Hi! I had a problem with VOP Softclient(www.voiceoperatorpanel.com) where the callee had one-way-audio after a call was REFERred to him. After lots of tracing, debugging and hairpulling i realized the problem lies in the changing rtp-timestamp when the RTP-stream is switched from the middleman to the initial call after the REFER. The Softclient was not able to cope with the changed timestamp and ignored the incoming RTP-packets, leading to no audio for the callee. I was eventually able to solve the problem by activating rtp-rewrite-timestamps on the profile(Also added it to http://wiki.freeswitch.org/wiki/RTP_Issues#Voiceoperatorpanel_VOP). However, i would like to know if FreeSWITCH/Sofia is working according to the RFC and if the Softclient is to blame for the problem(So i can file a bugreport with them). In this thread, Brian West states that it's ok to skip forward in timestamps as long as the marker-bit is set: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-July/060333.html However, the Marker-bit is not set by FreeSWITCH when the REFER occurs. I didn't find this stated in http://tools.ietf.org/html/rfc3550(But "timestamp" is mentioned a lot, so i may have missed it) but there's this bug-report for Asterisk, where the exact same problem is described and eventually handled: https://issues.asterisk.org/view.php?id=17007 Could someone with more insight please elaborate? Best regards, Christian