RTP Timestamp changes after REFER - Question on RFC conformity
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
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
The problem here is you can easily blur the lines on what is correct or incorrect and a possible fix might even break other things. http://sidious.freeswitch.org/patches/rtp_reset.diff Try this patch and see if its any better. It will send a mark and new SSRC every time a channel is in a new bridge situation. On Thu, Aug 23, 2012 at 10:31 AM, Christian Benke <benkokakao@gmail.com> wrote:
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
participants (2)
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Anthony Minessale
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Christian Benke