diff --git a/webrtc/src/opcodes/SelectProtocol.ts b/webrtc/src/opcodes/SelectProtocol.ts
index 08335ade..36527a8b 100644
--- a/webrtc/src/opcodes/SelectProtocol.ts
+++ b/webrtc/src/opcodes/SelectProtocol.ts
@@ -68,6 +68,8 @@ import * as sdpTransform from 'sdp-transform';
}
*/
+var test_hasMadeProducer = false;
+
export async function onSelectProtocol(this: Server, socket: WebSocket, data: Payload) {
const rtpCapabilities = this.mediasoupRouters[0].rtpCapabilities;
const codecs = rtpCapabilities.codecs as RtpCodecCapability[];
@@ -85,23 +87,33 @@ export async function onSelectProtocol(this: Server, socket: WebSocket, data: Pa
})),
*/
- const producer = await transport.produce({
- kind: "audio",
- rtpParameters: {
- mid: "audio",
- codecs: [{
- clockRate: 48000,
- payloadType: 111,
- mimeType: "audio/opus",
- channels: 2,
- }],
- headerExtensions: res.ext?.map(x => ({
- id: x.value,
- uri: x.uri,
- }))
- },
- paused: false,
- });
+ if (!test_hasMadeProducer) {
+ const producer = await transport.produce({
+ kind: "audio",
+ rtpParameters: {
+ mid: "audio",
+ codecs: [{
+ clockRate: 48000,
+ payloadType: 111,
+ mimeType: "audio/opus",
+ channels: 2,
+ }],
+ headerExtensions: res.ext?.map(x => ({
+ id: x.value,
+ uri: x.uri,
+ }))
+ },
+ paused: false,
+ });
+
+ const consumer = await transport.consume({
+ producerId: producer.id,
+ paused: false,
+ rtpCapabilities,
+ })
+
+ test_hasMadeProducer = true;
+ }
socket.send(JSON.stringify({
op: VoiceOPCodes.SESSION_DESCRIPTION,
|
