Media Gateway Control Protocol (MGCP)

Media Gateway Control Protocol is a protocol used in mainly in Voice Over IP (VoIP) systems. This was created to serve the needs of the carrier based IP telephony networks. The protocol corresponds to H.323 and SIP, devised as an internal system between the Media Gateway (MG) and Media gateway controller (MGC).

All call processing management by connecting to the IP network is done by MGC, carrying out continuous communications with IP signaling tool. SIP servers or H.323 gatekeeper would be examples.

MGCP is composed of MG (to perform media signal conversion between packets and circuits), one Call Agent (CA), and Signaling Gateway (SG). All of these connect to a Public Switched Telephone Network (PSTN). Mostly MGCP is within a decomposed multimedia gateway which has CA composed of the call control ‘nucleus’ and a media gateway which operates media functions.

MGs have multimedia endpoints with which the CA creates and manages media sessions with other endpoints. Endpoints are data sources or data sinks that can be either virtual or physical. Hardware installation is required to generate physical endpoints while creating virtual endpoints need software installation.

CAs have the capacity to produce new connections or alter an existing one. Broadly, a media gateway is a component that offers conversion between Internet data packets (or other network packets) and voice transmissions carried by phone lines. The CA gives instructions to endpoints to detect events and create signals. Endpoints intend to convey variations in service state to the CA mechanically. The CA then examines endpoints and the associations between endpoints.

Duplicate Message Handling in SMSCs

Duplicate message handling in SMSCs is a complex process than it appears to be. Currently what most of the SCs does is check the Message Reference (MR) figure and if it's the same, then reject the message classifying it as a duplicate message.

Above mentioned process is carried out if the value, TR-REJECT-DUPLICATE, is set to true by the Message Sending Entity (MSE).

Message Reference

TP-MR is a integer field submitted to SC by a Mobile Station (MS) as a SMS-SUBMIT or SMS-COMMAND. MS increments the MR by 1 for each Short Message (SM) sent. This unique number is obtained by reading the last used TP-MR valued from the SIM card of the MS and increment by 1. After each SM submitted to the network, the last used TP-MR value in the SIM is updated with the last MR and it ranges from 0 to 255.

Reject Duplicate

TP-Reject-Duplicate is a two bit field in SMS-SUBMIT indicating whether the duplicate SMs should be handled by the SC or not. If set to "True", SC checks to find out whether the SM is a duplicate or not and if set to "False", SC just forwards the SM to the required destination without checking for duplicates.

In the process of checking duplicate what happens first is the checking of the TP-Reject-Duplicate value, to check whether duplicate identification should be performed or not. If yes then with the current process, there is a possible loophole.

There are three scenarios of different MR receiving is involved in SC

1. When MR is larger than existing MR at SC
2. When MR is lower than existing MR at SC
3. When MR is equal to the existing MR at SC

1. When the MR is larger than existing MR at SC

This is the ideal preferred situation where MR is larger than the value in the SC database, where it indicates the SM is not a duplicate. This message is allowed to pass through the duplicate message handling filter provided in the SM TP-Reject-Duplicate parameter is set to true.

2. When the MR is lower than existing MR

This is a possible situation where the MR range restricts between 0 and 255. For the SMs sent after MR = 255, it starts from 0. Because of that message reference which is lower than the existing MR is allowed to flow through the filter safely.

3. When the MR is equal to the existing MR at SC

This is the problematic situation where MR value in the received SM is equal to the MR value of the previously sent SM's MR value. According to the definition, SM is rejected since it has the same value of MR.

Previously mentioned loophole occurs with the last point mentioned. In a situation where

• MS sent a SM thorough the SC where MR = x
• MS sent exactly 254 SMs through another Service Centre (SC)
• MS sending next SM through the relevant SC

In the above situation, SM is not duplicated, but still message would be discarded, categorizing as a duplicate SM.

Solution

To avoid this problem, SCs can make use of the timestamp applied at the SM receiving end. Since the timestamp has a accuracy of one second, it can be assumed that sending 254 messages per second from an ordinary MS is not possible. Sequential application of timestamp check once duplicating MR is identification would resolve the above mentioned problem of Message Duplication of SMSCs.

Sigtran

Sigtran refers to a protocol for the transport of switched circuit network signalling protocols over an IP network. It is the evolution of SS7 which defines adapters and core transport capabilities that mix SS7 and packet protocols to provide users both the technologies

Internet dial-up remote access, IP telephony inter working with PSTN are some of the most used applications of sigtran.

Key components in sigtran achitecture are as follows:

• MGC–Media Gateway Controller, responsible for mediating call control (between the SG and MG) and controlling access from the IP world to/from the PSTN.
• SG–Signaling Gateway, responsible for interfacing to the SS7 network and passing signaling messages to the IP nodes.
• MG–Media Gateway, responsible for packetization of voice traffic and transmitting the traffic towards the destination.
• IP SCP – an IP-enabled Service Control Point (SCP). This exists wholly within the IP network, but is addressable from the SS7 network.
• IP Phone–generically referred to as a “terminal.”

The interfaces relevant to signaling transport include SG to MGC, SG to SG. Signaling transport may potentially be applied to the MGC to MGC or MG to MGC interfaces as well, depending on requirements for transport of the associated signaling protocol.

For interworking with SS7-controlled SCN networks, the SG terminates the SS7 link and transfers the signaling information to the MGC using signaling transport. The MG terminates the interswitch trunk and controls the trunk based on the control signaling it receives from the MGC. For interworking with PSTN (Public Switched Telephone Network), IP networks will need to transport signaling such ISUP messages between IP nodes such as a Signaling Gateway and Media Gateway Controller or Media Gateway.

Stream Control Transmission Protocol (SCTP), a core protocol in the SIGTRAN protocol stack, provides transport layer services over IP. Examples of such transport include: transport of signaling between a Signaling Gateway and Media Gateway or Media Gateway Controller; transport of signaling ("backhaul") from a Media Gateway to a Media Gateway Controller; transport of TCAP between a Signaling Gateway and other IP nodes. SCTP is used by one of the following User Adaption layer protocols.

• SUA: Signalling Connection Control Part User Adaptation Layer
• IUA: ISDN Q.921-User Adaptation Layer
• M3UA: SS7 Message Transfer Part 3 (MTP3) User Adaptation layer
• M2UA: SS7 Message Transfer Part 2 (MTP2) User Adaptation layer
• M2PA: MTP2 Peer-to-peer user Adaptation layer
• V5UA: V5.2-User Adaptation Layer

REST API

The questions most of us have are "What exactly is REST (or REST-ful) API and what is the history behind it?, Where did it originate?" From the apperence it seems to be just URLs that usually return XML in some particular format with words REST standing for "REpresentational State Transfer".

A simple definition to start about REST is, it's the model of the web. For example when the browser makes a request to a URL and receives the response, the request can be either GET, PUT, POST and DELETE. The response can be of any format, HTML, image, PDF, XML. The process of requests and response is called REST.

According to the above definition, another question pops up abt the difference between HTTP and REST. The baseic answer would be HTTP is a application protocol whereas REST is a style of handling resources. HTTP displays many of the traits of the REST architectural style.

Rest Data Elements

Data Element	Modern Web Examples
resource	the intended conceptual target of a hypertext reference
resource identifier	URL, URN
representation	HTML document, JPEG image
representation metadata	media type, last-modified time
resource metadata	source link, alternates, vary
control data	if-modified-since, cache-control

Following are some expmaples of REST implementation

• Amazon.com offers a "REST" version of its main 'E-Commerce' developer interface;
• eBay offers a "REST" developer interface;
• Facebook offers a "REST" developer interface;
• Yahoo! offers a number of "REST" developer interfaces.
• Youtube offers a number of "REST" developer interfaces.
• Newscloud offers a number of "REST" interfaces.
• movideo offers a number of "REST" interfaces.
• SharePoint version 2010

An application or architecture considered RESTful or REST-style is characterized by:

• State and functionality are divided into distributed resources
• Every resource is uniquely addressable using a uniform and minimal set of commands (typically using HTTP commands of GET, POST, PUT, or DELETE over the Internet)
• The protocol is client/server, stateless, layered, and supports caching

This is essentially the architecture of the Internet and helps to explain the popularity and ease-of-use for REST.

SMSC Features Comparison

. Spatch Hyde - Gorgea Argela Istanbul WicoRe India Jinny Rome Squire UK CallUp Israel Netforce Denmark Motorola ZTE China Acision Huawei Jataayu . . CDMA support . Web to Phone . Email to text . Text to email . Delivery confirmation email . SMPP interface for applications . Voice mail count sent . Load balancing . SMS Voting . TAP input (pagers, software) . Web requests . . SMS erase if not delivered . Scheduled message delivery . . Caller identity restriction . Delivery to and from alias

. Flash/ Binary SMS . Unicode multi language support . . Sore min 5Mn msgs . UI roles, marketing, support, manager etc . EMS messaging . . Number portability . . Store and forward . . Multiple telco connectivity . Dashboard Service, Database, SMPP, Route statuses Delivery performance . Reports (user, date, string match on sender, text). . Message blocking by the subscribers .

One step ahead of VoIP

Voice over IP is basically a way of communicating using analog signals that we hear on the phone through digital signals in IP network. In the practical situation this can be used by anyone connected to a IP network such as internet, making free voice calls through a free VoIP software. available. As seen on the diagram below your personal computer can work as the same way your telephone works because of the VoIP technology.

There are three different types of VoIP services which are of common use today.

ATA

This is the most common and the simplest method, using a converter which converts analog signals to digital signals. Equipment like a modem would do this. It takes analog signals and converts in to digital signals and vice versa. Normal analog signals exist in traditional telephones can be converted into digital signals using this ATA instruments. In the normal internet connection context, modem converts the data transferred using analog signals to digital signals which is interpretable by the computer.

IP Phones

This instrument looks like a normal phone but with different connector. Instead of the normal RJ-11 connector these phones comprised of the Ethernet connector RJ-45 cable and directly connected to the router. The relevant hardware and software would be associated with it in order to ensure the guaranteed service throughout.

Computer to computer

The easiest way of communication, no payments involved. As the current context there are some software entities which provides software which supports that. The famous example would by Skype.

There are many advantages of using VoIP services

More than two person conversations

Cheap use of hardware and software

Extra features such as Caller ID, Contact list, Voice mail etc

Service Delivery Platform in different Angle

In service delivery to the customer, operators should be at least consider about two things, operator involvement in the service they provide without completely droppint it out of their reach and efficent, effective service provisioning to ther users with high security which builds the ultimate confidence in users. End to end service provisioning will differenciating the service provider and will take them to a higer status.

Looking at the long term business objective, the existing service aggregation to provide an ultimate satisfactory service to the customers wouild be a good option as the trend of product suite comes in to arena recently. Technology independant architecture in in sercice provisioning would be of utmost importantce.

Rapid convergent service creation environment would be a answer for the above facts mentioned, allowing service providers to create, control and execute sercices that they provide. It helps to standardize all the service provisionings, creating a horizontal platform which they can provision, control and bill for all the value added service provisionings whether the applications are developed by the third party application developers or the inhouse teams.

Benefits of a Service Delivery Platform

• Consistent, highly automated and reusable service environment
• Positive return on investment
• Service reuse promotion resulting less application development cost and less value added service charges
• Fast application creation
• Easy integrations to operator resourses by means of APIs

A new feature which could be added to this service delivery platforms would be cloud integration. Seemless expansion, extendibility, scalability as a result of SDP and cloud integrations would take the operators to the highst block of the competitiveness by differenciating themselves in the race.

OS Clustering

In the recent past there has been many implementations of successful cluster operating systems as a result of several research projects and R&D done for commercial purposes. GL unix are such examples. Clusters are increasingly becoming the solution for many sequential and parallel applications and nowadays it is widely used in server configurations.

The primary purpose of OS clustering which also known as aka hardware clustering, is to manage hardware and OS-level failures. What happens is it automatically stats up a backup server in the os-cluster when the primary server fails in such a way that any external person using the services would notice no difference accessing the services provided. It generally involves in re-assigning the failed server IP-address to the backup (IP-takeover), re-permissioning file system access with the backup if a shared file system is being used, and then running own script written to bootup the applications required for service offering.

OS-Clustering have the following advantages

• If you have a bunch of applications which must run on the same machine, OS clustering can ensure that all these must run on the primary node in the cluster
• If your applications are dependant on "local file system" like databases need to manage their files locally - the os cluster can ensure that this file system fails over with the primary node of the cluster
• If you dont have a NAS or a file server, and you need a shared file store, you can create a share file store on the OS cluster for use by machines outside the cluster. Of course if you have a file server or are storing information in a database, then its not valid

MAP

The Mobile Application Part (MAP), is one of protocols in the SS7 suite and it allows for the implementation of mobile network signaling infrastructure. The foundation behind MAP is to connect the distributed switching elements, called mobile switching centers (MSCs) with a master database, Home Location Register (HLR). The HLR dynamically stores the current location and profile of a mobile network subscriber. HLR is consulted during the processing of an incoming call. Conversely, the HLR is updated as the subscriber moves about the network and is thus serviced by different switches within the network.

MAP messages sent between mobile switches and databases to support user authentication, equipment identification, and roaming are carried by TCAP. In mobile networks when a mobile subscriber roams into a new mobile switching center (MSC) area, the integrated visitor location register (VLR) requests service profile information from the subscriber's HLR using MAP information carried within TCAP messages.

MAP has been evolving as wireless networks grow, from supporting strictly voice, to supporting packet data services as well. The fact that MAP is used to connect Next Generation elements such as the Gateway GPRS Support Node (GGSN) and Serving Gateway Support Node (SGSN) is a evidence to the sound design of the GSM signaling sytem.

Map has several basic fuctions;

Mechanism for a Gateway MSC (GMSC) to obtain a routing number for an incoming call

• Mechanism for an MSC via integrated VLR to update subscriber status and routing number
• Subscriber CAMEL, trigger data to switching elements via the VLR
• Subscriber supplementary service profile and data to switching elements via the VLR

MAP

The Mobile Application Part (MAP), is one of protocols in the SS7 suite and it allows for the implementation of mobile network signaling infrastructure. The foundation behind MAP is to connect the distributed switching elements, called mobile switching centers (MSCs) with a master database, Home Location Register (HLR). The HLR dynamically stores the current location and profile of a mobile network subscriber. HLR is consulted during the processing of an incoming call. Conversely, the HLR is updated as the subscriber moves about the network and is thus serviced by different switches within the network.

MAP messages sent between mobile switches and databases to support user authentication, equipment identification, and roaming are carried by TCAP. In mobile networks when a mobile subscriber roams into a new mobile switching center (MSC) area, the integrated visitor location register (VLR) requests service profile information from the subscriber's HLR using MAP information carried within TCAP messages.

MAP has been evolving as wireless networks grow, from supporting strictly voice, to supporting packet data services as well. The fact that MAP is used to connect Next Generation elements such as the Gateway GPRS Support node (GGSN) and Serving Gateway Support Node (SGSN) is a evidence to the sound design of the GSM signaling system.

MAP has several basic functions:

• Mechanism for a Gateway-MSC (GMSC) to obtain a routing number for an incoming call
• Mechanism for an MSC via integrated VLR to update subscriber status and routing number.
• Subscriber CAMEL trigger data to switching elements via the VLR
• Subscriber supplementary service profile and data to switching elements via the VLR.

MAP

The Mobile Application Part (MAP), is one of protocols in the SS7 suite and it allows for the implementation of mobile network signaling infrastructure. The foundation behind MAP is to connect the distributed switching elements, called mobile switching centers (MSCs) with a master database, Home Location Register (HLR). The HLR dynamically stores the current location and profile of a mobile network subscriber. HLR is consulted during the processing of an incoming call. Conversely, the HLR is updated as the subscriber moves about the network and is thus serviced by different switches within the network.

MAP messages sent between mobile switches and databases to support user authentication, equipment identification, and roaming are carried by TCAP. In mobile networks when a mobile subscriber roams into a new mobile switching center (MSC) area, the integrated visitor location register (VLR) requests service profile information from the subscriber's HLR using MAP information carried within TCAP messages.

MAP has been evolving as wireless networks grow, from supporting strictly voice, to supporting packet data services as well. The fact that MAP is used to connect Next Generation elements such as the Gateway GPRS Support node (GGSN) and Serving Gateway Support Node (SGSN) is a evidence to the sound design of the GSM signaling system.

MAP has several basic functions:

• Mechanism for a Gateway-MSC (GMSC) to obtain a routing number for an incoming call
• Mechanism for an MSC via integrated VLR to update subscriber status and routing number.
• Subscriber CAMEL trigger data to switching elements via the VLR
• Subscriber supplementary service profile and data to switching elements via the VLR.