Here is the Freeswitch log.
When caller hangup call, it seems FS didn't send bye signal to callee.
http://pastebin.freeswitch.org/21743

Did I misconfigure something or is it a bug of FS?

Jimmy

於 2013/12/12 下午 02:47, James Mortensen 提到:
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?  http://tryit.jssip.net

  1. __tsip_transport_ws_onmessage tsk_utils.js?svn=20:116
  2. Via: SIP/2.0/WS 10.0.199.33:5066;branch=z9hG4bKDgHH2Hg7yp0cD
  3. From: <sip:U1109903@10.0.199.33>;tag=t14jXmpU8cU0H
  4. To: "U1109902"<sip:U1109902@10.0.199.33>;tag=9iaJJuLuMJGOK7dE7Fsn
  5. Contact: <sip:U1109903@10.0.199.33:5060;transport=udp>
  6. Call-ID: fc24993f-d5cf-0ae3-a7a0-5f34be87d65f
  7. CSeq: 53036083 BYE
  8. Content-Length: 0
  9. Max-Forwards: 70
  10. User-Agent: FreeSWITCH-mod_sofia/1.5.8b+git~20131205T011809Z~4a2054a951~64bit
  11. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY,PUBLISH,SUBSCRIBE
  12. Supported: timer,precondition,path,replaces
  13. Reason: text="NORMAL_CLEARING";cause=16;text="NORMAL_CLEARING"


James



On Wed, Dec 11, 2013 at 7:25 PM, Jimmy Chang <chang33.tw@gmail.com> wrote:
Hi,

Thanks for your recommendation.
I had tried to disable this param but in vain.

I found that no matter what I set the webrtc-breaker param,
when a audio call was made, the caller and callee could hangup call fine.
When a video call was made, the caller and callee hangup call without notifying other side BYE signal.

Jimmy

2013/12/12 上午 05:14, Karsten Horsmann :
Hi,

try to disable webrtc-breaker in dubango sipml5.


2013/12/11 Jimmy Chang <chang33.tw@gmail.com>
Hi,

I'm using FreeSWITCH Version 1.5.8b+git~20131205T011809Z~4a2054a951~64bit.
Here are what I set
1) configure --with-openssl
2) uncomment for sip over websocket support (port 5066)

The caller and callee were all Chrome sipml5 clients and the audio and video were all fine.
The problem was that when caller or callee hangup the video call, the other side couldn't get BYE signal.

Here is the freeswitch log,
http://pastebin.freeswitch.org/21739

and here is the Chrome console log.
http://pastebin.freeswitch.org/21740

Any advice would be appreciated.

Jimmy

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--
Mit freundlichen Grüßen
*Karsten Horsmann*


_________________________________________________________________________
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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http://www.freeswitch.org




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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