rtp marker tweaks on webrtc talkback (#1187)

* set rtp marker * set marker if last packet was recevied 1s+ ago * fix after merge * reorder * set marker if last packet was recevied 1s+ ago
koush · Nov 18, 2023 · eaeae02 · eaeae02
1 parent 7460c71
commit eaeae02
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Showing 2 changed files with 9 additions and 3 deletions.
diff --git a/plugins/webrtc/src/ffmpeg-to-wrtc.ts b/plugins/webrtc/src/ffmpeg-to-wrtc.ts
@@ -217,6 +217,7 @@ export async function createTrackForwarder(options: {
     }
 
     let opusRepacketizer: OpusRepacketizer;
+    let lastPacketTs: number = 0;
     const audioRtpTrack: RtpTrack = {
         codecCopy: audioCodecCopy,
         onRtp: buffer => {
@@ -230,11 +231,13 @@ export async function createTrackForwarder(options: {
             }
             else {
                 const rtp = RtpPacket.deSerialize(buffer);
-                rtp.header.marker = false;
+                const now = Date.now();
+                rtp.header.marker = now - lastPacketTs > 1000; // set the marker if it's been more than 1s since the last packet
                 rtp.header.payloadType = audioTransceiver.sender.codec.payloadType;
                 // pcm audio can be concatenated.
                 // hikvision seems to send 40ms duration packets, so 25 packets per second.
                 audioTransceiver.sender.sendRtp(rtp.serialize());
+                lastPacketTs = now;
             }
         },
         encoderArguments: [
@@ -257,7 +260,7 @@ export async function createTrackForwarder(options: {
     // 1/9/2023:
     // 1378 is what homekit requests, regardless of local or remote network.
     // so setting 1378 as the fixed value seems wise, given apple probably has
-    // better knowledge of network capabilities, and also mirrors 
+    // better knowledge of network capabilities, and also mirrors
     // from my cursory research into ipv6, the MTU is no lesser than ipv4, in fact
     // the min mtu is larger.
     const videoPacketSize = 1378;

diff --git a/plugins/webrtc/src/wrtc-to-rtsp.ts b/plugins/webrtc/src/wrtc-to-rtsp.ts
@@ -274,15 +274,18 @@ export async function createRTCPeerConnectionSource(options: {
 
                 const ffmpegInput = await mediaManager.convertMediaObjectToJSON<FFmpegInput>(media, ScryptedMimeTypes.FFmpegInput);
 
+                let lastPacketTs: number = 0;
                 const { kill: destroy } = await startRtpForwarderProcess(console, ffmpegInput, {
                     audio: {
                         codecCopy: audioCodec.name,
                         encoderArguments: getFFmpegRtpAudioOutputArguments(ffmpegInput.mediaStreamOptions?.audio?.codec, audioTransceiver.sender.codec, maximumCompatibilityMode),
                         onRtp: (rtp) => {
                             const packet = RtpPacket.deSerialize(rtp);
+                            const now = Date.now();
                             packet.header.payloadType = audioCodec.payloadType;
-                            packet.header.marker = false;
+                            packet.header.marker = now - lastPacketTs > 1000; // set the marker if it's been more than 1s since the last packet
                             audioTransceiver.sender.sendRtp(packet.serialize());
+                            lastPacketTs = now;
                         },
                     },
                 });