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
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
_________________________________________________________________________ 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
-- Mit freundlichen Grüßen *Karsten Horsmann*
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 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
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> 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
_________________________________________________________________________ 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
-- Mit freundlichen Grüßen *Karsten Horsmann*
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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
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@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@10.0.199.33>;tag=jmB2eBSH0Bggm From: "U1109902" <sip:U1109902@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@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@10.0.199.33>;tag=jmB2eBSH0Bggm To: "U1109902" <sip:U1109902@10.0.199.33>;tag=ejrm0q06qe Call-ID: 3gc45bilg4rbt647ue0u CSeq: 53082708 BYE Contact: <sip:U1109903@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@10.0.199.33>;tag=ejrm0q06qe From: <sip:U1109903@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@10.0.199.33>;tag=ejrm0q06qe To: <sip:U1109903@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@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
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
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participants (3)
-
James Mortensen
-
Jimmy Chang
-
Karsten Horsmann