calling from X-Lite to webrtc sipml5 client using webrtcbreaker

[vineet srivastava]

hi,

 i m trying to establish call from X-Lite to sipml5 client running on google chrome.

 I am using SIP2SIP.INFO as the sip proxy.

 Registered one user on X-lite.

 Another one on sipml5 web-browser with pointing to webrtc2sip server running on local PC. SIP proxy is configured as SIP2SIP.INFO. Webrtc breaker is ON and media coder is ON in the setting.

 Both the clients have google STUN server configured as address.

 When I initiate the call from X-Lite to Web-browser then Call is received at Web-browser but it gets automatically rejected saying that FAILED TO GET LOCAL SDP.

 On seeing the java console of the web-browser we can see that its cribbing for DTLS fingerprint in the SDP hence it rejects the INVITE.

  As the INVITE is coming from X-Lite hence it doesnt have DTLS fingerprint.

  Please let me know how to achieve the call properly using the setup. 

Thanks,

Vineet 

[Mamadou DIOP]

https://code.google.com/p/webrtc2sip/wiki/FAQ#How_to_report_issue?

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[vineet srivastava]

hi,

 Thanks for the reply,

 i have attached the logs:

1. Webrtc2sip gateway logs.                                 <webrtc2sipGatewayLogs.txt>

2. config.xml of the running webrtc2sip gateway.     <config.xml>

3. Pcap logs of the webrtc2sip gateway.                 <webrtc2sipGatewayLogs.pcap>

4. Browser logs of the browser client.                     <BrowserClientLogs.txt>

Issue technical description remains the same as sent in previous query. 

Please have a look at the logs. 

Let me know if you need any more info.

Waiting for your reply.

Thanks,

Vineet 

[Mamadou DIOP]

- I'll go crazy …why browser logs are cut and all useful lines (version, browser, registration...) are missing?

- As I said a hundred times already, don’t change the values in confi.xml if you don’t don’t what they mean. Please revert all values (“sftp-type”, “srtp-mode”, …) except the ssl-certificates.

- You must enable rtcweb-breaker on sipml5.

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<webrtc2sipGatewayLogs.txt><config.xml><webrtc2sipGatewayLogs.pcap><BrowserClientLogs.txt>

[vineet srivastava]

HI Mamadou,

  We also saw that browser logs are cut hence took another set of logs.

  Please check the latest log files it shall have the in-between lines too.

  Providing the new set of logs by enabling webrtc breaker module. 

  Possiblity is more of being a setting issue as i am new to this WebRTC and have been banging my head for the last two weeks to understand the same hence my understanding is not full.

  Regarding version, browser why they are not there i don't know because i am a C,C++ developer only and not a browser developer. If you know then let me know. 

  Regarding the version of <srtp-mode> i had tried with optional mode too, then thought of trying with mandatory but i have reverted the same as per your suggesstion.

  Let me know if you need anymore info.

Thanks,

Vineet 

[Mamadou DIOP]

JS logs are still incomplete, must start with something like this:

SIPML5 API version = 1.4.217 SIPml-api.js?svn=224:1

location=http://sipml5.org/call.htm?svn=224 call.htm?svn=224:147

User-Agent=Mozilla/5.0 (Macintosh; Intel Mac OS X 10_9_2) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/35.0.1916.114 Safari/537.36 SIPml-api.js?svn=224:1

WebSocket supported = yes SIPml-api.js?svn=224:1

Navigator friendly name = chrome SIPml-api.js?svn=224:1

OS friendly name = mac SIPml-api.js?svn=224:1

Have WebRTC = yes SIPml-api.js?svn=224:1

Have GUM = yes SIPml-api.js?svn=224:1

Engine initialized

and you haven’t reverted your config.xml file:

*INFO: srtp-mode = mandatory

*INFO: srtp-type = dtls

*INFO: dtmf-type = rfc4733

<webrtc2sipGatewayLogs.txt><webrtc_client_console_logs_breaker_enabled.txt><webrtc2sipGatewayLogs.pcap>

[vineet srivastava]

hi Mamadou,

   Thanks for the reply.

   I have taken another set of logs for the issue and this time the JS logs are starting from the time we type sipml5 in the browser which gives the logs from starting.

   Just a note that i have tried browser to browser calling and it works. Hence the browser supports the functionality. Its google chrome.

   Attaching the files for your view.

  Waiting for your reply.

Thanks,

Vineet 

[Mamadou]

***ERROR: function: "tnet_transport_tls_set_certs()"
file: "src/tnet_transport.c"
line: "255"
MSG: SSL_CTX_use_certificate_file failed [0,error:0906D06C:PEM routines:PEM_read_bio:no start line]

This means the SSL certificates you're providing are not valid PEM files. Check the archive or wiki pages for how to generate self-signed certificates.

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Mamadou DIOP - Technology Evangelist
Doubango Telecom - Paris, France
http://www.doubango.org
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[vineet srivastava]

HI Mamadou,

  Thanks for the suggestion.

  I created self signed certificates and was able to make call from X-Lite to Web - browser and the call is working fine with audio as had selected OPUS in X-Lite.

  However vice-versa i.e. call from Web-browser is not reaching X-Lite. Looking into the issue.

Thanks,

Vineet