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Transport used for audio/video


Hari D
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Newbie here:

These questions are in the context of real time video transport with very low latency

1) What is the transport used to send video/audio from client to AntMedia Server? Is it Websocket or UDP. Websocket is TCP based so I think it is not a good choice if the latency has to be the lowest. 

2) Does Antmedia provide webrtc  C or C++  language SDK that can run on Linux/Ubuntu? 

Thanks

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Websocket is used for communication, and if you are using WebRTC for streaming, it is UDP.

If you are using HLS(HTTP Live Streaming) to watch the stream, it is of course uses TCP and latency is higher due to buffering but again if you use WebRTC to watch the stream, it is UDP again.

I don't think we have C or C++ based SDK's. I'll make this clear when i get a definitive answer.

Best Regards,
Enes.

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