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1. NETWORK ANALYSIS, ARCHITECTURE AND
DESIGNING
Course No: EETS 8331
Prof: Klyne Smith
Level 3- Service Control Point
Final Project-Team D
Name SMU ID
Shreyas Parmar 47214314
Udhav Pawar 47149366
Rohit Shriyan 47200352
Harsh Shah 47185935
Shail Shah 39714380
2. CONTENTS
1. Executive Summary
2. Frame Work
3. Customer Business Requirement
4. User Requirement
5. System Requirement
6. System Concept Diagram
7. Requirement Traceability Matrix
8. Test Cases
9. Risk Analysis and Mitigation
3. 1.Executive Summary
Thank you for allowing us (Project Team D) the opportunity to present our response to
Service Control Point for Level 3 Core Network’s RFP and to demonstrate our commitment
in assisting Level 3 Communications with this strategically important compliance project.
It is understood that Level 3 Communications is a reputed American Multinational
Telecommunications & Internet Service provider company, which is currently working to
provide Core transport, IP, Voice, Video & Content delivery for Internet carriers around the
world. Level 3 is also the largest Competitive Local Exchange Carrier (CLEC) & the 3rd
largest provider of Fibre Optic Internet access in the United States. Level 3 Communications
currently delivers Netflix & Apple Inc. music & video content over the Internet.
The Objective of this proposal is to showcase our understanding of your requirements & how
we will offer the solution efficiently & cost effectively. We have attempted to integrate the
information we have received from Level 3 Communications & we are committed to Level 3
Communication’s overall goals to provide a Service Control Point (SCP) function. We
understand that Service Control Point function should be able to integrate the services
currently provided by the Consolidated SCP. Also the C-SCP will be able to allow Level 3 to
make changes to the current services and also the new services that will be created in future.
We do plan to incorporate C-SCP with all the equipment that are necessary for supporting
Service creation, Service management, Service control, & media management.
Along with the above services, C-SCP will communicate directly via Session Initiation
Protocol (SIP) using the existing Level 3 Core routing engine & media gateways, also other
Application Servers, Class 4 Soft switches, Media Servers & SIP enabled data devices in the
Level 3 Core Voice network. The Solution will also provide self-service functionality to the
customers.
We appreciate the opportunity to provide this response where we have built our project
approach, work plan, cost estimates based upon the identified project needs and within the
timeline and manner requested.
5. 3. Customer Business Requirement
Implementing existing services.
Capable modifying the Services it manages or create new services and functionality
through Service Creation.
Capable of creating new services of modifying existing services or create new
services without the need to consult the Vendor.
Provisioning Portal to enable self-service functionality.
Number Translation
Information Delivery
Screening
Caller Interaction
Mid-Call Interaction
6. 4. User Requirement
User Requirement
Number
User Requirement Description
UR-1 All equipment vendors shall provide Airflow
documentation of test results in a densely populated
environment showing the heat dissipation cooling
characteristic of the chassis.
The system chassis shall be accessible via a local RS232
craft to be used by terminal, PC, or telemetry equipment.
It is desirable that access ports for this type of access be
available on both the front and back of the chassis.
UR-2 All line interface cards shall have LED indicators to
indicate when the line card has experienced a failure.
All cards shall have LED indicators to indicate when the
card software is loading or initializing.
C-SCP components shall have LED indicators on the
front panel, one for each alarm type (critical, major ,
minor and warning. ) The LED indicators will show
when there any of the above alarm types active
(standing) in the system.
In the event of a failure of one IP interface, the C-SCP
should continue functioning.
C-SCP shall support a local memory backup capability
that protects provisioned parameters and management
information in case of system failure.
This local memory backup shall be non-volatile.
C-SCP shall require that all management parameters and
operation is immediately restorable upon failure.
C-SCP shall support the ability to recover from the loss
of a single network interface card with no loss of
performance.
C-SCP shall immediately alarm when an IP port virtual
port or physical interface) has gone out of service.
C-SCP shall maintain and monitor the performance of all
internal bus facilities within each platform.
UR-3 Component hot swap-ability.
UR-4 The management IP interfaces shall support both ping
and traceroute functionality.
C-SCP shall support at least two Ethernet ports for
management access.
UR-5 C-SCP shall provide a means to query the device for all
the installed hardware, including chassis, fans disks,
cards, etc.
UR-6 The change in operational state of the system, card, or
7. interface will trigger an SNMP trap, detailing the reason.
UR-7 C-SCP will provide this information on a call-by-call
basis in the call-logging section.
UR-8 C-SCP will be able to report statistics in the following
formats: traffic and rates per time (bps, Kbps, Mbps, tps
cps, etc.), throughput (packets per second, Kpps and
Mpps), CPU utilization (percentage), memory and buffer
utilization.
C-SCP must generate a warning alarm when CPU/DSP
reaches 50% usage.
C-SCP must generate a warning alarm when CPU/DSP
reaches 80% usage.
C-SCP must generate a warning alarm when CPU/DSP
reaches 90% usage.
UR-9 C-SCP must support configurable threshold levels for
disk capacity alarms for minor, major and critical levels.
UR-10 Vendor should describe the SIP/RTP capture
functionality necessary for monitoring and
troubleshooting purposes.
UR-11 C-SCP shall have at least two IP interface cards that
share the network.
Power supplies shall be redundant
C-SCP must have DC power supplies for all HW
components.
UR-12 C-SCP must have a failover reliability of 99% or better.
C-SCP must completely failover within 2 seconds.
C-SCP must be able to completely failover within the
time specified during peak busy hour.
UR-13
C-SCP shall support remotely downloading software
upgrades and patches.
C-SCP shall be capable of performing software updates
and configuration changes without dropping established
calls and continue to process new call requests.
C-SCP must provide live software updates on all
components while continuing to process calls without
any degradation in performance.
UR-14 C-SCP must be able to process at least 5000 call
transactions/attempts per second at sustained rate for at
least 3 concurrent hours when operating in simplex
mode.
C-SCP shall support 200,000 simultaneous calls in
stateful proxy mode.
C-SCP shall support 10,000 simultaneous caller
interaction calls.
UR-15 C-SCP must be able to send and receive SIP-URIs in
globalized E.164 format for Globally Unique TN
(excepting 911 in the Request-URI and To Header).
8. Example sip:+14155551212@192.168.1.1
C-SCP should be able to send and receive SIP-URIs in
non-E.164 formats for non- Unique TN, such as DNIS .
UR-16 Multiple Access Levels per Login access through
TACACs or RADIUS.
Logging/debugging will be available via console and
terminals.
UR-17 Testing of the IP interface media using time domain
reflectometry or loopback tests.
It should support the ability for a managing system to
send an IP ping and traceroute to or from a destination C-
SCP to test network connectivity.
UR-18 C-SCP shall have the capability to store basic
configuration information for each virtual session (virtual
IP address and port pair) it can support.
UR-19 C-SCP shall have the capability to store basic
configuration information for each virtual session (virtual
IP address and port pair) it can support.
It must have a CLI for accessing configuration files.
The configuration files must be in ASCII text and display
in the exact same manner which they are inputted into
the system (i.e., one must be able to copy a display of the
configuration and paste it into a C-SCP).
C-SCP will have non-volatile storage onboard (either
removable or internal) with enough capacity to store
multiple software images and configurations.
It must support TFTP for receiving all software images
and configurations.
9. 5. System Requirement
Requirement
Number
System Requirement Description
SR-1 Design Requirement ➢ C-SCP cooling airflow should flow
from front bottom to back top.
➢ The individual chassis height shall not
exceed 75 inches, including any space
required
➢ In the event that multiple chassis are
installed in one rack, the combined
height shall also not exceed 75 inches.
➢ Cable connections to system interface
cards shall be on the back of the cards.
➢ C-SCP components shall have
Electrostatic Discharge (ESD) jacks on
the front and back.
SR-2 Power requirement ➢ The power supplies rating should be -48
VDC; however, they should have an
operating range of -40
➢ The DC power supplies rating may be
380VDC
➢ C-SCP shall support an option for AC
power supplies for all HW components
➢ The SC power supply rating shall be
220VAC
➢ C-SCP should not consume more than
500 watts per Rack Unit when fully
loaded.
➢ C-SCP should not consume more than
18.8 amps per Rack Unit when fully
loaded.
➢ C-SCP should have power switches or
breakers on the DC power supplies.
➢ Switchover of power supply modules
shall not cause disruption of service.
SR-3 Monitoring and Reporting ➢ C-SCP shall maintain and monitor call
processing
➢ C-SCP shall use a unique call identifier
to identify any call attempt.
➢ C-SCP shall respond to queries from
10. Element Management System (EMS)
on the availability of its interfaces.
➢ C-SCP must provide viewing/report
capability accessible to users.
➢ C-SCP shall support generation of
reports.
➢ C-SCP reports supported reports should
include:
○ Configuration Data
○ Provisioning Data
○ User/Security Data
○ System Health Statistics
○ System Performance Statistics
○ Call Statistics
○ Call Detail Records
SR-4 Service supportability
➢ C-SCP must support the Information
Delivery Service
SR-5 Performance, Capacity and Storage ➢ C-SCP must monitor disk capacity on
all components.
➢ C-SCP must provide enough local
storage of CDR data for 7 days of calls
at an average call rate of 400 calls per
second with 3-minute call hold times.
➢ C-SCP shall have enough local storage
capability for storing 7 days of CDRs at
a peak rate of 1000 cps at an average
call hold time of 3-minutes for a 3 hour
peak.
SR-6 Hot swappable requirement ➢ The cooling fan modules shall be "hot
swappable
➢ All system cards shall be hot
swappable, i.e. it must be possible to
replace a card without taking the C-
SCP out of service and without
affecting service.
➢ IP interface cards shall be hot
swappable.
➢ Power supplies shall be hot swappable.
➢ Disk drives, if any, shall be hot
swappable and RAID-capable.
SR-7 PBX requirement 1. Number translation
11. ➢ The Number Translation Service shall
support Time Dependent Routing.
➢ The Number Translation Service shall
support Percent Allocation Routing.
2.Mid Call Interaction:
➢ C-SCP must support the Mid-Call
Interaction Service
➢ Mid-Call Interaction Service shall
support Call Transfer using RFC 5589.
➢ Call Transfer feature shall support blind
call transfer capability to an entity.
➢ Call Transfer feature shall support
consultation call transfer capability to
an entity.
➢ Call Transfer feature shall support
attended call transfer capability to an
entity.
➢
3.Screening
➢ C-SCP must support the Screening
Service Screening Service shall allow
restrictions
➢ upon calls using: Whitelist,
Blacklist,Maximum Current Calls, or
Block.
➢ Screening Service shall make available
any element from the originating party
for screening.
➢ Whitelist Screening shall allow call
termination for calls that match
elements within the screening list.
➢ Whitelist Screening shall block all calls
that do not match any elements within
the list.
➢ Blocked calls from Whitelist Screening
shall either receive Treatment or an
Announcement before disconnecting.
➢ Possible Whitelist screening elements
shall include but not limited to Calling
Party (full or partial), Calling Party
Category Originating Line Information.
➢ Blacklist Screening shall block call
termination for calls that match
elements within the list.
➢ Blocked calls from Blacklist Screening
shall either receive Treatment or an
Announcement before disconnecting.
➢ Possible Blacklist screening elements
shall include but not limited to Calling
12. Party (full or partial), Calling Party
Category, or Originating Line
Information.
4. Call Interaction Service:
➢ Vendor should describe signaling used
between C-SCP and media resource
function in support of the Caller
Interaction Service.
➢ C-SCP must support the playing of
tones and announcements.
➢ Tones must include DTMF and MF
tones.
➢ Caller Interaction Service shall support
customized announcements for a
subscriber or group of subscribers.
➢ Caller Interaction Service shall support
prompting and collection of DTMF
digits
➢ from the originating party.
➢ Caller Interaction Service should
support prompting and collection of
information through speech recognition.
➢ Information collected by Speech
Recognition should include numbers,
letters, words or phrases.
➢ Caller Interaction Service should
support English, Spanish and
Portuguese languages.
➢ Caller Interaction Service shall support
menu routing.
➢ Information collected through Menu
Routing shall result in an
announcement, a termination route or
index to another service.
➢ The Menu value collected by the Caller
Interaction Service shall be recorded in
a call detail record.
➢ Caller Interaction Service shall support
Account Code Screening.
➢ Account Code Screening shall play an
announcement and collects digits from
the originating party to allow or block
the call from proceeding.
➢ The Account Code value collected by
the Caller Interaction Service shall be
recorded in a call detail record.
➢ The collection of the Account Code
value from the Caller Interaction
Service shall be either mandatory or
13. optional.
➢ The validation of the collected Account
Code value from the Caller Interaction
Service shall be mandatory or optional.
➢ C-SCP should support a configurable
inter-digit timer for the collection of
DTMF digits.
➢ C-SCP should support the collection of
up to 12 DTMF digits per account code.
➢ C-SCP should support configurable
attempts for the collection of Account
Codes.
➢ Calls that are blocked by the Account
Code Screening feature receive
treatment or an announcement before
being disconnected.
➢ Caller Interaction Service shall support
Authorization Code Screening.
➢ Authorization Code Screening shall
play an announcement and collects
digits from the originating party to
allow or block the call from proceeding.
➢ The validation of the collected
Authorization Code is mandatory.
➢ The Authorization Code value collected
by the Caller Interaction Service shall
be recorded in a call detail record.
➢ C-SCP should support configurable
attempts for the collection of
Authorization Codes.
➢ Calls that are blocked the Authorization
Code Screening feature receive
treatment or an announcement before
being disconnected.
➢ C-SCP should support a configurable
inter-digit timer for the collection of
DTMF digits.
➢ C-SCP should support the collection of
up to 12 DTMF digits per authorization
code.
➢ Caller Interaction Service shall support
PIN Routing.
➢ Pin Routing shall play an
announcement and collect PIN digits.
➢ The collected PIN digits shall be used
to determine the terminating destination
for the call.
➢ The collected PIN digits that do not
result in a valid destination shall receive
14. treatment or an announcement before
being disconnected.
➢
5. DNIS
➢ The Information Delivery Service shall
include DNIS, Calling Party
Information override, and Calling Name
Delivery.
➢ Information Delivery Service shall
support Dialed Number Information
Service (DNIS).
➢ DNIS values shall support 2-15 digit
lengths.
➢ DNIS number format shall include
E.164 formats.
➢ The Original Dialed Number should be
maintained in the TO or Diversion
header.
6.Billing Requirement:
➢ C-SCP CDRs must contain all
necessary attributes of the query in
order to bill:
○ Calling Party
○ Called Party
○ Charge Party
○ Originating GW
○ Originating trunkgroup
○ Destination GW
○ Destination trunkgroup
○ Route Choice selected
(completed calls only)
○ Call Initiation Time
○ Call Ringing Time
○ Call Answer Time
○ Call Termination Time
○ Release Cause Code
➢ Vendor should describe their CDR
format and contents
SR-8 Failover scenario and System
Backup
➢ C-SCP shall support the ability to
accept the request for a hard shutdown
of the entire C-SCP (i.e., take it
immediately out of service).
➢ C-SCP shall support the ability to
accept the request for a graceful or hard
shutdown of interfaces or ports.
SR-9 Codec and Security ➢ C-SCP must provide support for idle
15. session timeouts.
➢ All considered C-SCP will have their
platform subjected to Security Lab
testing which will consist of an array of
Denial of service attacks, including:
○ Ping, and TCP traffic floods
○ SYN attacks
○ Malformed UDP, TCP and
ICMP traffic
➢ C-SCP must support the ability to
modify the following IP stack variables:
○ TCP keepalive interval
○ TCP re-transmit interval
○ MinTCP re-transmit interval
○ MaxTCP close wait interval
○ TCP/IP abort interval
➢ The C-SCP must use RADIUS as the
primary means for authentication. Local
accounts must only be used when there
are no servers available For RADIUS
authentication.
➢ User account permission levels where
appropriate should be kept on the local
C-SCP rather than returning the values
via RADIUS attributes.
SR-10 SS7 interface Requirement ➢ C-SCP may support SS7 Signaling in
support of the platform and services.
SR-11 Routing requirement ➢ Time Dependent Routing elements shall
use: time of day, day of week, day of
year, or specific holidays to determine
the terminating destination.
➢ Time Dependent Routing shall function
across different time zones (areas).
➢ Time Dependent Routing shall have
minimum granularity of one minute (60
seconds).
➢ Percent Allocation Routing shall
support fixed weighted percentages to
determine the terminating destination.
➢ Percent Allocation Routing shall have
minimum granularity one percent. A
value of 0% indicates temporary
removal of a destination from route.
➢ The Number Translation Service shall
support interaction between the
features.(i.e. the features should be able
to index to one another).
16. ➢ The Number Translation Service should
be able to detect a circular reference
between features.
➢ The Number Translation Service shall
support querying the CRE using SIP for
LCR routing instruction when the
terminating destination is a PSTN.(See
Query CRE section for more details.)
➢ C-SCP must support querying Level 3
Core Routing Engine (CRE) using SIP
Invite in support of Least Cost Routing
(LCR).
SR-12 IPv4 and IPv6 ➢ C-SCP must support IPv4.
➢ C-SCP should support IPv6 both at IP
layer and in SIP messaging.
SR-13 SIP ➢ C-SCP must support SIP over UDP
➢ C-SCP must be RFC 3261 SIP
compliant.
➢ C-SCP must be compliant with the SIP
supporting RFCs: 2387, 3262, 3264
➢ C-SCP must function as a SIP Stateless
Redirect Server for call requests from /
tothe CRE or Level3 Gateways.
➢ C-SCP must function as a SIP Stateful
Proxy for call requests from / to Level3
Gateways.
➢ When C-SCP receives a SIP Invite from
another Gateway, C-SCP must be
capable of mapping a Destination Trunk
Group(DTG) tag to an IP Trunk Group
on the C-SCP.
➢ C-SCP must support SIP-T per RFC
3372 for communicating with Level 3
Gateways as necessary.
➢ C-SCP may support SIP Session Timers
per RFC 4028 for communicating with
Level3 Gateways.
➢ C-SCP must support SIP-I per ITU
Q.1912.5 for communicating with
Level3 Gateways as necessary.
➢ C-SCP should support SIP UPDATE
Method per RFC-3311.
SR-14 SNMP ➢ Access console via SNMP an alarm and
report for any failure of any sub-
component of C-SCP.
➢ C-SCP shall authenticate all SNMP
requests and check all address against
17. an access control list.
➢ C-SCP shall support an SNMP agent
that communicates with resources
within C-SCP, such as IP ports, to
update the configuration of these
resources, and to retrieve the status of
these resources.
➢ SNMP bit counters must all be 64 bit
counters
➢ C-SCP Media Resource Function must
support sending voice related security
failures, including packet drops, to a log
file and log host (SNMP, syslog, etc.)
SR-15 Latency Requirement ➢ After a soft reset, C-SCP must recover
the state of the Services within 5
minutes of reset.
➢ C-SCP shall support a boot time (ready
for calls) of < 3minutes
18. 6. System Concept Diagram
Figure 1: C-SCP as Stateful Proxy Server
Figure 2: C-SCP as Redirect Server
19. Figure 1 represents the diagram of stateful proxy server. A stateful proxy server keeps
track of every request and response that it receives. As per the requirement, it can use
the stored information in future. When it does not receive a response from the other
side, it can retransmit the request.
It can remember the request that it has forwarded, so it can associate the response with
other process. Sine, it maintains Transaction state, we call it as stateful proxy server.
Stateful proxy servers do not scale as much as stateless ones.
Whereas, a stateless proxy server simply forwards the message it receives. This type
of server does not store any information of the transaction.
Stateless proxies forget about the SIP request once it has been forwarded.
They are very fast as they can scale very well.
The different blocks in the diagram as explained below:
SIGNALING SYSTEM 7 (SS7):
SS7 is an international telecommunication’s standard that defines how network
elements in Public Switched Telephone Networks (PSTN) exchange information over
a digital signaling network.
Nodes in an SS7 network are called Signaling Points.
SS7 consists of channel such as Reserved or Dedicated, which are also known as
Signaling links.
Signaling points are of three types, namely Service Switching Points (SSPs), Signal
Transfer Points (STPs) and Service Control Points (SCPs).
SS7 uses out-of-band signaling, i.e. signaling information travels on a dedicated 56
kbps or 64 kbps channel.
It does not use same channel as in the case of a telephone call.
SS7 is used for setting up telephone calls more efficiently and services such as Call
forwarding and wireless roaming service can be easily added and managed.
PSTN consists of Telephone lines, Fiber Optic cables, Cellular networks,
Communication Satellites. These are all interconnected by switching centers, which
enables telephones to communicate with each other.
MEDIA GATEWAYS (MG):
It is a device such as Circuit switched IP gateway or channel bank. It can convert data
from the required format of one type of network to the required format of another.
Media gateways are used for Transcoding media between PSTN and IP networks.
Media gateways connect different types of networks. Functions such as Echo
cancelation, DTMF and Tone sender are also present in the media gateway.
Media gateway is also called a ‘Soft Switch’.
CORE ROUTING ENGINE (CRE):
Routing Engine handles all packets of routing protocols.
It frees the packet forwarding engine so that it can only handle Internet traffic packets.
Routing engine constructs and maintains one or more than one routing tables.
20. The routing engine derives table of active routes from the routing tables.
These are called forwarding tables, which are copied into the packet forwarding
engines.
SERVICE CONTROL POINT (SCP):
It is a standard component of the telephone system, which is used to control the
service.
SCPs are used in telecom industry for SS7 and SIP technologies.
SCP also has service data point (SDP), which is used to hold the data base.
Using this data base of the SDP, the SCP identifies the geographical number to which
the call is to be routed. SCP is also used to communicate with IP to play voice
messages or prompt for information to the user like prepaid long distance using
account codes.
SCP uses codes like ‘#’, which are used to terminate the input for a username or
password or it can be used for call forwarding.
SCPs are connected with SSPs or STPs. It depends on the network architecture that
the network service provider wants.
SCPs mostly used with STPs.
SERVICE BORDER CONTROLER (SBC):
It is a dedicated hardware device or software application which shows if the phone
calls are initiated, conducted and terminated on a Voice over internet protocol (VoIP)
network.
SBC acts as a router between the enterprise and carrier service and it allows only
authorized sessions to enter the connection point (border).
It defines and monitors the quality of the service (status) for all sessions.
By doing that, it ensures that the callers communicate with each other and the
emergency calls are delivered correctly and ensuring that the priority of those calla
are above all other.
SBC also acts as a firewall for session traffic. It does that by applying it’s own quality
of service rules and it also identifies specific incoming threats that can be caused to
the communication environments.
SBC provides measurements, access control and data conversion facilities for the calls
they control.
22. 8. Test Cases
REQUIREMENT TEST CASE
CPU memory utilization warning. ` The CSC-P must generate an alarm when the
CPU utilization reaches more than 50%. For this
the system must run a various number of
applications and then check whether the alarm
triggers above 50% utilization.
C-SCP must generate a major alarm when
memory fills to 80% capacity or above.
Fill the system's memory with data till the system
is 80% filled and check whether the warning
alarm rings or not.
C-SCP shall have at least two IP interface cards
that share the network.
Insert one interface card which is working and
one which is not working and check whether the
system is still operational on a single interface
card.
Local memory backup. For testing this we need to fail a system by any
means and check whether the all management
parameters and operation is immediately restored
in local backup.
99.999% service availability. Keep the system under observation for 1 month
and check that it is available 99.999% of times.
C-SCP will be able to report statistics such as
traffic and rates , throughput , memory and buffer
utilization.
We can check and calculate the traffic rate ,
throughput , memory utilization using the
software such as HP loadrunner , HTTP load ,
proxy sniffer.
C-SCP must monitor disk capacity on all
components.
The disk space of each component should be seen
by the network administrator.
Command line interface (CLI) The system must support CLI so that the
commands such as PING , TRACEROUTE ,
ARP can be be used in many applications.
Hot Swappable Test for the various components like system
cards, power by replacing the them and check
whether the system remains operational
10SNMP Test for Manager and Agent configuration and
Operation
Test for separate login Access to Manager and
Agent Test For MIB storage volatility.Higher
security to MIB for data integrity.
Load Sharing Provide high data and voice traffic and test for the
load sharing occurring through various network
entities
23. Remote software upgrade and patches Check for Firmware repository and provide
update through wireless or Software Defined
Networks.
Minimum Boot time and Reboot time Switch on and off the system on regular basis to
investigate for boot and reboot time.
Data service should not affect Call performance Flood the network with data exchange and voice
calls and put the network under surveillance for
24 hours to check for performance interference.
RAID capable Connect various hard drives and check for the
RAID compatibility and performance
SSH Access Test for Server connectivity.
Test for link and port security.
Test for Remote Validation of the user/network
administrator.
Implement ECHO message to check for the
availability of the remote host
Multiple Login Session Access Check for port security and also remote login
application to provide multiple login access and
multiple sessions as well.
TFTP Check for server connectivity and try to get file
via easy access
Simultaneous call support Make multiple calls per hour for a dedicated time
and check for voice performance of the system
TELNET Access Test for server connectivity.
Permit access to the remote host and check for
various network performance parameters
EMS Run the system to check for FCAPS model
Number Translation Service Make non-geographical calls and learn the
translation service.
Playing of tones and customized announcements
and tones must include DTMF and MF tones
Check that when user touches dial pad it
playsdifferent tones for each number (as well as
the "#" and "*").For instance, the number "1" is
represented by the frequencies 1209 Hz and 697
Hz and also is subscriber notified when the called
person is busy or when he is not available or any
customized greeting the person has kept.
Support speech recognization for collecting
information
Check if the system recognizes and collects the
information or options selected by the user and
accordingly proceed
Account Code Screening Test if system recognizes the account code from
the call interaction
Collection of tones and its attempt. Check if it collects the dtmf codes and allows
multiple attempts.
Authorize code screening Should authorize the call for any secure info
access
Support PIN routing Allow PIN routing to get direct access to accurate
destination
Call transfer Capability Transfer calls from one terminating point to other
according to menu option selection
24. Call Queing If busy the caller should be held in queue and
acknowledge him about wait time.
Hierarchical Access The call function must go in a hierarchy style i.e.
the main menu point and then subpoints
Support querying level 3 CRE and LCR Check if system supports 2 or more CRE for
queries using SIP and it should use least cost for
that
Attributes of Query Check that is it collection all necessary info of the
call i.e. calling party, charge party, originating
party etc.
CDR should be maintained and if destroyed
before consuming alarm should be generated
The CDR should be destroyed once before its
consumed to see if the alarm is generated and if it
doesn’t must take necessary steps.
Idle session timeouts During the non use it should go in sleep mode
until its used again.
Security lab testing Test for its security by testing with array of denial
of service attacks like its efficiency during traffic
floods, syn attacks, malformed udp, tcpetc
Use RADIUS as means for authorization and
remote monitoring.
Use RADIUS protocol for authorizing the caller
based on Tripple‘A’ -Accounting, authorization
and accounting. In case of any failure anywhere it
should support remote monitoring to resolve it
25. 9. RISK ANALYSIS
Attackers could exploit security holes in SS7 to track cell phone user’s, i.e.
subscriber’s movements and communication. They can also eavesdrop on
communication.
SS7 is becoming more available to the untrusted parties. This opens multiple
opportunities for initiating fraudulent transactions on behalf of the subscribers.
If an attacker intercepts a call, then there are various ways by which he can deny
services of various subscribers, like modifying user data, which results in loss of data
integrity. There can be severe cases, where the subscribe data is entirely deleted.
SS7 is slow to take advantage of higher bandwidth technology, which are widely
available in the computer industry, but are common in the Telecom infrastructure,
which is why SS7 shares little possibility with the computer industry.
In the near future, companies might stop using SBCs and Media Gateways, which
might lead to SBCs and Media Gateways becoming obsolete.
After system is employed, the standards might change very slowly, like in case of
PSTN, this happens when the carriers are not interconnected.
In case of PSTN, the ability to switch a call and maintaining a reliable system can
become very expensive.
With the growth of network multimedia services, the demand for Voice on Demand
(VoD) and Voice over Internet (VoI) is increasing rapidly. Due to lack of large scale
real world dataset, PSTN is not able to keep up with the changes.
There is a need of dedicated phone lines from the service providers because of the use
of PSTN. The analog telephone lines use 64kbps in each direction.
In PSTN, features like Music on hold, conferencing, caller ID, call waiting are all
available for an extra cost.
PSTN capabilities are difficult to upgrade and expand, because it involves several
additions of hardware and adding lines.
26. 10. MITIGATION
In order to stop today’s most pervasive attacks, a recommended set of actions called
as The Critical Security Controls (CSC) are used. These are very effective Cyber
Defensive actions, which provide very specific ways to stop the attacks.
The problems with security can be resolved by moving the subscriber to a separate
signaling channel, which will cause the subscriber to not have access to the signaling
channel and also switching traffic to alternate routes in case of failure.
In order to improve the detection of the attacks, various custom types of filters can be
used. For example, a Snort filter can be used, which can be used for detection of
attackers who track the location of subscribers.
Given the criticality of the network, regular internal/external penetration testing
should be conducted against the core network. It will provide us the advantage of
detection, isolation and troubleshooting of the problem/attack before it gets to the root
of network and causes major issues.
IP based communication technologies are picking up pace and they are accepted in
place of SBCs and Media Gateways. This is because IP based phone systems are
much more functional and cost effective, which makes them important for all
businesses. This will lead into transformation in the ways employees operate them on
daily basis.
In VOIP, we need only one internet connection to transmit all voice channels and
because VOIP uses compression, therefore it needs only 10kbps in each direction.
Using VOIP, features like Music on hold, conferencing, caller ID, call waiting are
available at free of cost.
VOIP can be easily enhanced by just upgrading the software and including additional
bandwidth.
Average call cost is cheaper in case of VOIP as compared to that in case of PSTN.
VOIP will allow addition of new features like HD voice and Over The Top (OTT). It
also helps in reducing the cost of interconnection.