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 ? 2013/12/12 ?? 02:47, James Mortensen ??:
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. recv=BYE sip:U1109902@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0 3. Via: SIP/2.0/WS 10.0.199.33:5066;branch=z9hG4bKDgHH2Hg7yp0cD 4. From: <sip:U1109903@10.0.199.33>;tag=t14jXmpU8cU0H 5. To: "U1109902"<sip:U1109902@10.0.199.33>;tag=9iaJJuLuMJGOK7dE7Fsn 6. Contact: <sip:U1109903@10.0.199.33:5060;transport=udp> 7. Call-ID: fc24993f-d5cf-0ae3-a7a0-5f34be87d65f 8. CSeq: 53036083 BYE 9. Content-Length: 0 10. Max-Forwards: 70 11. User-Agent: FreeSWITCH-mod_sofia/1.5.8b+git~20131205T011809Z~4a2054a951~64bit 12. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY,PUBLISH,SUBSCRIBE 13. Supported: timer,precondition,path,replaces 14. 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 <mailto: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 ?? 05:14, Karsten Horsmann :
Hi,
try to disable webrtc-breaker in dubango sipml5.
2013/12/11 Jimmy Chang <chang33.tw@gmail.com <mailto: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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Mit freundlichen Grüßen *Karsten Horsmann*
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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 FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
--- 由於 avast! 防毒軟體已開啟防護，所以此封電子郵件將不受病毒及惡意軟體的侵害。 http://www.avast.com