[Freeswitch-users] chrome sipml5 cant receive bye
Jimmy Chang
chang33.tw at gmail.com
Thu Dec 12 12:15:07 MSK 2013
Hi,
I had tried jssip and got same result.
Bye signal could be received in audio calls but not in video calls.
Here is what I found (both sipml5 and jssip.)
When client(U1109903) hangup video call, the BYE signal would reach to
FS but not to the other WebRTC client.
Then the other WebRTC cleint(U1109902) hangup the call, its Chrome log
shows all BYE signals are received in this time.
========================
BYE sip:U1109903 at 10.0.199.33:5060;transport=udp SIP/2.0
Via: SIP/2.0/WS l1ulafktoii1.invalid;branch=z9hG4bK9006154
Max-Forwards: 69
To: <sip:U1109903 at 10.0.199.33>;tag=jmB2eBSH0Bggm
From: "U1109902" <sip:U1109902 at 10.0.199.33>;tag=ejrm0q06qe
Call-ID: 3gc45bilg4rbt647ue0u
CSeq: 2213 BYE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0
...
BYE sip:7pmon84c at l1ulafktoii1.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 10.0.199.33:5066;branch=z9hG4bKtBSFaU4mScSZp
Max-Forwards: 70
From: <sip:U1109903 at 10.0.199.33>;tag=jmB2eBSH0Bggm
To: "U1109902" <sip:U1109902 at 10.0.199.33>;tag=ejrm0q06qe
Call-ID: 3gc45bilg4rbt647ue0u
CSeq: 53082708 BYE
Contact: <sip:U1109903 at 10.0.199.33:5060;transport=udp>
User-Agent:
FreeSWITCH-mod_sofia/1.5.8b+git~20131205T011809Z~4a2054a951~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
jssip-devel.js:671
JsSIP | TRANSPORT | sending WebSocket message:
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/WS 10.0.199.33:5066;branch=z9hG4bKtBSFaU4mScSZp
To: "U1109902" <sip:U1109902 at 10.0.199.33>;tag=ejrm0q06qe
From: <sip:U1109903 at 10.0.199.33>;tag=jmB2eBSH0Bggm
Call-ID: 3gc45bilg4rbt647ue0u
CSeq: 53082708 BYE
Content-Length: 0
jssip-devel.js:520
JsSIP | TRANSACTION | Timer J expired for non-INVITE server transaction
z9hG4bKtBSFaU4mScSZp jssip-devel.js:2073
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WS
l1ulafktoii1.invalid;branch=z9hG4bK9006154;received=10.0.1.69;rport=53126
From: "U1109902" <sip:U1109902 at 10.0.199.33>;tag=ejrm0q06qe
To: <sip:U1109903 at 10.0.199.33>;tag=jmB2eBSH0Bggm
Call-ID: 3gc45bilg4rbt647ue0u
CSeq: 2213 BYE
Content-Length: 0
========================
? 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 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
> <mailto: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 ?? 05:14, Karsten Horsmann :
>> Hi,
>>
>> try to disable webrtc-breaker in dubango sipml5.
>>
>>
>> 2013/12/11 Jimmy Chang <chang33.tw at gmail.com
>> <mailto: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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
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>>
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>>
>>
>>
>>
>> --
>> Mit freundlichen Grüßen
>> *Karsten Horsmann*
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>> Official FreeSWITCH Sites
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>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <mailto:FreeSWITCH-users at 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 at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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