webrtc: improve matching on the correct jitterbuffer

webrtc: improve matching on the correct jitterbuffer

The mapping between an RTP session and the SDP m= line is not always the
same, especially when BUNDLEing is used.

This causes a failure in a specific case where if when bundling,
if mline 0 is a data channel, and mline 1 an audio/video section,
then retrieving the transceiver at mline 0 (rtp session used) will fail
and cause an assertion.

This fix is actually potentially a regression for cases where the remote
part does not provide the a=ssrc: media level SDP attributes as is now
becoming common, especially when simulcast is involved.

The correct fix actually requires reading out header extensions as used
with bundle for signalling in the actual data, what media and therefore
transceiver is being used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2467>