gilles/go2rtc
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Merge branch 'master' into wyze

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This commit is contained in:
seydx
2026-01-17 20:31:05 +01:00
parent 160695857e 6a5deecfcc
commit 679454aedb
15 changed files with 260 additions and 69 deletions
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docker/Dockerfile
+1
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@@ -47,6 +47,7 @@ RUN if [ "${TARGETARCH}" = "amd64" ]; then apk add --no-cache libva-intel-driver
COPY --from=build /build/go2rtc /usr/local/bin/
EXPOSE 1984 8554 8555 8555/udp
ENTRYPOINT ["/sbin/tini", "--"]
VOLUME /config
WORKDIR /config
docker/hardware.Dockerfile
+1
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@@ -49,6 +49,7 @@ RUN --mount=type=cache,target=/var/cache/apt,sharing=locked --mount=type=cache,t
COPY --from=build /build/go2rtc /usr/local/bin/
EXPOSE 1984 8554 8555 8555/udp
ENTRYPOINT ["/usr/bin/tini", "--"]
VOLUME /config
WORKDIR /config
docker/rockchip.Dockerfile
+1
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@@ -43,6 +43,7 @@ RUN --mount=type=cache,target=/var/cache/apt,sharing=locked --mount=type=cache,t
COPY --from=build /build/go2rtc /usr/local/bin/
ADD --chmod=755 https://github.com/MarcA711/Rockchip-FFmpeg-Builds/releases/download/6.1-8-no_extra_dump/ffmpeg /usr/local/bin
EXPOSE 1984 8554 8555 8555/udp
ENTRYPOINT ["/usr/bin/tini", "--"]
VOLUME /config
WORKDIR /config
internal/doorbird/README.md
+21
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@@ -0,0 +1,21 @@
# Doorbird
*[added in v1.9.8](https://github.com/AlexxIT/go2rtc/releases/tag/v1.9.11)*
This source type supports [Doorbird](https://www.doorbird.com/) devices including MJPEG stream, audio stream as well as two-way audio.
It is recommended to create a sepearate user within your doorbird setup for go2rtc. Minimum permissions for the user are:
- Watch always
- API operator
## Configuration
```yaml
streams:
doorbird1:
- rtsp://admin:password@192.168.1.123:8557/mpeg/720p/media.amp # RTSP stream
- doorbird://admin:password@192.168.1.123?media=video # MJPEG stream
- doorbird://admin:password@192.168.1.123?media=audio # audio stream
- doorbird://admin:password@192.168.1.123 # two-way audio
```
internal/multitrans/README.md
+16
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@@ -0,0 +1,16 @@
# Multitrans
**added in v1.9.14** by [@forrestsocool](https://github.com/forrestsocool)
Two-way audio support for Chinese version of [TP-Link cameras](https://www.tp-link.com.cn/list_2549.html).
## Configuration
```yaml
streams:
tplink_cam:
# video use standard RTSP
- rtsp://admin:admin@192.168.1.202:554/stream1
# two-way audio use MULTITRANS schema
- multitrans://admin:admin@192.168.1.202:554
```
internal/multitrans/multitrans.go
+10
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@@ -0,0 +1,10 @@
package multitrans
import (
"github.com/AlexxIT/go2rtc/internal/streams"
"github.com/AlexxIT/go2rtc/pkg/multitrans"
)
func Init() {
streams.HandleFunc("multitrans", multitrans.Dial)
}
main.go
+2
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@@ -27,6 +27,7 @@ import (
"github.com/AlexxIT/go2rtc/internal/mjpeg"
"github.com/AlexxIT/go2rtc/internal/mp4"
"github.com/AlexxIT/go2rtc/internal/mpegts"
"github.com/AlexxIT/go2rtc/internal/multitrans"
"github.com/AlexxIT/go2rtc/internal/nest"
"github.com/AlexxIT/go2rtc/internal/ngrok"
"github.com/AlexxIT/go2rtc/internal/onvif"
@@ -96,6 +97,7 @@ func main() {
{"isapi", isapi.Init},
{"ivideon", ivideon.Init},
{"mpegts", mpegts.Init},
{"multitrans", multitrans.Init},
{"nest", nest.Init},
{"ring", ring.Init},
{"roborock", roborock.Init},
pkg/core/codec.go
+1 -1
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@@ -277,7 +277,7 @@ func ParseCodecString(s string) *Codec {
codec.ClockRate = uint32(Atoi(ss[1]))
}
if len(ss) >= 3 {
codec.Channels = uint8(Atoi(ss[1]))
codec.Channels = uint8(Atoi(ss[2]))
}
return &codec
pkg/multitrans/client.go
+203
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@@ -0,0 +1,203 @@
package multitrans
import (
"bufio"
"bytes"
"errors"
"fmt"
"net"
"net/http"
"net/url"
"github.com/AlexxIT/go2rtc/pkg/core"
"github.com/AlexxIT/go2rtc/pkg/tcp"
"github.com/google/uuid"
"github.com/pion/rtp"
)
type Client struct {
core.Connection
conn net.Conn
rd *bufio.Reader
closed core.Waiter
}
func Dial(rawURL string) (core.Producer, error) {
u, err := url.Parse(rawURL)
if err != nil {
return nil, err
}
if u.Port() == "" {
u.Host += ":554"
}
conn, err := net.DialTimeout("tcp", u.Host, core.ConnDialTimeout)
if err != nil {
return nil, err
}
c := &Client{
conn: conn,
rd: bufio.NewReader(conn),
}
if err = c.handshake(u); err != nil {
_ = conn.Close()
return nil, err
}
c.Connection = core.Connection{
ID: core.NewID(),
FormatName: "multitrans",
Protocol: "rtsp",
RemoteAddr: conn.RemoteAddr().String(),
Source: rawURL,
Medias: []*core.Media{
{
Kind: core.KindAudio,
Direction: core.DirectionSendonly,
Codecs: []*core.Codec{{Name: core.CodecPCMA, ClockRate: 8000, PayloadType: 8}},
},
},
Transport: conn,
}
return c, nil
}
func (c *Client) AddTrack(media *core.Media, _ *core.Codec, track *core.Receiver) error {
sender := core.NewSender(media, track.Codec)
sender.Handler = func(packet *rtp.Packet) {
clone := rtp.Packet{
Header: rtp.Header{
Version: 2,
Marker: packet.Marker,
PayloadType: 8,
SequenceNumber: packet.SequenceNumber,
Timestamp: packet.Timestamp,
SSRC: packet.SSRC,
},
Payload: packet.Payload,
}
// Encapsulate in RTSP Interleaved Frame (Channel 1)
// $ + Channel(1 byte) + Length(2 bytes) + Packet
size := 12 + len(clone.Payload)
b := make([]byte, 4+size)
b[0] = '$'
b[1] = 1 // Channel 1 for audio
b[2] = byte(size >> 8)
b[3] = byte(size)
if _, err := clone.MarshalTo(b[4:]); err != nil {
return
}
if _, err := c.conn.Write(b); err != nil {
return
}
}
sender.HandleRTP(track)
c.Senders = append(c.Senders, sender)
return nil
}
func (c *Client) handshake(u *url.URL) error {
// Step 1: Get Challenge
uid := uuid.New().String()
uri := fmt.Sprintf("rtsp://%s/multitrans", u.Host)
data := fmt.Sprintf("MULTITRANS %s RTSP/1.0\r\nCSeq: 0\r\nX-Client-UUID: %s\r\n\r\n", uri, uid)
if _, err := c.conn.Write([]byte(data)); err != nil {
return err
}
res, err := tcp.ReadResponse(c.rd)
if err != nil {
return err
}
if res.StatusCode != http.StatusUnauthorized {
return errors.New("multitrans: expected 401, got " + res.Status)
}
auth := res.Header.Get("WWW-Authenticate")
realm := tcp.Between(auth, `realm="`, `"`)
nonce := tcp.Between(auth, `nonce="`, `"`)
// Step 2: Send Auth
user := u.User.Username()
pass, _ := u.User.Password()
ha1 := tcp.HexMD5(user, realm, pass)
ha2 := tcp.HexMD5("MULTITRANS", uri)
response := tcp.HexMD5(ha1, nonce, ha2)
authHeader := fmt.Sprintf(`Digest username="%s", realm="%s", nonce="%s", uri="%s", response="%s"`,
user, realm, nonce, uri, response)
data = fmt.Sprintf("MULTITRANS %s RTSP/1.0\r\nCSeq: 1\r\nAuthorization: %s\r\nX-Client-UUID: %s\r\n\r\n",
uri, authHeader, uid)
if _, err = c.conn.Write([]byte(data)); err != nil {
return err
}
res, err = tcp.ReadResponse(c.rd)
if err != nil {
return err
}
if res.StatusCode != http.StatusOK {
return errors.New("multitrans: auth failed: " + res.Status)
}
// Session: 7116520596809429228
session := res.Header.Get("Session")
if session == "" {
return errors.New("multitrans: no session")
}
return c.openTalkChannel(uri, session)
}
func (c *Client) openTalkChannel(uri, session string) error {
payload := `{"type":"request","seq":0,"params":{"method":"get","talk":{"mode":"full_duplex"}}}`
data := fmt.Sprintf("MULTITRANS %s RTSP/1.0\r\nCSeq: 2\r\nSession: %s\r\nContent-Type: application/json\r\nContent-Length: %d\r\n\r\n%s",
uri, session, len(payload), payload)
if _, err := c.conn.Write([]byte(data)); err != nil {
return err
}
res, err := tcp.ReadResponse(c.rd)
if err != nil {
return err
}
if res.StatusCode != http.StatusOK {
return errors.New("multitrans: talkback failed: " + res.Status)
}
// Python checks for "error_code":0 in body.
if !bytes.Contains(res.Body, []byte(`"error_code":0`)) {
return fmt.Errorf("multitrans: talkback error: %s", string(res.Body))
}
return nil
}
func (c *Client) GetTrack(media *core.Media, codec *core.Codec) (*core.Receiver, error) {
return nil, core.ErrCantGetTrack
}
func (c *Client) Start() error {
_ = c.closed.Wait()
return nil
}
func (c *Client) Stop() error {
c.closed.Done(nil)
return c.Connection.Stop()
}
www/codecs.html
-65
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@@ -1,65 +0,0 @@
<!DOCTYPE html>
<html lang="en">
<head>
<meta name="viewport" content="width=device-width, initial-scale=1">
<title>codecs - go2rtc</title>
<style>
body {
margin: 0;
padding: 0;
}
html, body {
height: 100%;
width: 100%;
}
</style>
</head>
<body>
<div id="out"></div>
<script>
const out = document.getElementById('out');
const print = (name, caps) => {
out.innerText += name + '\n';
caps.codecs.forEach((codec) => {
out.innerText += [codec.mimeType, codec.channels, codec.clockRate, codec.sdpFmtpLine] + '\n';
});
out.innerText += '\n';
};
if (RTCRtpReceiver.getCapabilities) {
print('receiver video', RTCRtpReceiver.getCapabilities('video'));
print('receiver audio', RTCRtpReceiver.getCapabilities('audio'));
print('sender video', RTCRtpSender.getCapabilities('video'));
print('sender audio', RTCRtpSender.getCapabilities('audio'));
}
const types = [
'video/mp4; codecs="avc1.42401E"',
'video/mp4; codecs="avc1.42C01E"',
'video/mp4; codecs="avc1.42E01E"',
'video/mp4; codecs="avc1.42001E"',
'video/mp4; codecs="avc1.4D401E"',
'video/mp4; codecs="avc1.4D001E"',
'video/mp4; codecs="avc1.640032"',
'video/mp4; codecs="avc1.640C32"',
'video/mp4; codecs="avc1.F4001F"',
'video/mp4; codecs="hvc1.1.6.L93.B0"',
'video/mp4; codecs="hev1.1.6.L93.B0"',
'video/mp4; codecs="hev1.2.4.L120.B0"',
'video/mp4; codecs="flac"',
'video/mp4; codecs="opus"',
'video/mp4; codecs="mp3"',
'video/mp4; codecs="null"',
'application/vnd.apple.mpegurl',
];
const video = document.createElement('video');
out.innerText += 'video.canPlayType\n';
types.forEach(type => {
out.innerText += `${type} = ${'MediaSource' in window && MediaSource.isTypeSupported(type)} / ${video.canPlayType(type)}\n`;
});
</script>
</body>
</html>
www/config.html
+1 -1
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@@ -1205,7 +1205,7 @@
(async () => {
try {
const r = await fetch('https://go2rtc.org/schema.json', {cache: 'no-cache'});
const r = await fetch('schema.json', {cache: 'no-cache'});
if (r.ok) setupYamlHints(await r.json());
} catch (e) {
// ignore schema load errors
www/links.html
+1 -1
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@@ -87,7 +87,7 @@
if (data.setup_code === undefined) return;
const script = document.createElement('script');
script.src = 'https://cdnjs.cloudflare.com/ajax/libs/qrcodejs/1.0.0/qrcode.min.js';
script.src = 'https://cdn.jsdelivr.net/npm/qrcodejs@1.0.0/qrcode.min.js';
script.async = true;
script.onload = () => {
/* global BigInt */
www/net.html
+1 -1
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@@ -4,7 +4,7 @@
<meta charset="UTF-8">
<meta name="viewport" content="width=device-width, initial-scale=1.0">
<title>net - go2rtc</title>
<script src="https://unpkg.com/vis-network@9.1.9/standalone/umd/vis-network.min.js"></script>
<script src="https://cdn.jsdelivr.net/npm/vis-network@10.0.2/standalone/umd/vis-network.min.js"></script>
<style>
html, body, #network {
height: 100%;
website/schema.json → www/schema.json
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www/static.go
+1
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@@ -4,4 +4,5 @@ import "embed"
//go:embed *.html
//go:embed *.js
//go:embed *.json
var Static embed.FS