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<div class="moz-cite-prefix">Hi,<br>
I had tried jssip and got same result.<br>
Bye signal could be received in audio calls but not in video
calls.<br>
<br>
Here is what I found (both sipml5 and jssip.)<br>
When client(U1109903) hangup video call, the BYE signal would
reach to FS but not to the other WebRTC client.<br>
Then the other WebRTC cleint(U1109902) hangup the call, its Chrome
log shows all BYE signals are received in this time.<br>
<br>
========================<br>
BYE <a class="moz-txt-link-freetext" href="sip:U1109903@10.0.199.33:5060;transport=udp">sip:U1109903@10.0.199.33:5060;transport=udp</a> SIP/2.0<br>
Via: SIP/2.0/WS l1ulafktoii1.invalid;branch=z9hG4bK9006154<br>
Max-Forwards: 69<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:U1109903@10.0.199.33"><sip:U1109903@10.0.199.33></a>;tag=jmB2eBSH0Bggm<br>
From: "U1109902" <a class="moz-txt-link-rfc2396E" href="sip:U1109902@10.0.199.33"><sip:U1109902@10.0.199.33></a>;tag=ejrm0q06qe<br>
Call-ID: 3gc45bilg4rbt647ue0u<br>
CSeq: 2213 BYE<br>
Supported: path, outbound, gruu<br>
User-Agent: JsSIP 0.3.7<br>
Content-Length: 0<br>
<br>
...<br>
<br>
BYE <a class="moz-txt-link-freetext" href="sip:7pmon84c@l1ulafktoii1.invalid;transport=ws;ob">sip:7pmon84c@l1ulafktoii1.invalid;transport=ws;ob</a> SIP/2.0<br>
Via: SIP/2.0/WS 10.0.199.33:5066;branch=z9hG4bKtBSFaU4mScSZp<br>
Max-Forwards: 70<br>
From: <a class="moz-txt-link-rfc2396E" href="sip:U1109903@10.0.199.33"><sip:U1109903@10.0.199.33></a>;tag=jmB2eBSH0Bggm<br>
To: "U1109902" <a class="moz-txt-link-rfc2396E" href="sip:U1109902@10.0.199.33"><sip:U1109902@10.0.199.33></a>;tag=ejrm0q06qe<br>
Call-ID: 3gc45bilg4rbt647ue0u<br>
CSeq: 53082708 BYE<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:U1109903@10.0.199.33:5060;transport=udp"><sip:U1109903@10.0.199.33:5060;transport=udp></a><br>
User-Agent:
FreeSWITCH-mod_sofia/1.5.8b+git~20131205T011809Z~4a2054a951~64bit<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE<br>
Supported: timer, precondition, path, replaces<br>
Reason: Q.850;cause=16;text="NORMAL_CLEARING"<br>
Content-Length: 0<br>
<br>
<br>
jssip-devel.js:671<br>
JsSIP | TRANSPORT | sending WebSocket message:<br>
<br>
SIP/2.0 481 Call/Transaction Does Not Exist<br>
Via: SIP/2.0/WS 10.0.199.33:5066;branch=z9hG4bKtBSFaU4mScSZp<br>
To: "U1109902" <a class="moz-txt-link-rfc2396E" href="sip:U1109902@10.0.199.33"><sip:U1109902@10.0.199.33></a>;tag=ejrm0q06qe<br>
From: <a class="moz-txt-link-rfc2396E" href="sip:U1109903@10.0.199.33"><sip:U1109903@10.0.199.33></a>;tag=jmB2eBSH0Bggm<br>
Call-ID: 3gc45bilg4rbt647ue0u<br>
CSeq: 53082708 BYE<br>
Content-Length: 0<br>
<br>
<br>
jssip-devel.js:520<br>
JsSIP | TRANSACTION | Timer J expired for non-INVITE server
transaction z9hG4bKtBSFaU4mScSZp jssip-devel.js:2073<br>
JsSIP | TRANSPORT | received WebSocket text message:<br>
<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/WS
l1ulafktoii1.invalid;branch=z9hG4bK9006154;received=10.0.1.69;rport=53126<br>
From: "U1109902" <a class="moz-txt-link-rfc2396E" href="sip:U1109902@10.0.199.33"><sip:U1109902@10.0.199.33></a>;tag=ejrm0q06qe<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:U1109903@10.0.199.33"><sip:U1109903@10.0.199.33></a>;tag=jmB2eBSH0Bggm<br>
Call-ID: 3gc45bilg4rbt647ue0u<br>
CSeq: 2213 BYE<br>
Content-Length: 0<br>
========================<br>
<br>
<br>
於 2013/12/12 下午 02:47, James Mortensen 提到:<br>
</div>
<blockquote
cite="mid:CANhSRjKCr03yrWsHUVYRWutZ_XQbi_Xf_qGwSQP-O6WwedU8kA@mail.gmail.com"
type="cite">
<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
moz-do-not-send="true" 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
<a class="moz-txt-link-freetext" href="sip:U1109902@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws">sip:U1109902@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws</a>
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: <<a class="moz-txt-link-freetext" href="sip:U1109903@">sip:U1109903@</a><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"<<a class="moz-txt-link-freetext" href="sip:U1109902@">sip:U1109902@</a><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: <<a class="moz-txt-link-freetext" href="sip:U1109903@">sip:U1109903@</a><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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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>
Mit freundlichen Grüßen<br>
*Karsten Horsmann*<br>
</div>
<br>
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