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How Casino Prevents Delays and Monitors Flow Quality
1) Signal path map: where delay is born
Camera → Encoder. Low-latency settings: short GOP (1-2 s), limited B-frames, CBR/hard VBR, scheduled keyframes.
Encoder → Media Server. For interactive - WebRTC via SFU (Selective Forwarding Unit); for mass coverage - LL-HLS/DASH with segments of 200-500 ms.
Media Server → CDN. Edge caches segments, reducing the load on origin; WebRTC is not cached - emphasis on SFU channel width and smart fan-out.
Viewer network. ABR-ladder, jitter-buffer, frame/bitrate adaptation, quick profile switching without black screens.
Key idea: Delay is made up of small buffers along the way. To manage is to control each buffer and its "budget."
2) Basic principles of delay prevention
1. Segmentation under LL-HLS: short partial segments + low 'targetDuration'.
2. WebRTC profile: reduced deceiver buffer, prioritization of RTP streams, fast keyframes on demand.
3. Anti-jitter: adaptive jitter-buffer, NACK (lost packet retransmission), PLI/FIR (keyframe request), if necessary - FEC (forward error correction).
4. Backpressure in SFU: lowering the frame rate/bitrate and skipping non-priority layers (SVC) instead of a total drop.
5. Edge-proximity: routing viewers to the nearest PoP, origin-shield to offload the source.
6. Multi-CDN: RUM routing by real metrics (TTFB, error-rate), automatic fake.
3) What is "quality" in SLI/SLO terms
SLI (quality attributes):- e2e-delay (glass-to-glass)
- percentage buffering ratio and average duration of buffering drop-frame rate (lost frames)
- startup time
- bitrate-downgrade events
- WebRTC: RTT, packet loss, jitter, NACK/FEC share, TURN-relay share
- LL-HLS: segments on time (% of segments <1.5 s), manifest fetch errors
- 95p e2e WebRTC latency ≤ 2.5 s; LL-HLS ≤ 5 c rebuilding ratio <0.5% session; startup < 1,5 c (WebRTC) / < 2,5 c (LL-HLS)
- packet loss ≤ 1% (95p); RTT ≤ 120ms (95p)
- cache-hit CDN ≥ 80%, origin-egress ≤ 20% of total traffic
4) Active monitoring: how to catch problems before the player
Synthetic probes: robots connect to tables from different regions, measure startup, e2e-delay (by water timecodes), percentage of late-segments, WebRTC-RTT/packet loss.
Test beacons in video: an overlay with a → time stamp allows you to estimate e2e latency up to milliseconds.
Control tables/channels: one table "for monitoring" with a fixed scenario (card mill, "pendulum" for estimating frame gaps).
Periodic health-checks: provider/wallet API, TURN availability, TLS/certificate validity, IP-allowlist.
5) Passive monitoring: what is collected in real traffic
RUM (Real User Monitoring): The SDK on the client sends telemetry by segment/frame, buffers, profile changes, decoder errors.
WebRTC-stats: standard counters (inbound/outbound RTP, framesDropped, jitter, nackCount, pliCount, roundTripTime).
Player events: 'play', 'stall', 'recover', 'seek', 'qualitychange', 'fatal'.
Server metrics: CPU/GPU load of transcoders, egress to SFU/edge, QPS by manifest/segment, p95 API for debits/credits of bets.
Correlation: Peaks of 'late-bet' and disputed rounds often coincide with bursts of e2e delay - a signal to investigate.
6) Auto degradation without pain for the player
FPS reduction before resolution reduction. 60→48→30, then drop the 1080p→720p profile.
SVC/simulacast: sending several quality layers; SFU turns off the upper layers when overloaded.
Keyframe on demand: quick keyframe when changing profile to avoid "soap" and long resynchronization.
Buffer adaptation: temporarily expand the client buffer by 200-400 ms in case of an unstable network and return it back after stabilization.
Silent feedback: WebRTC → LL-HLS for "visual" feed in case of problems, blocking late bets.
7) Network and anti-loss: why "0% loss" does not happen
NACK/RTX: point retransmissions of lost packets.
FEC: redundancy at the RTP level - useful on dirty networks, but increases the bitrate.
Jitter-buffer adaptive: hold 60-150 ms; grow to 250-300 ms with bursts, then reduce.
DSCP/prioritization (where available): the priority of voice/video over bulk traffic in corporate networks.
TURN pool: white IP, geo-distribution, monitoring the share of relay sessions (if> 25% - check locks/firewalls/peering).
8) CDN architecture and origin protection
Origin-shield: Centre cache between edge and origin - dramatically reduces misses at peaks.
Multi-CDN: DNS/anycast router + RUM signals; automatic traffic flow when errors or TTFBs grow.
Manifests and segments: short TTL, prefetch of the next segment, priority channels for manifests (they are "more critical" than segments).
Protection: signed URLs, short TTL tokens, geo/ref restrictions, hotlink and restriction protection.
9) Encoders and transcoders: the more powerful, the more stable
CPU + GPU hybrid: ABR ladder on GPU (NVENC/Quick Sync), premium x264 CPU profile for quality.
Profiles for a mobile audience: 240p/360p/540p/720p - it is better to have a 540p "step" for medium-sized networks.
GOP/IDR frequency control: fast profile swap and accelerated recovery after losses.
Redundancy: hot reserve of transcoders; during overload - auto-off of "expensive" profiles (1080p60) with stability priority.
10) Incidents: How do they react as the round goes on
Real-time alerts: "95p e2e-delay> target," "rebuilding> threshold," "TURN-relay up> X%," "cache-hit down 1. Check region/PoP → switch to another CDN provider. 2. Include lean profiles (below FPS/bitrate). 3. Forced keyframe to speed up resynchronization. 4. Folback WebRTC → LL-HLS for viewers; on tables - a temporary lengthening of the betting window or a pause with a transparent announcement. Communication: banner in the player ("there is a stabilization of the stream"), incident log, post-mortem act. 11) Video-betting connection: Honesty trumps pixels Time synchronization: NTP/chrony on all nodes; events'round. result 'and' close bets' - with exact labels' video _ ts'. The "source of truth" is the round server. UI displays the result to the client only after server commit; replays are available for parsing. Anti-latent abuse: blocking bets when the viewer's e2e delay is above the threshold; if the thread degrades, the protection translates to "view only." 12) Dashboards: what is always at hand at NOC/VideoOps Video: e2e, startup, rebuffering, drop-frame, quality-switches, keyframes/min. WebRTC: RTT, loss, jitter, bitrate, NACK/PLI frequency, relay-ratio by TURN. CDN: cache-hit, TTFB, PoP/ASN errors, traffic/egress. Servers: transcoder CPU/GPU, egress SFU, sockets/FD, p95 API. Продукт: late-bet rate, dispute rate, session length, retention. 13) Safety and quality impact TLS termination on edge (minimum unnecessary cipher-hops). Short TTL tokens/URL: less chance of "hung" old manifestos for the client. IP-allowlist, mTLS for S2S: more stable connections, more transparent diagnostics. PII minimization: less processing overhead, easier cache strategy. 14) Live Quality Launch Checklist Delay prevention and quality control in live casinos is not one "magic setting," but a discipline: strict encoding profiles, smart media servers and ABRs, multi-CDN with origin-shield, anti-loss (NACK/FEC/PLI) and meticulous monitoring (RUM + synthetics) with understandable runbook s. When each layer knows its "delay budget," and the team sees the metrics in real time and knows how to gently degrade the quality, the player gets a stable stream and honest timing of bets - that's why the live format exists.
Network and CDN
Encoding and player
Monitoring
Operations