zishuo/webrtc
1
0
Fork 0
mirror of https://github.com/pion/webrtc.git synced 2025-10-12 18:40:05 +08:00
Code Issues Packages Projects Releases Wiki Activity

Don't blindly forward RTP Packets in rtp-to-webrtc

Browse Source 
ffmpeg produces packets that cause issues in Chromium. Instead of
validating/sanitizing just create a new packet.

Resolves #1514
This commit is contained in:
Sean DuBois
2021-01-13 09:36:11 -08:00
parent 52249252fb
commit 9439d820c5
2 changed files with 29 additions and 5 deletions
Download Patch File Download Diff File Expand all files Collapse all files

6
examples/rtp-to-webrtc/README.md
View File

@@ -38,6 +38,12 @@ gst-launch-1.0 videotestsrc ! video/x-raw,width=640,height=480,format=I420 ! vp8
ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 -vcodec libvpx -cpu-used 5 -deadline 1 -g 10 -error-resilient 1 -auto-alt-ref 1 -f rtp rtp://127.0.0.1:5004
```
If you wish to send audio replace both occurrences of `vp8` in `main.go` then run
```
ffmpeg -f lavfi -i "sine=frequency=1000" -c:a libopus -b:a 48000 -sample_fmt s16p -ssrc 1 -payload_type 111 -f rtp -max_delay 0 -application lowdelay rtp:/127.0.0.1:5004
```
### Input rtp-to-webrtc's SessionDescription into your browser
Copy the text that `rtp-to-webrtc` just emitted and copy into second text area

26
examples/rtp-to-webrtc/main.go
View File

@@ -7,8 +7,10 @@ import (
"net"
"github.com/pion/rtp"
"github.com/pion/rtp/codecs"
"github.com/pion/webrtc/v3"
"github.com/pion/webrtc/v3/examples/internal/signal"
"github.com/pion/webrtc/v3/pkg/media/samplebuilder"
)
func main() {
@@ -37,7 +39,7 @@ func main() {
fmt.Println("Waiting for RTP Packets, please run GStreamer or ffmpeg now")
// Listen for a single RTP Packet, we need this to determine the SSRC
inboundRTPPacket := make([]byte, 4096) // UDP MTU
inboundRTPPacket := make([]byte, 1500) // UDP MTU
n, _, err := listener.ReadFromUDP(inboundRTPPacket)
if err != nil {
panic(err)
@@ -50,7 +52,7 @@ func main() {
}
// Create a video track
videoTrack, err := webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{MimeType: "video/vp8"}, "video", "pion")
videoTrack, err := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: "video/vp8"}, "video", "pion")
if err != nil {
panic(err)
}
@@ -108,16 +110,32 @@ func main() {
// Output the answer in base64 so we can paste it in browser
fmt.Println(signal.Encode(*peerConnection.LocalDescription()))
videoBuilder := samplebuilder.New(10, &codecs.VP8Packet{}, 90000)
// Read RTP packets forever and send them to the WebRTC Client
for {
inboundRTPPacket = make([]byte, 1500) // UDP MTU
packet = &rtp.Packet{}
n, _, err := listener.ReadFrom(inboundRTPPacket)
if err != nil {
fmt.Printf("error during read: %s", err)
panic(fmt.Sprintf("error during read: %s", err))
}
if err = packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
panic(err)
}
if _, writeErr := videoTrack.Write(inboundRTPPacket[:n]); writeErr != nil {
videoBuilder.Push(packet)
for {
sample := videoBuilder.Pop()
if sample == nil {
break
}
if writeErr := videoTrack.WriteSample(*sample); writeErr != nil {
panic(writeErr)
}
}
}
}