diff --git a/plugins/webrtc/package-lock.json b/plugins/webrtc/package-lock.json
index d96df8182..045efc37a 100644
--- a/plugins/webrtc/package-lock.json
+++ b/plugins/webrtc/package-lock.json
@@ -1,12 +1,12 @@
 {
    "name": "@scrypted/webrtc",
-   "version": "0.0.112",
+   "version": "0.0.113",
    "lockfileVersion": 2,
    "requires": true,
    "packages": {
       "": {
          "name": "@scrypted/webrtc",
-         "version": "0.0.112",
+         "version": "0.0.113",
          "dependencies": {
             "@koush/werift": "file:../../external/werift/packages/webrtc",
             "@scrypted/common": "file:../../common",
diff --git a/plugins/webrtc/package.json b/plugins/webrtc/package.json
index df3e6782d..579adcb4e 100644
--- a/plugins/webrtc/package.json
+++ b/plugins/webrtc/package.json
@@ -1,6 +1,6 @@
 {
    "name": "@scrypted/webrtc",
-   "version": "0.0.112",
+   "version": "0.0.113",
    "scripts": {
       "scrypted-setup-project": "scrypted-setup-project",
       "prescrypted-setup-project": "scrypted-package-json",
diff --git a/plugins/webrtc/src/ffmpeg-to-wrtc.ts b/plugins/webrtc/src/ffmpeg-to-wrtc.ts
index 884cfc913..9a1f41b09 100644
--- a/plugins/webrtc/src/ffmpeg-to-wrtc.ts
+++ b/plugins/webrtc/src/ffmpeg-to-wrtc.ts
@@ -34,7 +34,7 @@ function getDebugModeH264EncoderArgs() {
 
 export async function createTrackForwarder(options: {
     timeStart: number,
-    isPrivate: boolean, destinationId: string,
+    isPrivate: boolean, destinationId: string, ipv4: boolean,
     requestMediaStream: RequestMediaStream,
     videoTransceiver: RTCRtpTransceiver, audioTransceiver: RTCRtpTransceiver,
     sessionSupportsH264High: boolean, maximumCompatibilityMode: boolean, transcodeWidth: number,
@@ -187,7 +187,11 @@ export async function createTrackForwarder(options: {
         },
     };
 
-    const videoPacketSize = 1300;
+    // ipv4 mtu is 1500
+    // so max usable packet size is 1500 - rtp header - tcp header - ip header
+    // 1500 - 12 - 20 - 20 = 1448.
+    // but set to 1440 cause that's what cameras seem to use for some reason.
+    const videoPacketSize = options.ipv4 ? 1440 : 1300;
     let h264Repacketizer: H264Repacketizer;
     let spsPps: ReturnType<typeof getSpsPps>;
 
diff --git a/plugins/webrtc/src/rtp-forwarders.ts b/plugins/webrtc/src/rtp-forwarders.ts
index 7895bec44..51147a40d 100644
--- a/plugins/webrtc/src/rtp-forwarders.ts
+++ b/plugins/webrtc/src/rtp-forwarders.ts
@@ -69,6 +69,7 @@ async function setupRtspClient(console: Console, rtspClient: RtspClient, channel
                 onRtp: (rtspHeader, rtp) => deliver(rtp),
             });
             console.log('rtsp/udp', section.codec, result);
+            return false;
         }
     }
     catch (e) {
diff --git a/plugins/webrtc/src/werift-util.ts b/plugins/webrtc/src/werift-util.ts
index 01fcc5cb5..a02226666 100644
--- a/plugins/webrtc/src/werift-util.ts
+++ b/plugins/webrtc/src/werift-util.ts
@@ -57,8 +57,11 @@ export function logIsPrivateIceTransport(console: Console, pc: RTCPeerConnection
             console.warn('Turn server is in use. For optimal performance ensure that your router supports source NAT.');
     }
     console.log('Connection is local network:', isPrivate);
+    const destinationId = pc.iceTransports[0]?.connection?.remoteAddr?.[0];
+    const ipv4 = ip.isV4Format(destinationId);
     return {
+        ipv4,
         isPrivate,
-        destinationId: pc.iceTransports[0]?.connection?.remoteAddr?.[0],
+        destinationId,
     };
 }