. 4. Session Advertisement : same as session announcement 5. Session Description : A well defined format for conveying sufficient information to discover and participate in a multimedia session

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