Chaos Monkey but for Audio Video Testing (WebRTC and UDP)
Details
- External ID
- 47119767
- Source
- HN
- Company
- —
- Product
- Chaos Monkey but for Audio Video Testing (WebRTC and UDP)
- Website domain
- github.com
- Launched
- Feb. 23, 2026
- Cohort
- —
- Upvotes
- 41
- Upvotes percentile
- 0.7897574123989218
- Tags
- —
- Fetched at
- Sept. 7, 2026, 9:25 p.m.
- Updated at
- Sept. 7, 2026, 9:25 p.m.
Description
It takes an input video and converts it into H.264/Opus RTP streams that you can blast at your video call systems (WebRTC, SFUs, etc.). It also injects network chaos like packet loss, jitter, and bitrate throttling to see how things breakIt scales from 1 to n participants, depending on the compute and memory of the host system Best part? It’s packaged with Nix, so it builds the same everywhere (Linux, macOS, ARM, x86). No dependency hellIt supports both UDP (with a relay chain for Kubernetes) and WebRTC (with containerized TURN servers). Chaos spikes can be distributed evenly, randomly, or front/back-loaded for different test scenarios. To change this, just edit the values in a single config file
Enrichment
- Theme
- audio and video developer utilities
- Vertical
- Horizontal
- Function
- Dev tools
- Audience
- Developer
- AI stance
- Not AI
- Project type
- Hobby / open-source project
- Normalized one-liner
- chaos testing for audio video webrtc and udp
- Manually corrected
- False
Could you build this?
Partial Creating a web dashboard or CLI is easy, but generating valid H.264/Opus RTP packet streams and simulating network chaos (loss, jitter, congestion) for real-time media requires deep multimedia and WebRTC expertise.
What it would actually take: Requires a media pipeline using GStreamer, FFmpeg, or Pion/libdatachannel integrated with low-level traffic control (Linux netem/tc or raw socket packet mangling). The hard part is generating synchronized RTP streams across dynamically scaled simulated peers while injecting controlled jitter, packet reordering, and MTU fragmentation without crashing media decoders. Demands deep domain expertise in real-time communication (WebRTC/RTP/RTCP) and network simulation.
Discussion
2 comments analyzed.
Concerns raised: Latency issues from connection speed changes in video channels, Video and data channel latency correlation problems
Feature requests: SIP protocol support
Competitors
Other products that read as similar to this one — 19 launches clear the similarity bar, closest 8 shown.
Attention rank: #6 of 20 (itself plus its competitors, highest first — normalized so YC and Product Hunt are compared fairly).
Launched 80 days after the earliest competitor.
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- turnrelay · github · 2026-09-14 · 8 upvotes · similarity 0.32
Other launches for this product
- No other launches for this product.
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