What is the standard approach to retrieve rtp packet loss and jitter metrics for calls routed through our singapore byoc trunks using the webrtc softphone sdk v3.2.1? the standard /api/v2/analytics/interactions/query endpoint returns basic call duration and disposition data, but the rtp_stats object is consistently null for these specific trunk associations. we have verified that the carrier is sending rtcp packets correctly via pcap analysis on the sbc. the issue seems isolated to the reporting pipeline when the webrtc leg is involved. we are using the javascript sdk for the softphone implementation, and the connection is stable with no ice failures. have others found a workaround to get granular qos metrics for byoc webrtc sessions? the current lack of rtp data makes it difficult to troubleshoot audio quality issues reported by agents in the asia pacific region. we need to correlate the softphone client logs with the server-side analytics, but the gap in the rtp_stats field prevents this. any insights on whether this is a known limitation or if there is an alternative endpoint we should be polling?