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DmitrySenashenko
Joined: Nov 29, 2013
Messages: 72
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Hi, Does anybody saw similar?
I have subj. No error in WebRTC.log. No error in WebRTCSampelApplication.log.
WebRTC is 3.0.3.0.303006. :cry:
If I look to Java console of my Chrome then I can see the following:
2015-4-21 18:28:11.104 INFO - connect -->
index.jsp:1 XMLHttpRequest cannot load https://cesip.avayacc.ru:9443/services/WebRTC/WebRtcServlet. No 'Access-Control-Allow-Origin' header is present on the requested resource. Origin 'https://cesip.avayacc.ru' is therefore not allowed access.
avayawebrtc-client.js:1601
2015-4-21 18:28:11.144 ERROR - Client:connect -- emit(ClientMessageConnect) failed. Socket errorClient.log @ avayawebrtc-client.js:1601(anonymous function) @ avayawebrtc-client.js:1498Socket.$.ajax.error @ avayawebrtc-client.js:907x.Callbacks.l @ jquery-2.0.3.min.js:4x.Callbacks.c.fireWith @ jquery-2.0.3.min.js:4k @ jquery-2.0.3.min.js:6x.ajaxTransport.x.support.cors.e.crossDomain.send.t @ jquery-2.0.3.min.js:6
index.jsp:1 XMLHttpRequest cannot load https://cesip.avayacc.ru:9443/services/WebRTC/WebRtcServlet?name=polling&sessionid=E5A09EDB-1CC0-402E-A2BB-D48CB04C9FF0. No 'Access-Control-Allow-Origin' header is present on the requested resource. Origin 'https://cesip.avayacc.ru' is therefore not allowed access.
avayawebrtc-client.js:1593
2015-4-21 18:28:11.149 INFO - socket disconnected

But my URL is https://cesip.avayacc.ru/services/WebRTCSampleApplication/index.jsp
They are in the same domain.
Looks like it is error in source code of WebRTCSampleApplication and need to change source code and compile it again.
Is'nt it?
Could somebody had similar? How to fix?

Best regards, Dmitry
JoelEzell
Joined: Nov 15, 2013
Messages: 780
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Hi Dmitri, even though you have the same domain, you still need to configure Cross-Origin Resource Sharing on your EDP server. Have you done so? If not, please follow the directions in the WebRTC Overview Guide and give it another shot. Let us know how it goes!
DmitrySenashenko
Joined: Nov 29, 2013
Messages: 72
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Yes, I have found fix of the issue. You are right. I lost some important settings.
The fix of the issue is Collaboration Environment -> Configuration -> HTTP Security
Thanks so much.
JoelEzell
Joined: Nov 15, 2013
Messages: 780
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I'm happy to hear that you got it working!
DmitrySenashenko
Joined: Nov 29, 2013
Messages: 72
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Hi again,
A small new question.
After call via WebRTC I have 10 sec delay after star of call. Over 10 sec (2 rings) I have call to phone.
I see this delay is in JavaScript. By trace I can see CE got call over 10 sec after start of call from WebRTC.
Look to trace:
2015-4-22 10:51:5.8 INFO - Call::onChangeState <-->.
avayawebrtc-client.js:1593 2015-4-22 10:51:5.8 INFO - Call::onChangeState: Peer connection signalling state changed callback, state:have-local-offer
avayawebrtc-client.js:1593 2015-4-22 10:51:5.13 INFO - Call::initiate -- set local peer connection session description succeeded
message_handler.js:15 dcePageScanner::scanPage::Offline
avayawebrtc-client.js:1593 2015-4-22 10:51:14.518 INFO - Call::initiate -- Candidate 0 - audio: candidate:414524848 1 udp 2122260223 192.168.62.1 49833 typ host generation 0

It means between Call Initiate and choose of IP for audio is delay 10 sec.
Do you have any ideas why? And how to fix this?
Did you see something similar?

And traditionally there are not any errors in WebRTC.log and WebRTCSampleApplication.log

Best regards, Dmitry
JoelEzell
Joined: Nov 15, 2013
Messages: 780
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Hi Dmitri, can you please verify that:

- You have the STUN Server Address correctly provisioned on the WebRTC snap-in (should be your Avaya SBC-E address)
- The PC you are using is able to access this address
- You are pressing the button to allow the browser to access your microphone

I believe that the browser will wait up to 10 seconds for a response from the STUN server (in the Avaya SBC) before timing out and proceeding without it.
DmitrySenashenko
Joined: Nov 29, 2013
Messages: 72
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Hi,
> - You have the STUN Server Address correctly provisioned on the WebRTC snap-in (should be your Avaya SBC-E address)
Yes, of course. But there is not way to check this connection from WebRTC site.
2015-04-23 17:21:11,703 [WebContainer : 3] WebRTC FINER - WebRTC-3.0.3.0.303006 - DispatchMessages finish. Session ID:C4F7CF50-6865-4247-B11B-05B1F55EE431
2015-04-23 17:21:11,703 [WebContainer : 3] WebRTC FINER - WebRTC-3.0.3.0.303006 - StartPolling finish. Session ID: C4F7CF50-6865-4247-B11B-05B1F55EE431
2015-04-23 17:21:11,703 [WebContainer : 3] WebRTC FINER - WebRTC-3.0.3.0.303006 - doGet finish
2015-04-23 17:21:21,255 [WebContainer : 3] WebRTC FINER - WebRTC-3.0.3.0.303006 - doPost start. Client:192.168.230.6

> - The PC you are using is able to access this address
Of course. It is working finally.

>- You are pressing the button to allow the browser to access your microphone
Of course. It is visible.

The idea with STUN server is good. I'll try to check this on SBC-E side.

Actually you can try this URL
https://sip.avayalab.ru/services/WebRTCSampleApplication/index.jsp
It is available from public internet. And look by your eyes. If you have problems with certificate then drop me email and I'll send it to you.
You can use number 55000. It is announcement.
Amit.Mishra
Joined: Dec 19, 2013
Messages: 34
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Hi Dmitry,
Once the SBC-E settings are verified and if you still see some issue, would it be possible to share some traces with us over email. We would send instructions on how to collect the traces if it is feasible.

Also, if possible, please share the complete SDP as seen in the console window of the browser for the local offer.

Thanks,
Amit.
JoelEzell
Joined: Nov 15, 2013
Messages: 780
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Hi Dmitri, just one other thing: it's the end-user's browser that must be able to access the STUN server address/port. You should be able to check this from your PC, right?

The JavaScript console clearly shows that the 10 second gap occurs when the ICE candidates are being collected. This definitely reinforces the theory that we're seeing a STUN timeout here.

2015-4-27 15:47:26.835 INFO - Call::initiate -- set local peer connection session description succeeded
2015-4-27 15:47:36.346 INFO - Call::initiate -- Candidate 0 - audio: candidate:2813519782 1 udp 2122257663 2002:8709:edeb::8709:edeb 52796 typ host generation 0
Yuichiro
Joined: Aug 21, 2015
Messages: 3
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Hi,
I am also troubled by the socket error.
CM, SM, SMGR, EDP, ASM is the manual to the original, on the status is the state no problem.

After a review of the log, I think that there are the following problems.
SM is sending an INVITE to the CM does not have a reaction.

Where there is a problem?
Could you tell me?

DipeshGandhi2 [Avatar]

Joined: Nov 14, 2011
Messages: 35
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Hello,

As your question requires log analysis and troubleshooting, please open a ticket opening a Devconnect technical support ticket by going to the DevConnect Portal page, and following the links through "Support & Forums" -> :Submit Support Ticket" -> "Technical Support" and we would be glad to assist with the issue. Be sure include detailed description of problem, the environment and log/trace files.


Thank you,
Dipesh


Dipesh Gandhi Software Engineer, Avaya DevConnect
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