Ready-to-use SRT / WebRTC / RTSP / RTMP / LL-HLS media server and media proxy that allows to read, publish, proxy, record and playback video and audio streams.
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3.8 KiB

package webrtc
import (
"github.com/pion/ice/v2"
"github.com/pion/interceptor"
"github.com/pion/webrtc/v3"
)
func stringInSlice(a string, list []string) bool {
for _, b := range list {
if b == a {
return true
}
}
return false
}
var videoCodecs = []webrtc.RTPCodecParameters{
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeAV1,
ClockRate: 90000,
},
PayloadType: 96,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeVP9,
ClockRate: 90000,
SDPFmtpLine: "profile-id=0",
},
PayloadType: 97,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeVP9,
ClockRate: 90000,
SDPFmtpLine: "profile-id=1",
},
PayloadType: 98,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeVP8,
ClockRate: 90000,
},
PayloadType: 99,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeH264,
ClockRate: 90000,
SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
},
PayloadType: 100,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeH264,
ClockRate: 90000,
SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f",
},
PayloadType: 101,
},
}
var audioCodecs = []webrtc.RTPCodecParameters{
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeOpus,
ClockRate: 48000,
Channels: 2,
SDPFmtpLine: "minptime=10;useinbandfec=1;stereo=1;sprop-stereo=1",
},
PayloadType: 111,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeG722,
ClockRate: 8000,
},
PayloadType: 9,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypePCMU,
ClockRate: 8000,
},
PayloadType: 0,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypePCMA,
ClockRate: 8000,
},
PayloadType: 8,
},
}
// APIConf is the configuration passed to NewAPI().
type APIConf struct {
ICEUDPMux ice.UDPMux
ICETCPMux ice.TCPMux
LocalRandomUDP bool
IPsFromInterfaces bool
IPsFromInterfacesList []string
AdditionalHosts []string
}
// NewAPI allocates a webrtc API.
func NewAPI(cnf APIConf) (*webrtc.API, error) {
settingsEngine := webrtc.SettingEngine{}
settingsEngine.SetInterfaceFilter(func(iface string) bool {
return cnf.IPsFromInterfaces && (len(cnf.IPsFromInterfacesList) == 0 ||
stringInSlice(iface, cnf.IPsFromInterfacesList))
})
settingsEngine.SetAdditionalHosts(cnf.AdditionalHosts)
var networkTypes []webrtc.NetworkType
// always enable UDP in order to support STUN/TURN
networkTypes = append(networkTypes, webrtc.NetworkTypeUDP4)
if cnf.ICEUDPMux != nil {
settingsEngine.SetICEUDPMux(cnf.ICEUDPMux)
}
if cnf.ICETCPMux != nil {
settingsEngine.SetICETCPMux(cnf.ICETCPMux)
networkTypes = append(networkTypes, webrtc.NetworkTypeTCP4)
}
if cnf.LocalRandomUDP {
settingsEngine.SetICEUDPRandom(true)
}
settingsEngine.SetNetworkTypes(networkTypes)
mediaEngine := &webrtc.MediaEngine{}
for _, codec := range videoCodecs {
err := mediaEngine.RegisterCodec(codec, webrtc.RTPCodecTypeVideo)
if err != nil {
return nil, err
}
}
for _, codec := range audioCodecs {
err := mediaEngine.RegisterCodec(codec, webrtc.RTPCodecTypeAudio)
if err != nil {
return nil, err
}
}
interceptorRegistry := &interceptor.Registry{}
err := webrtc.RegisterDefaultInterceptors(mediaEngine, interceptorRegistry)
if err != nil {
return nil, err
}
return webrtc.NewAPI(
webrtc.WithSettingEngine(settingsEngine),
webrtc.WithMediaEngine(mediaEngine),
webrtc.WithInterceptorRegistry(interceptorRegistry)), nil
}