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[liberationtech] AES-encyrpted telephony in Iran?

Adam Fisk a at
Wed Jun 13 15:59:37 PDT 2012

I would just point out Nathan that both Skype and PrivateGSM should be
doing all the NAT/firewall traversal necessary to create direct P2P
connections in about 90% of cases. If anything Skype's likely able to
cross NATs in more cases, so the participants in the conversation
should be observable even more frequently. I would argue that "it will
stream the RTP media stream through that same server" is only the case
in more like 10% of cases - both should (and certainly Skype does)
stream RTP directly between peers.

Totally agreed that the overall user experience with Skype should be
much much better though.


On Wed, Jun 13, 2012 at 4:09 AM, Nathan of Guardian
<nathan at> wrote:
> On 06/13/2012 01:24 AM, Naiz Mudin wrote:
>> We must be prepared, in my country, to communicate with one another
>> without fear of the govermnment having automatic access to our data, as
>> is the case currently, whenever we (for example) use Skype VOIP outside
>> of the country.
> One question Naiz - are you using Skype on mobile phones today to
> communicate, or only from desktops? Is this from 3G or wifi?
> I ask because I think that it is important to that while PrivateGSM
> software is technically very sound, there are some additional risks and
> perhaps limitations when it comes to running VoIP, and even more so
> secure VoIP, on a 3G network such as is available in Iran.
> Skype is tolerated in many places in the world where encryption and VoIP
> are considered illegal. Skype is allowed both because it is very
> difficult to block (it uses many different IPs/hosts to connect in a
> peer to peer like manner), but also because it is understood that most
> people are using it to talk with family and conduct business.
> If you use a VoIP system like PrivateGSM, or any SIP/RTP VoIP software,
> it will look very different on the network than Skype. It will most
> likely connect to one server, with a single IP address, for the SIP
> portion (registration, call setup), and then it will stream the RTP
> media stream through that same server. With secure VoIP that SIP stream
> will be secured using SSL or TLS, the RTP media stream using something
> like ZRTP or SRTP (basically the "AES encrypted" part you referenced).
> If two people are calling each other using this service, it will be very
> clear who is talking to whom and when. In some cases even, the RTP media
> stream will stream directly between the two people/devices
> communicating, clearly tying the two IP addresses together. If those IPs
> are connected to a registered SIM card, then you see how this can be
> problematic.
> With Skype, this is hidden in the cloud a bit more because there are
> millions of people, though obviously Skype has many, many security
> issues, and can suffer from this same direct IP-to-IP problem. Still,
> Skype is tolerated, while Secure VoIP may not be.
> Finally, there is another, more practical problem, which is speed or
> "latency". Skype is really good at low bandwidth connections, while
> Secure VoIP, mostly designed for enterprise/business use in the west, is
> not. PrivateGSM is very good as Jacob mentioned, so it may be able to
> adjust its call quality to the bandwidth/latency of your network, but
> your expectations should be set a bit lower, and amount of patience of
> your users higher... meaning, don't expect Skype simplicity and quality
> right away.
> Best,
>  Nathan
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Adam Fisk | |

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