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<p class="MsoNormal"><span lang="EN-US">Hello,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">I am trying to validate FS as a SIP to WebRTC Gateway in our lab environment.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">I started in middle October and have good result on it, but i cannot understand this One Way Audio trouble.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">I must thank the Freeswitch team and contributors for this very impressive work.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">FreeSWITCH Version 1.8.1-2-4f54cff36a~64bit (-2-4f54cff36a 64bit)<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">On system: SMP Debian 4.9.110-3+deb9u6 (2018-10-08) x86_64 GNU/Linux<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Openssl version : OpenSSL 1.1.0f 25 May 2017<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Chrome version: 69.0 (I tried with different version and with Firefox with the same trouble).</span><span lang="EN-US" style="font-size:10.0pt;font-family:"Segoe UI",sans-serif;color:#5F6368">
</span><span lang="EN-US"><o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">The wss part is ok with sip.js and verto.js<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">The Ice negotiation is ok, I use sometimes local networks and sometimes web, I have had to authorize networks in the candidate ACL and domain ACL (acl.conf.xml) The result is the same on both topology.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">DTLS negotiation is OK, and there is UDP streams between Chrome (or Firefox) and FS in both ways.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">There is no “audible” audio in the direction from FS to Chrome, the other direction is OK.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">The simplest test is to call the 5000 number from the Chrome client, I send you a paste bin and pcap trace for this call.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Should you give me information element to progress on this, what is really mandatory in the sip_profile/internal.xml and external.xml files, and in directory/default/1000.xml for a WebRTC call ?? What should be the good
options in fs_cli to see if the encryption of RTP packets is ok or not.. ?<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Do you think that I have to reinstall a Freeswitch from the current branch ?<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">@IP for FS : 192.168.145.67<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">@IP for Chrome : 10.70.54.43<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Link on the pastebin : </span><a href="https://pastebin.freeswitch.org/view/09a72087"><span lang="EN-US" style="font-size:10.0pt;font-family:"Helvetica",sans-serif;color:#005580;background:white">https://pastebin.freeswitch.org/view/09a72087</span></a><span lang="EN-US"><o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">I have a pcap on the same call that I can provide (3 Mb) if necessary..<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Thank you for helping me !!<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Best regards.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="color:#1F497D;mso-fareast-language:FR"><o:p> </o:p></span></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><a href="http://www.prosodie.com/"><span style="font-family:"Verdana",sans-serif;color:#1F497D;mso-fareast-language:FR;text-decoration:none"><img border="0" width="110" height="48" style="width:1.1458in;height:.5in" id="Image_x0020_1" src="cid:image001.gif@01D4752A.DFA892B0" alt="Prosodie-signature"></span></a><span style="font-family:"Verdana",sans-serif;color:#1F497D;mso-fareast-language:FR"><o:p></o:p></span></p>
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<p class="MsoNormal"><b><span lang="EN-US" style="font-size:9.0pt;font-family:"Verdana",sans-serif;color:#00468C;mso-fareast-language:FR">Rémi Marand – Product Owner – Pod Connect.<br>
</span></b><span style="font-size:9.0pt;font-family:"Verdana",sans-serif;color:#00468C;mso-fareast-language:FR">PROSODIE – Marketing & Produit<br>
Tél. : +33 (0)1.46.84.12.77 / 06.87.72.53.25<br>
</span><span style="font-size:9.0pt;font-family:"Verdana",sans-serif;color:#1F497D;mso-fareast-language:FR"><a href="mailto:rmarand@prosodie.com">rmarand@prosodie.com</a><o:p></o:p></span></p>
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