Mediasoup stopped working on Production

mediasoup libraries
1

Hi,

We have been using Mediasoup for nearly 3+ years, recently we upgraded our servers, and its been working fine for the last 6 Months, but today, we got an error, it seems to be some issue while creating a transport. Please refer to this image

image
image1920×535 90.7 KB

Please help with creating transport, nothing has been changed in terms of code, we tried to restart the server and deployment on a different machine altogether, but nothing seems to work.

Please help

2

Probably the Router on which you are creating the WebRtcTransport is already closed. Nothing relevant has changed in this regards in last months/years so the fact that you are experimenting this error once in 6 months doesn’t mean there is a bug in mediasoup. Best thing you can do is check your logic.

3

We have been using the platform on daily basis and we have configured the a version of mediasoup for the deployment, same error with that is coming but it’s working fine on local

 case "createWebRtcTransport": {
        // NOTE: Don't require that the Peer is joined here, so the client can
        // initiate mediasoup Transports and be ready when he later joins.

        const { forceTcp, producing, consuming, sctpCapabilities } =
          request.data;

        const webRtcTransportOptions = {
          ...config.mediasoup.webRtcTransportOptions,
          enableSctp: Boolean(sctpCapabilities),
          numSctpStreams: (sctpCapabilities || {}).numStreams,
          appData: { producing, consuming },
        };

        if (forceTcp) {
          webRtcTransportOptions.enableUdp = false;
          webRtcTransportOptions.enableTcp = true;
        }
        console.log("CREATEING WEB RTC TRANSPORT");
       
        try{  
          
          const transport = await this._mediasoupRouter.createWebRtcTransport({
            ...webRtcTransportOptions,
            webRtcServer: this._webRtcServer,
          });
          

          console.log("CREATED WEB RTC TRansport")
  
          transport.on("sctpstatechange", (sctpState) => {
            logger.debug(
              'WebRtcTransport "sctpstatechange" event [sctpState:%s]',
              sctpState
            );
          });
  
          transport.on("dtlsstatechange", (dtlsState) => {
            if (dtlsState === "failed" || dtlsState === "closed")
              logger.warn(
                'WebRtcTransport "dtlsstatechange" event [dtlsState:%s]',
                dtlsState
              );
          });
  
          // NOTE: For testing.
          // await transport.enableTraceEvent([ 'probation', 'bwe' ]);
          await transport.enableTraceEvent(["bwe"]);
  
          transport.on("trace", (trace) => {
            logger.debug(
              'transport "trace" event [transportId:%s, trace.type:%s, trace:%o]',
              transport.id,
              trace.type,
              trace
            );
  
            if (trace.type === "bwe" && trace.direction === "out") {
              peer
                .notify("downlinkBwe", {
                  desiredBitrate: trace.info.desiredBitrate,
                  effectiveDesiredBitrate: trace.info.effectiveDesiredBitrate,
                  availableBitrate: trace.info.availableBitrate,
                })
                .catch(() => { });
            }
          });
  
          // Store the WebRtcTransport into the protoo Peer data Object.
          peer.data.transports.set(transport.id, transport);
  
          accept({
            id: transport.id,
            iceParameters: transport.iceParameters,
            iceCandidates: transport.iceCandidates,
            dtlsParameters: transport.dtlsParameters,
            sctpParameters: transport.sctpParameters,
          });
        }catch(e){
          console.log("ERROR while create TRANSPORT");
          console.log('----------')
          console.log(e);
          console.log('----------')
        }

        const { maxIncomingBitrate } = config.mediasoup.webRtcTransportOptions;

        // If set, apply max incoming bitrate limit.
        if (maxIncomingBitrate) {
          try {
            await transport.setMaxIncomingBitrate(maxIncomingBitrate);
          } catch (error) { }
        }

        break;
      }

After this CREATED WEB RTC TRansport, catch block is console gets printed