US Service Request Email: To communicate with Chrome I have a FreeSwitch 1. I really don't know if just this is sufficient. Asked 2 years, 5 months ago. Alexander Kachkaev Alexander Kachkaev 6 6 silver badges 22 22 bronze badges. Make a call and and check the console logs on the browser for further details. The CudaTel Communication Server http: 
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Amsip SDK - webrtc vs sip - Antisip
But Chrome isn't on the same network, right? I've installed FreeSwitch following the default recipe, but instead of.
Post as a guest Name. Does it work on that same headset on webrtc. James My application server nginx is on the same network my FreeSwitch and Asterisk are, so I'm not using a Stun server. Hssip Fri, Oct 4, at On the FS console I can see that the ivr call follow is executed perfectly, but I can't hear anything on my headset.
@dskop/jssip
If you are looking for an example code, feel free to check out the implementation of react-sip a package that helps embed JsSIP into React apps. James On Thu, Oct 3, at 4: Here is roughly what you should add: If you're server is on Amazon EC2, make sure you're following the guide here: Does anyone have a example code for this project?
Email Required, but never shown. It is important to note that my client ChromeFreeSwitch and Asterisk were all at the same network while doing this tests.
Usage stats for jssip on npm -
Tornado Computer Systems, Inc. Thanks for the attention! Also, in Chrome, startup chrome from the command line with the options to enable debug logging:.
US Service Request Email: Free forum by Nabble. I'm working on a telecom company.
Unicorn Meta Zoo 9: I really don't know if just this is sufficient. Asked 2 years, 5 months ago. Also, in Chrome, startup chrome from the command line with the options to enable debug logging: In reply to this post by James Mortensen.
Improving the question-asking experience. This log file http: Then look to see if there are STUN binding errors. The call will disconnect with reason "RTP timeout" in that case.
Tryit jssip
The connection is established but I don't get any audio in both endpoints. STUN and other configuration settings will need to be adjusted in that case.
Sign up using Facebook. You didn't mention whether the server was in the cloud. My application server nginx is on the same network my FreeSwitch and Asterisk are, so I'm not using a Stun server. Thursday, October 03, 9: Make a call and and check jssjp console logs on the browser for further details.
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