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

Anthony Minessale anthony.minessale at gmail.com
Thu Aug 23 19:18:21 MSD 2012

Do you have a pcap of it not setting the mark bit when the timestamp
changes because the code will always set the mark bit when the packet
is about to send is not exactly the next packet it expected to send
after the previous one/

On Thu, Aug 23, 2012 at 10:04 AM, Christian Benke <benkokakao at gmail.com> wrote:
> 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
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Anthony Minessale II

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