Hi all, i am struggling with debugging a one-way audio situation. I analyzed everything related to SIP and SDP attributes and everything seems fine. I look at PJSIP logs in realtime during debugging. Particulary i am interested in: - see every RTP packets sent out - see to which address/port the RTP packets are sent out Is there a way to check RTP related information by enabling some macro and/or define? I am already running with PJSIP at maximum debug level. Fabio
