Created by ChatGPT o3
This concise primer refreshes the IP → UDP/TCP → NAT layers you’ll use constantly while building a peer‑to‑peer calling app.
1. Internet Protocol (IP)
1.1 Packets & Addressing
- IPv4: 32‑bit addresses (e.g.
192.168.1.42). About 4.3 billion total, now scarce. - IPv6: 128‑bit addresses (e.g.
2001:0db8::1). Practically inexhaustible. - Packet = header (metadata) + payload (upper‑layer data).
- IP is connectionless and best‑effort: no guarantees on delivery order, timing, or integrity.
1.2 Fragmentation
If a packet exceeds the network’s MTU (~1500 bytes on Ethernet), routers may split it into fragments—bad for real‑time media (extra delay & loss risk). Keep RTP payloads small.
1.3 Port Numbers
Ports live in UDP/TCP, not IP, but many tools show them together (e.g., 192.0.2.10:3478).
2. Transport Layer Choices
| Feature | UDP | TCP |
|---|---|---|
| Connection setup | None (stateless) | 3‑way handshake (SYN/ACK) |
| Reliability | None—packets may drop or reorder | Built‑in ACKs, retransmission |
| Ordering | Not guaranteed | Guaranteed (byte stream) |
| Overhead | Low (8‑byte header) | Higher (20‑byte header + control flags) |
| Latency impact | Minimal | Can spike (Nagle’s algorithm, congestion control) |
| Typical uses in VoIP | RTP media, STUN, DTLS‑SRTP | SIP over TCP/TLS, WebSocket signalling |
Takeaway: Real‑time audio/video prefers UDP—lost frames are cheaper than late ones. Signalling prefers TCP—small, reliable.
2.1 RTP over UDP
RTP rides atop UDP. RTP adds sequence numbers & timestamps so receivers can re‑sequence and sync playback.
2.2 TCP & TLS for Signalling
SIP can run on UDP, TCP, or TCP+TLS. WebRTC browsers usually tunnel signalling JSON over WebSocket (TCP) to traverse firewalls cleanly.
3. Network Address Translation (NAT)
3.1 Why NAT Exists
- Preserve scarce IPv4 addresses.
- Mask internal topologies for security.
3.2 How It Works
A home router keeps a translation table mapping:
private_IP:private_port ⇄ public_IP:public_port
Outgoing packets rewrite the source. Inbound replies must hit the exact 5‑tuple or they’re dropped.
3.3 NAT Types (RFC 4787)
| Type | Mapping Stability | Filtering Strictness | VoIP Impact |
|---|---|---|---|
| Full Cone | Fixed port | Loose | Easiest |
| Restricted | Fixed port | IP‑restricted | OK |
| Port‑Restricted | Fixed port | IP+port restricted | Tougher |
| Symmetric | New port per dest | Strict | Hard—needs TURN |
3.4 Traversal Tools
- STUN – learn your public IP/port (works unless symmetric NAT).
- ICE – gather candidates (host, srflx, relay) and perform connectivity checks.
- TURN – relay packets when P2P fails; last resort.
3.5 Voice Quality vs NAT
Each extra hop adds latency. TURN adds ~30–50 ms RTT in a local network; keep the relay as close as possible or avoid it.
4. Quick Checklist Before Coding
Master these basics and the SIP/WebRTC layers in your course will feel much easier.