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Live wav transmissions and fetching with RTSP receives by delay

Hello everyone,

I’m currently have the use case of having a .wav file, which is getting big by second by some application X, doesn’t really matter.

What I’m doing is using GStreamer to stream that wav file, using a RTSP pipeline so the wowza simply receives that file through RTSP while being encoded to AAC.

After initiating the streaming session to Wowza, I wish to start a new GStreamer session which fetches the same RTSP from Wowza.

When I’m doing this scenario, I’m receiving a 7 seconds delay, which means when I’m saying in the wav file recording source “1, 2, 3, 4, 5, 6, 7…” I’m hearing them with a 7 seconds delay.

Now the funny part… when I’m initiating the same session with a static file, which is not getting bigger by second and it’s whole when I’m starting the sessions, I don’t have delay of any some sory, and I’m hearing the audio at the same time.

I thought of searching about special flags that would help me out, with checking wowza in this manner, but I’m not an expert regarding Wowza, so I do will appreciate any help provided :slight_smile:

Thank you very much heads up!

It sounds like GStreamer is taking longer to encode an incomplete file. Wowza will only see the incoming live RTSP stream. I don’t anticipate there is a setting to optimize this.