<div dir="ltr"><br><br><div class="gmail_quote">---------- Forwarded message ----------<br>From: <b class="gmail_sendername">Herold Park</b> <span dir="ltr"><<a href="mailto:heroldpark@gmail.com">heroldpark@gmail.com</a>></span><br>
Date: 2014-08-15 16:09 GMT+09:00<br>Subject: SRTP problem on the FS<br>To: "<a href="mailto:freeswitch-users-request@lists.freeswitch.org">freeswitch-users-request@lists.freeswitch.org</a>" <<a href="mailto:freeswitch-users-request@lists.freeswitch.org">freeswitch-users-request@lists.freeswitch.org</a>><br>
<br><br><div dir="ltr"><span style="font-size:14px;color:rgb(0,0,0);font-family:Arial,Helvetica,'Luxi Sans',sans-serif;white-space:pre-wrap">Hello, everyone!
I'm having trouble with SRTP calls between two internal extensions.
Whenever I configure the phones to require SRTP calls has wrong.</span><span style="font-size:14px;color:rgb(0,0,0);font-family:Arial,Helvetica,'Luxi Sans',sans-serif;white-space:pre-wrap">
<br></span><div><span style="font-size:14px;color:rgb(0,0,0);font-family:Arial,Helvetica,'Luxi Sans',sans-serif;white-space:pre-wrap">If I select "allow SRTP" so that encryption is not mandatory.</span></div>
<div><span style="color:rgb(0,0,0);font-family:Arial,Helvetica,'Luxi Sans',sans-serif;font-size:14px;white-space:pre-wrap">I get only encryption on </span><span style="color:rgb(0,0,0);font-family:Arial,Helvetica,'Luxi Sans',sans-serif;font-size:14px;white-space:pre-wrap">phone to FS in a call.</span></div>
<div><span style="color:rgb(0,0,0);font-family:Arial,Helvetica,'Luxi Sans',sans-serif;font-size:14px;white-space:pre-wrap"><br></span></div><div><span style="font-size:14px;color:rgb(0,0,0);font-family:Arial,Helvetica,'Luxi Sans',sans-serif;white-space:pre-wrap">Here's an example of what's happening:</span></div>
<div><span style="font-size:14px;color:rgb(0,0,0);font-family:Arial,Helvetica,'Luxi Sans',sans-serif;white-space:pre-wrap">
Extension 1003 --- SRTP --> FreeSWITCH -- RTP --> 1004
Extension 1003 <-- RTP ---- FreeSWITCH -- SRTP <-- 1004</span><br clear="all"><div><br></div><div>Phones are LG-ERICSSON IP-8830E model. These phone is supported Secure RTP. These phones are configured to support SRTP, which is AES_CM_128_HMAC_SHA1_80.</div>
<div><br></div><div>FreeSWITCH version is FreeSWITCH-1.2.23.</div><div><br></div><div>conf/default.xml is configed.</div><div><br></div><div><img src="cid:ii_147d3f47aab300b1" alt="본문 이미지 1"><br></div><div><br></div><div>
<p style="padding:0px;margin:0px;color:rgb(0,0,255);font-family:Arial,Helvetica,'Luxi Sans',sans-serif;font-size:14.44444465637207px;white-space:pre-wrap">I've setup the sip_secure_media=true variable as instructed from
</p><p style="padding:0px;margin:0px;color:rgb(0,0,255);font-family:Arial,Helvetica,'Luxi Sans',sans-serif;font-size:14.44444465637207px;white-space:pre-wrap"><a href="https://wiki.freeswitch.org/wiki/Secure_RTP" rel="nofollow" style="color:rgb(0,0,153)" target="_blank">https://wiki.freeswitch.org/wiki/Secure_RTP</a>
</p><p style="padding:0px;margin:0px;color:rgb(0,0,255);font-family:Arial,Helvetica,'Luxi Sans',sans-serif;font-size:14.44444465637207px;white-space:pre-wrap">but it's still not working by SRTP. Any help is greatly appreciated!
</p><p style="padding:0px;margin:0px;color:rgb(0,0,255);font-family:Arial,Helvetica,'Luxi Sans',sans-serif;font-size:14.44444465637207px;white-space:pre-wrap">Kind regards,</p></div><span class="HOEnZb"><font color="#888888">-- <br>
세상은 아름답다.
</font></span></div></div>
</div><br><br clear="all"><div><br></div>-- <br>세상은 아름답다.
</div>