Sure this works, you can set rtp or srtp independently to every call leg (if FS is in media path) and even mix them in a conference.
Best regards Peter NOx-WHV schrieb: > Hi, > > i have a problem using SRTP Encrytion. All intern calls are SRTP encrypted. > Some of my Gateway don´t support SRTP encryption. > > In my dialplan I now set the sip_secure_media to false. > > <action application="set" data="sip_secure_media=false"/> > > It works. But is there any chance to encrypt the call on one side and use a > unencrypted call on the other side of the freeswitch? > > Phone -->(SRTP)--> Freeswitch -->(RTP)--> Gateway > > Thanks for help > > NOx > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
