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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;[email protected];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;[email protected];[email protected];mysecret</account-mail-->
<!--account-sip-caller>*;sip:[email protected];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 [email protected] on 5 Jan 2015 at 11:53
Original issue reported on code.google.com by
[email protected]
on 5 Jan 2015 at 11:53Attachments:
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