Yes and I would limit your cipher suites, If you send our full list to a Polycom it has a heart attack and mirrors back the SDP completely and the phone requires a reboot to fix itself. :) On Wed, Oct 26, 2016 at 12:49 PM, Tom Chen <chentom60@hotmail.com> wrote:
Hi Brian,
Thank you! Should I just add this line to vars.xml to "Accept/Offer SAVP/AVP with SAVP preferred"? <X-PRE-PROCESS cmd="set" data="rtp_secure_media=optional"/>
Tom
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 at hotmail.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> 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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