[Freedombox-discuss] FreedomBox VoIP/IM/Comms (SIP and Jabber), Jingle Nodes, SIP RELOAD

[daniel@pocock.com.au (Daniel Pocock)]

On 19/09/12 23:31, Wookey wrote:
> +++ Ramana Kumar [2012-09-19 15:37 +0100]:
>>    On Wed, Sep 19, 2012 at 3:22 PM, Daniel Pocock <[1]daniel at pocock.com.au>
>>    wrote:
>>
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>>      On 19/09/12 14:32, Ramana Kumar wrote:
>>      > This page might be mildly relevant:
>>      > [2]http://libreplanet.org/wiki/Group:Skype_Replacement
>>
>>      I heard about FSF Europe running a Skype Replacement event on the
>>      weekend, testing the different clients against each other.  Their goal
>>      is very general, moving people from closed source to Free software
>>
>>    Any more info about this? Did they find anything that worked?
> 
> Results here: etherpad.fsfe.org/RX7S6q45gQ
> 
> The testing was of a fairly basic nature - install stuff, get free
> SIP or jabber account (or use existing one) and see if you can
> talk/chat/video with people. representative of 'normal people' trying
> this out. The SIP/XMPP account/server used was not recorded in most
> cases, which I think is insuficient for repeatable tests.
> 
> More combinations failed than worked, but some worked. Most of my
> connecitons failed even though I have used my SIP hardphone
> successfully for years (seemed to be local DNS failure), and my n900
> SIP successfully in many other contexts.
> 
> It was a worthwhile start but I think there is enormous scope for
> further such sessions, including using servers under our control.
> 

I would certainly like to be involved in that and contribute what
resources I can to support it

I believe the testing needs to be a little bit more scientific and not
just take the `black box' approach, assessing each product on the
following perhaps:

- supported codecs (e.g. patent free, suitable for mobile, ...)  - and
which products support the codecs that other products use (matrix)?

- how easy is it for user to get the `right' codec for their call?  Is
it automatic (Skype has dynamic selection of codec based on bandwidth,
many free software products don't do this)

- which solutions support NAT traversal?  Is every permutation of NAT
and firewall environment tested?  ICE/STUN/TURN is good for this, but
client software support is not always 100% (e.g. Jitsi supports ICE with
Jabber, but not with SIP.  Lumicall supports ICE, but there are some
shortcomings, just look for the FIXMEs in the code to find out what they
are)

- how should users register for a truly `Free' VoIP network?  Virtually
all existing clients require users to both choose a provider and set up
a SIP account, and it is always more difficult than setting up Skype

- if there are many independent providers and small businesses running
their own private VoIP, and the client software does somehow allow the
users to connect to their chosen provider, they could be left in a
little island (that is often the case today).  How can they easily
interconnect to users with different providers?  This is one of the
questions I've been trying to solve with my `Federated VoIP' pages:
http://www.opentelecoms.org/federated-voip

- what solutions are suitable for both corporate and private use?  A
lasting solution must be universal.  Microsoft now has both the
corporate domain (Lync) and consumer (Skype) and will most likely try to
join them together more closely.  This is scary.