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 <chentom60@hotmail.com> wrote:
Hello,
By default, Freeswitch seems always translates incoming SRTP packets to outbound RTP only packets as the INVITE SIP message to Receiver from FS contains only RTP/AVP audio media in SDP section even if the Caller's INVITE has RTP/SAVP audio media only.
Caller to FS
Session Initiation Protocol (INVITE)
. . . .
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): 1001 625 1382 IN IP4 10.113.49.31
Session Attribute (a): rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
Media Description, name and address (m): audio 7078 *RTP/SAVP* 96 97 98 99 0 8 101 100 102
FS to Receiver
Session Description Protocol
Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): FreeSWITCH 1477472443 1477472444 IN IP4 172.17.10.23
Media Description, name and address (m): audio 25206 *RTP/AVP* 0 8 101 13
Is there a way to let FS send SRTP packets to receiver as the receiver can accept SRTP media only?
Tom
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