What steps will reproduce the problem?
1.SIP client send HTTP GET and webrtc2sip send back 101 Switching
2.Logs shows websocket connection accepted
3.When SIP over websocket packet is received by webrtc2sip it shows following
error and send TCP FIN to close connection:
After the handshaking (HTTP GET and HTTP 101), sipflex sends REGISTER message,
but webrtc2sip shows the Error message as follows:
***ERROR: function: "tsip_transport_layer_ws_cb()"
file: "src/transports/tsip_transport_layer.c"
line: "403"
MSG: WS handshaking not done yet
A new version same error but line #407
What is the expected output? What do you see instead?
SIP Over WebSocket should be accepted and there should not be TCP close
What version of the product are you using? On what operating system?
2.0 doubango
Please provide any additional information below.
Same issue has been discussed under following thread..
https://code.google.com/p/telepresence/issues/detail?id=22
https://groups.google.com/forum/#!searchin/doubango/WS$20handshaking$20not$20don
e$20yet$20/doubango/my1wDwTzf9Y/3K9uSnf2IfMJ
Original issue reported on code.google.com by daljitm...@gmail.com on 19 Feb 2015 at 7:15
Original issue reported on code.google.com by
daljitm...@gmail.com
on 19 Feb 2015 at 7:15