Network Communication & Streaming Protocols
VoIP Protocols & Ports
| Protocol | Purpose | Default Port(s) |
|---|---|---|
| SIP (Session Initiation Protocol) | Call signaling (setup, modify, and terminate calls) | 5060 (UDP/TCP, unencrypted) 5061 (TCP, encrypted with TLS) |
| RTP (Real-time Transport Protocol) | Transmits voice & video data | Dynamic range: 16384–32767 (UDP, varies by implementation) |
| RTCP (RTP Control Protocol) | Monitors RTP sessions for QoS | Same as RTP, alternating port numbers (UDP) |
Why These Ports Matter?
- SIP (5060/5061) → Handles call initiation, management, and termination.
- RTP (16384-32767) → Delivers actual voice/video packets.
- RTCP → Provides quality feedback (e.g., packet loss, jitter).
Firewalls & NAT Configurations Must Allow These Ports for Smooth VoIP Communication!
SIP (Session Initiation Protocol)
📌 What is SIP?
SIP (Session Initiation Protocol) is a signaling protocol used to establish, modify, and terminate real-time communication sessions such as voice, video, and messaging over IP networks. It is widely used in VoIP (Voice over IP) systems, video conferencing, and multimedia applications.
Key Features of SIP:
✅ Session Management
- Establishes, modifies, and ends VoIP calls and multimedia sessions.
✅ Supports Multiple Media Types
- Handles voice, video, messaging, and file transfers.
✅ Peer-to-Peer Communication
- Devices communicate directly after the session is established.
✅ User Mobility
- Calls can be redirected to different devices (e.g., from a desk phone to a mobile).
✅ Works with Other Protocols
- Uses RTP (Real-time Transport Protocol) for media transmission.
- Can integrate with H.323 for video conferencing.
✅ SIP Addressing
- Users have a SIP address similar to an email (sip:user@domain.com).
How SIP Works (Call Flow)
1️⃣ User A dials User B
2️⃣ SIP INVITE request is sent to User B’s SIP server
3️⃣ User B’s SIP server forwards the request
4️⃣ User B accepts the call (SIP 200 OK response)
5️⃣ Media (voice/video) is transmitted using RTP
6️⃣ Either user can send a BYE request to end the session
SIP in Network Directories
📌 SIP Servers (PBX Systems) Maintain Call Records
- Call logs and metadata (time, duration, participants)
- Voicemail and call recordings
📌 User Data is Stored on SIP Proxies or Registrars
- Registrar Server stores user locations
- Proxy Server routes SIP messages
📌 Example SIP Directory Structure:
/var/log/sip/
├── call_records/
├── voicemail/
├── sip_users.db
Comparison: SIP vs. Other Communication Protocols
| Feature | SIP | H.323 | WebRTC |
|---|---|---|---|
| Purpose | VoIP & multimedia | Video conferencing | Web-based real-time comms |
| Flexibility | Highly flexible | Rigid architecture | Browser-native |
| Network Type | IP-based | IP + PSTN | Web-based |
| Ease of Use | Easier to implement | Complex setup | Simple for developers |
SIP is a flexible, widely used protocol for VoIP, video calls, and real-time communications.
RTSP (Real-Time Streaming Protocol)
📌 What is RTSP?
RTSP (Real-Time Streaming Protocol) is a network control protocol used to establish and control streaming media sessions between clients and servers. It is commonly used for live video streaming, IP cameras, and media playback applications.
Key Features of RTSP:
✅ Media Control (Not Transmission)
- RTSP controls streaming sessions (play, pause, stop, seek) but does not transmit media itself.
- Uses RTP (Real-time Transport Protocol) or RTCP for media delivery.
✅ Supports On-Demand & Live Streaming
- Can stream pre-recorded media or live feeds (e.g., security cameras, live broadcasts).
✅ Client-Server Communication
- Clients send RTSP requests, and the server responds with stream control commands.
✅ Efficient Bandwidth Usage
- Clients can start/stop streaming without reloading the entire media file.
✅ Works Over TCP & UDP
- Uses TCP for control signaling and UDP (via RTP) for media transport.
RTSP Request Flow (Example for Video Playback)
1️⃣ Client sends SETUP request → Server responds with transport details.
2️⃣ Client sends PLAY request → Server starts streaming via RTP.
3️⃣ Client sends PAUSE request → Server temporarily halts streaming.
4️⃣ Client sends TEARDOWN request → Server ends the session.
RTSP in File Directories
📌 Media Server Directory Structure:
/media/server/
├── movies/
├── live_feeds/
├── recordings/
-
rtsp://192.168.1.10:554/live/stream (Streaming from an IP camera)-
rtsp://media.example.com:554/video.mp4 (On-demand video streaming)
Comparison: RTSP vs. Other Streaming Protocols
| Feature | RTSP | HLS (HTTP Live Streaming) | DASH (Dynamic Adaptive Streaming over HTTP) |
|---|---|---|---|
| Latency | Low (Real-time) | High (Chunk-based) | Medium |
| Transport Protocol | RTP (UDP/TCP) | HTTP (TCP) | HTTP (TCP) |
| Use Case | Live streaming, IP cameras | On-demand & live streaming | Adaptive streaming |
RTSP is ideal for real-time streaming in surveillance, conferencing, and live media applications.
VoIP (Voice over Internet Protocol)
📌 What is VoIP?
VoIP (Voice over Internet Protocol) is a technology that enables voice communication over the internet instead of traditional phone lines (PSTN). It converts voice into digital packets and transmits them using IP networks.
Key Features of VoIP:
✅ Uses IP Networks for Calls
- Voice is transmitted as data packets over the internet.
- Can work on Wi-Fi, Ethernet, or mobile data (4G/5G).
✅ Cost-Effective
- Eliminates traditional phone charges.
- Free or low-cost international calls.
✅ Flexible & Scalable
- Works on IP phones, computers, and mobile devices.
- Can integrate with business PBX (Private Branch Exchange) systems.
✅ Supports Multimedia
- Handles voice, video, and text messaging.
✅ Uses SIP for Call Signaling
- SIP (Session Initiation Protocol) is commonly used to manage VoIP calls.
✅ Requires RTP for Media Transmission
- RTP (Real-time Transport Protocol) is used for delivering voice data.
How VoIP Works (Call Flow)
1️⃣ User dials a number → SIP sends an INVITE request.
2️⃣ Call is established → Media is transmitted via RTP.
3️⃣ Voice is digitized & compressed → Sent in IP packets.
4️⃣ Call ends → SIP sends a BYE request to terminate.
VoIP & File Directories
📌 VoIP Servers Store Call Logs & Recordings:
/var/voip/
├── call_logs/
├── recordings/
├── sip_users.db
- SIP Server: Manages call signaling.
- Media Gateway: Converts VoIP to PSTN (if needed).
- RTP Streams: Handle actual voice data.
Comparison: VoIP vs. Traditional Telephony
| Feature | VoIP | PSTN (Traditional Phone) |
|---|---|---|
| Cost | Low (Internet-based) | High (Line rental, call charges) |
| Scalability | Easy (Software-based) | Limited (Requires physical lines) |
| Call Quality | Dependent on network | Consistent but costly |
| Features | Voice, video, messaging | Voice only |
| VoIP is the future of communication, offering flexible, cost-effective voice and multimedia services over IP networks. |