NGN Protocols

NGN PROTOCOLS amp STUDY MATERIAL page -1

N g n study guide

NEXT GENERATION NETWORK Compiled By- Syed irtiqa ali

BY- SYED IRTIQA ALI E-MAIL- irtiqaaligmailcom CELL-+92-34-55558984


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 TCPIP 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



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 (ie 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 softswitchThe feature server often built into a call agentsoftswitch 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 ldquoThe number that last called you was xxx-xxx-xxxx Press 1 to return this callrdquo 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 ldquo1rdquo the feature server instructs the call agent to establish a call between the user and the party that last called that user



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 H248-1) Both protocols follow the guidelines of the API Media Gateway Control Protocol Architecture and

ArchitectureThe 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

ldquowhat events should be reported to the Call Agent how endpoints should be connected together what signals should be played on endpointsrdquo

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 neednt 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



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 Agents 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 OverviewMGCP 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



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



H323 H323 is an umbrella Recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual 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 H32x 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

H323 Call Signaling is based on the ITU-T Recommendation Q931 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 ISDN-based PBX systems including transitions to IP-based Private Branch eXchanges (PBXs)

Within the context of H323 an IP-based PBX might be an H323 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 H323 excels at providing basic telephony functionality and interoperability H323rsquos strength lies in multimedia communication functionality designed specifically for IP networks



Contents

1 History

2 Protocols

3 Codecs

4 H323 Architecture

41 H323 Network Elements

411 Terminals

412 Multipoint Control Units

413 Gateways

414 Gatekeepers

415 Border Elements and Peer Elements

42 H323 Network Signaling

421 H2250 Call Signaling

422 RAS Signaling

423 H245 Call Control

4231 Capability Negotiation

4232 MasterSlave Determination

4233 Logical Channel Signaling

4234 Fast Connect

5 Use cases

51 H323 and Voice over IP services

52 H323 and Videoconference services



HistoryThe first version of H323 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 H323 has been revised and re-published with enhancements necessary to better-enable both voice and video functionality over Packet-switched networks with each version being backward-compatible with the previous version Recognizing that H323 was being used for communication not only on LANs but over WANs and within large carrier networks the title of H323 was changed when published in 1998 The title which has since remained unchanged is Packet-Based Multimedia Communications Systems The current version of H323 commonly referred to as H323v6 was published in 2006

One strength of H323 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

H323 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

ProtocolsH323 is a system specification that describes the use of several ITU-T and IETF protocols The protocols that comprise the core of almost any H323 system are

H2250 Registration Admission and Status (RAS) which is used between an H323 endpoint and a Gatekeeper to provide address resolution and admission control services

H2250 Call Signaling which is used between any two H323 entities in order to establish communication



H245 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 H323 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

H235 series describes security within H323 including security for both signaling and media

H239 describes dual stream use in videoconferencing usually one for live video the other for still images

H450 series describes various supplementary services

H460 series defines optional extensions that might be implemented by an endpoint or a Gatekeeper including ITU-T Recommendations H46017 H46018 and H46019 for Network address translation (NAT) Firewall (FW) traversal

In addition to those ITU-T Recommendations H323 utilizes various IETF Request for Comments for media transport and media packetization including Real-time Transport Protocol (RTP)

CodecsH323 utilizes both ITU-defined codecs and codecs defined outside the ITU Codecs that are widely implemented by H323 equipment include

Video codecs H261 H263 H264 Audio codecs G711 G729 (including G729a) G7231 G726 Text codecs T140



H323 ArchitectureThe H323 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 H323 deployments a gatekeeper is employed in order to among other things facilitate address resolution

H323 Network Elements Terminals Figure 1 - A complete sophisticated protocol stack

Terminals in an H323 network are the most fundamental elements in any H323 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 H323 terminal is something referred to as a protocol stack which implements the functionality defined by the H323 system The protocol stack would include an implementation of the basic protocol defined in ITU-T Recommendation H2250 and H245 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



most H323 systems do not implement such a wide array of capabilities but the logical arrangement is useful in understanding the relationships

Multipoint Control UnitsA 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 H323 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 H323 MCU the user might be able to see all of the other participants in the conference not only hear their voices



GatewaysGateways are devices that enable communication between H323 networks and other networks such as PSTN or ISDN networks If one party in a conversation is utilizing a terminal that is not an H323 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 H323 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 H320 and H324 to communicate with H323 systems Most of the third generation (3G) mobile networks deployed today utilize the H324 protocol and are able to communicate with H323-based terminals in corporate networks through such gateway devices

GatekeepersA Gatekeeper is an optional component in the H323 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



H323 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 H323 is referred to as a ldquozonerdquo 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 ldquointer-zonerdquo dialing so that two endpoints in separate zones can still communicate with each other

Border Elements and Peer ElementsFigure 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 H323 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



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

H323 Network SignalingH323 is defined as a binary protocol which allows for efficient message processing in network elements The syntax of the protocol is defined in ASN1 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 H323 systems

H2250 Call SignalingOnce the address of the remote endpoint is resolved the endpoint will use H2250 Call Signaling in order to establish communication with the remote entity H2250 messages are

Setup and Setup acknowledge Call Proceeding Connect



Alerting Information Release Complete Facility Progress Status and Status Inquiry Notify

Figure 3 - Establishment of an H323 call

In the simplest form an H323 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 H323 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 SignalingEndpoints 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)



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 systemrsquos 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



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)

H245 Call ControlOnce a call has initiated (but not necessarily fully connected) endpoints may initiate H245 call control signaling in order to provide more extensive control over the conference H245 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

H245 provides capabilities such as capability negotiation masterslave determination opening and closing of logical channels (ie 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 H245 capability negotiation is arguably the most important as it enables devices to communicate without having prior knowledge of the capabilities of the remote entity H245 enables rich multimedia capabilities including audio video text and data communication For transmission of audio video or text H323 devices utilize both ITU-defined codecs and codecs defined outside the ITU Codecs that are widely implemented by H323 equipment include

Video Codecs H261 H263 H264 Audio Codecs G711 G729 G729a G7231 G726 Text Codecs T140



H245 also enables real-time data conferencing capability through protocols like T120 T120-based applications generally operate in parallel with the H323 system but are integrated to provide the user with a seamless multimedia experience T120 provides such capabilities as application sharing T128 electronic whiteboard T126 file transfer T127 and text chat T134 within the context of the conference

When an H323 device initiates communication with a remote H323 device and when H245 communication is established between the two entities the Terminal Capability Set (TCS) message is the first message transmitted to the other side

MasterSlave Determination

After sending a TCS message H323 entities (through H245 exchanges) will attempt to determine which device is the master and which is the slave This process referred to as masterslave 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 masterslave 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



Fast ConnectFigure 5 - A typical H245 exchange

A typical H245 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 H323 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 H2250 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 H245 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 casesH323 and Voice over IP servicesVoice over Internet Protocol (VoIP) describes the transmission of voice using the Internet or other packet switched networks ITU-T Recommendation H323 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



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 H323 has been widely deployed by companies who wish to interconnect remote locations over IP using a number of various wired and wireless technologies

H323 and Videoconference servicesA 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 PCs 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 H323-based IP videoconferencing H323 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 ConferencesH323 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 ldquoMegaconferencerdquo

The Mega conferences are special non-profit world-wide events which use the H323 protocol to create a virtual conference involving hundreds of locations and thousands of people Everyone in the world with H323 equipment is invited to participate They are the worldrsquos 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 H323 technology vendor presentations roll calls musical events and open periods called mega conference Cafes where anyone can talk to anyoneA particularly



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)



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 Sprach- und Datennetz (German for Integrated Speech and Data Net) is a circuit-switched 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 kbits One of the major use cases is Internet access where ISDN typically provides a maximum of 128 kbits 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 1980s[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 ISDNs 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 peoples 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 providers (ISP) modem converts the digital content to analog signals before sending it and the users modem then converts those signals back to digital when receiving When connecting with ISDN there is no analog conversion



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 TCPIP model

5 Application layer

DHCP middot DNS middot FTP middot Gopher middot HTTP middot

IMAP4 middot IRC middot NNTP middot XMPP middot POP3 middot RTP

middot SIP middot SMTP middot SNMP middot SSH middot TELNET middot

RPC middot RTCP middot RTSP middot TLS (and SSL) middot

SDP middot SOAP middot GTP middot STUN middot NTP middot (more)

4 Transport layer

TCP middot UDP middot DCCP middot SCTP middot RSVP middot ECN middot (more)

3 Networkinternet layer

IP (IPv4 middot IPv6) middot OSPF middot IS-IS middot BGP middot IPsec middot ARP middot RARP middot RIP middot ICMP middot ICMPv6 middot IGMP middot (more)

2 Data link layer

80211 (WLAN) middot 80216 middot Wi-Fi middot WiMAX middot ATM middot DTM middot Token ring middot Ethernet middot FDDI middot Frame Relay middot GPRS middot EVDO middot HSPA middot HDLC middot PPP middot PPTP middot L2TP middot ISDN middot ARCnet middot LLTD middot (more)

1 Physical layer

Ethernet physical layer middot Modems middot PLC middot SONETSDH middot G709 middot Optical fiber middot Coaxial cable middot Twisted pair middot (mo



Message Transfer Part

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 Q701-Q705 Tests for the MTP are specified in the ITU-T recommendations Q781 for MTP2 and in Q782



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 Protocols

Application INAP MAP IS-41

TCAP CAP ISUP

Transport SCCP

Network MTP Level 3

Data link MTP Level 2

Physical MTP Level 1









MTP is formally defined in ITU-T recommendations Q701-Q705 Tests for the MTP are specified in the ITU-T recommendations Q781 for MTP2 and in Q782 for MTP3 These tests are used to validate the correct implementation of the MTP protocol


M3UA

The introduction to this article provides insufficient context for those unfamiliar with the subjectPlease help improve the article with a good introductory style

SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3





MTP-3

M3UA stands for MTP Level 3 (MTP3) User Adaptation Layer as defined by the IETF SIGTRAN working group M3UA enables the SS7 protocols User Parts (eg 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 OpenSS7s web site


5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer

Ethernet physical layer middot Modems middot PLC middot SONETSDH middot G709 middot Optical fiber middot Coaxial cable middot Twisted pair middot (more)



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 TCPmdashensuring 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 message-streams 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



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



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 highly-available 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



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 Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3



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



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

31 Class 0 Basic connectionless

32 Class 1 Sequenced

connectionless

33 Class 2 Basic connection-

oriented

34 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 Q711 to Q714 with additional information to implementors provided by Q715 and Q716 There are however regional variations defined by local standards bodies In the United States ANSI publishes its modifications to Q713 as ANSI T1112 or JT-Q711 to JT-Q714 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



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 CodeSubsystem 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 segmentingreassembly 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



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 re-assembly) to be detected and for SCCP to restart a connection should this occur



Transport over IP Networks

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-3mdashincluding 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

V51 (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 V51 interface relates to a single aggregate E1 (2 Mbits) link between a multiplexer and an exchange

V52 (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 V52 interface can control up to 16 E1 links at once and can include protection of the signalling channels

The Layer 3 protocols



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 V52 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

V51 only supports the Control PSTN and ISDN protocols V52 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












Feature server as a part of soft switch

The feature server often built into a call agentsoftswitch 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


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 2003Real 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 H323 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 H323 or SIP) making it the technical foundation of the Voice over IP industry It goes along with the RTCP and is built on top



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 H2250 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 amp RTCP

3 Mathematical background

4 Structure of RTPRTCP applications



Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31

Sequence Number Ver P X CC M PT0

Timestamp 32

SSRC identifier 64

CSRC identifiers 96

Extension header (optional) 96+(CCtimes32)

96+(CCtimes32) +

(Xtimes((EHL+1)times32))

Data

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



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 amp RTCPThe Real-time Transport Protocol (RTP) and the Real-time Transport Control Protocol (RTCP) are commonly used together RTP is used to transmit data (eg 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 (eg 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



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 RTCPThe 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 receivers 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



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 TCPIP 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 designSIP 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 voicevideo 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



(PSTN) SIP by itself does not define these features rather its focus is call-setup 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 end-user 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 (ie 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 H323 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 eg 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 H323 However some would counter that while SIP originally



had a goal of simplicity in its current state it has become as complex as H323 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 H323 SIP and H323 are not limited to voice communication but can mediate any kind of communication session from voice to video or future unrealized applications

SIP network elementsHardware endpoints mdash devices with the look feel and shape of a traditional telephone but that use SIP and RTP for communication mdash 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



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 peer-to-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 users 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 presenceA standard instant messaging protocol based on SIP called SIMPLE has been proposed and is under development SIMPLE can also carry presence information conveying a persons 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 Googles 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



Commercial applicationsFirewalls 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 SIPs 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



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 ie 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

NGN PROTOCOLS DOCUMENTATION

Bs-telecom members of yahoo

wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch



Softswitch













































NTFY - Notify













Contents

1 History

2 Protocols

3 Codecs

4 H323 Architecture


411 Terminals


413 Gateways

414 Gatekeepers




422 RAS Signaling





4234 Fast Connect

5 Use cases





















































































































ISDN ELEMENTS











5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer
















SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3













M3UA


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3





MTP-3




5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer






Contents


2 Benefits

3 Motivations


layers

5 Implementations

6 Packet structure

7 See also

8 External links









Benefits











Motivations


















Path MTU No Yes Yes




Bundling No No Yes


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3







Contents






connectionless


oriented


connection oriented










Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch







































NTFY - Notify













Contents

1 History

2 Protocols

3 Codecs

4 H323 Architecture


411 Terminals


413 Gateways

414 Gatekeepers




422 RAS Signaling





4234 Fast Connect

5 Use cases





















































































































ISDN ELEMENTS











5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer
















SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3













M3UA


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3





MTP-3




5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer






Contents


2 Benefits

3 Motivations


layers

5 Implementations

6 Packet structure

7 See also

8 External links









Benefits











Motivations


















Path MTU No Yes Yes




Bundling No No Yes


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3







Contents






connectionless


oriented


connection oriented










Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch











Contents

1 History

2 Protocols

3 Codecs

4 H323 Architecture


411 Terminals


413 Gateways

414 Gatekeepers




422 RAS Signaling





4234 Fast Connect

5 Use cases





















































































































ISDN ELEMENTS











5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer
















SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3













M3UA


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3





MTP-3




5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer






Contents


2 Benefits

3 Motivations


layers

5 Implementations

6 Packet structure

7 See also

8 External links









Benefits











Motivations


















Path MTU No Yes Yes




Bundling No No Yes


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3







Contents






connectionless


oriented


connection oriented










Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch



















































































































ISDN ELEMENTS











5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer
















SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3













M3UA


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3





MTP-3




5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer






Contents


2 Benefits

3 Motivations


layers

5 Implementations

6 Packet structure

7 See also

8 External links









Benefits











Motivations


















Path MTU No Yes Yes




Bundling No No Yes


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3







Contents






connectionless


oriented


connection oriented










Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch






5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer
















SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3













M3UA


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3





MTP-3




5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer






Contents


2 Benefits

3 Motivations


layers

5 Implementations

6 Packet structure

7 See also

8 External links









Benefits











Motivations


















Path MTU No Yes Yes




Bundling No No Yes


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3







Contents






connectionless


oriented


connection oriented










Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch















SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3













M3UA


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3





MTP-3




5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer






Contents


2 Benefits

3 Motivations


layers

5 Implementations

6 Packet structure

7 See also

8 External links









Benefits











Motivations


















Path MTU No Yes Yes




Bundling No No Yes


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3







Contents






connectionless


oriented


connection oriented










Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch







M3UA


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3





MTP-3




5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer






Contents


2 Benefits

3 Motivations


layers

5 Implementations

6 Packet structure

7 See also

8 External links









Benefits











Motivations


















Path MTU No Yes Yes




Bundling No No Yes


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3







Contents






connectionless


oriented


connection oriented










Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch




MTP-3




5 Application layer






4 Transport layer




2 Data link layer


1 Physical layer






Contents


2 Benefits

3 Motivations


layers

5 Implementations

6 Packet structure

7 See also

8 External links









Benefits











Motivations


















Path MTU No Yes Yes




Bundling No No Yes


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3







Contents






connectionless


oriented


connection oriented










Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch





Contents


2 Benefits

3 Motivations


layers

5 Implementations

6 Packet structure

7 See also

8 External links









Benefits











Motivations


















Path MTU No Yes Yes




Bundling No No Yes


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3







Contents






connectionless


oriented


connection oriented










Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch






Benefits











Motivations


















Path MTU No Yes Yes




Bundling No No Yes


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3







Contents






connectionless


oriented


connection oriented










Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch




Motivations


















Path MTU No Yes Yes




Bundling No No Yes


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3







Contents






connectionless


oriented


connection oriented










Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch







Path MTU No Yes Yes




Bundling No No Yes


SS7 protocol suite

Layer Protocols


TCAP CAP ISUP

Transport SCCP

Network MTP Level 3







Contents






connectionless


oriented


connection oriented










Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch




Contents






connectionless


oriented


connection oriented










Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch




Address Indicator


























V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch

















V5 interface













links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch







links





Softswitch






























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch



























Contents

[hide]

1 Packet structure






Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch



Packet structure

9-15+ Bits 0-1 2 3 4-7 8 16-31


Timestamp 32

SSRC identifier 64

CSRC identifiers 96


96+(CCtimes32) +


Data




X






























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch

























1 Protocol design











































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch





































wwwyahoogroupscom



H323

SIGTRAN

M3UA


V5 interface

Softswitch


NGN Protocols

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Transcript of NGN Protocols