Application Settings

Application settings control the behavior of individual applications running in Ant Media Server. These settings are stored in <AMS_DIR>/webapps/<AppName>/WEB-INF/red5-web.properties and can be configured through the web panel or by editing the properties file directly.

Streaming Settings

Recording Formats

boolean

default:"false"

Enable/disable MP4 recording. If enabled, an MP4 file is created in the <APP_DIR>/streams directory.

boolean

default:"false"

Enable/disable WebM recording.

boolean

default:"true"

Enable/disable HLS recording. If enabled, HLS files (m3u8 and ts) are created in the <APP_DIR>/streams directory and HLS playing is enabled.

boolean

default:"false"

Enable/disable DASH recording.

File Naming

boolean

default:"false"

Add date and time to MP4 file names. If enabled, timestamps are added to recorded MP4 files.

boolean

default:"false"

Add date and time to HLS file names (.m3u8 and .ts files).

string

default:"%r%b"

Format string for output file names. Supports the following placeholders:

%r: Resolution (e.g., “720p”)
%b: Bitrate (e.g., “1500kbps”)
{customText}: Custom text in curly braces

Examples:

"%r%b" → stream1_720p1500kbps
"{HD}%r%b" → stream1_HD720p1500kbps
"%r{custom}%b" → stream1_720pcustom1500kbps

File Cleanup

boolean

default:"true"

Delete HLS files (m3u8 and ts) after the broadcast has finished.

boolean

default:"true"

Delete DASH files (mpd and m4s) after the broadcast has finished.

HLS Configuration

string

default:"2"

Duration of segments in seconds. Segments will be cut on the next key frame after this time has passed.

string

default:"15"

Maximum number of playlist entries in m3u8 files. If 0, the list file will contain all segments.

string

default:""

HLS playlist type. Can be “event” or “vod”. Check HLS documentation for EXT-X-PLAYLIST-TYPE.

string

default:"mpegts"

HLS segment file type. Can be “mpegts” or “fmp4”. fmp4 is compatible for playing HEVC HLS streams.

string

default:"%09d"

Suffix format for HLS segment files.

Default: %09d (9 digit incremental)
With date: %Y%m%d-%s
Combined: %s-%%09d (requires +second_level_segment_index in hlsflags)

string

default:"delete_segments+program_date_time"

HLS flags for FFmpeg HLS Muxer. Separate multiple options with + or -.Example:

hlsflags=+program_date_time+round_durations+append_list

See FFmpeg HLS documentation for details.

string

default:""

HTTP endpoint to push the HLS stream.

string

default:""

Full path to HLS encryption key info file. Format:

key URI
key file path
IV (optional)

Example:

http://server/file.key
/path/to/file.key
0123456789ABCDEF0123456789ABCDEF

DASH Configuration

string

default:"6"

Duration of DASH segments in seconds.

string

default:"0.5"

Duration of DASH fragments. Fragments are a property of fragmented MP4 files (moof + mdat).

string

default:"3.5"

Target latency for DASH streaming in seconds.

string

default:"5"

Number of files in the DASH manifest.

string

default:"5"

Number of segments kept outside the manifest before removing from disk.

boolean

default:"true"

Enable low latency DASH. Effective if DASH is enabled.

boolean

default:"false"

Enable low latency HLS via DASH muxer. Effective if DASH is enabled.

boolean

default:"false"

Enable HLS through DASH muxer. Effective if DASH is enabled.

boolean

default:"false"

Use timeline in DASH muxing.

boolean

default:"true"

Use HTTP streaming in Low Latency DASH. If true, sends files through HTTP. If false, writes files to disk directly.

string

default:""

HTTP endpoint for DASH/CMAF streams.

WebRTC Settings

boolean

default:"true"

Enable/disable WebRTC playing.

int

default:"30"

Frame rate for video publishing to WebRTC players.

int

default:"50000"

Minimum port number for WebRTC connections. Must be less than webRTCPortRangeMax.

int

default:"60000"

Maximum port number for WebRTC connections. Must be greater than webRTCPortRangeMin.

string

default:"stun:stun1.l.google.com:19302"

STUN or TURN server URI for WebRTC ICE candidates. Can be a STUN or TURN URL.

string

default:""

TURN server username for WebRTC ICE candidates. Requires stunServerURI and turnServerCredential to be effective.

string

default:""

TURN server credential for WebRTC ICE candidates. Requires stunServerURI and turnServerUsername to be effective.

boolean

default:"false"

Enable TCP candidates for WebRTC connections. If false, only UDP will be used.

string

default:"unifiedPlan"

WebRTC SDP semantics. Can be “planB” or “unifiedPlan”.

int

default:"0"

Port allocator flags for WebRTC:

PORTALLOCATOR_DISABLE_UDP = 0x01
PORTALLOCATOR_DISABLE_STUN = 0x02
PORTALLOCATOR_DISABLE_RELAY = 0x04

boolean

default:"true"

Enable/disable IPv6 candidates for WebRTC.

int

default:"-1"

Application-level WebRTC viewer limit. -1 means unlimited.

int

default:"2000"

Interval in milliseconds for requesting key frames in SFU mode.

int

default:"10000"

Timeout in milliseconds for WebRTC client to start. Also used as reconnection timeout for publishers in fluctuating networks.

boolean

default:"false"

Use the original WebRTC stream in adaptive bitrate scenarios. If true and adaptive bitrate is enabled, both original and transcoded streams are available.

boolean

default:"false"

Replace ICE candidate address with server address. Requires serverName to be set in conf/red5.properties.

long

default:"2000"

Timeout in milliseconds for ICE gathering process, especially for WHIP ingestion.

boolean

default:"false"

Drop WebRTC ingest if no audio or video packets are received within webRTCClientStartTimeoutMs.

Adaptive Bitrate (ABR)

string

default:""

Encoder settings in JSON array format for adaptive bitrate streaming. If empty, SFU mode will be active.Example:

[
  {"height":480, "videoBitrate":500000, "audioBitrate":128000, "forceEncode":true},
  {"height":360, "videoBitrate":300000, "audioBitrate":96000, "forceEncode":true}
]

boolean

default:"true"

Add the original stream to the HLS playlist if adaptive bitrate is enabled.

boolean

default:"false"

Force stream decoding even if there is no adaptive bitrate setting.

Stats-Based ABR

boolean

default:"true"

Enable statistics-based adaptive bitrate switching algorithm.

float

default:"1"

Packet loss percentage threshold to switch to lower resolution.

float

default:"0.1"

Packet loss percentage threshold to switch to higher resolution.

int

default:"150"

Round trip time threshold in milliseconds to switch to higher resolution.

int

default:"30"

Jitter threshold in milliseconds to switch to higher resolution.

Excessive Bandwidth Algorithm

boolean

default:"false"

Enable the excessive bandwidth algorithm that attempts to switch to higher bitrate even when bandwidth seems insufficient.

int

default:"300000"

Bandwidth threshold in bps for the excessive bandwidth algorithm.

int

default:"3"

Number of consecutive calls needed to trigger bitrate switch.

int

default:"4"

Number of attempts before switching back to lower quality.

int

default:"10"

Packet loss difference threshold to immediately switch back to lower quality.

int

default:"20"

RTT measurement difference threshold to immediately switch back to lower quality.

Preview/Thumbnail Settings

int

default:"5000"

Period in milliseconds for creating preview images. Files are created in <APP_DIR>/preview directory.

boolean

default:"false"

If true, new stream previews with the same ID overwrite existing files. If false, previous files are saved with a suffix.

int

default:"480"

Height in pixels of the preview image.

boolean

default:"false"

Generate preview images if adaptive bitrate settings exist.

string

default:"png"

Preview image format. Can be “png”, “jpg”, or “webp”.

int

default:"75"

Preview quality for JPG and WEBP formats.

JPG: 2-31 (2 = best quality, 31 = worst quality). Recommended: 5
WEBP: 0-100 (0 = worst quality, 100 = best quality). Recommended: 75

Stream Sources

int

default:"0"

Period in seconds for automatically restarting stream fetchers. 0 means disabled.

boolean

default:"false"

Automatically start stream sources when server starts.

int

default:"0"

Buffer time in milliseconds for stream fetchers. 0 means no buffer.

string

default:"3"

RTSP transport type for pulling streams. Can be string or integer (OR combination):

“udp” or 1
“tcp” or 2
“udp_multicast” or 4
“http” or 256
“https” or 512

Default is 3 (udp OR tcp).

int

default:"5000"

Timeout in milliseconds for RTSP connections.

int

default:"1500"

Maximum analyze duration in milliseconds for determining video and audio existence in RTMP, SRT, and stream sources.

Stream Limits

int

default:"0"

Maximum acceptable FPS for incoming streams. 0 means no limit.

int

default:"0"

Maximum acceptable resolution for incoming streams. 0 means no limit.

int

default:"0"

Maximum acceptable bitrate for incoming streams. 0 means no limit.

int

default:"-1"

Application-level total incoming stream limit. -1 means unlimited.

int

default:"-1"

Maximum audio tracks in a multitrack playing connection. -1 means unlimited.

int

default:"-1"

Maximum video tracks in a multitrack playing connection. -1 means unlimited.

Data Channel

boolean

default:"true"

Enable/disable WebRTC data channel. When enabled, publishers can send messages to players.

string

default:"all"

Distribution mode for player messages:

"none": Player messages are delivered to nobody
"publisher": Player messages are delivered only to publisher
"all": Player messages are delivered to everyone (publisher and all players)

string

default:""

Webhook URL for data channel messages. All data channel messages are also sent to this endpoint.

boolean

default:"false"

Send audio level information from RTP headers to viewers via data channel. Useful for video conferencing to detect active speakers.

int

default:"120"

Audio level threshold for assigning audio tracks in limited track conferences. Range: 0-127 (0 = max level, 127 = silent).

Webhooks

string

default:""

Webhook URL for event notifications. See Webhook Integration for details.

string

default:""

Webhook URL for authentication when publishing streams.

string

default:""

Webhook URL for WebRTC play authentication.

int

default:"0"

Number of retry attempts on webhook POST failure.

long

default:"1000"

Delay in milliseconds between webhook retry attempts.

long

default:"-1"

Period in milliseconds for sending stream status updates to webhook. -1 means disabled. Minimum recommended: 5000ms.

string

default:"application/json"

Content type for webhook POST requests. For backward compatibility, can be set to “application/x-www-form-urlencoded”.

Scripts

string

default:""

Path to bash script called when muxing finishes.

string

default:""

Path to bash script called when streaming starts.

string

default:""

Path to bash script called when stream ends.

string

default:""

Path to bash script called when stream idle timeout occurs.

string

default:""

Path to bash script called when VoD upload finishes.

Storage

string

default:""

Subfolder path for storing media files. Supports placeholders:

%m: Main track ID
%s: Stream ID

Examples:

"%m" → streams/mainTrackId/0001.ts
"myStreams/%m/%s" → streams/myStreams/mainTrackId/streamId/0001.ts

string

default:"null"

Subfolder for recording files (MP4 and WebM).

VoD Settings

string

default:""

Directory for VoD files. Deprecated - use VoD import REST API instead.

Access Control

boolean

default:"false"

Only accept streams that are pre-registered in the database. Streams must be added via REST API before publishing.

boolean

default:"false"

Only accept rooms that are pre-registered in the database.

boolean

default:"true"

Prevent playing the same stream ID more than once in the same WebRTC session.

Multitrack & Conference

object

default:"{...}"

Visibility matrix defining which participant roles can see each other in conferences.Default configuration:

{
  "default": ["default"],
  "speaker": ["speaker", "active_attendee"],
  "attendee": ["speaker", "active_attendee"],
  "active_attendee": ["active_attendee", "speaker"]
}

RTMP Settings

long

default:"0"

RTMP ingesting buffer time in milliseconds. Server buffers this amount of video to compensate for stream interruptions.

boolean

default:"false"

Enable RTMP playback. Note: RTMP playback is deprecated and will be removed. Use CMAF/DASH instead.

int

default:"360"

Height of the stream transcoded from WebRTC to RTMP forwarding.

boolean

default:"true"

Relay RTMP metadata to muxers for playback synchronization.

SRT Settings

int

default:"150"

Time in milliseconds to wait for SRT packets to be received.

Misc Settings

boolean

default:"true"

Write viewer statistics (HLS, WebRTC) to the datastore.

boolean

default:"false"

Write subscriber connect/disconnect events to datastore. Alternatively available in analytics logs.

string

default:"127.0.0.1"

Comma-separated CIDR addresses allowed to access REST API. Must be in CIDR format (a.b.c.d/x).

string

default:""

Comma-separated CIDR addresses allowed to publish streams. Empty means accept from everywhere.

string

default:""

Forward HTTP requests with these extensions to httpForwardingBaseURL. Comma-separated (e.g., “mp4,m3u8”).

string

default:""

Base URL for forwarding HTTP requests.

string

default:""

Value for Content-Security-Policy HTTP response header to reduce XSS risks.

int

default:"2"

Maximum idle time in seconds between origin and edge connection before re-establishing.

boolean

default:"false"

Enable ID3 tag support for HLS.

int

default:"150"

Size of encoding queue for frames waiting to be encoded (default: 5 seconds of 30fps stream).

object

default:"{}"

Custom key-value settings for application-specific configuration.

Get Started

Core Concepts

Deployment

Configuration

Streaming

Features

Security

Storage

SDKs

Operations

Streaming Settings

Recording Formats

File Naming

File Cleanup

HLS Configuration

DASH Configuration

WebRTC Settings

Adaptive Bitrate (ABR)

Stats-Based ABR

Excessive Bandwidth Algorithm

Preview/Thumbnail Settings

Stream Sources

Stream Limits

Data Channel

Webhooks

Scripts

Storage

VoD Settings

Access Control

Multitrack & Conference

RTMP Settings

SRT Settings

Misc Settings

​Streaming Settings

​Recording Formats

​File Naming

​File Cleanup

​HLS Configuration

​DASH Configuration

​WebRTC Settings

​Adaptive Bitrate (ABR)

​Stats-Based ABR

​Excessive Bandwidth Algorithm

​Preview/Thumbnail Settings

​Stream Sources

​Stream Limits

​Data Channel

​Webhooks

​Scripts

​Storage

​VoD Settings

​Access Control

​Multitrack & Conference

​RTMP Settings

​SRT Settings

​Misc Settings

Streaming Settings

Recording Formats

File Naming

File Cleanup

HLS Configuration

DASH Configuration

WebRTC Settings

Adaptive Bitrate (ABR)

Stats-Based ABR

Excessive Bandwidth Algorithm

Preview/Thumbnail Settings

Stream Sources

Stream Limits

Data Channel

Webhooks

Scripts

Storage

VoD Settings

Access Control

Multitrack & Conference

RTMP Settings

SRT Settings

Misc Settings