[Freeswitch-users] chrome sipml5 cant receive bye

[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 at 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 at 10.0.199.33>;tag=t14jXmpU8cU0H
   5. To: "U1109902"<sip:U1109902 at 10.0.199.33>;tag=9iaJJuLuMJGOK7dE7Fsn
   6. Contact: <sip:U1109903 at 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 at 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 at 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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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
>>
>
>
>  --
> Mit freundlichen Grüßen
> *Karsten Horsmann*
>
>
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