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<DIV><FONT face=Arial size=2>Hi <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></BODY></HTML>