[Freedombox-discuss] FreedomBox VoIP/IM/Comms (SIP and Jabber), Jingle Nodes, SIP RELOAD
daniel at pocock.com.au
Thu Sep 27 11:21:51 UTC 2012
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I've proposed a session on this issue for the upcoming Paris Mini-Debconf:
It would be really useful to understand if other people with an
interest in this area will be attending and/or would like to
contribute to the planning/agenda for such a session
On 23/09/12 17:16, 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
>>> <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:
>>> 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
> 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
> - 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.
> _______________________________________________ Freedombox-discuss
> mailing list Freedombox-discuss at lists.alioth.debian.org
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