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@seydx/rtsp / BackchannelTarget

Interface: BackchannelTarget ​

Defined in: sinks/rtsp-server/backchannel-transcoder.ts:62

Target audio format the talkback stream is transcoded into.

Describes the encoder and output container the inbound viewer audio is converted to before being handed to the upstream camera. Choosing rtp for the format yields ready-to-send RTP packets, while a container format yields raw bytes suitable for other transports.

Properties ​

bitRate? ​

optional bitRate?: number

Defined in: sinks/rtsp-server/backchannel-transcoder.ts:104

Target encoder bitrate in bits per second.

When omitted the encoder picks its own default, which for narrowband mono AAC can exceed the per-frame ceiling and be clamped with a warning. Set an explicit value (for example 32000) to keep talkback frames small — important when the upstream multiplexes talkback onto a shared, rate-sensitive channel.

channels ​

channels: number

Defined in: sinks/rtsp-server/backchannel-transcoder.ts:94

Number of output audio channels (1 for mono, 2 for stereo).

Drives both the resample channel layout and the encoder configuration.

codec? ​

optional codec?: string

Defined in: sinks/rtsp-server/backchannel-transcoder.ts:79

Encoder codec name as understood by FFmpeg (for example aac, pcm_alaw).

A name-based alternative to codecId, letting callers select the target encoder without importing node-av's numeric AVCodecID constants. Resolved to a concrete encoder at startup, with an unknown name aborting startup.

codecId? ​

optional codecId?: AVCodecID

Defined in: sinks/rtsp-server/backchannel-transcoder.ts:70

node-av AVCodecID of the encoder to produce.

Identifies the codec the upstream camera expects on its backchannel; resolved to a concrete encoder at startup, with an unknown id aborting startup. Provide this or codec; when both are set, codec takes precedence.

format? ​

optional format?: string

Defined in: sinks/rtsp-server/backchannel-transcoder.ts:113

Output muxer format name.

Defaults to rtp, which emits RTP packets ready to forward to the upstream camera's sendRTSPPacket. A container format such as adts instead emits raw bytes (for example for a Node writable or proprietary talkback stream).

maxPacketSize? ​

optional maxPacketSize?: number

Defined in: sinks/rtsp-server/backchannel-transcoder.ts:121

RTP maximum packet size (MTU) in bytes.

Only consulted when format is rtp; bounds the size of emitted RTP packets. Defaults to 1200 bytes when omitted.

sampleRate ​

sampleRate: number

Defined in: sinks/rtsp-server/backchannel-transcoder.ts:87

Output sample rate in hertz the audio is resampled and encoded at.

Should match what the target codec and upstream camera expect (for example 8000 for G.711).