[Freeswitch-users] SIP TLS (and SRTP)

[Rupa Schomaker]

hmm...  When I put that section in, I put the wrong filename.  It should be
directory/default.xml.  I'll update the wiki.  Also, it is a valid
configuration to support tls but not srtp.  I'll put a bit of a discussion
in there talking about that.

Setting sip_secure_media to true requires the endpoint do srtp.  There is no
way (that I know of) to say "do srtp if possible but if not fallback to
clear".  zrtp does fallback to clear if it can't negotiate keys.  But zrtp
is supported by far fewer endpoints and no hardphones (as of yet).

On Wed, Jul 15, 2009 at 4:57 AM, Tzury Bar Yochay
<tzury.by at reguluslabs.com>wrote:

> Hi all,
>
> I was following the instruction found at
> http://wiki.freeswitch.org/wiki/SIP_TLS
> When I got to Step 4 I saw that instruction of editing the dial-string.
> However, in my conf/dialplan/default.xml I did not found any matched entry
> .
>
> Version I am using:
> typing version at my FS CLI yields: FreeSWITCH 1.0.trunk (14144M)
>
> thanks,
> Tzury Bar Yochay
>
>

-- 
-Rupa
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