1 Sep
2009
1 Sep
'09
10:19 a.m.
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