From f7d88273e48efbfa3dc4cdecff69599b8c9bed9a Mon Sep 17 00:00:00 2001
From: Koushik Dutta <koushd@gmail.com>
Date: Thu, 16 Nov 2023 21:17:01 -0800
Subject: [PATCH] webrtc: talkback fixes

---
 plugins/webrtc/package-lock.json   |  4 ++--
 plugins/webrtc/package.json        |  2 +-
 plugins/webrtc/src/wrtc-to-rtsp.ts | 11 ++++++++---
 3 files changed, 11 insertions(+), 6 deletions(-)

diff --git a/plugins/webrtc/package-lock.json b/plugins/webrtc/package-lock.json
index e3eabafdb..bad695772 100644
--- a/plugins/webrtc/package-lock.json
+++ b/plugins/webrtc/package-lock.json
@@ -1,12 +1,12 @@
 {
    "name": "@scrypted/webrtc",
-   "version": "0.1.76",
+   "version": "0.1.79",
    "lockfileVersion": 2,
    "requires": true,
    "packages": {
       "": {
          "name": "@scrypted/webrtc",
-         "version": "0.1.76",
+         "version": "0.1.79",
          "dependencies": {
             "@scrypted/common": "file:../../common",
             "@scrypted/sdk": "file:../../sdk",
diff --git a/plugins/webrtc/package.json b/plugins/webrtc/package.json
index 453cd7b5f..8288e65b5 100644
--- a/plugins/webrtc/package.json
+++ b/plugins/webrtc/package.json
@@ -1,6 +1,6 @@
 {
    "name": "@scrypted/webrtc",
-   "version": "0.1.76",
+   "version": "0.1.79",
    "scripts": {
       "scrypted-setup-project": "scrypted-setup-project",
       "prescrypted-setup-project": "scrypted-package-json",
diff --git a/plugins/webrtc/src/wrtc-to-rtsp.ts b/plugins/webrtc/src/wrtc-to-rtsp.ts
index c66165bc5..2be780488 100644
--- a/plugins/webrtc/src/wrtc-to-rtsp.ts
+++ b/plugins/webrtc/src/wrtc-to-rtsp.ts
@@ -261,14 +261,14 @@ export async function createRTCPeerConnectionSource(options: {
 
         let destroyProcess: () => void;
 
-        const track = audioTransceiver.sender.sendRtp;
+        const audioCodec = audioTransceiver?.sender?.codec;
 
         const ic: Intercom = {
             async startIntercom(media: MediaObject) {
                 if (!isPeerConnectionAlive(pc))
                     throw new Error('peer connection is closed');
 
-                if (!track)
+                if (!audioTransceiver?.sender?.sendRtp || !audioCodec)
                     throw new Error('peer connection does not support two way audio');
 
 
@@ -276,8 +276,13 @@ export async function createRTCPeerConnectionSource(options: {
 
                 const { kill: destroy } = await startRtpForwarderProcess(console, ffmpegInput, {
                     audio: {
+                        codecCopy: audioCodec.name,
                         encoderArguments: getFFmpegRtpAudioOutputArguments(ffmpegInput.mediaStreamOptions?.audio?.codec, audioTransceiver.sender.codec, maximumCompatibilityMode),
-                        onRtp: (rtp) => audioTransceiver.sender.sendRtp(rtp),
+                        onRtp: (rtp) => {
+                            const packet = RtpPacket.deSerialize(rtp);
+                            packet.header.payloadType = audioCodec.payloadType;
+                            audioTransceiver.sender.sendRtp(packet.serialize());
+                        },
                     },
                 });