09) Multimedia Networking

What are the three main types of multimedia networking applications?

1. Streaming stored audio/video (e.g. YouTube, Netflix) 2. Conversational voice/video over IP (e.g. Zoom, Teams) 3. Streaming live audio/video (e.g. live sports events)

How is analog audio digitized?

Sampled at a constant rate (e.g. 8000 samples/sec for telephone; 44100 for CD). Each sample is quantized and represented in bits (e.g. 8 bits = 256 values). Example: 8000 samples/sec x 8 bits = 64000 bps

What are common audio bit rates and quality levels?

64 kbps = telephone; 128 kbps = internet radio; 160 kbps = standard streaming; 320 kbps = max MP3; 1411 kbps = uncompressed CD quality (PCM/WAV)

What is spatial coding in video compression?

Instead of sending N repeated values of the same color, send only two values: the color and the number of repetitions (N)

What is temporal coding in video compression?

Instead of sending a complete frame, send only the differences from the previous frame

What is the difference between CBR and VBR video encoding?

CBR (Constant Bit Rate): encoding rate is fixed. VBR (Variable Bit Rate): encoding rate changes based on the amount of spatial/temporal coding needed

What is the continuous playout constraint in streaming video?

Once playback begins it must match the original timing. A client-side buffer is needed to absorb network jitter and ensure smooth playback

What happens when buffer fill rate x(t) < playout rate r? What about x > r?

x < r: buffer eventually empties causing video to freeze. x > r: buffer stays full with no starvation as long as initial playout delay is large enough

How does UDP streaming differ from HTTP/TCP streaming?

UDP: sends at encoding rate; short 2-5 sec playout delay; no congestion awareness; may be blocked by firewalls. HTTP/TCP: sends at max TCP rate; fill rate fluctuates due to congestion control; larger playout delay; passes through firewalls

What is DASH?

Dynamic Adaptive Streaming over HTTP. Server divides video into chunks stored at multiple quality levels with a manifest file. Client measures bandwidth and requests the highest sustainable quality chunk one at a time

What are the two CDN deployment strategies?

1. Enter Deep: push CDN servers deep into many access networks close to users (e.g. Akamai ~1700 locations). 2. Bring Home: fewer larger clusters near but not within access networks (e.g. Limelight)

How does DNS-based CDN content delivery work?

1. Client requests video URL. 2. DNS resolves to content provider. 3. Content provider DNS returns a CDN URL. 4. CDN DNS returns IP of best CDN server. 5. Client streams video from that CDN server

What are VoIP end-to-end delay requirements?

Less than 150 ms is good. Greater than 400 ms is bad and impairs interactivity. Includes packetization delay, playout delay, and network delay

What are the two types of VoIP packet loss?

1. Network loss: datagram lost due to router buffer overflow from congestion. 2. Delay loss: datagram arrives too late to be played out. Loss rates of 1-10% can typically be tolerated

What is the tradeoff in choosing VoIP fixed playout delay q?

Large q: fewer packets arrive too late (less loss) but worse interactivity. Small q: better interactivity but more packets arrive late and are lost

How does adaptive playout delay work in VoIP?

Estimates average delay using EWMA: d_i = (1-a)d_(i-1) + a(r_i - t_i). Also estimates deviation v_i. At start of each talk spurt sets: playout-time = t_i + d_i + K*v_i. Silent periods are compressed or elongated

How does a receiver detect the start of a new talk spurt?

Without loss: timestamp difference > 20 ms means new talk spurt. With possible loss: timestamp difference > 20 ms AND no gap in sequence numbers means new talk spurt

What are three VoIP packet loss recovery techniques?

1. Simple FEC: XOR n chunks to create a redundant chunk; can recover 1 lost chunk per group. 2. Piggyback lower-quality stream: send low-bitrate redundant copy alongside main stream. 3. Interleaving: spread small audio units across packets; no redundancy overhead but increases playout delay

What is RTP and what does it provide?

Real-Time Protocol (RFC 3550). Runs over UDP and provides: payload type ID (7 bits), sequence numbering (16 bits) to detect loss, timestamps (32 bits) for synchronization, and SSRC (32 bits) to identify the stream source

Does RTP provide QoS or guaranteed delivery?

No. RTP provides no QoS guarantees. Routers give RTP packets only best-effort service. RTP encapsulation is only visible at end systems

What is RTCP and what does it do?

Real-Time Control Protocol. Participants periodically send control packets containing sender reports (packets/bytes sent) and receiver reports (fraction lost, jitter, last seq#). Used for performance feedback and stream sync. Limited to ~5% of session bandwidth

How does RTCP synchronize audio and video streams?

Each RTCP sender report includes the RTP timestamp of the most recent packet and the wall-clock time it was created. Receivers use this mapping to align separate audio and video playout clocks

What is SIP and what services does it provide?

Session Initiation Protocol (RFC 3261). Handles call setup (INVITE/200 OK/ACK), codec negotiation, locating the callee's current IP, and call management (transfer, hold, add streams). Default port is 5060

What are the roles of a SIP Registrar vs a SIP Proxy?

Registrar: receives REGISTER messages and maps a user name to their current IP address. Proxy: routes INVITE messages toward the callee through one or more servers and returns the response with the callee's IP back to the caller

How does SIP compare to H.323?

H.323: complete vertically integrated suite from ITU with telephony flavor. SIP: single component from IETF with web/HTTP flavor; works with RTP but doesn't require it; uses the KISS (Keep It Simple Stupid) principle

What are the three approaches to network support for multimedia?

1. Best-effort: all traffic treated equally; no guarantees; low complexity; deployed everywhere. 2. Differentiated service: traffic classes with soft guarantees using marking/scheduling/policing. 3. Per-connection QoS: per-flow guarantees using admission control and scheduling; high complexity; rarely deployed

What are the four principles for QoS guarantees?

1. Packet marking: routers distinguish and treat traffic classes differently. 2. Policing/isolation: protect classes from misbehaving flows. 3. Efficient resource use: avoid wasting bandwidth with fixed non-sharable allocations. 4. Call admission: network blocks new flows if capacity is insufficient

What scheduling policies are used for QoS?

FIFO: serve in arrival order. Priority Scheduling: always serve highest-priority class first. Round Robin: cyclically serve one packet per class. Weighted Fair Queuing (WFQ): generalized RR where each class gets a weighted share of bandwidth

How does the Token Bucket policing mechanism work?

Bucket holds up to b tokens. Tokens added at rate r/sec (up to capacity). Each outgoing packet requires one token. Over interval t: packets admitted <= r*t + b. Controls both average rate (r) and burst size (b)

How do Token Bucket and WFQ together provide QoS guarantees?

Token bucket limits a flow to rate r and burst b. WFQ gives each flow a guaranteed minimum service rate R at the router. Together they provide a guaranteed upper bound on end-to-end delay