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Link Access Protocol Channel D

The objective of LAPD is to provide a secure, error-free connection between two end-points so as to
reliably transport Layer 3 messages. LAPD protocol provides framing, sequence control, error
detection, and recovery of multiple logical data links on the same D channel. The Layer 2 operation
starts with a Layer 3 request to establish a call, which prompts Layer 2 to demand Layer 1 to initiate
the start-up procedure. Afterwards Layer 2 initializes itself through the SABME ( Set Asynchronous
Balanced Mode Extended) procedure. The SABME is conducted between two peer processes by means
of unnumbered frames. Initialization guarantees that the correct sequence numbers are used in
processes on both ends.
Layer 2 protocol maintain the correct sequence of information due to many different types of error.
Firstly, if frames lost more than three consecutive times, the re-establishment of the LAP will be done
under the assumption that the connection has failed. Secondly, a timer mechanism and preset values
of time-out intervals detect lost frames in the transmit direction. Thirdly, frames received out of
sequence denote a lost frame in the receive direction. Finally, frame Check Sequence(FCS) value
control and discard frames corrupted by errors due to noise.
The LAPD protocol involves two major functions:
1. The bit-level protocol that deals with such operations as framing and cyclic redundancy error
checking ( CRC); and
2. The element of procedure that deals with the handshaking between peer entities at both
source and destination terminal equipment.

Framing in Data Transmissions:
A point-to-point connection between two computers or devices consists of a wire in which
data is transmitted as a stream of bits. However, these bits must be framed into visible
blocks of information. Framing is a function of the data link layer. It provides a way for
a sender to transmit a set of bits that are meaningful to the receiver. Ethernet, token ring,
frame relay, and other data link layer technologies have their own frame structures. Frames
have headers that contain information such as error-checking codes.
The advantage of using frames is that data is broken up into recoverable chunks
that can easily be checked for corruption. A glitch in the line during the transmission
will corrupt some frames. Only the lost frames and not the entire set of data needs to be
retransmitted.
The general format for frames is a header followed by a data payload area. The frame size
is usually fixed, while the data area is variable. Most data link networks use variable-size
framing, which has advantages (more efficient use of the network) and disadvantages
(unpredictable traffic flows and the inability to provide quality of service).
The preamble contains a set of bits that help the sender and receiver synchronize their
transmissions. The Ethernet bit pattern is 10101010, which produces a 10-MHz square wave
clocking signal for 5.6 microseconds. The SFD (start frame delimiter) indicates the actual
start of the frame. The destination and source addresses are self-explanatory. The LEN
(length) field indicates the length of the data portion of the frame. PAD (padding) is used for
fill-in bits. CRC (cyclic redundancy checksum) is used for error checking.
(BVCI) BSSGP Virtual Connection Identifier is used to identify the virtual circuits used in Gb
interface in carrying data between BSS and SGSN (Serving GPRS Support Node). It can point to
a cell, PTM entity or signaling entity.

The User Plane in charge of transporting the various users information (voice, data or video) to their
destination, according to the subscribed traffic contracts.
It relies on the ATM layer for multiplexing and switching, with guaranteed Quality of Services (QOS).
For that purpose, the user information (data protocol or particular media) is converted to ATM via the
"User Protocol or Media Adaptation" layer and the "ATM Adaptation Layer (AAL)".
The Control Plane for the connection set up and release according to subscribed traffic contracts.
Various types of connections are supported: symmetrical or asymmetrical point to point, multipoint,
multicast, unicast, etc.). It also uses the AAL, ATM and physical layers. A dedicated layer is used to
map the "control protocol" onto ATM in order to ensure high reliability.
The Management Plane for ATM monitoring and configuration.
Whats the difference between VRRP and HSRP?
Does VRRP sound a little like Hot Standby Router Protocol (HSRP)? There are many
similarities, such as load balancing and redundancy, but the greatest difference is that HSRP is a
Cisco proprietary method whereas VRRP is an industry standard (based on RFC 2338). Both of
these have some minor technical differences, but the resulting functionality is the same.
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: Signaling 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
- SIGTRAN: Signaling Transport Protocols


SIGTRAN Architecture

Signal Transfer Protocol:
SIGTRAN developed from Signaling System 7, or SS7. SS7 sets up calls and enables
transaction services such as caller ID. SIGTRAN takes it a step further, delivering more
calling functions. SIGTRAN has four components: Media Gateway Controller, the
Signaling Gateway, the Media Gateway and the Internet Protocol-Based Service
Control Point. Modern phone systems use SIGTRAN.





Erlang - a unit of traffic


An Erlang is a unit of telecommunications traffic measurement. Strictly speaking, an Erlang represents
the continuous use of one voice path. In practice, it is used to describe the total traffic volume of one
hour.
For example, if a group of user made 30 calls in one hour, and each call had an average call
duration of 5 minutes, then the number of Erlangs this represents is worked out as follows:
Minutes of traffic in the hour = number of calls x duration
Minutes of traffic in the hour = 30 x 5
Minutes of traffic in the hour = 150
Hours of traffic in the hour = 150 / 60
Hours of traffic in the hour = 2.5
Traffic figure = 2.5 Erlangs
Erlang traffic measurements are made in order to help telecommunications network designers
understand traffic patterns within their voice networks. This is essential if they are to
successfully design their network topology and establish the necessary trunk group sizes.
Erlang traffic measurements or estimates can be used to work out how many lines are required
between a telephone system and a central office (PSTN exchange lines), or between multiple
network locations.


Erlang traffic models


Several traffic models exist which share their name with the Erlang unit of traffic. They are formulae
which can be used to estimate the number of lines required in a network, or to a central office (PSTN
exchange lines). A formula also exists to model queuing situations, and lends itself well to estimating the
agent staffing requirements of call centers.
The main Erlang traffic model are listed below, with links to the free online calculators on this
Web site:
Erlang B
This is the most commonly used traffic model, and is used to work out how many lines are
required if the traffic figure (in Erlangs) during the busiest hour is known. The model assumes
that all blocked calls are immediately cleared.
Extended Erlang B
This model is similar to Erlang B, but takes into account that a percentage of calls are
immediately represented to the system if they encounter blocking (a busy signal). The retry
percentage can be specified.
Erlang C
This model assumes that all blocked calls stay in the system until they can be handled. This
model can be applied to the design of call center staffing arrangements where, if calls cannot be
immediately answered, they enter a queue.

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