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)

The WHEP signaller feature is available from version 0.15.0 of the published crate.

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	Environment Variable	Description	Default
-i, --input-url		WHEP source URL (required)	-
-o, --output-url		SRT output stream URL	srt://0.0.0.0:1234?mode=listener
--auth-token	WHEP_SRT_AUTH_TOKEN	Authorization token for WHEP endpoint	-
--latency	WHEP_SRT_JITTERBUFFER_LATENCY	Jitterbuffer latency in ms (sets rtpbin latency and liveadder min-upstream-latency)	200
--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:72):

• 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:81.

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)
• 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

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