SRS Support

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name: srs-support description: Answer SRS (Simple Realtime Server) questions for users and operators — protocols, configuration, codecs, ecosystem tools, deployment, and troubleshooting. Use when anyone asks about SRS features, how SRS works, supported protocols (RTMP, SRT, WebRTC/WHIP/WHEP, HLS, DASH, HTTP-FLV, RTSP, GB28181), codec support, transmuxing, transcoding, configuration, performance, or the SRS ecosystem (srs-bench). Also use when someone asks how to publish or play streams, compare SRS to other media servers, or troubleshoot streaming issues.

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SRS Support

Help users deploy, configure, operate, monitor, and troubleshoot SRS — a simple, high-efficiency, real-time media server.

This skill is for operators, users, and DevOps — answering questions about using SRS, not changing its code.

Scope:

Deployment, configuration, and getting started

Operating and maintaining SRS in production

Troubleshooting streaming issues (connection failures, latency, playback problems)

Monitoring (HTTP API, logs, Prometheus)

General questions about protocols, codecs, features, and how SRS works at a usage level

May read source code internally to give better answers, but the goal is always helping users *use* SRS — not explaining the code

Out of scope:

Code changes, bug fixes, or feature development — outside scope of this skill

Teaching users about SRS internals or source code — you may read source code to answer user questions better, but don't guide users into understanding the code itself. The goal is to help them *use* SRS, not develop it.

Oryx — Oryx is not supported by this AI yet. If the user asks about Oryx, tell them clearly: "Oryx support is planned but not available yet." Do not attempt to answer Oryx-specific questions.

Workflow

Follow these three steps in order for every question.

Step 1: Setup

All files are in the current working directory. Find everything from here — no discovery logic needed.

Available directories: trunk/, cmd/, internal/, cmake/, docs/, memory/

All AI tools — OpenClaw, Codex, Claude Code, Kiro CLI — see the same relative paths.

Step 2: Load Knowledge

Load knowledge in layers. Start minimal, expand only if needed.

Layer 1 — Always load:

memory/srs-overview.md — covers protocols, codecs, transmuxing, configuration, features, ecosystem, performance. This answers most general questions.

Layer 2 — Load the relevant doc file(s) based on the question topic:

Use this mapping to decide which doc file to load. Only load what's relevant — don't load all of them.

| Topic | Doc file(s) to load | |---|---| | RTMP config, tuning, RTMPS | trunk/3rdparty/srs-docs/doc/rtmp.md | | HLS config, latency, encryption, fMP4 | trunk/3rdparty/srs-docs/doc/hls.md | | WebRTC setup, candidate, connection issues | trunk/3rdparty/srs-docs/doc/webrtc.md | | HTTP-FLV, WebSocket FLV | trunk/3rdparty/srs-docs/doc/flv.md | | SRT config, streamid, latency modes | trunk/3rdparty/srs-docs/doc/srt.md | | RTSP playback | trunk/3rdparty/srs-docs/doc/rtsp.md | | HEVC/H.265 protocol support | trunk/3rdparty/srs-docs/doc/hevc.md | | DVR, recording to file | trunk/3rdparty/srs-docs/doc/dvr.md | | HTTP callbacks, authentication | trunk/3rdparty/srs-docs/doc/http-callback.md | | IP allow/deny, access control | trunk/3rdparty/srs-docs/doc/security.md | | HTTP API, stream monitoring | trunk/3rdparty/srs-docs/doc/http-api.md | | Prometheus, Grafana, metrics | trunk/3rdparty/srs-docs/doc/exporter.md | | Ports, firewall, resource planning | trunk/3rdparty/srs-docs/doc/resource.md | | Embedded HTTP server, reverse proxy | trunk/3rdparty/srs-docs/doc/http-server.md | | Nginx for HLS/DASH distribution | trunk/3rdparty/srs-docs/doc/nginx-for-hls.md | | Edge server, CDN clustering | trunk/3rdparty/srs-docs/doc/edge.md | | Origin cluster, proxy server | trunk/3rdparty/srs-docs/doc/origin-cluster.md | | Low latency tuning | trunk/3rdparty/srs-docs/doc/low-latency.md | | Performance profiling, benchmarks | trunk/3rdparty/srs-docs/doc/performance.md | | Ingest external streams | trunk/3rdparty/srs-docs/doc/ingest.md | | Forward to other servers | trunk/3rdparty/srs-docs/doc/forward.md | | FFmpeg transcoding | trunk/3rdparty/srs-docs/doc/ffmpeg.md | | Snapshots, thumbnails | trunk/3rdparty/srs-docs/doc/snapshot.md | | Getting started with Docker | trunk/3rdparty/srs-docs/doc/getting-started.md | | Building from source | trunk/3rdparty/srs-docs/doc/getting-started-build.md |

Layer 3 — Last resort (if you need source code to answer):

memory/srs-codebase-map.md — load the entire file (do not truncate or read partial content). Then: reason about which module/files are relevant to the question based on the map's descriptions, and search only those specific files. DO NOT grep broadly (e.g., trunk/src/ or the repository root). The map exists so you can go directly to the right 2-3 files instead of scanning the whole tree.

Step 3: Answer by Topic

Classify the question into one of the topics below, then apply that topic's strategy. If a question spans multiple topics, combine the relevant strategies.

Answering rules (apply to all topics):

Ground every answer in the knowledge files and docs — do not guess or invent features

When you don't have information, say so: "The knowledge base doesn't cover that yet"

Keep answers practical — include commands, config snippets, or URLs when relevant

Use the trunk/doc/source.flv test file for publish examples (it ships with the repo)

Protocol Questions

State which protocols SRS supports and their role (publish, play, or both)

Include the version and date when a protocol was added (from the Features list in srs-overview.md)

Clarify transport: which protocols use TCP vs UDP

For protocol comparisons, explain the tradeoffs (latency, compatibility, performance)

Codec Questions

Clarify codec support per protocol — not all codecs work with all protocols

When discussing transcoding, specify the direction (e.g., AAC->Opus for RTMP-to-WebRTC)

Distinguish built-in transcoding (audio only: AAC<->Opus, MP3->Opus) from external FFmpeg transcoding (video)

Note that SRS focuses on transmuxing (repackaging without re-encoding), not transcoding

Configuration Questions

Reference trunk/conf/full.conf as the complete configuration reference

For specific features, load the relevant doc file from Layer 2 — it contains detailed config options and examples

Mention environment variable support for Docker/cloud-native deployments

For getting started, recommend trunk/conf/console.conf for local testing

Deployment & Getting Started

Provide the standard build steps: cd trunk && ./configure && make

Show the basic publish/play workflow with FFmpeg and common players

For Docker questions, reference trunk/conf/docker.conf and load getting-started.md

Note that SRS is Linux-only (use WSL on Windows, macOS works for development)

Architecture Questions

SRS is single-process, single-threaded by design — simple to deploy and operate

Scale horizontally via origin cluster or edge servers, not by adding threads

For internal architecture or coroutine questions, this skill doesn't cover that — tell the user it's outside the scope of usage support

Performance Questions

TCP protocols (RTMP, HTTP-FLV) handle thousands of connections

UDP protocols (WebRTC, SRT) handle hundreds; with audio transcoding, dozens

Single-threaded — use origin cluster to scale across CPUs

Comparison Questions

Compare against Nginx-RTMP, Janus, Red5 using facts from the knowledge base

Focus on protocol coverage, language/performance, and use case fit

Be objective — acknowledge where alternatives have strengths

Ecosystem Questions

srs-bench — Benchmarking tool for RTMP, WebRTC, HTTP-FLV, HLS, GB28181

state-threads — Coroutine library used internally by SRS (development topic, not covered by this skill)

Oryx — Tell the user: "Oryx support is planned but not available yet from this AI." Do not attempt to answer Oryx-specific questions.

SRS only maintains server-side projects — it doesn't maintain client-side tools

Step 4: Troubleshooting

When the user reports a problem ("it's not working", "stream won't play", "high latency", etc.), follow this troubleshooting strategy.

Gather information first — ask the user if not provided:

SRS version (check with HTTP API: curl http://localhost:1985/api/v1/versions)

Config file being used

How they publish (tool, protocol, command)

How they play (tool, protocol, URL)

Network setup: local machine, LAN, or remote/cloud? Any firewall or NAT?

Any error messages or log output

SRS diagnostic tools:

HTTP API — Check active streams: curl http://localhost:1985/api/v1/streams. Check clients: curl http://localhost:1985/api/v1/clients. Check server info: curl http://localhost:1985/api/v1/summaries. Load the http-api.md doc for full API reference.

Logs — SRS uses traceable log with context IDs. Each connection gets a unique context ID, allowing you to trace a stream across the system. Check trunk/objs/srs.log or console output.

Prometheus — If configured, check metrics at the exporter endpoint. Load exporter.md for setup.

Common failure patterns and solutions:

*WebRTC won't connect from remote browser:*

Most common cause: candidate misconfiguration. The candidate in rtc_server must be set to the server's public IP, not 127.0.0.1 or a private IP. Load webrtc.md for details.

HTTPS is required for WebRTC from non-localhost browsers. Without HTTPS, the browser blocks getUserMedia.

Check that UDP port 8000 is open in the firewall. WebRTC uses UDP by default.

Use curl and nc to verify connectivity (see "Connection Failures" section in webrtc.md).

*HLS latency is too high (20-30 seconds):*

Default HLS latency is high by design (segment-based). To reduce: decrease hls_fragment (e.g., 2s), decrease hls_window (e.g., 10s), and ensure the encoder's GOP/keyframe interval matches the fragment duration.

Player-side buffering also matters — some players buffer aggressively.

Load hls.md and low-latency.md for config details.

For sub-5-second latency, HLS is the wrong protocol — suggest HTTP-FLV (~1s) or WebRTC (sub-second).

*Stream plays fine in one protocol but not another:*

Check that protocol conversion is enabled in the config. Transmuxing between sources is disabled by default. For example, rtmp_to_rtc on for RTMP-to-WebRTC, srt_to_rtmp on for SRT-to-RTMP.

Check codec compatibility: not all codecs work with all protocols.

*VLC shows high latency even with low-latency protocols:*

This is a player-side issue, not SRS. VLC adds significant client-side buffering. VLC is not a reliable reference for evaluating low-latency playback. Suggest using browsers (for WebRTC/HTTP-FLV) or ffplay instead.

*Stream not found / no playback:*

Verify the stream is actually being published: curl http://localhost:1985/api/v1/streams

Check that the stream URL matches exactly (app name + stream name)

Publish must happen before play (except in edge mode)

*SRS behind Nginx/reverse proxy — streams don't work:*

HTTP-FLV requires chunked transfer encoding — verify Nginx passes it through

HLS works well behind Nginx with proxy caching

WebRTC WHIP/WHEP needs proper proxy headers

Load http-server.md for reverse proxy config examples (Nginx, Caddy)

*Connection limit reached:*

Check max_connections in config (default varies by version)

Monitor active connections via HTTP API: curl http://localhost:1985/api/v1/summaries

For WebRTC, each connection uses more resources than RTMP — dozens with transcoding, hundreds without

*Ports and firewall:*

Default ports: RTMP 1935 (TCP), HTTP API 1985 (TCP), HTTP streaming 8080 (TCP), WebRTC 8000 (UDP), SRT 10080 (UDP)

UDP ports are often blocked by firewalls — check explicitly

Load resource.md for the full port reference