RTCP improvements

[gavv]

Follow-up for #14 and #675.

• Refine packet counter calculation. On receiver, derive it from packet_count in SR, and handle 32-bit wraps. On sender, derive it from ext_first_sn and ext_last_sn, and also handle 32-bit wraps. (add rtcp::PacketCounter)
• Add fract_loss calculation - loss ratio since last report. (add rtcp::LossEstimator)
• Use receive timestamp (RTS) as report time when processing RTCP report.
• Support multiple sources on sender (for cases like RaptorQ).
• Implement rtcp::IntervalComputer (computes packet generation interval according to algorithm from RFC 3550) and use in rtcp::Communicator instead of hard-coded 200ms report interval.
• Add sliding window to rtcp::RttEstimator. Compute moving minimum RTT, and average clock_shift based on it. Adjust communicator RTT tests to ensure that sliding window improves RTT & clock offset estimation precision.
• Implement RTCP BYE exchange. Handle exit by signal and ask pipeline to generate BYE. Processing of BYE is already implemented.
• Fix RTCP multicast support. If receiver RTCP endpoint is bound to multicast address, it should use Report_ToAddress instead of Report_Back mode.
• Always collect metrics for all receivers in FeedbackMonitor. If single-receiver mode is enabled, additionally create latency tuner, otherwise just collect metrics. Needed for Metrics API #681.
• Implement local SSRC change in FeedbackMonitor when latency goes out of bounds (to initiate session restart on remote side).
• Add tests with real captured packets.
• Schedule pipeline refresh() according to nearest deadline returned from RTCP.
• Handle system clock jumps in pipeline (since it breaks RTCP).