<html><head></head><body><div class="ydpe7aa7548yahoo-style-wrap" style="font-family:Helvetica Neue, Helvetica, Arial, sans-serif;font-size:16px;"><div></div>
<div><br></div><div dir="ltr" data-setdir="false">Dear <span>Sergey Safarov,</span></div><div dir="ltr" data-setdir="false"><br></div><div dir="ltr" data-setdir="false">As I said earlier there are several scenario that I'm working on hoping one of them to work. When I use internal profile and I set bypass-media to "true", all clients residing on the same network can have voice and video call with or without ZRTP and adding or removing video to/from call works excellent.</div><div dir="ltr" data-setdir="false">But, when using internal profile if I try to call one of clients on another network such as mobile data, the call takes place but no audio is gransmitted(no ZRTP).</div><div dir="ltr" data-setdir="false">And if I use external profile and set bypass-media and inbound-zrtp-passthru to "true", all clients regardless of NAT, Wi-fi or mobile data can call each other with ZRTP but no video! when I press the video button on Linphone nothing happens! and of course I've added H26x to codecs.</div><div dir="ltr" data-setdir="false">I've also reviewed Jira topics and commits you mentioned but they are not related to my problem.</div><div dir="ltr" data-setdir="false"><br></div><div dir="ltr" data-setdir="false">Thanks again for helping.<br></div>
</div><div id="ydp1ab834ceyahoo_quoted_2367367715" class="ydp1ab834ceyahoo_quoted">
<div style="font-family:'Helvetica Neue', Helvetica, Arial, sans-serif;font-size:13px;color:#26282a;"><div><br></div><div><br></div><div><br></div><div><div id="ydp1ab834ceymsg27258" class="ydp1ab834ceymsg9506408637"><div id="ydp1ab834ceyiv7953633980"><div dir="ltr">Is without ZRTP video/addition removing works well?<div><br></div><div>Case 1</div><div>Without ZRTP similar case fixed about two years ago. Sorry I cannot find related ticked created by me (jira login <span style="color:rgb(23,43,77);font-family:-apple-system, BlinkMacSystemFont, Roboto, Oxygen, Ubuntu, sans-serif;font-size:14px;">safarov</span>).<br>Please retest master and try bisect with commits 1-3 years ago.<br>Think you will find working version and where bug is introduced.<br><br>Case 2<br>Please look </div><div><a href="https://freeswitch.org/jira/browse/FS-9203" rel="nofollow" target="_blank">https://freeswitch.org/jira/browse/FS-9203</a><br><a href="https://freeswitch.org/jira/browse/FS-9206" rel="nofollow" target="_blank">https://freeswitch.org/jira/browse/FS-9206</a><br></div><div><br>Here you will find commit where it worked properly.<br>Just need bisect with master and find commit introduced bug.</div><div><br></div><div>Case 3</div><div>Useless for you. This not designed for users behind NAT.<br>So simple do not use.</div><div><br></div><div>Sergey</div></div><br><div class="ydp1ab834ceyiv7953633980gmail_quote"><div dir="ltr" class="ydp1ab834ceyiv7953633980gmail_attr">On Fri, Feb 21, 2020 at 7:22 AM Faisal Hanif <<a href="mailto:imfanee@gmail.com" rel="nofollow" target="_blank">imfanee@gmail.com</a>> wrote:<br></div><blockquote class="ydp1ab834ceyiv7953633980gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex;"><div>In you are using linphone mobile clients make sure you have added "<a href="http://stun.linphone.org" rel="nofollow" target="_blank">stun.linphone.org</a>" in account specific stun settings,<div>If you need both ZRTP & NAT handling keep media in proxy mode.</div></div><br><div class="ydp1ab834ceyiv7953633980gmail_quote"><div dir="ltr" class="ydp1ab834ceyiv7953633980gmail_attr">On Fri, Feb 21, 2020, 6:29 AM I K via FreeSWITCH-users <<a href="mailto:freeswitch-users@lists.freeswitch.org" rel="nofollow" target="_blank">freeswitch-users@lists.freeswitch.org</a>> wrote:<br></div><blockquote class="ydp1ab834ceyiv7953633980gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex;"><br><br><br>---------- Forwarded message ----------<br>From: I K <<a href="mailto:iriik@yahoo.com" rel="nofollow" target="_blank">iriik@yahoo.com</a>><br>To: "<a href="mailto:freeswitch-users@lists.freeswitch.org" rel="nofollow" target="_blank">freeswitch-users@lists.freeswitch.org</a>" <<a href="mailto:freeswitch-users@lists.freeswitch.org" rel="nofollow" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Bcc: <br>Date: Sat, 15 Feb 2020 22:11:12 +0000 (UTC)<br>Subject: RTP not transmitting over NAT<br><div><div style="font-family:Helvetica, Arial, sans-serif;font-size:16px;"><div dir="ltr">Hi every one,</div><div dir="ltr"><br></div><div dir="ltr">I'm new to Freeswitch and I'm having some difficulty configuring it, despite reading the documents. I have a Freeswitch running on a server with public IP and several Linphone android clients running behind NAT.</div><div dir="ltr">How should I configure it to transfer audio and video and use ZRTP if one client is on a LAN behind NAT and the other client is using mobile data?!</div><div dir="ltr"><br></div><div dir="ltr">Things I've acheived so far:</div><div dir="ltr">1- bypass-media = false and proxy-media = false:</div><div dir="ltr">In this mode audio works but when I turn on video no request is sent to the other phone. It other use just has to know when to turn on the camera to receive video. and of course ZRTP does not work well.<br></div><div dir="ltr">2- <span> bypass-media = false and proxy-media = true:</span></div><div dir="ltr"><span>audio works but video shows blank</span></div><div dir="ltr"><span>3-<span> bypass-media = true and proxy-media = false:</span></span></div><div dir="ltr"><span><span>If both clients are on the same LAN everything works fine, but if one client connects via mobile data, they can still call each other but no media is transferred and the call is dropped after 30s.</span></span></div><div dir="ltr"><span><span><br></span></span></div><div dir="ltr"><span><span>I've been working on this scenario for over a week and I'm desperate to make it work. Thanks in advance for your help.</span></span></div><div dir="ltr"><span><span><br></span></span></div><div dir="ltr"><span><span></span></span><br></div></div></div><br><br><br>---------- Forwarded message ----------<br>From: I K via FreeSWITCH-users <<a href="mailto:freeswitch-users@lists.freeswitch.org" rel="nofollow" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>To: "<a href="mailto:freeswitch-users@lists.freeswitch.org" rel="nofollow" target="_blank">freeswitch-users@lists.freeswitch.org</a>" <<a href="mailto:freeswitch-users@lists.freeswitch.org" rel="nofollow" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Bcc: <br>Date: Thu, 20 Feb 2020 17:29:02 -0800 (PST)<br>Subject: [Freeswitch-users] RTP not transmitting over NAT<br>_________________________________________________________________________<br>
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<a href="https://freeswitch.com" rel="nofollow" target="_blank">https://freeswitch.com</a></blockquote></div></div></div><div id="ydp1ab834ceymsg94717" class="ydp1ab834ceymsg9506408637"><div id="ydp1ab834ceyiv7953633980"><div><div>If your fs is behind nat, you should set sip-ext-ip and rtp-ext-ip.</div><div><br></div><div>This is done so that when fs sends an offer out it puts the sip-ext-ip and rtp ip in that offer.</div></div><div><br></div><div><div>On the client side set the stun</div><br><div class="ydp1ab834ceyiv7953633980gmail_quote"><div dir="ltr" class="ydp1ab834ceyiv7953633980gmail_attr">On Fri, 21 Feb 2020 at 04:38, Faisal Hanif <<a href="mailto:imfanee@gmail.com" rel="nofollow" target="_blank">imfanee@gmail.com</a>> wrote:<br></div><blockquote class="ydp1ab834ceyiv7953633980gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;padding-left:1ex;border-left-color:rgb(204,204,204);"><div>In you are using linphone mobile clients make sure you have added "<a href="http://stun.linphone.org" rel="nofollow" target="_blank">stun.linphone.org</a>" in account specific stun settings,<div>If you need both ZRTP & NAT handling keep media in proxy mode.</div></div><br><div class="ydp1ab834ceyiv7953633980gmail_quote"><div dir="ltr" class="ydp1ab834ceyiv7953633980gmail_attr">On Fri, Feb 21, 2020, 6:29 AM I K via FreeSWITCH-users <<a href="mailto:freeswitch-users@lists.freeswitch.org" rel="nofollow" target="_blank">freeswitch-users@lists.freeswitch.org</a>> wrote:<br></div><blockquote class="ydp1ab834ceyiv7953633980gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;padding-left:1ex;border-left-color:rgb(204,204,204);"><br><br><br>---------- Forwarded message ----------<br>From: I K <<a href="mailto:iriik@yahoo.com" rel="nofollow" target="_blank">iriik@yahoo.com</a>><br>To: "<a href="mailto:freeswitch-users@lists.freeswitch.org" rel="nofollow" target="_blank">freeswitch-users@lists.freeswitch.org</a>" <<a href="mailto:freeswitch-users@lists.freeswitch.org" rel="nofollow" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Bcc: <br>Date: Sat, 15 Feb 2020 22:11:12 +0000 (UTC)<br>Subject: RTP not transmitting over NAT<br><div><div style="font-family:Helvetica, Arial, sans-serif;font-size:16px;"><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;">Hi every one,</div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;"><br></div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;">I'm new to Freeswitch and I'm having some difficulty configuring it, despite reading the documents. I have a Freeswitch running on a server with public IP and several Linphone android clients running behind NAT.</div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;">How should I configure it to transfer audio and video and use ZRTP if one client is on a LAN behind NAT and the other client is using mobile data?!</div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;"><br></div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;">Things I've acheived so far:</div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;">1- bypass-media = false and proxy-media = false:</div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;">In this mode audio works but when I turn on video no request is sent to the other phone. It other use just has to know when to turn on the camera to receive video. and of course ZRTP does not work well.<br></div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;">2- <span style="font-family:Helvetica, Arial, sans-serif;"> bypass-media = false and proxy-media = true:</span></div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;"><span style="font-family:Helvetica, Arial, sans-serif;">audio works but video shows blank</span></div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;"><span style="font-family:Helvetica, Arial, sans-serif;">3-<span style="font-family:Helvetica, Arial, sans-serif;"> bypass-media = true and proxy-media = false:</span></span></div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;"><span style="font-family:Helvetica, Arial, sans-serif;"><span style="font-family:Helvetica, Arial, sans-serif;">If both clients are on the same LAN everything works fine, but if one client connects via mobile data, they can still call each other but no media is transferred and the call is dropped after 30s.</span></span></div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;"><span style="font-family:Helvetica, Arial, sans-serif;"><span style="font-family:Helvetica, Arial, sans-serif;"><br></span></span></div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;"><span style="font-family:Helvetica, Arial, sans-serif;"><span style="font-family:Helvetica, Arial, sans-serif;">I've been working on this scenario for over a week and I'm desperate to make it work. Thanks in advance for your help.</span></span></div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;"><span style="font-family:Helvetica, Arial, sans-serif;"><span style="font-family:Helvetica, Arial, sans-serif;"><br></span></span></div><div dir="ltr" style="font-family:Helvetica, Arial, sans-serif;"><span style="font-family:Helvetica, Arial, sans-serif;"><span style="font-family:Helvetica, Arial, sans-serif;"></span></span><br></div></div></div><br><br><br>---------- Forwarded message ----------<br>From: I K via FreeSWITCH-users <<a href="mailto:freeswitch-users@lists.freeswitch.org" rel="nofollow" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>To: "<a href="mailto:freeswitch-users@lists.freeswitch.org" rel="nofollow" target="_blank">freeswitch-users@lists.freeswitch.org</a>" <<a href="mailto:freeswitch-users@lists.freeswitch.org" rel="nofollow" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Bcc: <br>Date: Thu, 20 Feb 2020 17:29:02 -0800 (PST)<br>Subject: [Freeswitch-users] RTP not transmitting over NAT<br>_________________________________________________________________________<br>
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<a href="https://freeswitch.com" rel="nofollow" target="_blank">https://freeswitch.com</a></blockquote></div></div>-- <br><div dir="ltr" class="ydp1ab834ceyiv7953633980gmail_signature"><div dir="ltr"><div>Regards,</div><div><br></div>David Villasmil<div>email: <a href="mailto:david.villasmil.work@gmail.com" rel="nofollow" target="_blank">david.villasmil.work@gmail.com</a></div><div>phone: +34669448337</div></div></div></div></div>_________________________________________________________________________<br><br>The FreeSWITCH project is sponsored by SignalWire <a href="https://signalwire.com" rel="nofollow" target="_blank">https://signalwire.com</a><br>Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.<br>Build your next product on our scalable cloud platform.<br><br>Join our online community to chat in real time <a href="https://signalwire.community" rel="nofollow" target="_blank">https://signalwire.community</a><br><br>Professional FreeSWITCH Services<br><a href="mailto:sales@freeswitch.com" rel="nofollow" target="_blank">sales@freeswitch.com</a><br><a href="https://freeswitch.com" rel="nofollow" target="_blank">https://freeswitch.com</a><br><br>Official FreeSWITCH Sites<br><a href="https://freeswitch.com/oss" rel="nofollow" target="_blank">https://freeswitch.com/oss</a><br><a href="https://freeswitch.org/confluence" rel="nofollow" target="_blank">https://freeswitch.org/confluence</a><br><a href="https://cluecon.com" rel="nofollow" target="_blank">https://cluecon.com</a><br><br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org" rel="nofollow" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" rel="nofollow" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" rel="nofollow" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="https://freeswitch.com" rel="nofollow" target="_blank">https://freeswitch.com</a></div>
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