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webrtc: improve matching on the correct jitterbuffer

Matthew Waters requested to merge ystreet/gst-plugins-bad:webrtc-data-channel-mline0-bundle into master

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.

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