Sure!
Here's a list of commonly used Ubuntu (Linux) commands along
with their functions, grouped by category to help you understand them
better:
1. File & Directory Commands
Comman
Function
d
ls List directory contents
cd Change directory
pwd Show current directory path
mkdir Create a new directory
rmdir Delete an empty directory
rm Remove files or directories
cp Copy files or directories
Move or rename files or
mv
directories
touch Create an empty file
cat View contents of a file
more,
View long files page-by-page
less
find Search for files/directories
file Determine file type
2. User Management
Command Function
Display current logged-in
whoami
user
adduser
Add a new user
username
Command Function
passwd
Change a user's password
username
deluser
Delete a user
username
su Switch user
Run command with root
sudo
privileges
3. Permissions & Ownership
Comman
Function
d
chmod Change file permissions
chown Change file owner
chgrp Change group ownership
Show or set default permission
umask
mask
4. System Info & Monitoring
Command Function
Show disk usage in human-readable
df -h
format
du -sh
Show folder size
foldername
free -h Show memory usage
top or htop Show real-time system processes
uptime Show system uptime
uname -a Show system info
Command Function
hostname Show or set hostname
ps aux Show running processes
kill PID Kill a process
reboot Reboot the system
shutdown now Shutdown immediately
5. Network Commands
Command Function
ip a or ifconfig Show IP addresses
ping
Check network connectivity
[Link]
Show open ports and listening
netstat -tuln
services
ss -tuln Modern replacement for netstat
Show the path packets take to a
traceroute
host
nslookup Query DNS info
wget URL Download a file from a URL
curl URL Transfer data from or to a server
6. Package Management (APT)
Command Function
sudo apt update Update package lists
Upgrade all installed
sudo apt upgrade
packages
sudo apt install package- Install a package
Command Function
name
sudo apt remove package-
Remove a package
name
Remove unnecessary
sudo apt autoremove
packages
Install a .deb package
dpkg -i [Link]
manually
7. File Compression & Archiving
Command Function
tar -cvf [Link] Create tar
file archive
Extract tar
tar -xvf [Link]
archive
zip [Link] file Create zip file
unzip [Link] Extract zip file
gzip file Compress file
Decompress
gunzip [Link]
file
8. Editing Files
Comman
Function
d
nano file Open file with nano editor
vi file Open file with vi editor
Open file with GUI text editor (for
gedit file
Desktop only)
9. System Services (Systemd)
Command Function
systemctl status Show system services status
systemctl start
Start a service
servicename
systemctl stop
Stop a service
servicename
systemctl enable
Enable service to start at boot
servicename
systemctl disable Disable service from starting
servicename at boot
Software installation in Ubuntu occurs using the APT package
manager.
The main commands used are:
sudo apt update – Updates the package list.
sudo apt install package-name – Installs software.
sudo apt remove package-name – Removes installed software.
Package name means the exact name of the software as recognized by
Ubuntu’s package manager (APT).
For example:
To install Chrome: google-chrome-stable
To install Python: python3
To install VLC: vlc
So when you run:
sudo apt install vlc
"vlc" is the package name.
OSPF (Open Shortest Path First) serves as the backbone routing
protocol, operating at the Network Layer (Layer 3).
Access Control Lists (ACLs) operate at the Network Layer (Layer 3) of
the OSI model.
At the Application Layer (Layer 7), DHCP (Dynamic Host
Configuration Protocol) automates IP address management for all
devices in each department. Using UDP at the Transport Layer
(Layer 4), DHCP follows the DORA process (Discover, Offer, Request,
Acknowledge) to assign IPs from department-specific pools (e.g.,
[Link]/24 for HR). The inclusion of Option 150 ensures IP phones
receive the TFTP server address for firmware updates. Meanwhile, SIP
(Session Initiation Protocol), another Layer 7 protocol, establishes
and manages VoIP calls between extensions (e.g., HR's 101 to Sales'
601) through text-based messages like INVITE and ACK. SIP works in
conjunction with RTP (Real-time Transport Protocol) at
the Session/Presentation Layer (Layers 5-6), which handles the
actual voice packet transmission with timestamps and sequence
numbers for proper playback.
ICMP (Internet Control Message Protocol) at the Network Layer
(Layer 3) plays a crucial role
At the Data Link Layer (Layer 2), ARP (Address Resolution
Protocol) maintains the crucial mapping between IP addresses and MAC
addresses within each subnet.
RTP operates at the Transport Layer (Layer 4)
Is SSH on Layer 6 or Layer 7?
SSH works at the Application Layer (Layer 7) of the OSI model.
Layer 6 (Presentation) deals with encryption/decryption and format
conversion (e.g., JPEG, MP3).
SSH uses Layer 6 functions internally (for encryption), but as a
protocol, it’s considered a Layer 7 service because users interact
with it like an application (e.g., SSH client).
How Codecs Are Used in VoIP?
Codecs compress and decompress audio signals in VoIP to optimize
bandwidth and maintain voice quality.
Purpose of Codecs in VoIP
Reduce bandwidth usage.
Improve voice quality.
Support different network conditions.
What is a PBX System?
A PBX (Private Branch Exchange) is a telephone system that manages
internal and external calls within an organization.
How PBX is Used for VoIP?
IP-PBX connects VoIP calls using SIP trunks over the internet.
Routes calls between SIP phones and external networks.
Common software: Asterisk, FreePBX, 3CX.
Purpose of PBX in VoIP
Call routing and management.
Reduces telecom costs with VoIP.
Supports voicemail, call forwarding, and conferencing.
You said:
How cost reduced? To the point
ChatGPT said:
VoIP reduces costs by using the internet instead of traditional phone lines,
eliminating long-distance charges, reducing hardware costs, and
allowing free internal calls within the same network.
You said:
How PBX use internet? Only to the point
ChatGPT said:
PBX uses the internet via SIP trunks to connect calls over VoIP, replacing
traditional phone lines with IP-based communication.
Call Establishes Using RTP → Once connected, RTP (Real-time
Transport Protocol) carries the voice data.
Voice is Converted into Data Packets → The microphone captures the
voice, codecs compress it, and the data is sent over the internet.
You mean packets sequence number is given in RTP? [Link] to the point
ChatGPT said:
Yes, RTP assigns sequence numbers to packets to ensure they are received
and played in the correct order.
Client Sends Request (SIP Signaling) → Happens on the VoIP client
(softphone/IP phone) and is processed by the SIP server.
Server Connects Clients → The SIP server finds the receiver and
establishes the call session.
RTP Handles Voice Transmission → After connection, RTP manages
real-time voice data between clients.
Codecs Compress & Decompress Audio → This occurs inside VoIP
endpoints (softphones/IP phones) to optimize bandwidth.
QoS (Quality of Service) → Applied on network routers/switches to
prioritize VoIP traffic.
Jitter Buffering & Packet Loss Handling → Happens on the VoIP
endpoints to smooth out voice quality.
NAT Traversal (STUN, TURN, ICE) → Happens at firewalls, NAT
devices, and VoIP servers to handle private/public IPs.
Encryption (SRTP/TLS) → Applied on VoIP endpoints and SIP servers
for secure communication.
What is a Proxy in Networking?
A proxy is an intermediary server between a client (user) and the
internet, handling requests and responses.
Purpose of a Proxy
1. Security → Hides IP addresses and filters malicious content.
2. Anonymity → Masks user identity for privacy.
3. Content Filtering → Blocks or allows specific websites.
4. Caching → Stores frequently accessed data to speed up browsing.
5. Load Balancing → Distributes traffic to multiple servers for efficiency.
6. Access Control → Restricts or grants access to certain users.
IPsec (Internet Protocol Security)
Creates a secure tunnel between two devices/networks.
Encrypts data before sending over the internet.
Uses two main protocols:
o AH (Authentication Header) → Ensures data integrity (not
changed).
o ESP (Encapsulating Security Payload) → Encrypts and
secures data.
SSH (Secure Shell)
Encrypts login sessions to prevent hacking.
Used to remotely access and control devices like routers/servers.
Works by authenticating the user using passwords or SSH keys.
SSL (Secure Sockets Layer)
Encrypts web traffic between a browser and a website.
Uses SSL certificates to verify website authenticity.
Security Topics & Their OSI Layers
1. Firewalls → Layer 3 & 4 (Network & Transport) (Next-Gen Firewalls
also work on Layer 7)
2. VPNs (IPsec, SSL VPN) → Layer 3, 4, & 7
3. Intrusion Detection & Prevention (IDS/IPS) → Layer 3, 4, & 7
4. Access Control (ACLs, 802.1X, NAC) → Layer 2, 3, & 7
5. Network Encryption (TLS, IPsec, MACsec) → Layer 2, 3, & 7
6. Authentication & Authorization (AAA, RADIUS, TACACS+) →
Layer 2 & 7
Layer 3 (Network Layer) – Filters & Routes Traffic
Firewalls (Basic ACLs) → Blocks or allows IP-based traffic.
IPsec VPN → Encrypts IP packets for secure communication.
Layer 4 (Transport Layer) – Controls Connections
Stateful Firewalls & IDS/IPS → Monitors TCP/UDP traffic for threats.
Layer 7 (Application Layer) – Secures User-Level Communication
SSL/TLS Encryption → Secures HTTPS, VoIP, and email traffic.
AAA (RADIUS, TACACS+) → Manages user authentication for network
devices.
802.1Q is used for VLAN tagging to identify and separate network traffic
on switches.
802.1X → Port-based network access control (NAC)
802.1Q → VLAN tagging
802.1D → Spanning Tree Protocol (STP) for loop prevention
802.3 → Ethernet standard
802.11 → Wi-Fi (Wireless LANs)
VLAN Trunking → Allows multiple VLANs to pass through a single link
using 802.1Q tagging.
Inter-VLAN Routing → Enables communication between VLANs using a
router or Layer 3 switch.
No, VLAN Trunking and Link Aggregation Control Protocol (LACP) are
different:
VLAN Trunking (802.1Q) → Carries multiple VLANs over a single
link by tagging traffic.
LACP (802.3ad) → Combines multiple physical links into one
logical link for higher bandwidth and redundancy.
Redundancy ensures network reliability by using backup links,
devices, or systems to prevent failures and maintain connectivity.
. Microwave for VoIP
Microwave links are used in wireless backhaul networks to carry
VoIP traffic over long distances.
They provide high-speed, point-to-point connectivity between
telecom towers.
Used in areas where fiber or wired connections are not available.
VoIP traffic travels over microwave using IP-based transmission.
How Voice over LTE (VoLTE) Works (4G Calling)
VoLTE (Voice over LTE) allows voice calls over 4G LTE networks
instead of using traditional 2G/3G.
It uses IP packets instead of circuit switching, making calls faster
and clearer.
Works with IMS (IP Multimedia Subsystem) to manage VoIP
traffic on LTE.
Provides better voice quality (HD Voice) and supports
simultaneous voice and data usage.
. Does Video Calling Exist in VoIP?
Yes, VoIP supports video calling using RTP and video codecs
(H.264, VP8, etc.).
Packet switching is used in VoLTE. ✅ (Correct)
VoLTE uses packet switching, but VLANs are not used in mobile
networks. VLANs are for wired networks (Ethernet).
IMS (IP Multimedia Subsystem) manages VoIP traffic and multimedia
services like voice and video calls.
Instagram VoIP first establishes a connection via a signaling
server.
The server finds the recipient’s IP address and helps set up a P2P or
relay connection.
P2P (Peer-to-Peer) means direct communication between two
users, without a middle server forwarding data. However, if P2P fails,
a relay server is used.
P2P does NOT mean the connection remains open when no call is
made—it is established only during the call.
PSTN is an analog circuit-switched network, while VoIP is digital
and packet-switched.
VoIP over fiber means voice is transmitted as IP packets using
fiber-optic internet.
Audio Codecs (G.711, G.729) for voice compression
Video Codecs (H.264, VP8, VP9) for video compression
1. Mobile Network – Is It Wireless?
Yes, a mobile network is wireless because:
It uses radio signals to connect mobile devices to the nearest cell
tower (BTS).
BTS forwards data to the core network (MSC/SGW/PGW).
Wireless technologies like 2G, 3G, 4G, and 5G enable mobile
communication.
Backhaul connections (microwave or fiber) carry data from the
tower to the core network.
2. Multimedia in VoLTE – Which Gets More Preference, Voice or
Video?
In VoLTE, Voice gets more priority than video because:
VoLTE is mainly designed for high-quality voice calls.
QoS (Quality of Service) prioritizes voice traffic to reduce delays.
Video is also supported (ViLTE = Video over LTE), but it’s not as
optimized as voice.
IMS (IP Multimedia Subsystem) manages both voice and video,
but voice is always prioritized.
Yes, Asterisk is a VoIP server that handles SIP signaling and manages
VoIP calls. It can be used as a private PBX system or a VoIP gateway.
(B) How Is a Tunnel Created in VoIP?
When a user sends a VoIP call request, a tunnel is created using:
1. SIP signaling to establish the connection.
2. NAT traversal techniques (STUN, TURN, ICE) to pass through
firewalls.
What Is a Relay Server?
A relay server is used when direct P2P (Peer-to-Peer)
communication fails due to NAT (Network Address Translation)
or firewalls.
The relay server forwards VoIP packets between users instead of
direct P2P.
Example: TURN (Traversal Using Relay NAT) servers handle this in
VoIP.
VoIP works through the internet.
If the internet is down, VoIP calls go through PSTN.
VoIP over fiber means fiber is used as an internet connection to carry VoIP
packets.
If the internet fails, calls can fall back to PSTN (circuit-switched network).
Fiber itself does NOT use circuit switching—it carries IP packets using packet
switching.
Compression is used to reduce bandwidth usage while maintaining
acceptable quality.
Voice compression (e.g., G.729, Opus) helps VoIP work on low-
bandwidth networks.
Video compression (e.g., H.264, VP8) reduces file size so video
calls can run smoothly.
Is Multimedia Used in RTP (Real-Time Protocol)?
✅ Yes, RTP is used for multimedia streaming, including voice
and video.
RTP is responsible for real-time transmission of audio and video in
VoIP, video conferencing, and live streaming.
Voice codecs (G.711, G.729, Opus) and video codecs (H.264,
VP8) work inside RTP.
RTP itself does not guarantee delivery—it relies on UDP for fast
transmission and QoS for priority handling.
What Is PBX, and How Does It Work?
PBX (Private Branch Exchange) Definition:
PBX is a private telephone system used within an organization to
manage internal and external calls.
PBX Working from Start to End:
1. User Dials a Number – The IP phone or softphone sends a request to
the PBX.
2. PBX Receives the Request – It decides whether the call is internal
(same PBX) or external (VoIP/PSTN).
3. PBX Routes the Call –
o If internal, the PBX directly connects both users.
o If external, the PBX sends the call to a VoIP gateway or VoIP
server (e.g., Asterisk, FreePBX).
4. Call Is Converted – If needed, the PBX translates VoIP packets to
PSTN signals for traditional networks.
5. Call Is Established – The two users communicate.
VoIP Tunnel (Including VPN & Firewall Bypass)
How Is a VoIP Tunnel Created?
1. VoIP Client (User 1) Requests Connection.
2. A VPN Tunnel (IPSec, TLS, or L2TP) Is Created for encryption.
3. The Tunnel Passes Through Firewalls & NAT to reach the VoIP
server.
4. The VoIP Server Routes the Call to the destination (PSTN or another
VoIP client)
TLS (Transport Layer Security) works at Layer 4 (Transport Layer)
and provides encryption, authentication, and data integrity for
SIP signaling.
Authentication mainly works at Layer 7 (Application Layer) but can
also involve Layer 2 (Data Link Layer) and Layer 4 (Transport
Layer) in some cases.
Layer 7 (Application Layer) → Authentication in SIP, HTTPS, SSH,
etc. using usernames, passwords, or certificates.
2️⃣ Layer 4 (Transport Layer) → Authentication in TLS, SSL for
secure communication.
3️⃣ Layer 2 (Data Link Layer) → Authentication in 802.1X (Network
Access Control) for device verification before network access.
1️⃣ VoIP uses both UDP and TCP → Correct. SIP signaling can use
TCP or UDP, while RTP (media stream) mostly uses UDP for low
latency.
2️⃣ Video codecs compress → Resolution, frame rate, and bitrate
to reduce bandwidth usage (e.g., H.264, VP8).
3️⃣ PBX is a server → Yes, it manages internal VoIP calls and connects
to external networks (PSTN, SIP trunks, or the internet).
4️⃣ PBX can create a tunnel → Yes, if using VPN or encrypted SIP
trunks. The main VoIP server (SIP provider or cloud PBX) routes
external calls to the destination.
5️⃣ Proxy in VoIP → Used between clients and VoIP servers to
handle SIP signaling, NAT traversal, or load balancing.
Internet is ISP or router base routing
Fiber VoIP Flow
1. User initiates VoIP call.
2. Call request sent to ISP through fiber connection.
3. ISP routes call request to VoIP provider.
4. VoIP provider establishes connection with the destination.
5. RTP stream initiated for audio/video transmission.
6. VoIP user initiates a call.
7. VoIP server converts SIP request into PSTN format.
8. Call routed to PSTN gateway.
9. PSTN network connects call to landline/mobile number.
10. RTP used between VoIP user and PSTN gateway.
Media Gateways (MG) Usage:
Step 1: Acts as a bridge between PSTN (Public Switched Telephone
Network) and the IP network.
Step 2: Converts speech samples between PSTN (PCM) and IP (RTP
with different codecs).
Step 3: Communicates with the Call Agent (Media Gateway
Controller - MGC) using MEGACO/MGCP for call handling.
Signaling Gateways (SG) Usage:
Step 1: Acts as a network component responsible for signaling
message exchange.
Step 2: Transfers call setup, billing, location, and other control
information between different networks.
Step 3: Converts SS7 (used in PSTN) to IP signaling (SIGTRAN).
Step 4: Example: A SIGTRAN Signaling Gateway translates SS7
signaling into SIGTRAN (IP-based signaling).
Converts voice into digital packets and transmits them over an IP network.
Uses protocols such as SIP (Session Initiation Protocol) and H.323.
Media Servers: Handle features like voicemail and conferencing.
H.323: An older protocol for call setup and control.
MGCP (Media Gateway Control Protocol): Controls media
gateways between different networks.
Advanced Features: Voicemail, call forwarding, video calls, and more.
VoIP Implementations
On-Premises VoIP: Businesses set up their own VoIP system with
PBX.
Hosted VoIP: A third-party provider manages VoIP services.
Cloud VoIP: Fully online, managed by cloud service providers
Authentication: SIP authentication ensures only authorized users make calls.
H. Protocol (H.323) vs. SIP
You're right—H.323 is not just for video compression. It is a complete
VoIP protocol suite that handles call setup, control, and media
transmission (voice, video, and data). It includes multiple protocols:
H.225 (Call signaling and registration)
H.245 (Negotiates call capabilities)
RTP (Real-time Transport Protocol) for media transmission
H.264 (Video compression)
Then why do we need SIP?
SIP (Session Initiation Protocol) is simpler and more flexible
than H.323. It is mainly used for VoIP call setup and management.
SIP is text-based (like HTTP) and easier to implement than the binary-
based H.323.
H.323 was widely used in the 1990s but SIP became dominant because
of its flexibility and ease of integration with other technologies (VoIP,
video conferencing, etc.).
So, both H.323 and SIP can set up VoIP calls, but SIP is now more
common.
If the receiver is using a traditional phone, the VoIP provider
routes the call to a PSTN gateway, which converts the VoIP call into
a format understood by the traditional phone network.
Why Does VoIP Require Less Physical Infrastructure?
1. Traditional PSTN phones need dedicated wiring for every call. VoIP
uses existing internet connections.
2. VoIP doesn't require massive central office switching equipment
like PSTN.
How Businesses Use PBX Servers?
Some companies buy and set up their own PBX for complete
control.
Others use a hosted/cloud PBX where a VoIP provider manages it for
them.
Hosted PBX requires only VoIP phones and an internet connection, as
the provider handles the backend.
Why doesn’t everyone move to cloud VoIP?
Some businesses prefer on-premise PBX for security, control, and
lower long-term costs.
Cloud VoIP requires good internet quality, which isn’t available
everywhere.
Security in VoIP (From Start to End Call Setup)
Security must be applied at different points in the VoIP process:
Step 1: Call Request & Authentication
The client (caller) sends a SIP INVITE to the VoIP server.
The server checks authentication (username, password).
Security measures: TLS (Transport Layer Security) encrypts this
request.
Step 2: Firewall & NAT Traversal
Firewalls check if the request is allowed.
If the call is blocked due to NAT (Network Address Translation),
STUN/TURN/ICE protocols may be used to establish the call.
Step 3: Call Routing & Encryption
If the call is going through the internet, VPNs or SRTP (Secure RTP)
encrypt the audio/video stream.
The VoIP server routes the call to the recipient.
H.225 is NOT the server itself, but it communicates with the
server to register users and establish calls.
2. H.245 = Call Negotiation?
H.245 ensures that both sender and receiver "speak the same
language" for audio and video transmission.
3. SIP is Like HTTP? How?
Both use headers to define information.
Both are human-readable (you can open a SIP packet and read the
text easily).
H.323 messages are encoded in binary format, making them
harder to read and debug.
SIP, in contrast, uses plain text (like HTTP), which is easier for
humans to read.
6. Security in VoIP Server?
Yes, security measures start at the VoIP server when a client sends
a request. Security involves:
1. Authentication:
o Verifies username/password before allowing the call.
2. What is a Redirect Server?
2. A Redirect Server is used when the original Proxy Server does not forward the request
itself but instead tells the caller where to send the request.
Tell the location
3. What is a Location Server?
A Location Server stores the current IP addresses of users
registered in the VoIP network.
When a Proxy or Redirect Server needs to find a user, it queries
the Location Server.
The Location Server does NOT handle calls—it only stores where
users are located.
What Does "Fork Request" in a Proxy Server Mean?
Forking means a Proxy sends the call to multiple locations at the same time.
PSTN Protocols:
PSTN is the traditional telephone system, and its protocols ensure
voice communication through copper wires or fiber optics:
SS7 (Signaling System 7): A signaling protocol used for routing calls
in PSTN.
ISUP (ISDN User Part): A signaling protocol that helps establish and
control connections in the PSTN.
In VoIP, a phone number is typically converted into an IP address using
DNS (Domain Name System) or SIP Proxy Server. Here's how the
process works:
1. User dials a phone number: The VoIP system (like a softphone or IP
phone) receives the phone number.
2. DNS Lookup or SIP Proxy: The phone number is converted to an IP
address through a DNS lookup or via the SIP proxy server.
RAS (Registration, Admission, and Status) in VoIP is a protocol
used in H.323-based networks to handle communication between
endpoints (such as IP phones) and gatekeepers. It manages:
1. Registration: The process where a VoIP device registers with the
gatekeeper.
2. Admission: It controls whether a device is allowed to initiate a call.
3. Status: Keeps track of the device’s current status or availability for
calls.
A Multipoint Control Unit (MCU) in VoIP is a device used in video
conferencing to manage communication between multiple endpoints
(such as phones or computers). It:
1. Coordinates audio and video streams between several participants.
2. Mixes the media streams (audio/video) and ensures they are correctly
transmitted to each participant.
3. Helps in connecting more than two users in a video or voice
conference.
✅ 1. Your VoIP understanding:
You said:
VoIP used in wireless (tower to tower via microwave/radio) → core network →
internet server → other client.
And also used in fiber/wired → internet/cloud server.
✅ Answer: Yes, you are correct.
VoIP over wireless: via microwave links, radio signals, or mobile
data (LTE/5G) → goes to core network → internet.
VoIP over wired/fiber: through Ethernet/fiber lines → connects to
local/cloud VoIP servers.
✅ 2. Is the Internet a physical place or wire-based?
The Internet is a global network of physical infrastructure
(cables, fiber optics, routers, etc.).
So yes, it is wire-based (physical cables under oceans, in cities),
even if access is wireless (like Wi-Fi or 4G).
✅ 3. Is PBX a physical server?
It can be:
o ✅ Physical PBX (hardware box in office).
o ✅ Or software PBX (installed on a cloud server or PC, like
Asterisk/FreeSWITCH).
✅ 4. AAA & Syslog Server — Which Layer? SIP Request over
Wireless?
AAA (Authentication, Authorization, Accounting):
🔸 Used at Application Layer (Layer 7).
Syslog Server (for logging):
🔸 Also works at Application Layer (Layer 7).
SIP requests go over wireless?
🔸 ✅ Yes, SIP can go over Wi-Fi, 4G/5G, or any wireless medium.
✅ 5. VoIP uses internet, PSTN uses telephone lines — What
about wires?
VoIP: Uses internet wires/fiber or wireless (Wi-Fi, LTE, etc.).
PSTN: Uses copper wires or fiber (traditional telephone lines).
🔸 Both use wires — difference is how data is transmitted (packet-
switched vs. circuit-switched).
✅ 6. How SIP Authentication works?
SIP uses username + password (digest authentication).
The SIP server challenges the user with a message → user must
respond with encrypted credentials.
Only valid users can make/receive calls.
UMTS (Universal Mobile Telecommunications System)
A 3G mobile communication technology that provides higher data
rates than 2G (GSM).
Supports voice, video calls, and mobile internet.
GMSC (Gateway Mobile Switching Center)
A core network component in mobile communication.
Acts as a gateway between mobile networks and external
networks (e.g., PSTN, VoIP).
Code Division Multiple Access (CDMA) is a multiple access
technique where multiple users share the same frequency band by
using unique code sequences to differentiate their signals.
Backhaul is the part of a telecom network that connects Radio
Access Network (RAN) (base stations) to the Core Network.
LOS means the transmitting and receiving antennas must have a
clear, unobstructed path for signal transmission. Any obstruction
(hills, buildings) can cause signal degradation or loss.
What is the Frequency Range of Microwave Communication?
Answer:
Microwave frequencies are divided into:
Lower bands (6 GHz – 11 GHz): Long-distance transmission.
Higher bands (18 GHz – 80 GHz): Short-range, high-capacity links.
Why is Microwave Used in Mobile Backhaul?
Answer:
Microwave is used because:
Faster Deployment: No need for fiber laying.
High Capacity: Supports Gbps data rates.
Reliable Performance: Works well in remote or urban areas where
fiber is costly.
Backhall is= to connect tower to core network without fiber mean
wireless radio signal used. Core network is =msc to internet
What Are the Frequency Ranges of Microwave?
Low-frequency bands (6 GHz – 11 GHz): Long-distance
transmission, less affected by rain.
Mid-frequency bands (15 GHz – 23 GHz): Balanced range and
capacity.
High-frequency bands (26 GHz – 80 GHz): Short-range, high-
capacity, used in 5G.
How is Microwave Used for RAN?
Microwave links act as backhaul to transport data from RAN (base
stations) to the Core Network.
Used where fiber is not available, ensuring fast and cost-effective
connectivity.
3G
2G 4G 5G
KPI (UMTS/WCDM
(GSM) (LTE) (NR)
A)
100-500 10-50 <1
Latency 50-100 ms
ms ms ms
300
Peak
9.6-384 2 Mbps (HSPA: Mbps 10-20
Data
Kbps 42 Mbps) -1 Gbps
Rate
Gbps
700 600
Spectru 900/180 MHz – MHz –
900/2100 MHz
m 0 MHz 3.5 52
GHz GHz
Up to Up to Up to
Mobility 250 Up to 350 km/h 500 1000
km/h km/h km/h
Spectral
Very
Efficienc Low Medium High
High
y
What is RAN (Radio Access Network)?
RAN is the part of a mobile network that connects user devices
(phones, IoT) to the core network using radio signals.
Purpose of RAN:
Provides wireless connectivity between users and the network.
Includes base stations (eNodeB/gNodeB), antennas, and radio
equipment.
Yes, this is a 2G (GSM) architecture. Here are the functions of each
part:
1. Mobile Station (MS - Phone/Car)
User device that sends/receives calls and data via radio signals.
Communicates with BTS (Base Transceiver Station).
Radio signals are used here to connect to BTS.
2. BTS (Base Transceiver Station)
Handles radio communication with mobile devices.
Converts radio signals to digital data and vice versa.
Connects to BSC via Abis interface.
3. BSC (Base Station Controller)
Manages multiple BTSs.
Controls handover, power control, and frequency allocation.
4. MSC (Mobile Switching Center)
Main switch that connects calls to other networks.
Manages call routing, handovers, and mobility management.
5. VLR (Visitor Location Register)
Stores temporary subscriber information for roaming users.
6. HLR (Home Location Register)
Stores permanent subscriber details (SIM number, services, etc.).
7. AUC (Authentication Center)
Provides encryption keys for secure communication.
8. EIR (Equipment Identity Register)
Maintains a database of valid, stolen, and blacklisted devices.
9. OMC (Operation & Maintenance Center)
Monitors and manages the network’s performance and faults.
10. NMC (Network Management Center)
Higher-level network monitoring and optimization.
Where Are Radio Signals Used?
Between Mobile Station (MS) and BTS (over the Um-interface).
Radio signals enable wireless communication in the GSM
network.
Yes, this is 3G (UMTS) network architecture.
Microwave Usage
Radio Signals: Used between Mobile Phone and Node B (cell
tower).
Microwave Backhaul: Used from Node B to RNC (if fiber is not
available) to connect to the core network.
Function of Each Part
1. Mobile Phone
Communicates with Node B using radio signals.
2. Radio Network
Node B: Equivalent to BTS in 2G; handles radio communication with
mobile devices.
RNC (Radio Network Controller): Manages multiple Node Bs,
handles handovers, and controls power levels.
3. Core Network
MSC/VLR (Mobile Switching Center / Visitor Location Register):
Handles call switching and mobility management.
HLR (Home Location Register): Stores subscriber information and
authentication details.
AuC (Authentication Center): Provides security by verifying users.
EIR (Equipment Identity Register): Stores information about mobile
devices to prevent stolen phones from accessing the network.
SGSN (Serving GPRS Support Node): Handles packet-switched data
(mobile internet).
GGSN (Gateway GPRS Support Node): Connects the mobile
network to external data networks like the internet.
GMSC (Gateway Mobile Switching Center): Connects the mobile
network to external voice networks.
Microwave Usage in 4G LTE Network
Microwave is used for backhaul between eNodeB and the Core
Network (EPC) if fiber is unavailable.
Functions of Each Part
Mobile Phone (UE - User Equipment)
Connects to the network using LTE radio signals.
Radio Network
eNodeB (Evolved Node B):
o Handles radio communication with the UE.
o Directly connects to the Core Network (no RNC in 4G).
Core Network (Evolved Packet Core - EPC)
MME (Mobility Management Entity):
o Manages signaling, authentication, and mobility.
S-GW (Serving Gateway):
o Routes user data between eNodeB and P-GW.
HSS (Home Subscriber Server):
o Stores user profiles and authentication details.
P-GW (Packet Data Network Gateway):
o Connects the mobile network to external data networks
(Internet).
PCRF (Policy and Charging Rules Function):
o Controls QoS and charging policies.
In the 5G Standalone Network diagram:
Dotted Line (Between 5G UE and gNB) → Represents signaling
(control plane), such as mobility management and session setup.
Solid Lines (Between gNB, Core Network, and Internet) →
Represent data transmission (user plane), meaning actual voice,
video, and internet traffic.
The Transport Network connects the 5G Radio Network (gNB) with
the 5G Core (AMF/UPF) using fiber or microwave links.
5G UE (User Equipment) - Mobile Device
A smartphone or any 5G-compatible device that connects to the mobile
network.
Communicates with the base station (gNB) using radio signals.
gNB (Next-Generation Node B - 5G Radio Network)
The base station in a 5G network.
Handles radio communication with 5G devices.
Splits its functions into:
o CU (Central Unit) – Handles control plane tasks (e.g., mobility,
handovers).
o DU (Distributed Unit) – Handles user plane tasks (e.g., packet
forwarding).
Microwave is used here for backhaul when fiber is unavailable,
connecting the gNB to the transport network.
Transport Network
Connects the radio network (gNB) to the core network.
Uses fiber or microwave links, depending on availability.
Microwave is used here as a backhaul option when fiber is too costly
or impractical.
5G Core Network (AMF/UPF)
AMF (Access and Mobility Management Function) → Manages
signaling, authentication, and handovers.
UPF (User Plane Function) → Routes user data traffic towards the
internet or private networks.
Feature Microwave Fiber
Cost Lower Higher
Speed Up to 10 Gbps Higher speeds
Latency Low Very low
Deployme
Fast Time-consuming
nt
Remote/rural Urban/high-density
Best for
areas areas
Yes, the Radio Access Network (RAN) lies between the BTS (or
NodeB/eNodeB/gNB) and the Core Network.
Yes, a BTS can handle multiple frequencies but it is assigned a
specific frequency for each mobile connection in its coverage area.
So, it operates on multiple frequencies but assigns each mobile to
a specific one.
No problem, here’s a very simple explanation:
SVI (Switched Virtual Interface) is used in a switch to:
👉 Give IP address to a VLAN
👉 Allow communication between VLANs (like a router)
Example:
If VLAN 10 has computers, SVI lets them talk to other VLANs or go to the internet.
It acts like a default gateway for that VLAN.
NLOS (Non-Line-of-Sight) → Works even if there’s an obstacle (like a
WiFi signal passing through walls).
Microwave uses different frequencies, such as 6 GHz, 11 GHz, 18
GHz, and E-band (70/80 GHz).
Lower frequencies (e.g., 6 GHz) travel longer distances.
Higher frequencies (e.g., E-band) give faster speeds but for
short distances.
1) SS7 Protocol (Signaling System No. 7)
SS7 is a signaling protocol used in telecommunications networks for
call setup, routing, and control. It enables different network elements
(like switches and databases) to communicate and manage voice calls,
SMS, and other services.
Key Functions of SS7:
Call Setup & Teardown: Establishes and terminates calls in PSTN and
mobile networks.
SMS Handling: Facilitates text message delivery.
Number Portability: Helps in mobile number portability.
Billing & Fraud Detection: Assists in real-time billing and fraud
prevention.
SS7 Vulnerabilities & Security:
SS7 can be exploited for call/SMS interception and location tracking.
Securing SS7 requires firewalls, encryption, and anomaly
detection systems.
Maximum Theoretical
Technology
Speed
2G (GPRS/EDGE) 56-236 Kbps
3G (HSPA+) 42 Mbps
4G (LTE-A) 1 Gbps
5G 10 Gbps
6G (Future) 100-1000 Gbps
Technolog TDM FDM
CDMA OFDMA
y A A
2G (GSM) ✅ ✅ ❌ ❌
3G
❌ ❌ ✅ ❌
(WCDMA)
4G (LTE) ❌ ❌ ❌ ✅
5G (NR) ❌ ❌ ❌ ✅
Maximum Theoretical
Technology
Speed
6G ❌ ❌ ❌ ✅
7) Throughput of Optical Fiber
Single-Mode Fiber (SMF): Up to 10 Tbps.
Multi-Mode Fiber (MMF): Up to 100 Gbps (shorter distances).
8) STP (Spanning Tree Protocol)
STP prevents network loops in layer 2 networks.
Uses BPDU (Bridge Protocol Data Units) to select a root bridge.
Types: STP, RSTP, MSTP (Rapid STP is faster).
9) SNMP (Simple Network Management Protocol)
Used for monitoring & managing network devices.
Works on UDP Port 161 (queries) & 162 (traps).
Versions: SNMPv1 (basic), SNMPv2c (enhanced), SNMPv3
(secure).
10) How Many Users Can a Router Handle?
Home Routers: 10-50 users.
Enterprise Routers: 500-10,000 users.
Carrier-Grade Routers: 100,000+ users.
11) How Many Users Can an OLT Handle?
OLT (Optical Line Terminal) in GPON networks can serve 64-128
ONUs per PON port.
Large-scale OLTs can handle thousands of users with multiple PON
ports.
12) Network Management (FCAPS Model)
FCAPS is a network management framework that stands for:
1. Fault Management (F) – Detect and troubleshoot network issues.
2. Configuration Management (C) – Manage device settings and
configurations.
3. Accounting Management (A) – Monitor usage for billing.
4. Performance Management (P) – Optimize network efficiency and
performance.
5. Security Management (S) – Implement policies for network security.
1) NetBIOS (Network Basic Input/Output System)
NetBIOS is a protocol that allows applications on different computers to
communicate over a local network. It provides:
Name Service: Identifies devices on a network.
Session Service: Manages communication between two devices.
Datagram Service: Sends data without requiring a session.
Common NetBIOS Ports:
UDP 137: Name resolution.
UDP 138: Datagram service.
TCP 139: Session service.
2) What is LTE?
LTE (Long-Term Evolution) is a 4G wireless technology that
provides high-speed internet and voice services.
Key Features:
Faster Speeds: Up to 1 Gbps.
Low Latency: ~10ms.
Uses OFDMA & MIMO: Improves spectral efficiency.
Supports VoLTE (Voice over LTE): Enables HD voice calls.
3) Codes in VoIP Practical
VoIP uses different codecs for voice compression and transmission.
Common VoIP Codecs:
G.711: High quality, uncompressed (64 Kbps).
G.729: Compressed, lower bandwidth (~8 Kbps).
G.722: HD voice quality (~64 Kbps).
5) TFTP in VoIP Practical
TFTP (Trivial File Transfer Protocol) is used in VoIP to:
Download configuration files for IP phones.
Load firmware updates for VoIP devices.
Work in lightweight, UDP-based mode (Port 69).
6) IP Helper Address
IP Helper Address is a feature in Cisco routers that forwards DHCP
broadcasts from one subnet to another.
It helps VoIP phones get IP addresses even when the DHCP server is on a different
network.
8) Analog Phone vs Cisco IP Phone 7960
Feature Analog Phone Cisco IP Phone 7960
Connection Type Uses RJ-11 (POTS) Uses RJ-45 (Ethernet)
Call Signaling PSTN SIP, SCCP
9) Frequency Spectrum Allocation Agencies
Pakistan: PTA (Pakistan Telecommunication Authority)
USA: FCC (Federal Communications Commission)
India: TRAI (Telecom Regulatory Authority of India)
UK: OFCOM (Office of Communications)
Global Coordination: ITU (International Telecommunication
Union)
1) Range of LAN & WAN
Network Type Range Usage
LAN (Local Area Up to 100 meters (Wi-Fi) or a
Used in homes, offices, and campuses.
Network) few kilometers (wired)
WAN (Wide Area Kilometers to thousands of Used to connect cities, countries, and
Network) kilometers continents (e.g., the Internet).
2) Amplitude Modulation (AM), Frequency Modulation (FM),
Phase Modulation (PM)
These modulation techniques are used in communication to
transmit signals over a medium.
Modulation
Definition Usage
Type
Varies the
Amplitude Used in AM radio,
amplitude of the
Modulation aviation
signal while keeping
(AM) communication.
frequency constant.
Varies the
Frequency
frequency while Used in FM radio, TV
Modulation
keeping amplitude broadcasting.
(FM)
constant.
Phase Used in digital
Varies the phase of
Modulation communications like
the carrier signal.
(PM) Wi-Fi, Bluetooth.
Purpose: Not encryption, but for efficient signal transmission over
different mediums (air, cables, fiber).
It is Used for data representation.
3) 5G (NR - New Radio)
5G New Radio (NR) is the latest standard for 5G mobile
communication, offering:
Higher speeds (up to 10 Gbps).
Low latency (~1ms) for real-time applications.
IoT (Internet of Things) operates across multiple OSI layers:
Physical Layer: Sensors, RFID, ZigBee, Bluetooth, Wi-Fi.
Data Link Layer: MAC addresses for device communication.
Network Layer: IPv6, 6LoWPAN for IoT addressing.
Application Layer: MQTT, CoAP for IoT communication.
5) DNS Works at Which OSI Layer?
DNS (Domain Name System) operates at the Application Layer
(Layer 7).
It translates domain names (e.g., [Link]) into IP addresses.
Uses UDP Port 53 for fast resolution and TCP Port 53 for large
queries.
In VoIP, DSCP (Differentiated Services Code Point) marking is
used to prioritize voice traffic by assigning specific QoS (Quality of
Service) values in the IP packet header.
Data Transfer Rate Terms
1. Bandwidth: The maximum amount of data that can be transmitted
over a network in a given amount of time, usually measured in bits per
second (bps).
2. Throughput: The actual amount of data successfully transferred over
the network, typically less than the bandwidth due to overheads.
3. Bitrate: The number of bits transmitted per second, commonly used in
streaming or communication systems.
4. Baud Rate: Refers to the number of signal units transmitted per
second, often related to modulation.
5. Gigabit Ethernet (GbE): Refers to network speeds of 1 Gbps or
higher.
Latency and Response Speed Terms
6. Latency: The time it takes for a packet to travel from the source to the
destination, measured in milliseconds (ms).
7. Round Trip Time (RTT): The time taken for a signal to travel to the
destination and back to the source.
8. Propagation Delay: The time taken for a signal to traverse the
physical medium.
Wireless Networking Speed Terms
9. Data Rate (Wi-Fi): The theoretical maximum speed of a wireless
network, such as 802.11ac (1.3 Gbps) or 802.11ax (Wi-Fi 6, up to 9.6
Gbps).
10. Channel Bandwidth: The frequency range within which data
can be transmitted, e.g., 20 MHz, 40 MHz, etc.
Other Speed-Related Concepts
11. Connection Speed: The rate negotiated between two network
devices during communication setup.
12. Download/Upload Speed: The rate at which data is
downloaded from or uploaded to a server, often measured in Mbps.
13. Clock Speed: Refers to the frequency at which a network device
(e.g., CPU or router) processes data, measured in GHz.
14. Packet Switching Speed: The rate at which network devices
(e.g., switches, routers) can process and forward packets.