I created certificates with command ./ast_tls_cert -C [my_real_IP_lan _address] -O "My Super Company" - d /etc/asterisk/keys then checked is created in destination directory. After that i configured http.conf, restart core of Asterisk and try get http://my_real_lan_ip_address:8089. But i got nothing and in CLI i got this
OK. I choose Asterisk HTTPS server for testing. Its up and running but I dont see even my local camera. There is only this error
sdp-interop-sl-1.4.0.js:1613 Uncaught TypeError: Cannot read property 'some' of undefined
at module.exports (sdp-interop-sl-1.4.0.js:1613)
at Object.toPlanB (sdp-interop-sl-1.4.0.js:690)
at RTCSession.<anonymous> (cyber_mega_phone.js:98)
at RTCSession.EventEmitter.emit (jssip-3.0.13.js:21730)
at RTCSession.receiveInviteResponse (jssip-3.0.13.js:15906)
at RTCSession.receiveResponse (jssip-3.0.13.js:15698)
at RequestSender.receiveResponse (jssip-3.0.13.js:17663)
at InviteClientTransaction.receiveResponse (jssip-3.0.13.js:18701)
at UA.onTransportData (jssip-3.0.13.js:20595)
at Transport.onData (jssip-3.0.13.js:19317)
I don’t have experience in that area so I can’t really help. It’s possible the browsers changed something or did something which has caused a problem. With WebRTC if you are expecting to use it that’s something you have to take into account when deploying it, you have to know the technology and react to any changes that occur.
Have you followed the guide exactly for it? Including configuration and other things?
Ok, I re-read everything again and made some changes acording to tutorial and it is working. I’m wondering is any windows/client which will collaborate with persons who are use web-based video-conference ?
I’m trying also to open the wss and ping from https://www.websocket.org/echo.html but i’m getting these errors and after trying to look at a correct solution i’m totally lost right now.
Anyone cal help with this?
Thanks
ERROR[12086]: iostream.c:633 ast_iostream_start_tls: Problem setting up ssl connection: error:00000001:lib(0):func(0):reason(1), Internal SSL error
ERROR[12086]: tcptls.c:157 handle_tcptls_connection: Unable to set up ssl connection with peer ‘192.168.1.1:34574’
ERROR[12086]: iostream.c:538 ast_iostream_close: SSL_shutdown() failed: error:00000001:lib(0):func(0):reason(1), Internal SSL error
Hi,
I’m trying to configure asterisk for webrtc and start using multi stream (SFU). Right now I according to guide at this site https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone
Things are going fine when i access https://my_IP:8089/phone. I’m getting redirected to Cyber Mega Phone Page.The strange thing is the log written in asterisk:
[May 4 13:38:18] ERROR[3896]: tcptls.c:157 handle_tcptls_connection: Unable to set up ssl connection with peer ‘IP:50089’
[May 4 13:38:18] ERROR[3896]: iostream.c:538 ast_iostream_close: SSL_shutdown() failed: error:00000001:lib(0):func(0):reason(1), Internal SSL error
[May 4 13:38:23] ERROR[3895]: iostream.c:633 ast_iostream_start_tls: Problem setting up ssl connection: error:00000001:lib(0):func(0):reason(1), Internal SSL error
Video conference also works fine, but the log is still writing these errors.Can you help me with the cause of this?