webrtcbin: Reported jitter is nonsense
The "jitter" reported for incoming streams is nonsense (infinite, smaller than a clock tick and/or negative).
It seems that the
clock-rate used for the
CLOCK_RATE_VALUE_TO_SECONDS division in
_get_stats_from_rtp_source_stats is bad (zero, or some very large positive or negative value). This can be confirmed by inserting some logging.
clock-rate obtained in
_get_codec_stats_from_pad can paper this over, but the question remains: Why do the RTP source stats not contain a valid clock rate in the first place?