WHEP to SRT Bridge

A GStreamer-based application that bridges WebRTC streams from WHEP (WebRTC HTTP Egress Protocol) endpoints to SRT (Secure Reliable Transport) output streams.

Overview

This tool consumes WebRTC media from a WHEP endpoint and re-streams it as SRT, enabling integration between WebRTC and SRT-based workflows. This is particularly useful for:

• Converting WebRTC streams to professional broadcast formats
• Integrating WebRTC sources into SRT-based production pipelines
• Low-latency streaming to SRT consumers
• Building bridges between web-based and broadcast infrastructure

Features

• WHEP Input: Consumes WebRTC streams via the WHEP protocol
• SRT Output: Outputs to SRT with configurable parameters
• Audio Processing: Automatically handles audio decoding, conversion, and AAC encoding
• Multi-track Support: Handles multiple audio tracks via audio mixing (liveadder)
• Continuous Output: Silent audio source ensures continuous stream even without input
• Docker Support: Ready-to-use Docker image with all dependencies included
• Flexible Configuration: Supports both whepsrc and whepclientsrc implementations

Prerequisites

Native Build Requirements

• Rust (1.83+ recommended, using 2024 edition)
• GStreamer 1.24+ with the following plugins:
  • gstreamer-plugins-base
  • gstreamer-plugins-good
  • gstreamer-plugins-bad
  • gstreamer-plugins-ugly
  • gstreamer-libav
  • gstreamer-nice
• Development libraries:
  • libssl-dev
  • libgstreamer1.0-dev
  • libgstreamer-plugins-base1.0-dev
  • libgstreamer-plugins-bad1.0-dev

GStreamer Rust Plugins

This project requires GStreamer Rust plugins from gst-plugins-rs:

• gst-plugin-webrtc (provides whepclientsrc with WHEP feature)

Note: The Cargo.toml currently uses a git dependency pinned to a specific commit SHA (e136005b108ec85bdc8bc533c551f56ef978e950) because the WHEP signaller feature is not yet available in the official crate release. This will be updated to use the published crate once the feature is available in the next official release.

Building

From Source

# Clone the repository
git clone <repository-url>
cd whep-srt

# Build the project
cargo build --release

# The binary will be at target/release/whep-srt

Using Docker

# Build the Docker image
docker build -t whep-srt .

# Run the container
docker run -it whep-srt -i <WHEP_ENDPOINT_URL> -o <SRT_OUTPUT_URL>

Usage

Basic Usage

./whep-srt -i <WHEP_INPUT_URL> -o <SRT_OUTPUT_URL>

Command Line Options

Option	Description	Default
-i, --input-url	WHEP source URL (required)	-
-o, --output-url	SRT output stream URL	srt://0.0.0.0:1234?mode=listener
--dot-debug	Output debug .dot files of the pipeline	false

Examples

Listen for SRT connections on port 1234 (default):

./whep-srt -i http://localhost:8889/mystream/whep

Push to a specific SRT destination:

./whep-srt -i http://localhost:8889/mystream/whep \
  -o "srt://192.168.1.100:5000?mode=caller"

Using Docker with port mapping:

docker run -p 1234:1234/udp whep-srt \
  -i http://host.docker.internal:8889/mystream/whep \
  -o "srt://0.0.0.0:1234?mode=listener"

Running the included debug script:

# Edit run.sh to configure your WHEP endpoint
./run.sh

Pipeline Architecture

The application dynamically constructs a GStreamer pipeline that:

1. WHEP Source: Connects to the WHEP endpoint using whepsrc or whepclientsrc (configurable)
2. Dynamic Pad Handling: Detects and handles audio/video tracks as they become available
3. Audio Processing Chain:
   • Decodes incoming audio tracks using decodebin
   • Converts audio to F32LE format at 48kHz
   • Mixes multiple audio tracks using liveadder
   • Adds a silent audio test source to ensure continuous output
   • Encodes to AAC using avenc_aac
4. Output Chain:
   • Muxes audio into MPEG-TS using mpegtsmux
   • Sends to SRT destination via srtsink

Pipeline String (when using whepsrc):

whepsrc → [dynamic audio pads] → decodebin → audioconvert → audioresample →
capsfilter → liveadder ← audiotestsrc (silence) → avenc_aac → aacparse →
mpegtsmux → queue → srtsink

Pipeline String (when using whepclientsrc):

whepclientsrc → [dynamic audio pads] → decodebin → audioconvert → audioresample →
capsfilter → liveadder ← audiotestsrc (silence) → avenc_aac → aacparse →
mpegtsmux → queue → srtsink

Configuration

SRT Parameters

The SRT output URL supports standard SRT URI parameters:

• mode=listener - Wait for incoming connections (default)
• mode=caller - Connect to a remote SRT receiver
• latency=<ms> - Set SRT latency buffer (default: 100ms)
• Additional parameters supported by GStreamer's srtsink element

WHEP Source Selection

The application supports two WHEP source implementations (configurable in src/main.rs:56):

• whepclientsrc (currently enabled) - From gst-plugin-webrtc - Newer implementation using signaller interface (will eventually replace whepsrc)
• whepsrc - From gst-plugin-webrtchttp - Original WebRTC implementation based on webrtcbin

Toggle between them by changing the whepsrc boolean variable in the code. Note: whepclientsrc requires the plugin to be registered via gstrswebrtc::plugin_register_static() as shown in src/main.rs:65.

Supported Codecs

Audio Input (via RTP):

• OPUS (default, 48kHz)

Video Input (via RTP):

• VP8, VP9 (default)
• H.264, H.265
• AV1

Note: Video tracks are currently sent to fakesink and not included in SRT output.

Development

Debug Logging

Enable GStreamer debug output using environment variables:

# Show all debug output
GST_DEBUG=*:DEBUG ./whep-srt -i <WHEP_URL>

# Show WHEP-specific debug output
GST_DEBUG=*whep*:DEBUG ./whep-srt -i <WHEP_URL>

# Save debug log to file
GST_DEBUG_FILE=debug.log GST_DEBUG=*:DEBUG ./whep-srt -i <WHEP_URL>

# Generate pipeline visualization (DOT files) using the --dot-debug flag
./whep-srt -i <WHEP_URL> --dot-debug

# Or set the environment variable directly
GST_DEBUG_DUMP_DOT_DIR=./ ./whep-srt -i <WHEP_URL>

Pipeline Visualization

The application automatically generates GraphViz DOT files of the pipeline on state changes and errors when the --dot-debug flag is used. The files are timestamped with the format <epoch>-<state>.dot (e.g., 1729000000-Playing.dot, 1729000000-error.dot). Convert them to SVG for visualization:

# Convert a DOT file to SVG
dot -Tsvg 1729000000-error.dot -o pipeline.svg

# Or use xdot for interactive viewing
xdot 1729000000-error.dot

Code Structure

• src/main.rs - Main application logic
  • Command-line argument parsing (Args struct)
  • Pipeline construction and management
  • Dynamic pad handling for audio/video tracks
  • Event loop and error handling
  • Debug pipeline visualization (debug_pipeline function)

Known Issues & Limitations

• Video handling: Video tracks are currently discarded (sent to fakesink)
• Git dependency: Uses pinned git commit for gst-plugin-webrtc until WHEP feature is available in published crate
• Audio-only output: Only audio is currently muxed to SRT output

Troubleshooting

Missing GStreamer elements: If you get errors about missing elements, ensure all required GStreamer plugins are installed:

gst-inspect-1.0 whepclientsrc
gst-inspect-1.0 srtsink
gst-inspect-1.0 avenc_aac

SRT connection issues: Check your firewall settings and ensure the SRT port (default 1234/udp) is accessible.

No audio output: Enable debug logging to see if audio pads are being created and linked correctly.

Future Improvements

• Add video support to SRT output

• Support for published crates.io versions of gst-plugins-rs

License

See the LICENSE file for details.

Author

Per Enstedt <per.enstedt@eyevinn.se>

Developed at Eyevinn Technology

Contributing

Contributions are welcome! Please feel free to submit issues or pull requests.

Related Projects

• GStreamer - Multimedia framework
• gst-plugins-rs - GStreamer plugins written in Rust
• WHEP Specification - WebRTC HTTP Egress Protocol
• SRT Alliance - Secure Reliable Transport protocol