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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