[Freeswitch-users] SRTP support

[Brian West]

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> 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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-- 

*Brian West*
brian at freeswitch.org


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