Webrtcsink latency

That’s expected. You would add a queue of suitable size to allow for compensating the latency difference between the different streams (e.g. if your audio stream has 500ms latency and the video stream 200ms, then the video stream needs an additional 300ms of buffering at least). This will not increase latency. See this explanation about the min/max latency and generally how latency works in GStreamer.

That means that at this point at least latency configuration was not successful (the LATENCY event was rejected by upstream). The debug logs will contain more information about what went wrong there.

Most likely you’ll get a LATENCY message on the bus at a later time, and then via gst_bin_recalculate_latency() it should end up with a valid latency configuration. Or possibly even without you having to do anything at all.