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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package core
import (
"encoding/json"
"testing"
"time"
"github.com/aler9/gortsplib/v2"
"github.com/aler9/gortsplib/v2/pkg/format"
"github.com/aler9/gortsplib/v2/pkg/media"
"github.com/gorilla/websocket"
"github.com/pion/rtp"
"github.com/pion/webrtc/v3"
"github.com/stretchr/testify/require"
)
func TestWebRTCServer(t *testing.T) {
p, ok := newInstance("paths:\n" +
" all:\n")
require.Equal(t, true, ok)
defer p.Close()
medi := &media.Media{
Type: media.TypeVideo,
Formats: []format.Format{&format.H264{
PayloadTyp: 96,
PacketizationMode: 1,
}},
}
v := gortsplib.TransportTCP
source := gortsplib.Client{
Transport: &v,
}
err := source.StartRecording("rtsp://localhost:8554/stream", media.Medias{medi})
require.NoError(t, err)
defer source.Close()
c, _, err := websocket.DefaultDialer.Dial("ws://localhost:8889/stream/ws", nil) //nolint:bodyclose
require.NoError(t, err)
defer c.Close()
_, msg, err := c.ReadMessage()
require.NoError(t, err)
var iceServers []webrtc.ICEServer
err = json.Unmarshal(msg, &iceServers)
require.NoError(t, err)
pc, err := newPeerConnection(webrtc.Configuration{
ICEServers: iceServers,
})
require.NoError(t, err)
defer pc.Close()
pc.OnICECandidate(func(i *webrtc.ICECandidate) {
if i != nil {
enc, _ := json.Marshal(i.ToJSON())
c.WriteMessage(websocket.TextMessage, enc)
}
})
connected := make(chan struct{})
pc.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
if state == webrtc.PeerConnectionStateConnected {
close(connected)
}
})
track := make(chan *webrtc.TrackRemote)
pc.OnTrack(func(trak *webrtc.TrackRemote, recv *webrtc.RTPReceiver) {
track <- trak
})
_, err = pc.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo)
require.NoError(t, err)
localOffer, err := pc.CreateOffer(nil)
require.NoError(t, err)
enc, err := json.Marshal(localOffer)
require.NoError(t, err)
err = c.WriteMessage(websocket.TextMessage, enc)
require.NoError(t, err)
err = pc.SetLocalDescription(localOffer)
require.NoError(t, err)
_, msg, err = c.ReadMessage()
require.NoError(t, err)
var remoteOffer webrtc.SessionDescription
err = json.Unmarshal(msg, &remoteOffer)
require.NoError(t, err)
err = pc.SetRemoteDescription(remoteOffer)
require.NoError(t, err)
go func() {
for {
_, msg, err := c.ReadMessage()
if err != nil {
return
}
var candidate webrtc.ICECandidateInit
err = json.Unmarshal(msg, &candidate)
if err != nil {
return
}
pc.AddICECandidate(candidate)
}
}()
<-connected
time.Sleep(500 * time.Millisecond)
source.WritePacketRTP(medi, &rtp.Packet{
Header: rtp.Header{
Version: 2,
Marker: true,
PayloadType: 96,
SequenceNumber: 123,
Timestamp: 45343,
SSRC: 563423,
},
Payload: []byte{0x01, 0x02, 0x03, 0x04},
})
trak := <-track
pkt, _, err := trak.ReadRTP()
require.NoError(t, err)
require.Equal(t, &rtp.Packet{
Header: rtp.Header{
Version: 2,
Marker: true,
PayloadType: 102,
SequenceNumber: pkt.SequenceNumber,
Timestamp: pkt.Timestamp,
SSRC: pkt.SSRC,
CSRC: []uint32{},
},
Payload: []byte{0x01, 0x02, 0x03, 0x04},
}, pkt)
}