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Copy file name to clipboardExpand all lines: guides/for_developers/fd_sdp.md
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# WebRTC SDP
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WebRTC uses SDP offer/answer to negotiate session parameters (numer of audio/video tracks, their directions, codecs, etc.).
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WebRTC uses SDP offer/answer to negotiate session parameters (number of audio/video tracks, their directions, codecs, etc.).
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The way they are exchanged between both sides is not standardized.
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Very often it is a websocket.
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WebRTC was standardized by (among others) Google, Cisco and Mozilla.
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* when transceiver is stopped, port number in mline is set to 0
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* port number in mline set to 9 means that connection address will be set dynamically via ICE
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* SDP can include ICE candidates but it doesn't have to.
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In particular, when you create the first offer it won't have any ICE candidates, but if you wait a couple of seconds and read peerconnection.localDescription it will contain ICE candidates that were gatherd throughout this time.
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In particular, when you create the first offer it won't have any ICE candidates, but if you wait a couple of seconds and read peerconnection.localDescription it will contain ICE candidates that were gathered throughout this time.
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* offerer can offer to both send and receive
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* mline includes a list of supported codecs.
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They are sorted in preference order
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Client sends SDP offer including both its own tracks and server's tracks.
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This requires a single negotiation.
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* client sends SDP offer only including its own tracks.
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After this is negotiated succsessfully, server sends its SDP offer.
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This requries two negotiations.
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After this is negotiated successfully, server sends its SDP offer.
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This requires two negotiations.
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* we use two separate peer connections, one for sending and one for receiving.
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This way client and server can send their offers in parallel.
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