[Free-RTC] Escape from PRISM!

[Lee Azzarello]

Awesome! Glad it worked. If you wanna ring me, I'm lee at ostel.co.

The ZRTP linphone support is allegedly compiled in the binary
distributed via f-droid. I'm working on getting a version for iOS out.

-lee

On Tue, Oct 8, 2013 at 10:36 AM, Sylvain Berfini
<sylvain.berfini at linphone.org> wrote:
> Hi Lee,
>
> I tried your service with Linphone (one I compiled myself with ZRTP enabled,
> the one in the Google Play Store doesn't embed it yet), and it works great
> (at least with your echo service).
>
> Regards,
> Sylvain
>
>
> 2013/6/14 Lee Azzarello <lee at rockingtiger.com>
>>
>> I operate a secure SIP service at https://ostel.co
>>
>> I'm using rtpproxy for NAT traversal. Since the audio stream is
>> encrypted with ZRTP, the server doesn't have access to the content of
>> calls.
>>
>> During the course of this project, I have discovered numerous
>> complications. The nut I'm trying to crack right now is SIP
>> federation. I have alias calling working via SRV records but calling
>> between SIP domains is pending. The closest I came so far is a call
>> between ostel.co and ekiga.net succeeded but one of the two providers
>> is mangling the rtp stream so that the ZRTP key agreement never
>> happens.
>>
>> I'll post my progress when I get that nailed down. I also have a
>> milestone on my roadmap to build a Chef cookbook to automate the
>> assembly of a full secure SIP stack.
>>
>> Regards,
>> Lee
>>
>> On Tue, Jun 11, 2013 at 11:12 AM, MJ Ray <mjr at phonecoop.coop> wrote:
>> > Daniel Pocock <daniel at pocock.com.au>
>> >> There are also some sore points, e.g. the Debian 7 release includes an
>> >> Empathy version that only works with Google's gmail TURN server and
>> >> not the Debian packaged TURN server.  Things like that are going to
>> >> continue frustrating users for some time to come but will eventually
>> >> be ironed out.
>> >
>> > I think it's a far bigger frustration that we can't actually call
>> > anyone who's not using the same system yet, and sometimes it won't
>> > even work on the same system, while the debugging logs on either
>> > client or provider often seem to be insufficient to actually find out
>> > why something doesn't work.
>> >
>> > I've got jitsi working over XMPP, but damned if I can see how to let
>> > others call in, or call outside the same system with it :-/
>> >
>> > The lumicall site says things like "If the VoIP provider supports SIP,
>> > TLS and ICE, it should work" which is great, but what UK VoIP DDI
>> > provider supports all three?  In short, part of the necessary
>> > information for mass use seems to be missing still.  If anyone has it
>> > working, please post step-by-step instructions so others can set
>> > things up and help spread the spoken word.
>> >
>> > Thanks,
>> > --
>> > MJ Ray (slef), member of www.software.coop, a for-more-than-profit
>> > co-op.
>> > http://koha-community.org supporter, web and library systems developer.
>> > In My Opinion Only: see http://mjr.towers.org.uk/email.html
>> > Available for hire (including development) at http://www.software.coop/
>> > _______________________________________________
>> > Free-RTC mailing list
>> > Free-RTC at lists.fsfe.org
>> > https://lists.fsfe.org/mailman/listinfo/free-rtc
>> _______________________________________________
>> Free-RTC mailing list
>> Free-RTC at lists.fsfe.org
>> https://lists.fsfe.org/mailman/listinfo/free-rtc
>
>
>
> _______________________________________________
> Free-RTC mailing list
> Free-RTC at lists.fsfe.org
> https://lists.fsfe.org/mailman/listinfo/free-rtc
>