[Free-RTC] Escape from PRISM!

Emil Ivov emcho at jitsi.org
Fri Jun 14 12:55:01 CEST 2013


On Fri, Jun 14, 2013 at 12:36 PM, MJ Ray <mjr at phonecoop.coop> wrote:
> Emil Ivov <emcho at jitsi.org>
>> That said, I don't remember seeing you post on the Jitsi mailing lists,
>> so if you are having any issues, that would be a good place to start
>> looking for help.
>
> I've not posted to the lists because I fear the root problem is that
> Jitsi developers disagree with most SIP-POTS services about STUN and
> they're probably actually correct,

Very, very few VoIP providers rely on standalone STUN for NAT
traversal. I believe ekiga.net and 1&1 are the only ones I've heard of
and they are not really representative (one is mostly meant for a
specific client, the other for a specific network). Most use hosted
NAT traversal and latching. There are two reasons for this:

a) standalone STUN doesn't work
b) when calling POTS lines you are using server-side bandwidth anyway
so there is no point whatsoever in bothering with STUN.

> but that doesn't really help solve
> my problem of having some way to call in/out that doesn't suck like
> ippi did last time I looked (more expensive than a fixed line!)

I assume you are saying this in comparison with Skype, in which case I
don't think what you are saying is accurate. Personally, I've been
using them happily for a while now but I suppose prices depend on
destinations and if the ones you are interested in tend to be more
expensive with one provider, then just try another one. Or use
several. That's the beauty of it all.

There are numerous other options. OVH, onSIP,  sipgate ... I really
think we can fill in a list of about a 100 within a day.

Still, there's a major gotcha with POTS services: if your point was to
escape surveillance as this thread suggests, then there is no way any
kind of POTS termination could be a solution.

Emil

--
https://jitsi.org


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