The SRT Target is an output component that sends video and audio from Composer to any SRT (Secure Reliable Transport) receiver. SRT is designed for low-latency, high-quality video transmission over unpredictable networks, making it ideal for live production workflows.
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When to Use SRT Target
- Live streaming to SRT receivers: Send content to media servers, encoders, or CDNs that accept SRT input
- Contribution feeds: Deliver high-quality video over the internet with low latency
- Point-to-point streaming: Direct streaming between locations with reliable delivery
- Backup or redundant streams: Create additional output paths for failover scenarios
If you need more than 4 stereo audio outputs (8 channels total), or want to route audio from Audio Channel Strip components, use SRT Target 16 Channel Audio instead.
The standard SRT Target only uses the main audio bus (channels 1–8) and supports up to four stereo encoders.
Setup
- Add SRT Target to your project outputs
- Configure the SRT address: Enter the destination server address in SRT URL format
- Set SRT options: Configure packet size and latency parameters
- Configure video settings: Set bitrate and encoding options appropriate for your use case
- Configure audio channels: Enable and map audio channels as needed
- Start sending: Use the Start sending command to begin streaming
Parameters
Options marked (Advanced) are only visible in the UI when Show advanced options is set to Yes.
General Settings
| Parameter | Default | Description |
|---|---|---|
| Autostart when application starts | Enabled | When enabled, the SRT stream will automatically start when the Composer project is started |
| Show advanced options | No | Toggle to reveal advanced configuration options for video encoding, additional audio channels, and output buffer settings |
Addresses
| Parameter | Default | Description |
|---|---|---|
| SRT base server address (srt://ip:port) | srt://[ipadress]:[port] |
The destination SRT server address in URL format. Replace with your actual server IP and port |
| SRT options (packet size, latency etc) | pkt_size=1316&latency=200000 |
URL query parameters for SRT connection settings. Default configures 1316-byte packets and 200ms latency |
| SRT stream id (optional, may be required by CDN) | (empty) | Optional stream identifier required by some CDNs or SRT servers for stream routing |
| SRT connection string | (auto-generated) | Read-only field showing the complete connection string built from the above parameters |
Dynamic Placeholders
The SRT base server address field supports dynamic placeholders that are resolved at connection time:
| Placeholder | Description |
|---|---|
@@HostName() |
Replaced with the local machine's hostname |
SRT Connection Modes
You can add mode=caller or mode=listener to the SRT options field to control the connection behavior:
| Mode | Description |
|---|---|
mode=caller |
Composer connects to a remote SRT server (default behavior) |
mode=listener |
Composer listens for incoming SRT connections on the specified port |
If no mode is specified in the SRT options field, Composer automatically uses mode=caller.
Example: To set Composer as a listener with 200ms latency, enter: pkt_size=1316&latency=200000&mode=listener
Video Encoder Configuration
| Parameter | Default | Description |
|---|---|---|
| Video bitrate | 3.00 Mbit/s | Target video bitrate for the H.264/HEVC encoder |
| H264 B frame distance (Advanced) | Zero | Number of B-frames between reference frames. Zero disables B-frames for lowest latency |
| H264 Gop size (Advanced) | Same as double processing framerate | GOP (Group of Pictures) length. Controls keyframe interval |
| H264 profile (Advanced) | baseline | H.264 encoding profile. Baseline offers widest compatibility |
| Use CBR configuration (Advanced) | Enabled | Enable Constant Bit Rate mode for consistent bandwidth usage |
| Max video rate (multiplier) (Advanced) | 1.000 | Maximum bitrate multiplier relative to target bitrate |
| Min video rate (multiplier) (Advanced) | 1.000 | Minimum bitrate multiplier relative to target bitrate |
| Buffer size (multiplier) (Advanced) | 1.000 | Encoder buffer size multiplier relative to target bitrate |
| Quality/Speed control (Advanced) | veryfast | Encoder preset balancing quality vs. encoding speed |
| Tune (Advanced) | zerolatency | Encoder tuning optimized for specific content types or latency requirements |
| H264 threads (Advanced) | Two | Number of CPU threads for software encoding (only used when NVENC is disabled) |
| Use NVENC as video encoder | Enabled | Use NVIDIA hardware encoder for better performance |
| Use HEVC as video codec | Disabled | Use HEVC/H.265 codec instead of H.264 |
| Advanced options (json) (Advanced) | {"no-scenecut": "1","zerolatency": "1","strict_gop": "0","gpu": "any","preset": "p4","delay": "0","rc-lookahead": "0","surfaces": "1", "rc": "cbr"} |
JSON configuration for advanced NVENC encoder options |
Video Bitrate Options
Available bitrates range from 50 kbit/s to 30.00 Mbit/s.
H264 B Frame Distance Options (Advanced)
| Value | Description |
|---|---|
| Zero | No B-frames (lowest latency) |
| One | One B-frame between reference frames |
| Two | Two B-frames between reference frames |
H264 Gop Size Options (Advanced)
| Value |
|---|
| Same as processing framerate |
| Same as double processing framerate |
| 25 frames |
| 30 frames |
| 50 frames |
| 60 frames |
| 90 frames |
| 100 frames |
| 120 frames |
| 150 frames |
| 180 frames |
| 200 frames |
| 250 frames |
H264 Profile Options (Advanced)
| Value | Description |
|---|---|
| baseline | Basic profile with widest device compatibility |
| main | Main profile with improved compression |
| high | High profile for best quality at given bitrate |
| high10 | High 10-bit profile |
| high422 | High 4:2:2 chroma profile |
| high444 | High 4:4:4 chroma profile |
Quality/Speed Control Options (Advanced)
| Value | Description |
|---|---|
| ultrafast | Fastest encoding, lowest quality |
| superfast | Very fast encoding |
| veryfast | Fast encoding with reasonable quality |
| faster | Faster than default |
| fast | Fast encoding |
| medium | Balanced speed and quality |
| slow | Slower encoding, better quality |
| slower | Much slower encoding |
| veryslow | Very slow encoding, high quality |
| placebo | Maximum quality, extremely slow |
Tune Options (Advanced)
| Value | Description |
|---|---|
| film | Optimized for film content |
| animation | Optimized for animated content |
| grain | Preserves film grain |
| stillimage | Optimized for static images |
| psnr | Optimized for PSNR metric |
| ssim | Optimized for SSIM metric |
| fastdecode | Optimized for fast decoding |
| zerolatency | Optimized for lowest latency streaming |
H264 Threads Options (Advanced)
Available values range from Zero to Eight.
Audio Encoders Configuration
The SRT Target supports up to four independent audio channels, each with its own encoder, channel mapping, bitrate, and language metadata.
Audio channels must be enabled sequentially without gaps. For example, to use Audio Channel Three, you must also enable Audio Channel One and Two. You cannot skip channels.
Audio Channel One
| Parameter | Default | Description |
|---|---|---|
| Audio channel one encoder | AAC Stereo (L/R) | Audio encoder for channel one |
| Audio channel one mapping | Channels 1 & 2 | Source audio channels to encode |
| Audio channel one bitrate | 384 kbit/s | Audio bitrate for channel one |
| Audio channel one language description | (empty) | Language metadata tag for channel one |
Audio Channel Two (Advanced)
| Parameter | Default | Description |
|---|---|---|
| Audio channel two encoder | None (disabled) | Audio encoder for channel two |
| Audio channel two mapping | Channels 3 & 4 | Source audio channels to encode |
| Audio channel two bitrate | 384 kbit/s | Audio bitrate for channel two |
| Audio channel two language description | (empty) | Language metadata tag for channel two |
Audio Channel Three (Advanced)
| Parameter | Default | Description |
|---|---|---|
| Audio channel three encoder | None (disabled) | Audio encoder for channel three |
| Audio channel three mapping | Channels 5 & 6 | Source audio channels to encode |
| Audio channel three bitrate | 128 kbit/s | Audio bitrate for channel three |
| Audio channel three language description | (empty) | Language metadata tag for channel three |
Audio Channel Four (Advanced)
| Parameter | Default | Description |
|---|---|---|
| Audio channel four encoder | None (disabled) | Audio encoder for channel four |
| Audio channel four mapping | Channels 7 & 8 | Source audio channels to encode |
| Audio channel four bitrate | 96 kbit/s | Audio bitrate for channel four |
| Audio channel four language description | (empty) | Language metadata tag for channel four |
Audio Encoder Options
These options apply to all audio channels.
| Value | Description |
|---|---|
| None (disabled) | Audio channel is disabled |
| AAC Stereo (L/R) | Stereo AAC encoding with left/right channels |
| AAC Mono downmix (L+R) | Mono AAC encoding with left+right downmix |
Audio Channel Mapping Options
| Value | Description |
|---|---|
| Channels 1 & 2 | Maps input audio channels 1 and 2 |
| Channels 3 & 4 | Maps input audio channels 3 and 4 |
| Channels 5 & 6 | Maps input audio channels 5 and 6 |
| Channels 7 & 8 | Maps input audio channels 7 and 8 |
Audio Bitrate Options
Available bitrates range from 8 kbit/s to 384 kbit/s.
Options (Advanced)
| Parameter | Default | Description |
|---|---|---|
| Warn on low average bitrate (<10%) | Enabled | Display warnings when average bitrate falls below 10% of target |
| Reconnect Interval | 5 seconds | Time to wait before attempting automatic reconnection after disconnection |
| Max output queue size (packets) | 150 | Maximum number of packets to buffer before dropping. Range: 150-500 |
| Allow frame skipping | Disabled | When enabled, allows dropping frames if output queue warnings occur |
Reconnect Interval Options (Advanced)
| Value | Description |
|---|---|
| (Never) | Automatic reconnection disabled |
| Instant | Reconnect immediately |
| 1 second | Wait 1 second before reconnecting |
| 3 seconds | Wait 3 seconds before reconnecting |
| 5 seconds | Wait 5 seconds before reconnecting |
| 10 seconds | Wait 10 seconds before reconnecting |
| 15 seconds | Wait 15 seconds before reconnecting |
| 20 seconds | Wait 20 seconds before reconnecting |
| 30 seconds | Wait 30 seconds before reconnecting |
Commands
| Command | Description |
|---|---|
| Start sending | Starts the video encoder and begins streaming to the SRT destination |
| Stop sending | Stops the encoder and disconnects from the SRT destination |
| Reconnect | Forces a reconnection to the SRT destination |
Status Information
These fields display real-time information about the SRT stream and are read-only.
| Field | Description |
|---|---|
| Log | Recent log messages from the SRT Target |
| Uptime since start | Duration since the stream was started |
| Outgoing bitrate | Current measured outgoing bitrate |
| SRT message time | Timestamp of the last SRT status message |
| Number of audio channels used | Count and description of active audio channels |
Performance and Properties
These fields display connection statistics and diagnostics and are read-only.
| Field | Description |
|---|---|
| Number of reconnects | Count of automatic reconnections performed |
| Current SRT endpoint | The currently connected SRT server address |
| Error counter | Total number of errors encountered |
| Last Error DateTime | Timestamp of the most recent error |
| Warning counter | Total number of warnings encountered |
| Last Warning DateTime | Timestamp of the most recent warning |
Workflow Tips
Basic Setup
- Enter your SRT server address in the format
srt://ip:port - Configure SRT options for your network conditions (default 200ms latency works for most cases)
- Set video bitrate appropriate for your content and available bandwidth
- Enable at least one audio channel with appropriate mapping
- Click Start sending to begin streaming
Low Latency Streaming
- Keep Use NVENC as video encoder enabled for hardware acceleration
- Use zerolatency tune setting (default)
- Set H264 B frame distance to Zero (default)
- Keep H264 Gop size at a low value
- Reduce SRT latency in options if network conditions allow
High Quality Streaming
- Increase Video bitrate to match your bandwidth capacity
- Consider using high profile for better compression efficiency
- Use slower Quality/Speed control presets if CPU/GPU resources allow
Multi-Language Audio
- Enable multiple audio channels (up to 4)
- Map each channel to different input channel pairs
- Set appropriate language description for each channel (e.g., "eng", "spa", "fra")
- Adjust bitrates based on content requirements
Network Reliability
- Set Reconnect Interval to automatically recover from disconnections
- Increase SRT latency in options for unreliable networks
- Enable Warn on low average bitrate to monitor stream health
- Monitor Error counter and Warning counter for issues
Troubleshooting
Stream fails to connect:
- Verify the SRT server address is correct and reachable
- Check that the destination port is open and the SRT server is running
- Ensure SRT stream id is provided if required by the destination
High latency or buffering:
- Reduce video bitrate if bandwidth is limited
- Increase SRT latency parameter for unstable networks
- Check Outgoing bitrate matches your target bitrate
Video quality issues:
- Increase video bitrate
- Try a slower Quality/Speed control preset
Audio not working:
- Verify at least one audio encoder is not set to "None (disabled)"
- Check audio channel mapping matches your input audio configuration
- Ensure source audio is present in Composer
Frequent disconnections:
- Increase SRT latency for network instability
- Check network connectivity and bandwidth
- Set appropriate Reconnect Interval for automatic recovery
- Review error and warning counters for patterns
NVENC encoder not available:
- Verify NVIDIA GPU is installed and drivers are up to date
- Disable Use NVENC as video encoder to fall back to software encoding
- Check GPU is not overloaded by other applications
Output queue warnings:
- Reduce video bitrate
- Enable Allow frame skipping as temporary measure
- Check CPU/GPU load and network bandwidth
- Increase Max output queue size if needed