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<p class="MsoNormal"><span style="color:#1F497D">Hello again,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">The following log message coincides with the initiation of the issue:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">2017-02-13 12:19:52.549786 [INFO] switch_rtp.c:5867 Auto Changing port from 172.17.85.240:58638 to 172.17.85.240:58632<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Version shows: FreeSWITCH Version 1.4.26-37~64bit (-37 64bit)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Tried adding in the default dialplan, under the inbound extension, just before the bridge command:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><action application="set" data="disable_rtp_auto_adjust=true"/><o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/><o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Where is the correct place to add them?<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Are these parameters valid for my exact version?<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">How can I verify if they become active? Adding a line like the one below shows blank.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><action application="log" data="WARNING ~ Inbound disable-rtp-auto-adjust value:${disable-rtp-auto-adjust}"/><o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Thank you<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]
<b>On Behalf Of </b>Bejan, Alex<br>
<b>Sent:</b> Monday, January 30, 2017 4:12 PM<br>
<b>To:</b> freeswitch-users@lists.freeswitch.org<br>
<b>Subject:</b> Re: [Freeswitch-users] one way speech path after attempting voice path redirecting in new invite<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Hello,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">This is my first posting into this list although I monitored activities in the past.<o:p></o:p></p>
<p class="MsoNormal">Freeswitch behaves great in our environment – thank you all who contribute!<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">We started recently observing some one way speech path issues in the following scenario and I would like to ask for suggestions, please:<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">We have incoming calls from provider to FS that further bridge to different Standard SIP PBX.<o:p></o:p></p>
<p class="MsoNormal">Call is established to SIP device connected on the PBX.<o:p></o:p></p>
<p class="MsoNormal">The PBX immediately invites back attempting to redirect rtp path to start recording the call.<o:p></o:p></p>
<p class="MsoNormal">FS starts sending RTP to the new destination.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">There are 2 different very similar PBX environments connected to the same FS:
<o:p></o:p></p>
<p class="MsoListParagraph" style="text-indent:-.25in">-<span style="font-size:7.0pt;font-family:"Times New Roman","serif"">
</span>In one the PBX starts sending RTP even before the connection is ACKed. Recording succeeds.<o:p></o:p></p>
<p class="MsoListParagraph" style="text-indent:-.25in">-<span style="font-size:7.0pt;font-family:"Times New Roman","serif"">
</span>In the second one, PBX starts sending RTP almost one second later. It seems FS reverts back to the original speech path, while the other devices remain unaware, resulting in one way speech path.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">The rtp-timeout-sec <span style="color:#1F497D">and </span>rtp-hold-timeout-sec are on the default values
<o:p></o:p></p>
<p class="MsoNormal"> <!-- rtp inactivity timeout --><o:p></o:p></p>
<p class="MsoNormal"> <param name="rtp-timeout-sec" value="300"/><o:p></o:p></p>
<p class="MsoNormal"> <param name="rtp-hold-timeout-sec" value="1800"/><o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Are there any other parameters that could have such an effect?<o:p></o:p></p>
<p class="MsoNormal">Your suggestions on further troubleshooting or how to make this work even when RTP is not being received will be greatly appreciated!<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thank you very much!<o:p></o:p></p>
<p class="MsoNormal">Regards<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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