WebRTC module

MediaBridge::Options

[Configuration]({#ref classicy_1_1Configuration #}) options for the WebRTC media bridge.

Options

#include <icy/webrtc/mediabridge.h>

Configuration options for the WebRTC media bridge.

Public Attributes

ReturnNameDescription
av::VideoCodecvideoCodecVideo codec for the send track. Leave both name and encoder empty to skip creating a video track.
av::AudioCodecaudioCodecAudio codec for the send track. Leave both name and encoder empty to skip creating an audio track.
uint32_tvideoSsrc0 = auto-generate
uint32_taudioSsrc0 = auto-generate
intvideoPayloadTypeReuse negotiated offer payload type when answering, -1 = default.
intaudioPayloadTypeReuse negotiated offer payload type when answering, -1 = default.
std::stringcnameCNAME for RTCP (auto if empty)
std::stringvideoMidExplicit MID for the negotiated video m-line when answering an offer.
std::stringaudioMidExplicit MID for the negotiated audio m-line when answering an offer.
rtc::Description::DirectionvideoDirection
rtc::Description::DirectionaudioDirection
unsignednackBufferSizeMax RTP packets retained for video NACK retransmission.

videoCodec

av::VideoCodec videoCodec

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


audioCodec

av::AudioCodec audioCodec

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


videoSsrc

uint32_t videoSsrc = 0

0 = auto-generate


audioSsrc

uint32_t audioSsrc = 0

0 = auto-generate


videoPayloadType

int videoPayloadType = -1

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


audioPayloadType

int audioPayloadType = -1

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


cname

std::string cname

CNAME for RTCP (auto if empty)


videoMid

std::string videoMid

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


audioMid

std::string audioMid

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


videoDirection

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

audioDirection

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

nackBufferSize

unsigned nackBufferSize = 512

Max RTP packets retained for video NACK retransmission.