webrtc2sip + sipml5 : MSG: WS handshaking not done yet

What steps will reproduce the problem?
1. Asterisk and webrtc2sip is installed on centOS 6.6
2. Created certificates for config.xml 
3. Try to make call from sipml5 client

What is the expected output? What do you see instead?
Expecting successful registration and start call from sipml5 client to linphone.

What version of the product are you using? On what operating system?
1. CentOS 6.6, Asterisk 11.6 certified version, latest versions of SipML5 and 
WebRTC2sip

Please provide server logs with DEBUG level equal to INFO

webrtc2sip.log file is attached

When log-level is changed to INFO, error is not found in webrtc2sip logs. 
However
if I change log level to ERROR then below error is found:

***ERROR: function: "tsip_transport_layer_ws_cb()"
file: "src/transports/tsip_transport_layer.c"
line: "403"
MSG: WS handshaking not done yet
***ERROR: function: "tsip_transport_layer_ws_cb()"
file: "src/transports/tsip_transport_layer.c"
line: "403"
MSG: WS handshaking not done yet

Please provide browser logs

browser.log file is attached.

Below is config.xml file:

<config>

  <debug-level>INFO</debug-level>

  <transport>udp;*;10060</transport>
  <transport>ws;*;10060</transport>
  <transport>wss;*;10062</transport>
  <!--transport>tcp;*;10063</transport-->
  <!--transport>tls;*;10064</transport-->

  <enable-rtp-symetric>yes</enable-rtp-symetric>
  <enable-100rel>no</enable-100rel>
  <enable-media-coder>yes</enable-media-coder>
  <enable-videojb>no</enable-videojb>
  <video-size-pref>vga</video-size-pref>
  <rtp-buffsize>65535</rtp-buffsize>
  <avpf-tail-length>100;400</avpf-tail-length>
  <srtp-mode>optional</srtp-mode>
  <srtp-type>sdes;dtls</srtp-type>
  <dtmf-type>rfc4733</dtmf-type>

  <codecs>opus;pcma;pcmu;gsm;vp8;h264-bp;h264-mp;h263;h263+</codecs>
  <codec-opus-maxrates>48000;48000</codec-opus-maxrates>

  <stun-server>stun.l.google.com;19302;stun-user@doubango.org;stun-password</stun-server>
  <enable-icestun>yes</enable-icestun>

  <max-fds>-1</max-fds>

  <!--nameserver>66.66.66.6</nameserver-->

  <ssl-certificates>
    /certs/privkey.pem;
    /certs/newcert.pem;
    *;
    no
  </ssl-certificates>

  <!-- ***CLICK-TO-CALL SERVICE*** -->

  <transport>c2c;*;10070</transport>
  <transport>c2cs;*;10072</transport>
  <database>sqlite;*</database>
  <!--account-mail>smtps;*;*;auth.smtp.1and1.fr;465;noreply@example.com;noreply@example.com;mysecret</account-mail-->
  <!--account-sip-caller>*;sip:a@example.com;a;example.com;mysecret</account-sip-caller-->

</config>

I want to check make call and receive call between SipML5 and linphone.

I am not sure what wrong I am doing for certificates. Do I really need to use 
certificates for ws protocol? And do I need to install certificate to client 
browser also? Please check the logs and let me know the issue with my 
configurations. I have tried multiple time and through multiple references but
every-time I blocked because of affricate issues.

Thanks and Regards
Vinod Pandey
Original issue reported on code.google.com by pandey.g...@gmail.com on 5 Jan 2015 at 11:53
Attachments:
andresetevejob / webrtc2sip

webrtc2sip + sipml5 : MSG: WS handshaking not done yet #172
