mirror of
https://github.com/aler9/rtsp-simple-server
synced 2025-10-05 15:46:58 +08:00
a1dc9f45f5
IMPORTANT NOTE: this doesn't allow to read H265 tracks with WebRTC, just to publish them. The inability to read H265 tracks with WebRTC is not in any way related to the server but depends on browsers and on the fact that they are not legally entitled to embed a H265 decoder inside them.
489 lines
10 KiB
Go
489 lines
10 KiB
Go
package webrtc
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import (
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"context"
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"testing"
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"time"
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"github.com/bluenviron/gortsplib/v4/pkg/format"
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"github.com/bluenviron/mediamtx/internal/conf"
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"github.com/bluenviron/mediamtx/internal/test"
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"github.com/pion/rtp"
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"github.com/pion/sdp/v3"
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"github.com/pion/webrtc/v3"
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"github.com/stretchr/testify/require"
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)
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func TestPeerConnectionCloseImmediately(t *testing.T) {
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pc := &PeerConnection{
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HandshakeTimeout: conf.StringDuration(10 * time.Second),
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TrackGatherTimeout: conf.StringDuration(2 * time.Second),
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LocalRandomUDP: true,
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IPsFromInterfaces: true,
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Publish: false,
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Log: test.NilLogger,
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}
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err := pc.Start()
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require.NoError(t, err)
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defer pc.Close()
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_, err = pc.CreatePartialOffer()
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require.NoError(t, err)
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// wait for ICE candidates to be generated
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time.Sleep(500 * time.Millisecond)
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pc.Close()
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}
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func TestPeerConnectionPublishRead(t *testing.T) {
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for _, ca := range []struct {
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name string
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in format.Format
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webrtcOut webrtc.RTPCodecCapability
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out format.Format
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}{
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{
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"av1",
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&format.AV1{
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PayloadTyp: 96,
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},
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webrtc.RTPCodecCapability{
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MimeType: "video/AV1",
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ClockRate: 90000,
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},
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&format.AV1{
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PayloadTyp: 96,
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},
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},
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{
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"vp9",
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&format.VP9{
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PayloadTyp: 96,
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},
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webrtc.RTPCodecCapability{
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MimeType: "video/VP9",
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ClockRate: 90000,
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SDPFmtpLine: "profile-id=0",
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},
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&format.VP9{
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PayloadTyp: 96,
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},
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},
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{
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"vp8",
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&format.VP8{
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PayloadTyp: 96,
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},
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webrtc.RTPCodecCapability{
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MimeType: "video/VP8",
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ClockRate: 90000,
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},
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&format.VP8{
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PayloadTyp: 96,
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},
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},
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{
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"h265",
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test.FormatH265,
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webrtc.RTPCodecCapability{
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MimeType: "video/H265",
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ClockRate: 90000,
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},
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&format.H265{
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PayloadTyp: 96,
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},
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},
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{
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"h264",
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test.FormatH264,
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webrtc.RTPCodecCapability{
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MimeType: "video/H264",
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ClockRate: 90000,
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SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f",
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},
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&format.H264{
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PayloadTyp: 96,
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PacketizationMode: 1,
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},
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},
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{
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"opus multichannel",
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&format.Opus{
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PayloadTyp: 112,
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ChannelCount: 6,
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},
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webrtc.RTPCodecCapability{
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MimeType: "audio/multiopus",
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ClockRate: 48000,
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Channels: 6,
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SDPFmtpLine: "channel_mapping=0,4,1,2,3,5;num_streams=4;coupled_streams=2",
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},
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&format.Opus{
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PayloadTyp: 96,
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ChannelCount: 6,
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},
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},
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{
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"opus stereo",
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&format.Opus{
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PayloadTyp: 111,
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ChannelCount: 2,
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},
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webrtc.RTPCodecCapability{
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MimeType: "audio/opus",
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ClockRate: 48000,
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Channels: 2,
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SDPFmtpLine: "minptime=10;useinbandfec=1;stereo=1;sprop-stereo=1",
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},
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&format.Opus{
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PayloadTyp: 96,
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ChannelCount: 2,
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},
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},
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{
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"opus mono",
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&format.Opus{
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PayloadTyp: 111,
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ChannelCount: 1,
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},
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webrtc.RTPCodecCapability{
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MimeType: "audio/opus",
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ClockRate: 48000,
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Channels: 2,
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SDPFmtpLine: "minptime=10;useinbandfec=1",
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},
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&format.Opus{
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PayloadTyp: 96,
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ChannelCount: 1,
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},
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},
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{
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"g722",
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&format.G722{},
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webrtc.RTPCodecCapability{
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MimeType: "audio/G722",
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ClockRate: 8000,
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},
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&format.G722{},
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},
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{
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"g711 pcma 8khz mono",
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&format.G711{
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PayloadTyp: 8,
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SampleRate: 8000,
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ChannelCount: 1,
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},
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webrtc.RTPCodecCapability{
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MimeType: "audio/PCMA",
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ClockRate: 8000,
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},
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&format.G711{
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PayloadTyp: 8,
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SampleRate: 8000,
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ChannelCount: 1,
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},
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},
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{
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"g711 pcmu 8khz mono",
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&format.G711{
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MULaw: true,
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PayloadTyp: 0,
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SampleRate: 8000,
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ChannelCount: 1,
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},
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webrtc.RTPCodecCapability{
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MimeType: "audio/PCMU",
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ClockRate: 8000,
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},
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&format.G711{
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MULaw: true,
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PayloadTyp: 0,
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SampleRate: 8000,
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ChannelCount: 1,
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},
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},
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{
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"g711 pcma 8khz stereo",
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&format.G711{
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PayloadTyp: 96,
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SampleRate: 8000,
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ChannelCount: 2,
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},
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webrtc.RTPCodecCapability{
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MimeType: "audio/PCMA",
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ClockRate: 8000,
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Channels: 2,
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},
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&format.G711{
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PayloadTyp: 119,
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SampleRate: 8000,
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ChannelCount: 2,
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},
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},
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{
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"g711 pcmu 8khz stereo",
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&format.G711{
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MULaw: true,
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PayloadTyp: 96,
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SampleRate: 8000,
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ChannelCount: 2,
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},
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webrtc.RTPCodecCapability{
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MimeType: "audio/PCMU",
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ClockRate: 8000,
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Channels: 2,
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},
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&format.G711{
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MULaw: true,
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PayloadTyp: 118,
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SampleRate: 8000,
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ChannelCount: 2,
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},
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},
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{
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"g711 pcma 16khz stereo",
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&format.G711{
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PayloadTyp: 96,
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SampleRate: 16000,
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ChannelCount: 2,
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},
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webrtc.RTPCodecCapability{
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MimeType: "audio/L16",
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ClockRate: 16000,
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Channels: 2,
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},
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&format.LPCM{
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PayloadTyp: 96,
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BitDepth: 16,
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SampleRate: 16000,
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ChannelCount: 2,
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},
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},
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{
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"g711 pcmu 16khz stereo",
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&format.G711{
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MULaw: true,
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PayloadTyp: 96,
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SampleRate: 16000,
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ChannelCount: 2,
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},
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webrtc.RTPCodecCapability{
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MimeType: "audio/L16",
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ClockRate: 16000,
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Channels: 2,
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},
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&format.LPCM{
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PayloadTyp: 96,
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BitDepth: 16,
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SampleRate: 16000,
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ChannelCount: 2,
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},
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},
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{
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"l16 8khz stereo",
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&format.LPCM{
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PayloadTyp: 96,
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BitDepth: 16,
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SampleRate: 8000,
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ChannelCount: 2,
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},
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webrtc.RTPCodecCapability{
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MimeType: "audio/L16",
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ClockRate: 8000,
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Channels: 2,
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},
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&format.LPCM{
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PayloadTyp: 96,
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BitDepth: 16,
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SampleRate: 8000,
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ChannelCount: 2,
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},
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},
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{
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"l16 16khz stereo",
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&format.LPCM{
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PayloadTyp: 96,
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BitDepth: 16,
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SampleRate: 16000,
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ChannelCount: 2,
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},
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webrtc.RTPCodecCapability{
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MimeType: "audio/L16",
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ClockRate: 16000,
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Channels: 2,
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},
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&format.LPCM{
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PayloadTyp: 96,
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BitDepth: 16,
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SampleRate: 16000,
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ChannelCount: 2,
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},
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},
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{
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"l16 48khz stereo",
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&format.LPCM{
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PayloadTyp: 96,
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BitDepth: 16,
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SampleRate: 48000,
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ChannelCount: 2,
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},
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webrtc.RTPCodecCapability{
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MimeType: "audio/L16",
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ClockRate: 48000,
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Channels: 2,
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},
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&format.LPCM{
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PayloadTyp: 96,
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BitDepth: 16,
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SampleRate: 48000,
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ChannelCount: 2,
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},
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},
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} {
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t.Run(ca.name, func(t *testing.T) {
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pc1 := &PeerConnection{
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HandshakeTimeout: conf.StringDuration(10 * time.Second),
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TrackGatherTimeout: conf.StringDuration(2 * time.Second),
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LocalRandomUDP: true,
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IPsFromInterfaces: true,
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Publish: true,
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OutgoingTracks: []*OutgoingTrack{{
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Format: ca.in,
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}},
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Log: test.NilLogger,
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}
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err := pc1.Start()
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require.NoError(t, err)
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defer pc1.Close()
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pc2 := &PeerConnection{
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HandshakeTimeout: conf.StringDuration(10 * time.Second),
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TrackGatherTimeout: conf.StringDuration(2 * time.Second),
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LocalRandomUDP: true,
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IPsFromInterfaces: true,
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Publish: false,
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Log: test.NilLogger,
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}
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err = pc2.Start()
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require.NoError(t, err)
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defer pc2.Close()
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offer, err := pc1.CreatePartialOffer()
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require.NoError(t, err)
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answer, err := pc2.CreateFullAnswer(context.Background(), offer)
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require.NoError(t, err)
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err = pc1.SetAnswer(answer)
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require.NoError(t, err)
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go func() {
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for {
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select {
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case cnd := <-pc1.NewLocalCandidate():
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err2 := pc2.AddRemoteCandidate(cnd)
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require.NoError(t, err2)
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case <-pc1.Connected():
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return
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}
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}
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}()
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err = pc1.WaitUntilConnected(context.Background())
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require.NoError(t, err)
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err = pc2.WaitUntilConnected(context.Background())
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require.NoError(t, err)
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err = pc1.OutgoingTracks[0].WriteRTP(&rtp.Packet{
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Header: rtp.Header{
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Version: 2,
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Marker: true,
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PayloadType: 111,
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SequenceNumber: 1123,
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Timestamp: 45343,
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SSRC: 563424,
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},
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Payload: []byte{5, 2},
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})
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require.NoError(t, err)
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inc, err := pc2.GatherIncomingTracks(context.Background())
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require.NoError(t, err)
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exp := ca.webrtcOut
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exp.RTCPFeedback = inc[0].track.Codec().RTPCodecCapability.RTCPFeedback
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require.Equal(t, exp, inc[0].track.Codec().RTPCodecCapability)
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require.Equal(t, ca.out, inc[0].Format())
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})
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}
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}
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// test that an audio codec is present regardless of the fact that an audio track is not.
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func TestPeerConnectionFallbackCodecs(t *testing.T) {
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pc1 := &PeerConnection{
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HandshakeTimeout: conf.StringDuration(10 * time.Second),
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TrackGatherTimeout: conf.StringDuration(2 * time.Second),
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LocalRandomUDP: true,
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IPsFromInterfaces: true,
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Publish: false,
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Log: test.NilLogger,
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}
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err := pc1.Start()
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require.NoError(t, err)
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defer pc1.Close()
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pc2 := &PeerConnection{
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HandshakeTimeout: conf.StringDuration(10 * time.Second),
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TrackGatherTimeout: conf.StringDuration(2 * time.Second),
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LocalRandomUDP: true,
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IPsFromInterfaces: true,
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Publish: true,
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OutgoingTracks: []*OutgoingTrack{{
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Format: &format.AV1{
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PayloadTyp: 96,
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},
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}},
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Log: test.NilLogger,
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}
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err = pc2.Start()
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require.NoError(t, err)
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defer pc2.Close()
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offer, err := pc1.CreatePartialOffer()
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require.NoError(t, err)
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answer, err := pc2.CreateFullAnswer(context.Background(), offer)
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require.NoError(t, err)
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var s sdp.SessionDescription
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err = s.Unmarshal([]byte(answer.SDP))
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require.NoError(t, err)
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require.Equal(t, []*sdp.MediaDescription{
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{
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MediaName: sdp.MediaName{
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Media: "video",
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Port: sdp.RangedPort{Value: 9},
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Protos: []string{"UDP", "TLS", "RTP", "SAVPF"},
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Formats: []string{"97"},
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},
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ConnectionInformation: s.MediaDescriptions[0].ConnectionInformation,
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Attributes: s.MediaDescriptions[0].Attributes,
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},
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{
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MediaName: sdp.MediaName{
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Media: "audio",
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Port: sdp.RangedPort{Value: 9},
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Protos: []string{"UDP", "TLS", "RTP", "SAVPF"},
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Formats: []string{"0"},
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},
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ConnectionInformation: s.MediaDescriptions[1].ConnectionInformation,
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Attributes: s.MediaDescriptions[1].Attributes,
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},
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}, s.MediaDescriptions)
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}
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