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refactor: move media channel handling to common/ and setup duplex channel

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a duplex media channel should be more realistic, you generally both send and receive media when doing video call and
stuff
This commit is contained in:
KokaKiwi 2025-08-16 18:48:43 +02:00
parent ff738ff045
commit 01bde142d4
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GPG key ID: FD333F84686EFE78
5 changed files with 187 additions and 102 deletions
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90
client/lib/webrtc.go
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@ -15,9 +15,9 @@ import (
"github.com/pion/transport/v3"
"github.com/pion/transport/v3/stdnet"
"github.com/pion/webrtc/v4"
"github.com/pion/webrtc/v4/pkg/media"
"gitlab.torproject.org/tpo/anti-censorship/pluggable-transports/snowflake/v2/common/event"
"gitlab.torproject.org/tpo/anti-censorship/pluggable-transports/snowflake/v2/common/media"
"gitlab.torproject.org/tpo/anti-censorship/pluggable-transports/snowflake/v2/common/proxy"
"gitlab.torproject.org/tpo/anti-censorship/pluggable-transports/snowflake/v2/common/util"
)
@ -44,6 +44,7 @@ type WebRTCPeer struct {
bytesLogger bytesLogger
eventsLogger event.SnowflakeEventReceiver
proxy *url.URL
mediaChannel *media.MediaChannel
}
// Deprecated: Use NewWebRTCPeerWithNatPolicyAndEventsAndProxy Instead.
@ -107,6 +108,7 @@ func NewWebRTCPeerWithNatPolicyAndEventsAndProxy(
connection.eventsLogger = eventsLogger
connection.proxy = proxy
connection.mediaChannel = media.NewMediaChannel()
err := connection.connect(config, broker, natPolicy)
if err != nil {
@ -342,7 +344,10 @@ func (c *WebRTCPeer) preparePeerConnection(
c.open = make(chan struct{})
log.Println("WebRTC: DataChannel created")
c.openMediaTrack()
err = c.mediaChannel.Start(c.pc)
if err != nil {
log.Printf("Failed to setup media channel: %v", err)
}
offer, err := c.pc.CreateOffer(nil)
// TODO: Potentially timeout and retry if ICE isn't working.
@ -369,85 +374,12 @@ func (c *WebRTCPeer) preparePeerConnection(
return nil
}
func (c *WebRTCPeer) openMediaTrack() {
videoTrack, err := webrtc.NewTrackLocalStaticSample(
webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeAV1}, "video", "pion",
)
if err != nil {
log.Printf("webrtc.NewTrackLocalStaticSample ERROR: %s", err)
return
}
rtpSender, err := c.pc.AddTrack(videoTrack)
if err != nil {
log.Printf("webrtc.AddTrack ERROR: %s", err)
return
}
go func() {
rtcpBuf := make([]byte, 1500)
for {
if _, _, rtcpErr := rtpSender.Read(rtcpBuf); rtcpErr != nil {
return
}
}
}()
go func() {
ticker := time.NewTicker(time.Second)
defer ticker.Stop()
for ; true; <-ticker.C {
// Add jitter to simulate "realistic" media patterns
jitterDelay := time.Duration(randomInt(0, 200)) * time.Millisecond
time.Sleep(jitterDelay)
// Vary packet sizes for specific frames types
var bufSize int
frameType := randomInt(1, 100)
switch {
case frameType <= 5: // I-frames: 5% chance, larger
bufSize = randomInt(8000, 15000)
case frameType <= 35: // P-frames: 30% chance, medium
bufSize = randomInt(2000, 5000)
default: // B-frames: 65% chance, smaller
bufSize = randomInt(500, 2000)
}
buf := make([]byte, bufSize)
// Add some timing variation
frameDuration := time.Duration(randomInt(900, 1100)) * time.Millisecond
err = videoTrack.WriteSample(media.Sample{Data: buf, Duration: frameDuration})
if err != nil {
log.Printf("webrtc.WriteSample ERROR: %s", err)
}
// Simulate some burst of smaller packets
if randomInt(1, 10) == 1 { // 10% chance
burstCount := randomInt(2, 5)
for i := 0; i < burstCount; i++ {
smallBuf := make([]byte, randomInt(100, 400))
time.Sleep(time.Duration(randomInt(10, 50)) * time.Millisecond)
frameDuration = time.Duration(randomInt(16, 33)) * time.Millisecond
err = videoTrack.WriteSample(media.Sample{Data: smallBuf, Duration: frameDuration})
if err != nil {
log.Printf("webrtc.WriteSample burst ERROR: %s", err)
break
}
}
}
}
}()
log.Println("WebRTC: Media track opened")
}
// cleanup closes all channels and transports
func (c *WebRTCPeer) cleanup() {
// Stop media channel
if c.mediaChannel != nil {
c.mediaChannel.Stop()
}
// Close this side of the SOCKS pipe.
if c.writePipe != nil { // c.writePipe can be nil in tests.
c.writePipe.Close()