MediaBridge::Options

Options

#include <icy/webrtc/mediabridge.h>

Configuration options for the WebRTC media bridge.

Return	Name	Description
`av::VideoCodec`	`videoCodec`	Video codec for the send track. Leave both name and encoder empty to skip creating a video track.
`av::AudioCodec`	`audioCodec`	Audio codec for the send track. Leave both name and encoder empty to skip creating an audio track.
`uint32_t`	`videoSsrc`	0 = auto-generate
`uint32_t`	`audioSsrc`	0 = auto-generate
`int`	`videoPayloadType`	Reuse negotiated offer payload type when answering, -1 = default.
`int`	`audioPayloadType`	Reuse negotiated offer payload type when answering, -1 = default.
`std::string`	`cname`	CNAME for RTCP (auto if empty)
`std::string`	`videoMid`	Explicit MID for the negotiated video m-line when answering an offer.
`std::string`	`audioMid`	Explicit MID for the negotiated audio m-line when answering an offer.
`rtc::Description::Direction`	`videoDirection`
`rtc::Description::Direction`	`audioDirection`
`unsigned`	`nackBufferSize`	Max RTP packets retained for video NACK retransmission.
`JitterBufferConfig`	`videoJitterBuffer`	Receive-side playout buffering for depacketized video frames.
`JitterBufferConfig`	`audioJitterBuffer`	Receive-side playout buffering for depacketized audio frames.

av::VideoCodec videoCodec

Video codec for the send track. Leave both name and encoder empty to skip creating a video track.

av::AudioCodec audioCodec

Audio codec for the send track. Leave both name and encoder empty to skip creating an audio track.

uint32_t videoSsrc = 0

0 = auto-generate

uint32_t audioSsrc = 0

0 = auto-generate

int videoPayloadType = -1

Reuse negotiated offer payload type when answering, -1 = default.

int audioPayloadType = -1

Reuse negotiated offer payload type when answering, -1 = default.

std::string cname

CNAME for RTCP (auto if empty)

std::string videoMid

Explicit MID for the negotiated video m-line when answering an offer.

std::string audioMid

Explicit MID for the negotiated audio m-line when answering an offer.

rtc::Description::Direction videoDirection = rtc::Description::Direction::SendRecv

rtc::Description::Direction audioDirection = rtc::Description::Direction::SendRecv

unsigned nackBufferSize = 512

Max RTP packets retained for video NACK retransmission.

JitterBufferConfig videoJitterBuffer

Receive-side playout buffering for depacketized video frames.

JitterBufferConfig audioJitterBuffer

Receive-side playout buffering for depacketized audio frames.