<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Can you post a sip trace of this entire call, the 19 means we are rejecting that m= line, are there 2 m lines, AVP and SAVP to indicate optional secure?<div><br></div><div>Mike</div><div><br><div><html>On Apr 23, 2008, at 3:01 PM, Krzysiek wrote:</html><br class="Apple-interchange-newline"><blockquote type="cite"><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0; "><div bgcolor="#ffffff"><div><font face="Arial" size="2">Hi<span class="Apple-converted-space"> </span><br>I have 2 softphones PhonerLite (they support SRTP via SDES ) and the freeswitch (windows RC1 version) server and I wanted to make secure call between those two endpoints (SRTP).<br>I spend whole day on testing this scenario and my conclusions are:<br>- when the option: <action application="export" data="sip_secure_media=true"/> is uncommented, and both enpoints have enabled SRTP then:<br>1) Initiator of the session sends SIP Invite with a=crypto paramter and supported codecs<br>2) Freeswitch receives SIP Invite and sends SIP Invite to the receiver (also with the crypto)<br>3) Receiver receives the SIP Invite with the a=crypto parameter and he sends back supported codecs with 200 OK message (but without a=crypto parametr. Is that ok? I'm afraid not)<br>4) Freeswitch sends 200 OK message but witout any codecs: m=audio 0 RTP/AVP 19 and no a= parameters!<br>5) Final result is that the second leg of the session between Freeswitch and receiver has SRTP transport enbaled and the first leg (initiator- Freeswitch) doesn't hear anything - no codecs! However Freeswitch is sending RTP (not SRTP) pacekets to the initiator.</font></div><div> </div><div><font face="Arial" size="2">Could someone explain to me, what is going on, and why freeswitch doesn't forward codecs accepted by the receiver to the initiator?<br>Is it a PhonerLite's bug or freeswitch's? Maybe someone has tested SRTP with the PhonerLite softphone or any other free softphone with srtp support?</font></div><div> </div><div><font face="Arial" size="2">When I uncommented: <param name="Inbound-no-media" value="true"><br>everything works fine. The parameter <action application="export" data="sip_secure_media=true"/> doesn't change anything then (but i cound miss something).</font></div><div> </div><div><font face="Arial" size="2">Thanks for help<br>Chris</font></div>_______________________________________________<br>Freeswitch-users mailing list<br><a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org">http://www.freeswitch.org</a><br></div></span></blockquote></div><br></div></body></html>