feat: add query to push and play

README.en.md

@@ -67,7 +67,13 @@ Response Body: `SDP`
 ### Push Test Page
 
 `/webrtc/test/publish`
+- `?streamPath=xxx` The streamPath to publish, default is `live/webrtc`
+- you can add other query parameters to the URL
+### Play Test Page
 
+`/webrtc/test/subscribe`
+- `?streamPath=xxx` The streamPath to play, default is `live/webrtc`
+- you can add other query parameters to the URL
 ## WHIP
 
 WebRTC-HTTP ingestion protocol

README.md

@@ -64,7 +64,13 @@ Response Body: `SDP`
 ### 推流测试页面
 
 `/webrtc/test/publish`
+- 可增加参数`?streamPath=xxx`指定推流地址，默认为`live/webrtc`
+- 可以增加其他推流参数
+### 播放测试页面
 
+`/webrtc/test/subscribe`
+- 可增加参数`?streamPath=xxx`指定播放地址，默认为`live/webrtc`
+- 可以增加其他播放参数
 ## WHIP
 WebRTC-HTTP ingestion protocol
 用于WebRTC交换SDP信息的规范

main.go

@@ -155,6 +155,9 @@ func (conf *WebRTCConfig) Play_(w http.ResponseWriter, r *http.Request) {
 
 func (conf *WebRTCConfig) Push_(w http.ResponseWriter, r *http.Request) {
 	streamPath := r.URL.Path[len("/push/"):]
+	if r.URL.RawQuery != "" {
+		streamPath += "?" + r.URL.RawQuery
+	}
 	w.Header().Set("Content-Type", "application/sdp")
 	bytes, err := ioutil.ReadAll(r.Body)
 	var puber WebRTCPublisher

publish.html

@@ -12,7 +12,7 @@
   <video id="video" width="640" height="480" autoplay muted>
   </video>
   <!-- <button id="sw" onclick="action()" type="button" style="width:100px;height:30px;display: block;">unpublish</button> -->
-<pre>
+  <pre>
   <code id="remoteSdp">
 
   </code>
@@ -39,9 +39,12 @@
     pc.onicecandidate = (e) => {
       console.log('onicecandidate', e.candidate);
     };
-    mediaStream.id = 'live/webrtc';
+    const searchParams = new URLSearchParams(location.search);
+    const streamPath = searchParams.get('streamPath') || 'live/webrtc';
+    searchParams.delete('streamPath')
+    mediaStream.id = streamPath;
     mediaStream.getTracks().forEach((t) => {
-        pc.addTrack(t, mediaStream);
+      pc.addTrack(t, mediaStream);
     });
     // const videoTransceiver = pc.addTransceiver(mediaStream.getVideoTracks()[0], { direction: 'sendonly' });
     // const audioTransceiver = pc.addTransceiver(mediaStream.getAudioTracks()[0], { direction: 'sendonly' });
@@ -49,7 +52,7 @@
     const offer = await pc.createOffer();
     await pc.setLocalDescription(offer);
     const result = await fetch(
-      `/webrtc/push/live/webrtc`,
+      `/webrtc/push/${streamPath}${location.search?`?${searchParams.toString()}`:''}`,
       {
         method: 'POST',
         mode: 'cors',

publisher.go

@@ -46,7 +46,7 @@ func (puber *WebRTCPublisher) onTrack(track *TrackRemote, receiver *RTPReceiver)
 				return
 			}
 			rtpItem := puber.AudioTrack.GetRTPFromPool()
-			if i, _, err := track.Read(rtpItem.Value.Raw[:1460]); err == nil {
+			if i, _, err := track.Read(rtpItem.Value.Raw); err == nil {
 				rtpItem.Value.Unmarshal(rtpItem.Value.Raw[:i])
 				puber.AudioTrack.WriteRTP(rtpItem)
 			} else {
@@ -66,7 +66,7 @@ func (puber *WebRTCPublisher) onTrack(track *TrackRemote, receiver *RTPReceiver)
 				return
 			}
 			rtpItem := puber.VideoTrack.GetRTPFromPool()
-			if i, _, err := track.Read(rtpItem.Value.Raw[:1460]); err == nil {
+			if i, _, err := track.Read(rtpItem.Value.Raw); err == nil {
 				rtpItem.Value.Unmarshal(rtpItem.Value.Raw[:i])
 				puber.VideoTrack.WriteRTP(rtpItem)
 			} else {

subscribe.html

@@ -40,8 +40,11 @@
     // const dc = pc.createDataChannel('sdp');
     const offer = await pc.createOffer();
     await pc.setLocalDescription(offer);
+    const searchParams = new URLSearchParams(location.search);
+    const streamPath = searchParams.get('streamPath') || 'live/webrtc';
+    searchParams.delete('streamPath')
     const result = await fetch(
-      `/webrtc/play/live/webrtc`,
+      `/webrtc/play/${streamPath}${location.search?`?${searchParams.toString()}`:''}`,
       {
         method: 'POST',
         mode: 'cors',