Jump to content
  • 0

Transport used for audio/video

Hari D


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? 


Link to comment
Share on other sites

  • Answers 1
  • Created
  • Last Reply

Top Posters For This Question

Popular Days

Top Posters For This Question

1 answer to this question

Recommended Posts

  • 0

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,

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.

Answer this question...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.


  • Create New...