The Real Time Control Protocol (RTCP) is used by Real Time Protocol (RTP) to report the performance of a media stream. RTCP utilizes Sender Report (SR) and Receiver Report (RR) messages to this end. Senders use the RTCP SRs to assist in synchronizing the media streams, while RTCP RRs are used by the Receivers to inform about current loss fraction, jitter, sequence numbers, and to the round-trip time (RTT).
In a bridge architecture scenario, RTCP RRs must be enabled in the bridge to be able to visualize RTT values on the callstats.io dashboard. There are some bridges which might not have this setting enabled by default, so please be sure to verify this (see an example for Asterisk here).
On the other hand, codecs can also play an important role in RTT reporting. In the case of FreeSwitch, OPUS codec has to be enabled in order to get RTT information on the dashboard. This is because OPUS enables RTCP since it needs that information to tune itself.
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