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1) Why telecom power supply has -48V dc power supply?
• • • Negative sign indicates The positive grounded

he negative voltage on the line was better than positive to prevent electro-chemical reactions from destroying the copper cable quickly, if cables get wet Low enough not to cause serious danger if somebody touches the telephone wires

• • •

2)In 1989, AT&T introduced its 900 services to the general public. What was its 900 number service option called?
MultiQuest 3) What is the need of CPG message in ISUP protocol? Consider any call fowarding scenario.... As soon as switch receives Setup, it checks that the Called_Party has forwarded option activated and so tries to forward the call. Meanwhile, Switch sends CPG message towards originator saying that Call is in Progress. A message, sent in either direction during the setup or active phase of the call(ex.inbetween ACM and ANM), indicating that an event, which is of significance, and should be relayed to the originating or terminating access, has occurred. 4) What is a superheterodyne receiver? superhetrodyne receiver is one which has same carrier frequency as the transmitter otherwise the original signal recovery will not be possible. 5) What is protocol testing ? Types of tool used in telecom testing ? Protocol testing means to test the functionality of the node(piece of software) which should compliance to some standard message flow. For this one should take a tester(Testing Unit)

which should send standard messages to the node(Item under test) Different types of telecom testing Tools used GlomoSim simulator: It has the ability to test 100 nodes on single cpu and for thousands of nodes multiprocessor. Abacus5000 for SIP testing Etherpeek IP Nethawk SS7 Wireshark SS7/IP K1297-G20(Tektronics) 6) What is MTU ? In computer networking, the term Maximum Transmission Unit (MTU) refers to the size (in bytes) of the largest packet that a given layer of a communications protocol can pass onwards. MTU parameters usually appear in association with a communications interface (NIC, serial port, etc.). The MTU may be fixed by standards (as is the case with Ethernet) or decided at connect time (as is usually the case with point-to-point serial links). A higher MTU brings higher bandwidth efficiency. However large packets can block up a slow interface for some time, increasing the lag on other packets. 7) A 2MB PCM(pulse code modulation) has how many channels? 30 voice channels, 1 signaling channel, & 1 synchronization channel. 8) What is the nominal voltage required in subscriber loop connected to local exchange? - 48 volts. This -48 volts is required by the copper wires to prevent it from corroded & eluded 9) VOICE is sampled at which Frequency? 8 KHz as the sampling frequency will be double the maximum frequency component for no aliasing effect, as per the nyquist theorm. as the maximum frequency allowed by the filter is just 4khz. so the sampling frequency is 8KHz 9) how many Max number of satellite hops allowed in voice communication? 2 hops 10) what is the significance of SGM (segmentation) message in ISUP? When messages exceed the maximum size of the SS7 packet (272 octets) the message must be segmented. This message type is used to send an additional segment to the destination signaling point. 11)Protocol used in Google Talk.

a)SIP b)H.323 c)XSALT d)XMPP __________________ Re: What is the difference between IAM and SAM? Answer #2

No if IAM is more that 272 octect it will be fragmented but name of that message is segmentation. SAM is used for overlap calls. When user dials some digits and SS7 node can find the next hop without receiving all the digits from user, in this case ss7 will send the IAM with those digits and other dialled digits will be sent in SAM.
Re: If there is no RLC in response to REL what will the intermediate switch will do? Answer #2

Hi This is Shashank, after the Time out there is no RELC then the Switch Will hold the Call unless and untill the (I) in Progress Call Again asks for the RELC. Then after the Time Set by the Switch Timer of the Call is released or made released by the Switch.
Re: WHAT IS IPDSLAM AND WHAT IS FUNCTION OF IT. Answer #1

A Digital Subscriber Line Access Multiplexer (DSLAM, often pronounced dee-slam) allows telephone lines to make faster connections to the Internet. ...
)

Re: WHAT IS IPDSLAM AND WHAT IS FUNCTION OF IT.

There are two types of DSLAMs. 1. ATM DSLAM works with fibre optics ATM technology. 2. IPDSLAM work in ip technology.
What are the different functionalities of MTP3 layer? Answer #3

The MTP3 Layer sits below the UP (User Parts) and is known as the network management functions. It offers two types of functions: 1- Message handling functions: Use the label (DPC, OPC and SLS) for: - Message distribution: This function is used to distribute the message to the proper UP by means of checking the destination of the message. - Message discrimination: This function checks if the message if the SP is indeed the right receptor of the message - Message routing: Used in order to determine the signaling link on which the message should be routed to be sent to the destination 2- Network Management functions: Are used to - Restore failed links - Activate idle links - Deactivate aligned links - Maintaining service if signaling links or SPs are lost and in case of congestion (Using Changeover, Changeback, forced rerouting, controlled rerouting and signaling traffic flow control)

Elaborate ss7 stack Answer #1

hope the below link will be useful to know the ss7 stack http://www.cisco.com/en/US/products/hw/switches/ps2 246/produ cts_preinstallation_guide_chapter09186a00800ea6d2.html
What is the difference between SCCP and MTP3? Answer #1

While MTP L3 can only identify and distribute messages to specific points and user parts, SCCP can route messages to specific applications that reside at signaling points.(with help of GTT)
What is the difference between SCCP and MTP3? Answer #4

regardless the fact that MTP3 as well as SCCP both layers are responsible for the routing of the message. the only difference is that SCCP is doing end to end routing where MTP3 is doing point to point routing.

Signaling System No 7
From Wikipedia, the free encyclopedia

SS7 protocol suite
OSI layer Application Network Data link Physical TCAP, CAP, ISUP, ... MTP Level 3 + SCCP MTP Level 2 MTP Level 1 SS7 protocols INAP, MAP, IS-41...

Signalling System No. 7 (SS7) is a set of telephony signaling protocols which are used to set up most of the world's public switched telephone network 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. It is usually referenced as Signalling System No. 7 or Signalling System #7, or simply abbreviated to SS7. In North America it is often referred to as CCSS7, an acronym for Common Channel Signalling System 7. In some European countries, specifically the United Kingdom, it is sometimes called C7 (CCITT number 7) and is also known as number 7 and CCIS7 (Common Channel Interoffice Signaling 7).

There is only one international SS7 protocol defined by ITU-T in its Q.700-series recommendations.[1] There are however, many national variants of the SS7 protocols. Most national variants are based on two widely deployed national variants as standardized by ANSI and ETSI, which are in turn based on the international protocol defined by ITU-T. Each national variant has its own unique characteristics. Some national variants with rather striking characteristics are the China (PRC) and Japan (TTC) national variants. The Internet Engineering Task Force (IETF) has also defined level 2, 3, and 4 protocols that are compatible with SS7: • • • MTP2 (M2UA and M2PA) MTP3 (M3UA) Signalling Connection Control Part (SCCP) (SUA)

but use a Stream Control Transmission Protocol (SCTP) transport mechanism. This suite of protocols is called SIGTRAN.

Contents
[hide] • • • • • • 1 History 2 Functionality o 2.1 Signaling modes 3 Physical network 4 SS7 protocol suite 5 References 6 External links

[edit] History
Common Channel Signaling protocols have been developed by major telephone companies and the ITU-T since 1975 and the first international Common Channel Signaling protocol was defined by the ITU-T as Signalling System No. 6 (SS6) in 1977.[2] Signalling System No. 7 was defined as an international standard by ITU-T in its 1980 (Yellow Book) Q.7XX-series recommendations.[1] SS7 was designed to replace SS6, which had a restricted 28-bit signal unit that was both limited in function and not amenable to digital systems.[2] SS7 has substantially replaced SS6, Signalling System No. 5 (SS5), R1 and R2, with the exception that R1 and R2 variants are still used in numerous nations. SS5 and earlier systems used in-band signaling, in which the call-setup information was sent by playing special multi-frequency tones into the telephone lines, known as bearer channels in the parlance of the telecom industry. This led to security problems with blue boxes. SS6 and SS7 implement out-of-band signaling protocols, carried in a separate signaling channel,[3] explicitly keep the end-user's audio path—the so-called speech path—separate from the signaling phase to eliminate the possibility that end users may introduce tones that would be mistaken for those used for signaling. See falsing. SS6 and SS7 are referred to as so-called Common Channel Interoffice Signalling Systems (CCIS) or Common Channel Signaling (CCS) due to their hard separation of signaling and bearer channels. This required a separate channel dedicated solely to signaling, but the greater speed of signaling decreased the holding time of the bearer channels, and the number of available channels was rapidly increasing anyway at the time SS7 was implemented. The common channel signaling paradigm was translated to IP via the SIGTRAN protocols as defined by the IETF. While running on a transport based upon IP, the SIGTRAN protocols are not an SS7 variant, but simply transport existing national and international variants of SS7.[4]

[edit] Functionality
The term signaling, when used in telephony, refers to the exchange of control information associated with the establishment of a telephone call on a telecommunications circuit.[5] An example of this control information is the digits dialed by the caller, the caller's billing number, and other call-related information. When the signaling is performed on the same circuit that will ultimately carry the conversation of the call, it is termed Channel Associated Signaling (CAS). This is the case for earlier analogue trunks, MF and R2 digital trunks, and DSS1/DASS PBX trunks. In contrast, SS7 signaling is termed Common Channel Signaling (CCS) in that the path and facility used by the signaling is separate and distinct from the telecommunications channels that will ultimately carry the telephone conversation. With CCS, it becomes possible to exchange signaling without first seizing a facility, leading to significant savings and performance increases in both signaling and facility usage. Because of the mechanisms used by signaling methods prior to SS7 (battery reversal, multi-frequency digit outpulsing, A- and B-bit signaling), these older methods could not communicate much signaling information. Usually only the dialed digits were signaled, and only during call setup. For charged calls, dialed digits and charge number digits were outpulsed. SS7, being a high-speed and high-performance packet-based communications protocol, can communicate significant amounts of information when setting up a call, during the call, and at the end of the call. This permits rich call-related services to be developed. Some of the first such services were call management related services that many take for granted today: call forwarding (busy and no answer), voice mail, call waiting, conference calling, calling name and number display, call screening, malicious caller identification, busy callback.[6] The earliest deployed upper layer protocols in the SS7 signaling suite were dedicated to the setup, maintenance, and release of telephone calls.[7] The Telephone User Part (TUP) was adopted in Europe and the Integrated Services Digital Network (ISDN) User Part (ISUP) adapted for Public Switched Telephone Network (PSTN) calls was adopted in North America. ISUP was later used in Europe when the European networks upgraded to the ISDN. (North America never accomplished full upgrade to the ISDN and the predominant telephone service is still the older POTS.) Due to its richness and the need for an out-of-band channel for its operation, SS7 signaling is mostly used for signaling between telephone switches and not for signaling between local exchanges and customer-premises equipment (CPE). Because SS7 signaling does not require seizure of a channel for a conversation prior to the exchange of control information, Non-Facility Associated Signalling (NFAS) became possible. NFAS is signaling that is not directly associated with the path that a conversation will traverse and may concern other information located at a centralized database such as service subscription, feature activation, and service logic. This makes possible a set of network-based services that do not rely upon the call being routed to a particular subscription switch at which service logic would be executed, but permits service logic to be distributed throughout the telephone network and executed more expediently at originating switches far in advance of call routing. It also permits the subscriber increased mobility due to the decoupling of service logic from the subscription switch. Another characteristic of ISUP made possible by SS7 with NFAS is the exchange of signaling information during the middle of a call.[5] Also possible with SS7 is Non-Call-Associated Signaling, which is signaling that is not directly related to the establishment of a telephone call.[8] An example of this is the exchange of the registration information used between a mobile telephone and a Home Location Register (HLR) database: a database that tracks the location of the mobile. Other examples include Intelligent Network and local number portability databases.[9]

[edit] Signaling modes

As well as providing for signaling with these various degrees of association with call set up and the facilities used to carry calls, SS7 is designed to operate in two modes: Associated Mode and QuasiAssociated Mode.[10] When operating in the Associated Mode, SS7 signaling progresses from switch to switch through the PSTN following the same path as the associated facilities that carry the telephone call. This mode is more economical for small networks. The Associated Mode of signaling is not the predominant choice of modes in North America.[11] When operating in the Quasi-Associated Mode, SS7 signaling progresses from the originating switch to the terminating switch, following a path through a separate SS7 signaling network composed of Signal Transfer Points. This mode is more economical for large networks with lightly loaded signaling links. The Quasi-Associated Mode of signaling is the predominant choice of modes in North America.[12]

[edit] Physical network
SS7 is an out-of-band signaling protocol, i.e. seperate from the bearer channels that carry the voice or data. This seperation extends onto the physical network by having circuits that are solely dedicated to carrying the SS7 links. Clearly splitting the signaling plane and voice circuits. An SS7 network has to be made up of SS7-capable equipment from end to end in order to provide its full functionality. The network is made up of several link types (A, B, C, D, E, and F) and three signaling nodes - Service switching point (SSPs), signal transfer point (STPs), and Service Control Point (SCPs). Each node is identified on the network by a number, a point code. Extended services are provided by a database interface at the SCP level using the SS7 network. The links between nodes are full-duplex 56, 64, 1,536, or 1,984 kbit/s graded communications channels. In Europe they are usually one (64 kbit/s) or all (1,984 kbit/s) timeslots (DS0s) within an E1 facility; in North America one (56 or 64 kbit/s) or all (1,536 kbit/s) timeslots (DS0As or DS0s) within a T1 facility. One or more signaling links can be connected to the same two endpoints that together form a signaling link set. Signaling links are added to link sets to increase the signaling capacity of the link set. In Europe, SS7 links normally are directly connected between switching exchanges using F-links. This direct connection is called associated signaling. In North America, SS7 links are normally indirectly connected between switching exchanges using an intervening network of STPs. This indirect connection is called quasi-associated signaling. Quasi-associated signaling reduces the number of SS7 links necessary to interconnect all switching exchanges and SCPs in an SS7 signaling network.[13] SS7 links at higher signaling capacity (1.536 and 1.984 Mbit/s, simply referred to as the 1.5 Mbit/s and 2.0 Mbit/s rates) are called High Speed Links (HSL) in contrast to the low speed (56 and 64 kbit/s) links. High Speed Links (HSL) are specified in ITU-T Recommendation Q.703 for the 1.5 Mbit/s and 2.0 Mbit/s rates, and ANSI Standard T1.111.3 for the 1.536 Mbit/s rate. There are differences between the specifications for the 1.5 Mbit/s rate. High Speed Links utilize the entire bandwidth of a T1 (1.536 Mbit/s) or E1 (1.984 Mbit/s) transmission facility for the transport of SS7 signaling messages.[14] SIGTRAN provides signaling using SCTP associations over the Internet Protocol.[15] The protocols for SIGTRAN are M2PA, M2UA, M3UA and SUA.

[edit] SS7 protocol suite
SS7 protocol suite
OSI layer Application TCAP, CAP, ISUP, ... SS7 protocols INAP, MAP, IS-41...

Network Data link Physical

MTP Level 3 + SCCP MTP Level 2 MTP Level 1

The SS7 protocol stack borrows partially from the OSI Model of a packetized digital protocol stack. OSI layers 1 to 3 are provided by the Message Transfer Part (MTP) and the Signalling Connection Control Part (SCCP) of the SS7 protocol (together referred to as the Network Service Part (NSP)); for circuit related signaling, such as the Telephone User Part (TUP) or the ISDN User Part (ISUP), the User Part provides layer 7. Currently there are no protocol components that provide OSI layers 4 through 6.[1] The Transaction Capabilities Application Part (TCAP) is the primary SCCP User in the Core Network, using SCCP in connectionless mode. SCCP in connection oriented mode provides the transport layer for air interface protocols such as BSSAP and RANAP. TCAP provides transaction capabilities to its Users (TC-Users), such as the Mobile Application Part, the Intelligent Network Application Part and the CAMEL Application Part. The Message Transfer Part (MTP) covers a portion of the functions of the OSI network layer including: network interface, information transfer, message handling and routing to the higher levels. Signalling Connection Control Part (SCCP) is at functional Level 4. Together with MTP Level 3 it is called the Network Service Part (NSP). SCCP completes the functions of the OSI network layer: end-to-end addressing and routing, connectionless messages (UDTs), and management services for users of the Network Service Part (NSP).[16] Telephone User Part (TUP) is a link-by-link signaling system used to connect calls. ISDN User Part (ISUP) is the key user part, providing a circuit-based protocol to establish, maintain, and end the connections for calls. Transaction Capabilities Application Part (TCAP) is used to create database queries and invoke advanced network functionality, or links to Intelligent Network Application Part (INAP) for intelligent networks, or Mobile Application Part (MAP) for mobile services. International Point Code Numbering International Signalling Point Codes (ISPCs) are 14-bit binary codes used to establish direct SS7 signalling links and interconnection with overseas networks. The 14 bits of the ISPC are commonly represented by three decimal numbers (e.g. 5-047-0): 1. 2. 3. the first decimal, with the range of 0 to 7, represents the three (3) most significant bits; It identifies world geographical zone the second decimal string, with the range of 000 to 255 represents the following eight (8) bits; The sub-field of 8 bits identify a geographical area or network in a specific zone. and the third decimal, with the range of 0 to 7, represents the three least significant bits. The sub-field of 3 bits (CBA) should identify a signalling point in a specific geographical area or network. The combination of the first and second sub-fields could be regarded as a signalling area/network code (SANC).

14 bit Point code = <3 bits World Geographic Zone><8 bit Geographical Area or Network><3 bits Signalling Nodes identifier in a specific geographical area or network>

Following are rules for International Point Codes allocation as defined by CCITT : 1. 2. 3. 4. Each country (or geographical area) should be assigned at least one signalling area/network code (SANC). Two of the zone identifications, namely 0 and 1 codes, are reserved for future allocation. The system of international signalling point codes (ISPC) will provide for 6 * 256 * 8 (12288) ISPCs. If a country (or geographical area) should require more than 8 international signalling points, one or more additional signalling area/network code(s) (SANC) would be assigned to it. The assignment of signalling area/network codes (SANC) is to be administered by the CCITT. The assignment of signalling point identifications in the sub-field (CBA) will be made by each

5.

country (or geographical area) and the CCITT Secretariat notified.

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