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<h2><a href="https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial?focusedCommentId=11338315#comment-11338315">Secure Calling Tutorial</a></h2>
<h4>Comment edited by <a href="https://wiki.asterisk.org/wiki/display/~gilles">gilles</a>
:</h4>
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<h4>Changes (1)</h4>
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<tr><td class="diff-snipped" >...<br></td></tr>
<tr><td class="diff-unchanged" >But in Wireshark, I can see that, on the server side, the signaling goes through port 5061. <br> <br></td></tr>
<tr><td class="diff-added-lines" style="background-color: #dfd;">And last but not least, many people use the following variables in their dialplan when setting sRTP : <br>_SIP_SRTP_SDES=1 <br>_SIPSRTP=enable <br>_SIPSRTP_CRYPTO=enable <br>Why don't you use it ? Are they unnecessary ? <br>Do you know if they are documented somewhere ? <br> <br>When I set SRTP (both on clients and server sides), the only difference I can see with TLS is that the media goes through the server and not straight to the other client. Like with TLS only, for the media, I don't see the UDP/RTP layer anymore, but UDP/Data. Also, with or without sRTP set, UDP/Data begins with 800 hex code, like with UDP, that makes me think sRTP doesn't work for me. Also, I always get this error message : <br>[Jan 28 17:29:14] WARNING[9646]: res_srtp.c:338 ast_srtp_unprotect: SRTP unprotect: authentication failure <br>What does it mean ? What do I need to do to make it work ? <br> <br></td></tr>
<tr><td class="diff-unchanged" >Thanks for your help, <br>Gilles <br></td></tr>
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</div> <h4>Full Content</h4>
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<p>I tried my registered biz version (2.30) and the free version of Zoipper (2.36) for Windows, in both this TLS Certificate file option isn't there. However, I can see it disabled in the Linux version (1.18). So I'm now using PhonerLite where I could find it. Thanks for that.</p>
<p>But, by default, TLS works fine without the user certificate.<br/>
And I can't see any client certificate request from the server in Wireshark, should I set it somewhere ?<br/>
I couldn't find anything in sip.conf or in Asterisk 1.8 doc about it.</p>
<p>Also, when my client registers, I get something like :<br/>
– Registered SIP 'phonerlite' at 10.100.5.61:49296<br/>
But in Wireshark, I can see that, on the server side, the signaling goes through port 5061.</p>
<p>And last but not least, many people use the following variables in their dialplan when setting sRTP :<br/>
_SIP_SRTP_SDES=1<br/>
_SIPSRTP=enable<br/>
_SIPSRTP_CRYPTO=enable<br/>
Why don't you use it ? Are they unnecessary ?<br/>
Do you know if they are documented somewhere ?</p>
<p>When I set SRTP (both on clients and server sides), the only difference I can see with TLS is that the media goes through the server and not straight to the other client. Like with TLS only, for the media, I don't see the UDP/RTP layer anymore, but UDP/Data. Also, with or without sRTP set, UDP/Data begins with 800 hex code, like with UDP, that makes me think sRTP doesn't work for me. Also, I always get this error message :<br/>
<span class="error">[Jan 28 17:29:14]</span> WARNING<span class="error">[9646]</span>: res_srtp.c:338 ast_srtp_unprotect: SRTP unprotect: authentication failure<br/>
What does it mean ? What do I need to do to make it work ?</p>
<p>Thanks for your help,<br/>
Gilles</p>
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