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<p class="MsoNormal">Hello,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Sorry, i come back with this case.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I understand that the browsers behaviour is to use DTLS… but in my case it is not the case.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I have to set this constraints in my client javascript, and i do not find how to code it with sip.js….<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Should you please give me a way to set correctly that on the Freeswitch side, a configuration option to specify « WebRTC Media » on connection ??<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<pre><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">I tried with </span><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"><action application="set" data="sip_secure_media=false"/> in the dialplan/default.xml file, but it does not change, i tried before with </span><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#24292E;background:white">{media_webrtc=true} in the extension file without success.<o:p></o:p></span></pre>
<pre><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#24292E;background:white"><o:p> </o:p></span></pre>
<pre><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#24292E;background:white">Thanks for help.<o:p></o:p></span></pre>
<pre><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#24292E;background:white"><o:p> </o:p></span></pre>
<pre><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#24292E;background:white">Rémi Marand.<o:p></o:p></span></pre>
<pre><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#24292E;background:white">Capgemini/Prosodie.<o:p></o:p></span></pre>
<pre><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"><o:p> </o:p></span></pre>
<pre><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"><o:p> </o:p></span></pre>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><b><span style="mso-fareast-language:FR">De :</span></b><span style="mso-fareast-language:FR"> FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org>
<b>De la part de</b> Jeremy Lainé<br>
<b>Envoyé :</b> lundi 19 novembre 2018 09:18<br>
<b>À :</b> freeswitch-users@lists.freeswitch.org<br>
<b>Objet :</b> Re: [Freeswitch-users] One way audio case between FS and Chrome on WebRTC<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p>That is very odd, the default behaviour in browsers *is* to use DTLS for SRTP keying instead of the legacy SDES. In fact SDES's status is "MUST NOT implement" since 2013, and the DtlsSrtpKeyAgreement constraint is scheduled for retirement:<span style="mso-fareast-language:FR"><o:p></o:p></span></p>
<p><a href="https://bugs.chromium.org/p/chromium/issues/detail?id=804275">https://bugs.chromium.org/p/chromium/issues/detail?id=804275</a><o:p></o:p></p>
<p>Concerning SIP.js I have never experienced the need to set this constraint, it works fine against freeswitch without it.<o:p></o:p></p>
<p>Jeremy<o:p></o:p></p>
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<p class="MsoNormal">On 11/13/18 8:55 AM, MARAND, Remi wrote:<o:p></o:p></p>
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<p class="MsoNormal">Hello,<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">I finally was able to solve my problem.<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">For information, on this trouble, the parameter :<o:p></o:p></p>
<p class="MsoNormal"><i><span style="font-size:10.0pt">RTCConstraints: {"optional": [{'DtlsSrtpKeyAgreement': 'true'}]}</span></i><o:p></o:p></p>
<p class="MsoNormal">Is to add at the good place in jquery.FSRTC.js (or verto-min.js (I think that's already the case in this script)) i did not find how to code in the sip.js version but it should be possible.<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Thanks to those who answered my question, and sorry for the 3 Mb of pcap file i sent to the user-list !!!<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Perhaps this DtlsSrtpKeyAgreement parameter role should be added and explain in the Verto/WebRTC examples availables on Websites, i suppose that in 2014, it was not mandatory but now with the lasts versions of Chrome and FF it seems to
be.<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Regards.<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Remi Marand.<o:p></o:p></p>
<p class="MsoNormal"><a href="mailto:rmarand@prosodie.com">rmarand@prosodie.com</a><o:p></o:p></p>
<p class="MsoNormal">+33687725325.<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
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<p class="MsoNormal"><b><span style="mso-fareast-language:FR">De :</span></b><span style="mso-fareast-language:FR"> MARAND, Remi
<br>
<b>Envoyé :</b> lundi 5 novembre 2018 18:01<br>
<b>À :</b> <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
<b>Objet :</b> One way audio case between FS and Chrome on WebRTC</span><o:p></o:p></p>
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<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US">Hello,</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US">I am trying to validate FS as a SIP to WebRTC Gateway in our lab environment.</span><o:p></o:p></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.</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US">I must thank the Freeswitch team and contributors for this very impressive work.</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US">FreeSWITCH Version 1.8.1-2-4f54cff36a~64bit (-2-4f54cff36a 64bit)</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US">On system: SMP Debian 4.9.110-3+deb9u6 (2018-10-08) x86_64 GNU/Linux</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US">Openssl version : OpenSSL 1.1.0f 25 May 2017</span><o:p></o:p></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">
</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US">The wss part is ok with sip.js and verto.js</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></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.</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></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.</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></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.</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></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.</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></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.. ?</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US">Do you think that I have to reinstall a Freeswitch from the current branch ?</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US">@IP for FS : 192.168.145.67</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US">@IP for Chrome : 10.70.54.43</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></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><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US">I have a pcap on the same call that I can provide (3 Mb) if necessary..</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US">Thank you for helping me !!</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US"> </span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US">Best regards.</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-US" style="color:#1F497D;mso-fareast-language:FR"> </span><o:p></o:p></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:1.15in;height:.5in" id="Image_x0020_1" src="cid:image001.gif@01D49249.6C7398D0" alt="Prosodie-signature"></span></a><o:p></o:p></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></span><o:p></o:p></p>
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