I'm using native c++ API to build client app to connect to AMS and stream (play) published audio and video content. I hit a snag where signaling completes successfully but then server responds with "play_finished" message. The logs on the server show the following:
2021-01-20 19:58:38,105 [vert.x-eventloop-thread-1] WARN i.a.enterprise.webrtc.WebRTCClient - It's still not streaming so stop the webrtc client for stream: raf and client:988791721
2021-01-20 19:58:38,106 [vert.x-eventloop-thread-0] INFO i.a.enterprise.webrtc.WebRTCClient - Stop is called for raf for 988791721
2021-01-20 19:58:38,106 [vert.x-eventloop-thread-0] INFO i.a.enterprise.webrtc.WebRTCClient - Disposing audio source for stream Id raf
What am I missing on the client side to cause this message?
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Rafal Kulczycki
I'm using native c++ API to build client app to connect to AMS and stream (play) published audio and video content. I hit a snag where signaling completes successfully but then server responds with "play_finished" message. The logs on the server show the following:
2021-01-20 19:58:38,105 [vert.x-eventloop-thread-1] WARN i.a.enterprise.webrtc.WebRTCClient - It's still not streaming so stop the webrtc client for stream: raf and client:988791721
2021-01-20 19:58:38,106 [vert.x-eventloop-thread-0] INFO i.a.enterprise.webrtc.WebRTCClient - Stop is called for raf for 988791721
2021-01-20 19:58:38,106 [vert.x-eventloop-thread-0] INFO i.a.enterprise.webrtc.WebRTCClient - Disposing audio source for stream Id raf
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