[Freeswitch-users] RTP Timestamp changes after REFER - Question on RFC conformity

[Christian Benke]

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