diff --git a/plugins/webrtc/package-lock.json b/plugins/webrtc/package-lock.json
index 43d816e10..7af111629 100644
--- a/plugins/webrtc/package-lock.json
+++ b/plugins/webrtc/package-lock.json
@@ -1,12 +1,12 @@
 {
    "name": "@scrypted/webrtc",
-   "version": "0.0.39",
+   "version": "0.0.40",
    "lockfileVersion": 2,
    "requires": true,
    "packages": {
       "": {
          "name": "@scrypted/webrtc",
-         "version": "0.0.39",
+         "version": "0.0.40",
          "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 afc8cd4f4..e781b8238 100644
--- a/plugins/webrtc/package.json
+++ b/plugins/webrtc/package.json
@@ -1,6 +1,6 @@
 {
    "name": "@scrypted/webrtc",
-   "version": "0.0.39",
+   "version": "0.0.40",
    "scripts": {
       "prepublishOnly": "NODE_ENV=production scrypted-webpack",
       "prescrypted-vscode-launch": "scrypted-webpack",
diff --git a/plugins/webrtc/src/ffmpeg-to-wrtc.ts b/plugins/webrtc/src/ffmpeg-to-wrtc.ts
index 0ec7f2131..63f2ab177 100644
--- a/plugins/webrtc/src/ffmpeg-to-wrtc.ts
+++ b/plugins/webrtc/src/ffmpeg-to-wrtc.ts
@@ -5,13 +5,12 @@ import { connectRTCSignalingClients } from "@scrypted/common/src/rtc-signaling";
 import { RtspServer } from "@scrypted/common/src/rtsp-server";
 import { createSdpInput, parseSdp } from "@scrypted/common/src/sdp-utils";
 import sdk, { FFmpegInput, Intercom, MediaStreamDestination, MediaStreamTool, RTCAVSignalingSetup, RTCSignalingSession } from "@scrypted/sdk";
-import ip from 'ip';
 import { WeriftOutputSignalingSession } from "./output-signaling-session";
 import { waitConnected } from "./peerconnection-util";
 import { getFFmpegRtpAudioOutputArguments, RtpTrack, RtpTracks, startRtpForwarderProcess } from "./rtp-forwarders";
 import { ScryptedSessionControl } from "./session-control";
 import { requiredAudioCodecs, requiredVideoCodec } from "./webrtc-required-codecs";
-import { isPeerConnectionAlive } from "./werift-util";
+import { isPeerConnectionAlive, logIsPrivateIceTransport } from "./werift-util";
 
 const { mediaManager } = sdk;
 
@@ -169,15 +168,7 @@ export async function createRTCPeerConnectionSink(
         await waitConnected(pc);
 
         console.log('connected', Date.now() - timeStart);
-
-        let isPrivate = true;
-        for (const ice of pc.iceTransports) {
-            const [address, port] = ice.connection.remoteAddr;
-            isPrivate = isPrivate && ip.isPrivate(address);
-            console.log('ice transport ip', address);
-        }
-
-        console.log('Connection is local network:', isPrivate);
+        const isPrivate = logIsPrivateIceTransport(console, pc);
 
         // should really inspect the session description here.
 
diff --git a/plugins/webrtc/src/werift-util.ts b/plugins/webrtc/src/werift-util.ts
index 6187e20c2..cd6cffc48 100644
--- a/plugins/webrtc/src/werift-util.ts
+++ b/plugins/webrtc/src/werift-util.ts
@@ -1,4 +1,5 @@
 import { RTCIceServer, RTCPeerConnection, RTCSessionDescription } from "@koush/werift";
+import ip from 'ip';
 
 export function createRawResponse(response: RTCSessionDescription): RTCSessionDescriptionInit {
     return {
@@ -40,3 +41,14 @@ export function getWeriftIceServers(configuration: RTCConfiguration): RTCIceServ
 
     return ret;
 }
+
+export function logIsPrivateIceTransport(console: Console, pc: RTCPeerConnection) {
+    let isPrivate = true;
+    for (const ice of pc.iceTransports) {
+        const [address, port] = ice.connection.remoteAddr;
+        isPrivate = isPrivate && ip.isPrivate(address);
+        console.log('ice transport ip', address);
+    }
+    console.log('Connection is local network:', isPrivate);
+    return isPrivate;
+}
diff --git a/plugins/webrtc/src/wrtc-to-rtsp.ts b/plugins/webrtc/src/wrtc-to-rtsp.ts
index 5eaa12dfd..413450a4d 100644
--- a/plugins/webrtc/src/wrtc-to-rtsp.ts
+++ b/plugins/webrtc/src/wrtc-to-rtsp.ts
@@ -1,7 +1,7 @@
-import { BundlePolicy, Output, Pipeline, RTCPeerConnection, RtcpPacket, RtcpPayloadSpecificFeedback, RTCRtpTransceiver, RTCSessionDescription, RtpPacket, uint16Add } from "@koush/werift";
+import { BundlePolicy, Pipeline, RTCPeerConnection, RtcpPacket, RtcpPayloadSpecificFeedback, RTCRtpTransceiver, RtpPacket, uint16Add } from "@koush/werift";
 import { FullIntraRequest } from "@koush/werift/lib/rtp/src/rtcp/psfb/fullIntraRequest";
 import { listenZeroSingleClient } from "@scrypted/common/src/listen-cluster";
-import { findH264NaluType, findH264NaluTypeInNalu, getNaluTypesInNalu, H264_NAL_TYPE_IDR, RtspServer } from "@scrypted/common/src/rtsp-server";
+import { getNaluTypesInNalu, RtspServer } from "@scrypted/common/src/rtsp-server";
 import { createSdpInput, parseSdp } from '@scrypted/common/src/sdp-utils';
 import sdk, { FFmpegInput, Intercom, MediaObject, MediaStreamUrl, ResponseMediaStreamOptions, RTCAVSignalingSetup, RTCSessionControl, RTCSignalingChannel, RTCSignalingOptions, RTCSignalingSendIceCandidate, RTCSignalingSession, ScryptedMimeTypes } from "@scrypted/sdk";
 import dgram from 'dgram';
@@ -9,7 +9,7 @@ import { Socket } from "net";
 import { waitConnected } from "./peerconnection-util";
 import { getFFmpegRtpAudioOutputArguments, startRtpForwarderProcess } from "./rtp-forwarders";
 import { requiredAudioCodecs, requiredVideoCodec } from "./webrtc-required-codecs";
-import { createRawResponse, getWeriftIceServers, isPeerConnectionAlive } from "./werift-util";
+import { createRawResponse, getWeriftIceServers, isPeerConnectionAlive, logIsPrivateIceTransport } from "./werift-util";
 
 const { mediaManager } = sdk;
 
@@ -98,6 +98,7 @@ export async function createRTCPeerConnectionSource(options: {
 
             waitConnected(pc).then(() => {
                 console.log('connected', Date.now() - timeStart);
+                logIsPrivateIceTransport(console, pc);
                 sessionControl.startSession().catch(() => { });
             });
 
@@ -297,6 +298,7 @@ export async function createRTCPeerConnectionSource(options: {
     const url = `rtsp://127.0.0.1:${port}`;
     const mediaStreamUrl: MediaStreamUrl = {
         url,
+        container: 'rtsp',
         mediaStreamOptions,
     };