Course Objective & Scope
Streaming audio, video and control events with millisecond latency between browsers, mobile apps or telephony lines and multimodal models requires solid systems and network engineering. Choosing the wrong transport or mishandling buffers causes audio dropouts, packet loss bursts and server memory exhaustion.
In this technical course participants position WebSockets, Server-Sent Events (SSE) and WebRTC at the right architectural tiers, build bidirectional client–edge–model pipelines, implement ephemeral token security, and produce a horizontal scaling and latency measurement plan.
Who Should Attend?
- Backend, platform and systems engineers building real-time AI infrastructure
- WebRTC, media server and low-latency API developers
- Full-stack engineers working on real-time interactive applications
- Prerequisite: Python/Node.js/Go, networking fundamentals and async programming experience
Key Takeaways
- Choose correctly between WebRTC at the client edge, WebSockets server-to-server and SSE for text/event streams
- Manage OpenAI Realtime API and Gemini Live API session and conversation state events end to end
- Implement PCM/Opus audio framing, jitter buffers and backpressure handling
- Secure direct browser-to-model WebRTC connections using ephemeral tokens and SDP handshakes
- Scale stateful long-lived streaming sessions across load balancers and media workers
Curriculum
Streaming Transports: SSE, WebSockets and WebRTC
- SSE for text/events, WebSockets for server-to-server and WebRTC for last-mile media
- TCP head-of-line blocking versus UDP/RTP media transport
OpenAI Realtime and Gemini Live Wire Protocols
- Client and server event lifecycles (input_audio_buffer, response.create, conversation.item)
- Ephemeral API keys, WebRTC data channels and secure session bootstrapping
Audio Framing, Codecs and Buffer Management
- PCM16, G.711 (mu-law/a-law) and Opus transcoding and resampling
- Ring buffers, jitter handling and preventing audio underruns under network jitter
Scaling Stateful Streaming Services
- LiveKit / SFU media server topologies and agent worker pool orchestration
- Edge routing, graceful draining and session reconnection
Telemetry, Load Testing and Latency Measurement Plan
- Simulating concurrent voice streams and packet-loss conditions
- Tracking p50/p95/p99 latency across network, model and playback stages
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