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

[Marc Petit-Huguenin]

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On 09/23/2012 08:16 AM, Daniel Pocock wrote:
> 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)?

WebRTC did the right thing in mandating G.711 (uLaw & Alaw) and Opus in all
implementations:

https://www.ietf.org/mail-archive/web/rtcweb/current/msg05267.html

If all open source/free software implementers pledge to implement those two
codecs in their implementation, then we hopefully can stop having this
discussion and focus on the other points.

[...]

> 
> - 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)

I would like to point out that there is currently a call for comments in
MMUSIC on revising the ICE specification, following the experience accumulated
this last few years:

https://www.ietf.org/mail-archive/web/mmusic/current/msg09558.html

- -- 
Marc Petit-Huguenin
Email: marc at petit-huguenin.org
Blog: http://blog.marc.petit-huguenin.org
Profile: http://www.linkedin.com/in/petithug
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