Hi! Here's a trace: http://poab.org/misc/one_way_audio_internal_g711.pcap (or http://filedump.org/files/yMnIRQUC1345735448.html for the archive) The REFER happens at Paket 1305, Time 25.662729 The RTP-Paket that are sent to the C-endpoint after the REFER are continuous from A's point of view(As it is the continuation of the A-B-stream), but not from C's point of view. Regards, Christian On 23 August 2012 17:18, Anthony Minessale <anthony.minessale@gmail.com> wrote:
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@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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