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<p>Brian,</p>
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<p>Thank you for the suggestion!</p>
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<p>Tom</p>
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<div id="divRplyFwdMsg" dir="ltr"><font face="Calibri, sans-serif" color="#000000" style="font-size:11pt"><b>From:</b> Brian West <brian@freeswitch.org><br>
<b>Sent:</b> Wednesday, October 26, 2016 10:50 AM<br>
<b>To:</b> Tom Chen<br>
<b>Cc:</b> freeswitch-users@lists.freeswitch.org<br>
<b>Subject:</b> Re: SRTP support</font>
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<div dir="ltr">Yes and I would limit your cipher suites, If you send our full list to a Polycom it has a heart attack and mirrors back the SDP completely and the phone requires a reboot to fix itself. :)</div>
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<div class="gmail_quote">On Wed, Oct 26, 2016 at 12:49 PM, Tom Chen <span dir="ltr">
<<a href="mailto:chentom60@hotmail.com" target="_blank">chentom60@hotmail.com</a>></span> wrote:<br>
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<pre>Hi Brian,</pre>
<pre>Thank you! Should I just add this line to vars.xml to "<span>Accept/Offer SAVP/AVP with SAVP preferred"?</span><div><X-PRE-PROCESS cmd="set" data="rtp_secure_media=<wbr>optional"/> </div><div><br></div></pre>
<pre>Tom</pre>
<pre>>Read the very long description on SRTP in the vars.xml file, It has some
>extensive documentation on this topic. (Hint: were not a proxy.)
On Wed, Oct 26, 2016 at 11:42 AM, Tom Chen <<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">chentom60 at hotmail.com</a>> wrote:
><i> Hello,
</i><span class="">><i>
</i>><i>
</i>><i> By default, Freeswitch seems always translates incoming SRTP packets to
</i>><i> outbound RTP only packets as the INVITE SIP message to Receiver from FS
</i>><i> contains only RTP/AVP audio media in SDP section even if the Caller's
</i>><i> INVITE has RTP/SAVP audio media only.
</i>><i>
</i>><i>
</i>><i> Caller to FS
</i>><i>
</i>><i> Session Initiation Protocol (INVITE)
</i>><i>
</i>><i> . . . .
</i>><i>
</i>><i> Session Description Protocol
</i>><i>
</i>><i> Session Description Protocol Version (v): 0
</i>><i>
</i>><i> Owner/Creator, Session Id (o): 1001 625 1382 IN IP4 10.113.49.31
</i>><i>
</i>><i> Session Attribute (a): rtcp-xr:rcvr-rtt=all:10000
</i>><i> stat-summary=loss,dup,jitt,TTL voip-metrics
</i>><i>
</i></span>><i> Media Description, name and address (m): audio 7078 *RTP/SAVP* 96 97 98
</i><span class="">><i> 99 0 8 101 100 102
</i>><i>
</i>><i>
</i>><i> FS to Receiver
</i>><i>
</i>><i> Session Description Protocol
</i>><i>
</i>><i> Session Description Protocol Version (v): 0
</i>><i> Owner/Creator, Session Id (o): FreeSWITCH 1477472443 1477472444 IN IP4
</i>><i> 172.17.10.23
</i>><i>
</i></span>><i> Media Description, name and address (m): audio 25206 *RTP/AVP* 0 8 101
</i><span class="">><i> 13
</i>><i>
</i>><i>
</i>><i> Is there a way to let FS send SRTP packets to receiver as the receiver can
</i>><i> accept SRTP media only?
</i>><i>
</i>><i>
</i>><i> Tom
</i>><i>
</i>><i>
</i>><i> ______</i></span></pre>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br>
<span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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