<div dir="ltr">For your video call, I can see at the end FreeSWITCH both receives and sends a BYE. Are you sure that's not a sipml5 problem? Even in the Chrome logs there's a BYE. Are you sure this isn't a sipml5 bug? What happens if you use the JsSIP Try It demo instead, just as a sanity check? <a href="http://tryit.jssip.net">http://tryit.jssip.net</a><div>
<br></div><div><ol style="color:rgb(51,51,51);font-family:monospace;font-size:11px"><li class="" style="background-color:rgb(244,244,244)"><div class="" style="font-family:'Courier New',Courier,monospace">__tsip_transport_ws_onmessage tsk_utils.js?svn=<span class="">20</span>:<span class="">116</span></div>
</li><li class=""><div class="" style="font-family:'Courier New',Courier,monospace">recv=BYE sip:U1109902@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/<span class="">2.0</span></div></li>
<li class="" style="background-color:rgb(244,244,244)"><div class="" style="font-family:'Courier New',Courier,monospace">Via: SIP/<span class="">2.0</span>/WS <span class="">10.0</span><span class="">.199</span><span class="">.33</span>:<span class="">5066</span>;branch=z9hG4bKDgHH2Hg7yp0cD</div>
</li><li class=""><div class="" style="font-family:'Courier New',Courier,monospace">From: <sip:U1109903@<span class="">10.0</span><span class="">.199</span><span class="">.33</span>>;tag=t14jXmpU8cU0H</div>
</li>
<li class="" style="background-color:rgb(244,244,244)"><div class="" style="font-family:'Courier New',Courier,monospace">To: "U1109902"<sip:U1109902@<span class="">10.0</span><span class="">.199</span><span class="">.33</span>>;tag=9iaJJuLuMJGOK7dE7Fsn</div>
</li><li class=""><div class="" style="font-family:'Courier New',Courier,monospace">Contact: <sip:U1109903@<span class="">10.0</span><span class="">.199</span><span class="">.33</span>:<span class="">5060</span>;transport=udp></div>
</li><li class="" style="background-color:rgb(244,244,244)"><div class="" style="font-family:'Courier New',Courier,monospace">Call-ID: fc24993f-d5cf-0ae3-a7a0-5f34be87d65f</div></li><li class=""><div class="" style="font-family:'Courier New',Courier,monospace">
CSeq: <span class="">53036083</span> BYE</div></li><li class="" style="background-color:rgb(244,244,244)"><div class="" style="font-family:'Courier New',Courier,monospace">Content-Length: <span class="">0</span></div>
</li><li class=""><div class="" style="font-family:'Courier New',Courier,monospace">Max-Forwards: <span class="">70</span></div></li><li class="" style="background-color:rgb(244,244,244)"><div class="" style="font-family:'Courier New',Courier,monospace">
User-Agent: FreeSWITCH-mod_sofia/<span class="">1.5</span>.8b+git~20131205T011809Z~4a2054a951~64bit</div></li><li class=""><div class="" style="font-family:'Courier New',Courier,monospace">Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY,PUBLISH,SUBSCRIBE</div>
</li><li class="" style="background-color:rgb(244,244,244)"><div class="" style="font-family:'Courier New',Courier,monospace">Supported: timer,precondition,path,replaces</div></li><li class=""><div class="" style="font-family:'Courier New',Courier,monospace">
Reason: text="NORMAL_CLEARING";cause=<span class="">16</span>;text="NORMAL_CLEARING"</div></li></ol></div><div class="gmail_extra"><br clear="all"><div><div dir="ltr"><div><br></div><div>James<br></div>
<div><br><div><br></div></div></div></div><br><div class="gmail_quote">On Wed, Dec 11, 2013 at 7:25 PM, Jimmy Chang <span dir="ltr"><<a href="mailto:chang33.tw@gmail.com" target="_blank">chang33.tw@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 text="#000000" bgcolor="#FFFFFF">
<div>Hi,<br>
<br>
Thanks for your recommendation.<br>
I had tried to disable this param but in vain.<br>
<br>
I found that no matter what I set the webrtc-breaker param, <br>
when a audio call was made, the caller and callee could hangup
call fine.<br>
When a video call was made, the caller and callee hangup call
without notifying other side BYE signal.<br>
<br>
Jimmy<br>
<br>
2013/12/12 上午 05:14, Karsten Horsmann :<br>
</div><div><div class="h5">
<blockquote type="cite">
<div dir="ltr">Hi,
<div><br>
</div>
<div>try to disable webrtc-breaker in dubango sipml5.</div>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">2013/12/11 Jimmy Chang <span dir="ltr"><<a href="mailto:chang33.tw@gmail.com" target="_blank">chang33.tw@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> <font face="Arial">Hi,<br>
<br>
I'm using FreeSWITCH Version
1.5.8b+git~20131205T011809Z~4a2054a951~64bit.<br>
Here are what I set<br>
1) configure --with-openssl<br>
2) uncomment for sip over websocket support (port 5066)<br>
<br>
The caller and callee were all Chrome sipml5 clients and
the audio and video were all fine.<br>
The problem was that when caller or callee hangup the
video call, the other side couldn't get BYE signal.<br>
<br>
Here is the freeswitch log,<br>
<a href="http://pastebin.freeswitch.org/21739" target="_blank">http://pastebin.freeswitch.org/21739</a><br>
<br>
and here is the Chrome console log.<br>
<a href="http://pastebin.freeswitch.org/21740" target="_blank">http://pastebin.freeswitch.org/21740</a><br>
<br>
Any advice would be appreciated.<span><font color="#888888"><br>
<br>
Jimmy</font></span></font> </div>
<br>
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<br>
<br clear="all">
<div><br>
</div>
-- <br>
Mit freundlichen Grüßen<br>
*Karsten Horsmann*<br>
</div>
<br>
<fieldset></fieldset>
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