[Freeswitch-users] SRTP support

[Brian West]

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 at 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
> *>>>* ______*
>
>
>


-- 

*Brian West*
brian at freeswitch.org


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