Telecom Network Protocols — Beginner's Guide

Telecom network protocols are the standardized rules that communication devices follow to establish, maintain, and terminate connections — from the legacy SS7 system that routes phone calls to modern SIP that powers VoIP communications.

What You’ll Learn

• What SS7 is and how it routes traditional phone calls
• How SIP establishes and tears down VoIP calls
• The difference between signaling and media in telecom
• Call flow diagrams for both SS7 and SIP

Why Network Protocols Matter

Without protocols, phones wouldn’t know how to talk to each other. Telecom protocols are the secretaries of the phone world — they handle the introductions, negotiate the terms, and clean up after the conversation ends. Understanding them is essential for anyone working with telecom, VoIP, or network infrastructure.

Doda Browser uses SIP-inspired signaling for establishing secure connections. Durga Antivirus Pro applies SS7-style transaction management for tracking threat intelligence events across distributed agents.

Learning Path

    flowchart LR
  A[Telecom Basics] --> B[Network Protocols<br/>You are here]
  B --> C[VoIP & SIP]
  C --> D[4G/LTE Architecture]
  D --> E[5G Networks]
  

Signaling vs Media: The Two Conversations

Every phone call or video call has two separate data streams that travel alongside each other:

Think of signaling as the waiters in a restaurant. They take your order, tell the kitchen, bring your food, and bring the bill. The media is the food itself.

Signaling protocols (SS7, SIP) handle the waiter work. Media protocols (RTP) carry the actual voice/video.

SS7 — Signaling System 7

SS7 is the protocol suite that has run the world’s telephone networks since the 1970s. It’s the hidden intelligence behind every phone call.

What SS7 Does

• Call setup: Finds the recipient and establishes a path
• Routing: Determines the best path through the network
• Number translation: Converts dialed numbers to routing addresses (like DNS for phones)
• Billing: Tracks call duration for charging
• Text messaging: Carries SMS between carriers
• Call forwarding/ringing: Manages supplementary services

How SS7 Routes a Call

[Caller Phone] → [SSP] → [STP] → [SCP] → [STP] → [SSP] → [Receiver Phone]

Let’s trace a call:

1. SSP (Service Switching Point): Your local exchange detects you dialing a number
2. STP (Signal Transfer Point): Routes the signaling message through the network (like a router)
3. SCP (Service Control Point): The database that contains number translations and service logic. It asks: “Where is this number? What features does the caller have?”
4. The path is established: The SSP at the destination rings the recipient’s phone

SS7 Layers

SS7 has a layered architecture similar to OSI:

Real-World SS7 Example: Incoming Call

When someone calls your mobile:

1. The calling network sends an SS7 IAM (Initial Address Message) to the destination network
2. The destination network queries its HLR (Home Location Register) via SS7 to find your current location
3. Your phone rings
4. When you answer, SS7 exchanges ANM (Answer Message)
5. Billing starts
6. When one party hangs up, SS7 sends REL (Release) followed by RLC (Release Complete)

    sequenceDiagram
    Caller->>SSP: Dial number
    SSP->>STP: IAM (Initial Address Message)
    STP->>SCP: Query (where is this number?)
    SCP->>STP: Response (route to this switch)
    STP->>Destination SSP: IAM
    Destination SSP->>Phone: Ring
    Phone->>Destination SSP: Answer
    Destination SSP->>STP: ANM (Answer Message)
    Note over Caller,Phone: Conversation in progress
    Phone->>Destination SSP: Hang up
    Destination SSP->>STP: REL (Release)
    STP->>SSP: REL
    SSP->>STP: RLC (Release Complete)
  

SIP — Session Initiation Protocol

SIP is the modern protocol that sets up, manages, and tears down multimedia sessions — phone calls, video calls, instant messaging, and conferencing. It’s the backbone of VoIP.

How SIP Differs from SS7

SIP Messages

SIP uses text-based messages, similar to HTTP:

SIP Responses

Like HTTP, SIP uses status codes:

A SIP Call Flow

INVITE sip:bob@company.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060
From: <sip:alice@company.com>
To: <sip:bob@company.com>
Call-ID: 12345@192.168.1.10
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length: 150

v=0
o=alice 12345 67890 IN IP4 192.168.1.10
c=IN IP4 192.168.1.10
m=audio 49170 RTP/AVP 0

This INVITE message says: “Bob, Alice wants to talk to you. Here’s her IP address and the audio format she supports.”

The response flow:

Alice's Phone              SIP Server              Bob's Phone
    |                          |                       |
    |--- INVITE -------------->|                       |
    |                          |--- INVITE ----------->|
    |                          |<-- 100 Trying --------|
    |                          |<-- 180 Ringing -------|
    |<-- 180 Ringing ----------|                       |
    |                          |<-- 200 OK ------------|
    |<-- 200 OK ---------------|                       |
    |--- ACK ----------------->|--- ACK -------------->|
    |                          |                       |
    |<========== RTP MEDIA (voice) ==================>|
    |                          |                       |
    |--- BYE ----------------->|--- BYE -------------->|
    |                          |<-- 200 OK ------------|
    |<-- 200 OK ---------------|                       |

RTP — Real-Time Transport Protocol

While SS7 and SIP handle signaling, RTP carries the actual voice and video.

RTP packets contain:

• A timestamp (for synchronization)
• A sequence number (to reorder packets)
• The payload (encoded voice/video)

RTCP (RTP Control Protocol) monitors quality — packet loss, jitter, and round-trip time.

SDP — Session Description Protocol

SDP is like a negotiation menu within SIP. It describes:

• What media types are offered (audio, video)
• What codecs are available (G.711, G.729, Opus)
• What IP/port to send media to

v=0
o=- 12345 67890 IN IP4 192.168.1.10
s=-
c=IN IP4 192.168.1.10
t=0 0
m=audio 49170 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

This says: “I can accept audio on port 49170. I support G.711 μ-law (0), G.711 A-law (8), and telephone events (101).”

Security in Telecom Protocols

SS7 Security Vulnerabilities

SS7 was designed when only trusted carriers had access. Today, SS7 has serious security flaws:

• Location tracking: Anyone with SS7 access can locate any mobile phone
• Interception: Calls and SMS can be intercepted
• Fraud: SMS interception enables two-factor authentication bypass

SIP Security

SIP can be secured with:

• TLS (Transport Layer Security): Encrypts the SIP signaling (SIPS)
• SRTP (Secure RTP): Encrypts voice/video media
• Authentication: SIP REGISTER requires password or certificate
• IP-based access control: Firewalls restrict which IPs can send SIP traffic

Durga Antivirus Pro applies SIP-inspired session management for secure agent-to-console communication, using TLS-encrypted signaling channels similar to SIPS.

Real-World: How a VoIP Call Routes

When you call someone using WhatsApp, Skype, or a business phone system:

1. Registration: Your phone registers with the SIP server
2. Call initiation: You dial a number; your phone sends a SIP INVITE
3. DNS resolution: The SIP server performs DNS lookups to locate the destination (sip:user@company.com → IP address)
4. Routing: The SIP server routes the INVITE toward the destination
5. Negotiation: SDP exchange determines which codecs both sides support
6. Media establishment: RTP streams flow directly between both parties (or through a relay if NAT is involved)
7. Monitoring: RTCP tracks quality — if packet loss exceeds threshold, the call quality indicator turns red

Common Mistakes

1. Confusing signaling with media

Signaling sets up the call (SIP/SS7). Media carries the conversation (RTP). They are separate streams.

2. Thinking SS7 is fully replaced by SIP

SS7 still handles mobile network signaling (4G/LTE core, SMS). SIP is used in VoIP but hasn’t replaced SS7 in traditional telecom cores.

3. Ignoring NAT traversal in VoIP

SIP and RTP have trouble with NAT (Network Address Translation). STUN/TURN/ICE protocols are needed for calls across NAT.

4. Not securing SIP endpoints

Default SIP configurations often lack encryption. Unsecured SIP can be hijacked for toll fraud.

5. Assuming all codecs are equal

Audio codecs differ in bandwidth, quality, and CPU usage. G.711 uses 64 kbps; Opus uses 6-510 kbps with better quality at lower bitrates.

Practice Questions

1. What is the difference between SS7 and SIP? SS7 is the legacy circuit-switched signaling protocol for PSTN. SIP is the modern IP-based protocol for VoIP and multimedia sessions.

2. What does an SS7 IAM message do? IAM (Initial Address Message) initiates a call by sending the called number and routing information through the network.

3. What SIP method starts a call? INVITE.

4. What is the purpose of SDP in a SIP call? To negotiate media capabilities — codecs, IP addresses, and port numbers for the audio/video stream.

5. How do you secure SIP communications? Use TLS for signaling (SIPS), SRTP for media encryption, and strong authentication for registration.

Challenge: Draw a complete SIP call flow diagram for a call between two users on different VoIP providers. Include DNS lookup, SIP proxy routing, SDP negotiation, RTP media exchange, and call termination with BYE.

FAQ

Try It Yourself

You can inspect SIP-like communication using curl and Bash:

# Simulate a SIP OPTIONS query (check if a SIP server is alive)
# This sends an OPTIONS request to a public SIP server
echo "OPTIONS sip:example.com SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060
From: <sip:test@example.com>
To: <sip:test@example.com>
Call-ID: test@127.0.0.1
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0" | nc -w 3 -u example.com 5060

Or analyze VoIP with Wireshark:

# Install Wireshark (GUI-based protocol analyzer)
sudo apt-get install wireshark

# Or use tshark (command-line)
sudo tshark -i eth0 -f "port 5060 or port 5061"

What’s Next

Built by the developers of Doda Browser, DodaZIP, and Durga Antivirus Pro. Updated 2026-06-06.

What’s Next

Congratulations on completing this Network Protocols tutorial! Here’s where to go from here:

• Practice daily — Consistency is more important than long study sessions
• Build a project — Apply what you learned by building something real
• Explore related topics — Check out other tutorials in the same category
• Join the community — Discuss with other learners and share your progress

Remember: every expert was once a beginner. Keep coding!