Multimedia Networking — VoIP, Streaming, CDN
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# Multimedia Networking — VoIP, Streaming, CDN ## 🎯 Learning Objectives - Explain SIP and RTP protocol operation - Describe video streaming and adaptive bitrate - Analyze CDN architecture and routing - Understand QoS requirements for multimedia * * * ## 1. VoIP Protocols ### 1.1 SIP (Session Initiation Protocol) *(...
Multimedia Networking — VoIP, Streaming, CDN
🎯 Learning Objectives
- Explain SIP and RTP protocol operation
- Describe video streaming and adaptive bitrate
- Analyze CDN architecture and routing
- Understand QoS requirements for multimedia
1. VoIP Protocols
1.1 SIP (Session Initiation Protocol)
(Diagram)
1.2 RTP (Real-time Transport Protocol)
- Header: Version, sequence number, timestamp, SSRC
- Runs over UDP (ports 16384-32767)
- RTCP provides feedback (packet loss, jitter)
2. Video Streaming
2.1 Adaptive Bitrate (ABR)
Client switches between quality levels based on bandwidth.
| Quality | Bitrate | Resolution | Buffer Time |
|---|---|---|---|
| 4K | 25 Mbps | 3840×2160 | 10s buffer |
| 1080p | 5 Mbps | 1920×1080 | 10s buffer |
| 720p | 2.5 Mbps | 1280×720 | 10s buffer |
| 480p | 1 Mbps | 854×480 | 10s buffer |
Protocols: HLS (Apple), DASH (MPEG), Smooth Streaming (Microsoft)
3. CDN (Content Delivery Network)
Architecture:
- Edge servers cache content near users
- Origin server holds master content
- DNS routes users to nearest edge (Diagram)
3.1 CDN Routing
- DNS-based: User resolves name → CDN DNS → returns nearest edge IP
- Anycast: Same IP advertised from multiple locations → BGP routes to nearest
- HTTP redirect: Edge returns 302 to better edge
4. Common Pitfalls
Pitfall: Confusing SIP with RTP
The mistake: Thinking SIP carries media.
Correct approach: SIP sets up and tears down sessions (control plane). RTP carries the actual audio/video data (data plane). They use different ports and protocols.
5. Key Concepts Reference
| Protocol | Port | Function |
|---|---|---|
| SIP | 5060/5061 (TLS) | Session setup/teardown |
| RTP | 16384-32767 | Media transport |
| RTCP | RTP+1 | Quality feedback |
| HLS | 80/443 | Adaptive streaming |
| DASH | 80/443 | Adaptive streaming |
6. 📝 Practice Questions
Q1: How does adaptive bitrate streaming prevent buffering?Answer: The client monitors download speed and buffer occupancy. If buffer drops below threshold, client requests lower bitrate segment (e.g., 1080p→720p). If bandwidth increases, client upgrades. This prevents buffer underrun at the cost of temporary quality reduction. YouTube uses this: you'll notice quality drops when network degrades. Q2: Why does RTP use UDP instead of TCP?Answer: Real-time applications prefer timeliness over reliability. TCP retransmissions cause delay and head-of-line blocking — a lost packet retransmitted 100ms later is useless for real-time audio. UDP accepts some packet loss (concealed by codecs) for lower latency. RTP runs over UDP; quality feedback is sent via RTCP. Q3: How does a CDN know which edge server is "nearest" to a user?Answer: Multiple techniques: (1) DNS geolocation — query resolver's IP to estimate user location. (2) Anycast — BGP routing automatically chooses closest edge. (3) Latency probing — measure RTT to possible edges. (4) Client IP subnet — /24 or /32 granularity. Cloudflare uses anycast; Akamai uses DNS-based routing with detailed maps.
7. 🔗 Cross-References
- Week 2 - TCP/UDP: Transport decisions
- Week 5 - Application Layer: HTTP-based streaming
- BSCS4003 (Privacy & Security): Secure streaming Join Discord PreviousNetwork ManagementNextNetwork Troubleshooting