diff --git a/plugins/webrtc/src/rtp-forwarders.ts b/plugins/webrtc/src/rtp-forwarders.ts
index 0cfbe8ab4..84365aeeb 100644
--- a/plugins/webrtc/src/rtp-forwarders.ts
+++ b/plugins/webrtc/src/rtp-forwarders.ts
@@ -293,7 +293,7 @@ export async function startRtpForwarderProcess(console: Console, ffmpegInput: FF
                             }
                         });
 
-                        const audioClient = await listenZeroSingleClient();
+                        const audioClient = await listenZeroSingleClient('127.0.0.1');
                         let audioPipe: Writable;
                         killDeferred.promise.finally(() => audioClient.clientPromise.then(client => client.destroy()));
                         let rtspServer: RtspServer;
@@ -421,7 +421,7 @@ export async function startRtpForwarderProcess(console: Console, ffmpegInput: FF
             // seems better to use udp for audio timing/chop.
             const useUdp = rtspMode === 'udp';
 
-            const serverPort = await listenZeroSingleClient();
+            const serverPort = await listenZeroSingleClient('127.0.0.1');
 
             args.push(
                 '-rtsp_transport',
diff --git a/server/src/listen-zero.ts b/server/src/listen-zero.ts
index d20353efa..9e707c5ef 100644
--- a/server/src/listen-zero.ts
+++ b/server/src/listen-zero.ts
@@ -7,13 +7,13 @@ export class ListenZeroSingleClientTimeoutError extends Error {
     }
 }
 
-export async function listenZero(server: net.Server, hostname?: string) {
+export async function listenZero(server: net.Server, hostname: string) {
     server.listen(0, hostname);
     await once(server, 'listening');
     return (server.address() as net.AddressInfo).port;
 }
 
-export async function listenZeroSingleClient(hostname?: string, options?: net.ServerOpts, listenTimeout = 30000) {
+export async function listenZeroSingleClient(hostname: string, options?: net.ServerOpts, listenTimeout = 30000) {
     const server = new net.Server(options);
     const port = await listenZero(server, hostname);