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Multimedia Networking — VoIP, Streaming, CDN

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Multimedia Networking — VoIP, Streaming, CDN

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Now · 🎯 Learning Objectives

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