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webrtc: add a few comments on bundle and src pad exposure
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1 changed files with 18 additions and 1 deletions
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@ -62,7 +62,21 @@
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*
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*
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* On the receiving side, RTPTransceiver's are created in response to setting
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* On the receiving side, RTPTransceiver's are created in response to setting
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* a remote description. Output pads for the receiving streams in the set
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* a remote description. Output pads for the receiving streams in the set
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* description are also created.
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* description are also created when data is received.
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*
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* A TransportStream is created when needed in order to transport the data over
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* the necessary DTLS/ICE channel to the peer. The exact configuration depends
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* on the negotiated SDP's between the peers based on the bundle and rtcp
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* configuration. Some cases are outlined below for a simple single
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* audio/video/data session:
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*
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* - max-bundle (requires rtcp-muxing) uses a single transport for all
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* media/data transported. Renegotiation involves adding/removing the
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* necessary streams to the existing transports.
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* - max-compat without rtcp-mux involves two TransportStream per media stream
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* to transport the rtp and the rtcp packets and a single TransportStream for
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* all data channels. Each stream change involves modifying the associated
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* TransportStream/s as necessary.
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*/
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*/
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/*
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/*
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@ -2908,6 +2922,9 @@ _connect_output_stream (GstWebRTCBin * webrtc,
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g_free (pad_name);
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g_free (pad_name);
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gst_element_sync_state_with_parent (GST_ELEMENT (stream->receive_bin));
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gst_element_sync_state_with_parent (GST_ELEMENT (stream->receive_bin));
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/* The webrtcbin src_%u output pads will be created when rtpbin receives
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* data on that stream in on_rtpbin_pad_added() */
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}
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}
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typedef struct
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typedef struct
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