Don't use connect(), since that is incompatible with multicast.
Instead, drop received packets that are from senders we do not want.
Also set multicast loopback = false so we don't receive RTCP RRs from
ourselves and interpret them as RTCP SRs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
If we only send a single Transport in the Transports header, then the
server is allowed to omit it in the response. This has some strange
consequences for UDP transport: specifically, we have no idea what
addr/port we will get the packets from.
In those cases, we connect() on the socket when we receive the first
packet, so we can send RTCP RRs, and also so we can ensure that we
ignore data from other addresses.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
GST_PLUGIN_FEATURE_RANK=rtspsrc2:1 gst-play-1.0 [URI]
Features:
* Live streaming N audio and N video
- With RTCP-based A/V sync
* Lower transports: TCP, UDP, UDP-Multicast
* RTP, RTCP SR, RTCP RR
* OPTIONS DESCRIBE SETUP PLAY TEARDOWN
* Custom UDP socket management, does not use udpsrc/udpsink
* Supports both rtpbin and the rtpbin2 rust rewrite
- Set USE_RTPBIN2=1 to use rtpbin2 (needs other MRs)
* Properties:
- protocols selection and priority (NEW!)
- location supports rtsp[ut]://
- port-start instead of port-range
Co-Authored-by: Tim-Philipp Müller <tim@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
