<div dir="ltr">Hi Giovanni,<div><br></div><div>I'm not sure now since call set up works in cantina.</div><div>I'm using standard FS setup with 7443 port for wss and simple (50 lines) of sip.js code, which I can share. Using google stun servers.</div><div>So this setup still works with older versions.</div><div><br></div><div>ICE candidates are saved normally on new version, but then after:</div><div><div><font face="times new roman, serif">2016-07-27 09:43:11.092485 [INFO] switch_core_media.c:7059 Activating Video ICE</font></div></div><div>I get:</div><div><div><font face="times new roman, serif">2016-07-27 09:43:11.092485 [ALERT] switch_rtp.c:7171 Skip sending video packet 0 bytes (ice not ready @ line 4366!)</font></div></div><div><br></div><div>and after Callstate Change RINGING -> EARLY it fails on both audio and video port:</div><div><br></div><div><div><font face="times new roman, serif">2016-07-27 09:43:11.092485 [ALERT] switch_rtp.c:7171 Skip sending video packet 0 bytes (ice not ready @ line 4366!)</font></div><div><font face="times new roman, serif">2016-07-27 09:43:11.352476 [ALERT] switch_rtp.c:1210 STUN from <a href="http://89.184.168.162:65369">89.184.168.162:65369</a> EXPECTED<br></font></div><div><font face="times new roman, serif">2016-07-27 09:43:11.362494 [ALERT] switch_rtp.c:1210 STUN from <a href="http://89.184.168.162:65361">89.184.168.162:65361</a> EXPECTED</font></div><div><font face="times new roman, serif">2016-07-27 09:43:11.392477 [ALERT] switch_rtp.c:1210 STUN from <a href="http://89.184.168.162:65369">89.184.168.162:65369</a> EXPECTED</font></div><div><font face="times new roman, serif">2016-07-27 09:43:11.402477 [ALERT] switch_rtp.c:1210 STUN from <a href="http://89.184.168.162:65361">89.184.168.162:65361</a> EXPECTED</font></div><div><font face="times new roman, serif">2016-07-27 09:43:11.442488 [ALERT] switch_rtp.c:1210 STUN from <a href="http://89.184.168.162:65369">89.184.168.162:65369</a> EXPECTED</font></div><div><font face="times new roman, serif">2016-07-27 09:43:11.442488 [ALERT] switch_rtp.c:1210 STUN from <a href="http://89.184.168.162:65361">89.184.168.162:65361</a> EXPECTED</font></div><div><font face="times new roman, serif">2016-07-27 09:43:11.492482 [ALERT] switch_rtp.c:1210 STUN from <a href="http://89.184.168.162:65369">89.184.168.162:65369</a> EXPECTED</font></div><div><font face="times new roman, serif">2016-07-27 09:43:11.502520 [ALERT] switch_rtp.c:1210 STUN from <a href="http://89.184.168.162:65361">89.184.168.162:65361</a> EXPECTED</font></div><div><font face="times new roman, serif">2016-07-27 09:43:11.512489 [ERR] switch_rtp.c:3112 video Handshake failure 1</font></div><div><font face="times new roman, serif">2016-07-27 09:43:11.512489 [INFO] switch_rtp.c:3113 Changing video DTLS state from HANDSHAKE to FAIL</font></div><div><font face="times new roman, serif">2016-07-27 09:43:11.542511 [ERR] switch_rtp.c:3112 audio Handshake failure 1</font></div><div><font face="times new roman, serif">2016-07-27 09:43:11.542511 [INFO] switch_rtp.c:3113 Changing audio DTLS state from HANDSHAKE to FAIL</font></div></div><div><font face="times new roman, serif"><br></font></div><div><font face="times new roman, serif"><br></font></div><div><font face="times new roman, serif">Maybe the google stun server is failing to send initial rtp.</font></div><div><font face="times new roman, serif"><br></font></div><div><font face="times new roman, serif">Thanks,</font></div><div><font face="times new roman, serif">mirko</font></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Jul 27, 2016 at 9:10 AM, Giovanni Maruzzelli <span dir="ltr"><<a href="mailto:gmaruzz@gmail.com" target="_blank">gmaruzz@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div>Hi Mirko,<br><br></div>how can we reproduce the problem?<br><br></div>I just connected to <a href="https://cantina.freeswitch.org/vc/" target="_blank">https://cantina.freeswitch.org/vc/</a> , then dialed 888, and I was connected to conference, with video and all, from chrome Version 52.0.2743.82 (64-bit) Linux, no problem.<br><br></div>-giovanni<br><br></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="h5">On Wed, Jul 27, 2016 at 9:03 AM, Mirko Brankovic <span dir="ltr"><<a href="mailto:mirkobrankovic@gmail.com" target="_blank">mirkobrankovic@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><div dir="ltr">Hi,<div>Has anyone encountered this new problem with new version of Chrome.</div><div>Maybe it is a question for the Chrome dev but just to ask :)</div><div><br></div><div>Chrome v52 gives me bad AV handshake on FS:</div><div><br></div><div><div><font face="times new roman, serif">2016-07-27 08:52:20.952477 [ERR] switch_rtp.c:3112 video Handshake failure 1</font></div><div><font face="times new roman, serif">2016-07-27 08:52:20.952477 [INFO] switch_rtp.c:3113 Changing video DTLS state from HANDSHAKE to FAIL</font></div><div><font face="times new roman, serif">2016-07-27 08:52:20.982513 [ERR] switch_rtp.c:3112 audio Handshake failure 1</font></div><div><font face="times new roman, serif">2016-07-27 08:52:20.982513 [INFO] switch_rtp.c:3113 Changing audio DTLS state from HANDSHAKE to FAIL</font></div></div><div><br></div><div><div>On v51 fingerprint is verified:</div><div><br></div><div><div><font face="times new roman, serif">2016-07-27 09:03:02.072474 [INFO] switch_rtp.c:3119 Changing video DTLS state from HANDSHAKE to SETUP</font></div><div><font face="times new roman, serif">2016-07-27 09:03:02.122569 [INFO] switch_rtp.c:3028 video Fingerprint Verified.</font></div><div><font face="times new roman, serif">2016-07-27 09:03:02.122569 [INFO] switch_rtp.c:3644 Activating Video Secure RTP SEND</font></div></div><span><font color="#888888"><div><br></div><div><br></div>-- <br><div><div dir="ltr"><div><div dir="ltr">Regards,<div>Mirko</div></div></div></div></div>
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