Signalling :- Some Important questions?

1. What is the diffetrence between circuit signaling and non circuit related signalling??
Ans.:--
Signalling was initially introduced to initiate the call process i.e. setting up and clearing down connection between trunks.Traditionally, once a circuit was set up, no other signaling was performed apart from releasing the call; therefore, all calls were simple, basic telephone service calls. However, modern telephone networks can perform signaling while a call is in progress, especially for supplementary services—for example, to introduce another called party into the call, or to signal the arrival of another incoming call (call waiting) to one of the parties.

When there is signalling when there is no call to be establish then this signalling is known as  non circuit related signalling. It is primarily used for query and response with telecommunications databases to support cellular networks, intelligent networks, and supplementary services. For example, in Public Land Mobile Networks (PLMNs), the visitor location register (VLR) that is in charge of the area into which the subscriber has roamed updates the home location register (HLR) of the subscriber's location.

Purging Time

As we know to allow roaming in a GSM  network VLR uses the temporary data of subscriber form HLR. this data remains as long as the subscriber is connected to the BSS that is  apart of that MSC/VLR area. But interesting thing is that the subscriber data remins even after the mobiel system is switched.
But the question arises, upto what time??
yes that is what Purging comes out of the box.
After an extended period of the MS being switched off, the subscription data willbe purged from the MSC.
Purging is that time , which defines the data retainment within MSC.

Multihoming

Multi homing is a method to connect a single devide with more than one IP addresses to ensure reliability in the network.

Multihoming is a mechanism used to configure one computer with more than one network interface and multiple IP addresses. It provides enhanced and reliable Internet connectivity without compromising efficient performance. The multihoming computer is known as the host and is directly or indirectly connected to more than one network.


Point to remember:-

1.) Multihoming is intended to increase the reliability of network applications but it does not necessarily improve their performance. 
2.)

Stream Control Transmission Protocol

SCTP protocols is  transport layer protocol that is same as TCP or UDp, but supports the service features of both:
1.)message-oriented like UDP and
2.)in-sequence transport of messages with congestion control like TCP.


Features

Features of SCTP include:

    Multihoming support in which 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 chunks within independent streams eliminate unnecessary head-of-line blocking, as opposed to TCP byte-stream delivery.
    Path selection and monitoring select a primary data transmission path and test the connectivity of the transmission path.
    Validation and acknowledgment mechanisms protect against flooding attacks and provide notification of duplicated or missing data chunks.
    Improved error detection suitable for Ethernet jumbo frames.

The designers of SCTP originally intended it for the transport of telephony (Signaling System 7) over Internet Protocol, 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).

Intelligent Network Application Part (INAP)

Intelligent Network Application Part (INAP) is the signaling protocol used in Intelligent Networking.INAP is a signaling protocol between a service switching point (SSP), network media resources (intelligent peripherals), and a centralized network database called a service control point (SCP). The SCP consists of operator or 3rd party derived service logic programs and data.

Address Resolution Protocol (ARP)

Address Resolution Protocol (ARP) is a protocol for mapping an Internet Protocol address (IP address) to a physical machine address that is recognized in the local network. For example, in IP Version 4, the most common level of IP in use today, an address is 32 bits long. In an Ethernet local area network, however, addresses for attached devices are 48 bits long. (The physical machine address is also known as a Media Access Control or MAC address.) A table, usually called the ARP cache, is used to maintain a correlation between each MAC address and its corresponding IP address. ARP provides the protocol rules for making this correlation and providing address conversion in both directions.
How ARP Works
When an incoming packet destined for a host machine on a particular local area network arrives at a gateway, the gateway asks the ARP program to find a physical host or MAC address that matches the IP address. The ARP program looks in the ARP cache and, if it finds the address, provides it so that the packet can be converted to the right packet length and format and sent to the machine. If no entry is found for the IP address, ARP broadcasts a request packet in a special format to all the machines on the LAN to see if one machine knows that it has that IP address associated with it. A machine that recognizes the IP address as its own returns a reply so indicating. ARP updates the ARP cache for future reference and then sends the packet to the MAC address that replied.

Since protocol details differ for each type of local area network, there are separate ARP Requests for Comments (RFC) for Ethernet, ATM, Fiber Distributed-Data Interface, HIPPI, and other protocols.

There is a Reverse ARP (RARP) for host machines that don't know their IP address. RARP enables them to request their IP address from the gateway's ARP cache.

Rehoming and resources sharing in network

A method, system, and apparatus enables automatic radio base station (RBS) rehoming from a first radio network controller (RNC) to a second RNC when the first RNC becomes overloaded and/or automatic resource sharing between/among RNCs when one RNC becomes overloaded. A managing/controlling entity receives load measuring reports from RNCs and analyzes them. If a first RNC is frequently and/or constantly experiencing an overcapacity condition, then an RBS of the RNC may be rehomed to a second RNC without operator intervention. If a first RNC temporarily experiences an overcapacity condition, then the workload of the first RNC may be shared by a second RNC also without operator intervention. When rehoming an RBS or sharing a load of a first RNC, ongoing connections are completed by the first RNC where the connection was initiated, but new connections are (for the rehoming case) or new connections may be (for load sharing cases) initiated using a second RNC.

TSC Transit Switching Center

 A number of products are available to build transit solutions:

The Transit Switching Center (also called TSC Classic) can be used to build hierarchical networks with centralized interconnect using TDM based technology.

In addition to the transit function, other optional functions can also be located in the Transit Switching Center, adding additional benefits and possibilities to create an optimized network solution:

The Gateway Mobile Switching Center (GMSC) for the interrogation of HLRService Switching Function (SSF) to interwork with an SCP nodeShort Message Service GMSC (SMS-GMSC), receiving all terminating Short Messages from the Short Message Service Centre 

Benefits

By introducing Transit Switching Centers, we can offer operators PLMNs that are easier and more cost effective to manage. Capacity can be released in the existing nodes, as the TSC is an integrated part of the PLMN.

Centralized interconnect to other PSTN/PLMNStructure large networks to interconnect regionsA more flexible network with reduced O&M costScalable capacity and comprehensive service capabilities maximize revenue potential

Circuit Switched FallBack (CSFB)

Circuit Switched FallBack (CSFB) is a technology whereby voice and SMS services are delivered to LTE devices through the use of GSM or another circuit-switched network.
Circuit Switched FallBack is needed because LTE is a packet-based all-IP network that cannot support circuit-switched calls. When an LTE device is used to make or receive a voice call or SMS, the device "falls back" to the 3G or 2G network to complete the call or to deliver the SMS text message.
CSFB was specified in 3rd Generation Partnership Project (3GPP) Release 8. CSFB requires a software upgrade of the operators core and radio network.
CSFB is often seen as an interim solution for LTE operators. Voice over LTE (VoLTE) is considered to be the long-term goal for the delivery of voice services on LTE networks.

Abbreviations in Telecom

IMS IP multimedia system
MME Mobility management identities
RAT Radio Access Technologies

Stream Control Transmission Protocol (SCTP)

High-Speed Downlink Packet Access (HSDPA) 

Universal Mobile Telecommunications System (UMTS)  

Transmission Control Protocol (TCP)
 User Datagram Protocol (UDP)

LINK TYPES

LINK TYPES
North American Signaling Network links are characterized by their
usage and grouped into the following classes:
• A-link (“access”): connects signaling end-point (for example,
local exchange or MSC in GSM) to STP
• C-link (“cross”): interconnects mated STP nodes
• B-link (“bridge”) and D-link (“diagonal”): interconnect peer
STP pairs
• E-link (“extended”): connects signaling end-point to another
STP pair for enhanced reliability
• F-link (“fully associated”): connects two signaling end-points

Circuit identification code (CIC)

In the Quasi-associated Signaling Mode the signaling messages
belonging to one call may take different paths than speech. The
speech goes directly from node A to B. The signals go via node C
which acts as a transit node. Hence node C is called the Signal
Transfer Point (STP).
The call set-up data sent in a message is not used in the STP node
C. The STP simply reroutes the messages to node B. Therefore, it is
required that the MSU carries some sort of label. The label
indicates not only the source and destination of the MSU, but also
the identity of the speech trunk the message is referring to. This
identity is called the Circuit Identity Code (CIC).

The CIC provides information about where the voice part of the call is carried - on which trunk and in which timeslot.

Integrated Services Digital Network

Integrated Services for Digital Network (ISDN) is a set of communication standards for simultaneous digital transmission of voice, video, data, and other network services over the traditional circuits of the public switched telephone network. ISDN is a circuit-switched telephone network system, which also provides access to packet switched networks, designed to allow digital transmission of voice and data over ordinary telephone copper wires, resulting in potentially better voice quality than an analog phone can provide.

D channel

D channel (delta channel) is a telecommunications term which refers to the ISDN channel in which the control and signalling information is carried.

The bit rate of the D channel of a basic rate interface is 16 kbit/s, whereas it amounts to 64 kbit/s on a primary rate interface.

For DSS1 signalling, the D channel layer 2 protocol is Q.921 also called LAPD and it carries Layer 3 messages according to Q.931 protocol

Dual-tone multi-frequency signaling

Dual-tone multi-frequency signaling (DTMF) is used for telecommunication signaling over analog telephone lines in the voice-frequency band between telephone handsets and other communications devices and the switching center. The version of DTMF that is used in push-button telephones for tone dialing is known as Touch-Tone. It was developed by Western Electric and first used by the Bell System in commerce, using that name as a registered trademark.

Signalling Tones

1. Dial Tone : Indicates that exchange is ready to accept the dialed digits  from subscriber.
2. Ringing Tone  :- The tone send to called party after establishing the call.
3. Busy Tone:-A bursty 400  Hertz signal to inform  the calling party that subscriber is busy.
4. Number unobtainable tone  :- Called party line is out of order
5. Call in Progress tone(Routing tone ):-When a call progress through no. of exchanges subscriber listens a number of tone due to different exchanges involved in the call routing.

Answer-seizure ratio

Answer-seizure ratio

The answer-seizure ratio (ASR) is a measurement of network quality[1] and call success rates in telecommunications. It is the percentage of answered telephone calls with respect to the total call volume.
Definition

In telecommunication an attempted call is termed a seizure. The answer-seizure ratio is defined as 100 times the ratio of answered calls, i.e. the number of seizures resulting in an answer signal, divided by the total number of seizures:

ASR = 100 \ \frac {answered \ calls}{seizures}

Busy signals and other call rejections by the telephone network count as call failures. However, the inclusion in the ASR accounting of some failed calls varies in practical applications. This makes the ASR highly dependent on end-user action. Low answer-seizure ratios may be caused by far-end switch congestion, not answering by called parties and busy destination circuits.

RJ -45 (Registered Jack -45)

RJ45 is a standard type of connector for network cables. RJ45 connectors are most commonly seen with Ethernet cables and networks.

RJ45 connectors feature eight pins to which the wire strands of a cable interface electrically. Standard RJ-45 pinouts define the arrangement of the individual wires needed when attaching connectors to a cable.

Several other kinds of connectors closely resemble RJ45 and can be easily confused for each other. The RJ-11 connectors used with telephone cables, for example, are only slightly smaller (narrower) than RJ-45 connectors.

OSI model

Partitioning the whole network structures into the abstract layers ISO established a 7 layered architecture named as :-------

7. Application  Network process to application
6. Presentation Data representation, encryption and decryption, convert machine dependent data to machine independent data
5. Session -    Interhost communication, managing sessions between applications
4. Transport     Reliable delivery of packets between points on a network.
3. Network     Addressing, routing and (not necessarily reliable) delivery of datagrams between points on a network.
2. Data link     A reliable direct point-to-point data connection.
1. Physical     A (not necessarily reliable) direct point-to-point data connection.

Confusions to confuse you???

>>The Session Initiation Protocol (SIP) is a protocol related to which layer??
 A) Session
B) Application

Answer -- Application

What is RS in RS-232 ?

RS-232 is defined as the. “Interface between data terminal equipment and data communications equipment using serial binary data exchange.

RS-232 (Recommended Standard - 232) is a telecommunications standard for binary serial communications between devices.

Global Title vs. Point Code

GT AND SPC are using for addressing and routing of the message in SS7 network. But GT is only used for routing messages at SCCP level. GT is formated as national number.


In simple terms Global Title is used at SCCP level in a SS7 stack and Point code is used at MTP level.
GT is an unique identity of a Network Element while Point code is unique with in a particular SS7 network.
lets say for example : there are two Telecom operators Airtel, Idea . no two GT will be same in these two networks but we can have same Point codes in both networks.

Signalling System No. 7

Signalling System No. 7 (SS7)

> Not  a single protocol but a set of  telephony signaling protocols which are being used to set up telephone calls.

>The main purpose is to set up and tear down telephone calls.

>Other uses include number translation, local number portability, prepaid billing mechanisms, short message service (SMS), and a variety of other mass market services.

Validity period of an SMS message

An SMS message is stored temporarily in the SMS center if the recipient mobile phone is unavailable. It is possible on most mobile handset to specify an expiry period after which the SMS message will be deleted from the SMS center. Once deleted, the SMS message will no longer be available for dispatch to the recipient mobile phone (even if it becomes online). The validity period should be regarded by the handset user as a request, as the SMSC itself can be configured to ignore or otherwise handle message delivery schedules.

Short message service center(SMSC)

A short message service center (SMSC) is a network element in the mobile telephone network. Its purpose is to store, forward, convert and deliver SMS messages.
The full designation of an SMSC according to 3GPP is Short Message Service - Service Centre (SMS-SC)
Basic Trajectories of SMS are

1. From Mobile to another Mobile, this trajectory is referred as MO-MT (Mobile Originated - Mobile Terminated)
2. From Mobile to a content provider (also called as Large Account / ESME), this trajectory is referred as MO-AT (Mobile Originated - Application Terminated)
3. From Application to a Mobile, this trajectory is referred as AO-MT (Application Originated - Mobile Terminated)

An SMS center (SMSC) is responsible for handling the SMS operations of a wireless network.

1. When an SMS message is sent from a mobile phone, it will reach an SMS center first.
2. The SMS center then forwards the SMS message towards the destination.
3. The main duty of an SMSC is to route SMS messages and regulate the process. If the recipient is unavailable (for example, when the mobile phone is switched off), the SMSC will store the SMS message.
4. It will forward the SMS message when the recipient is available or when the message's expiry period is exceeded.