[Spce-user] having trouble with provisioning jitsi on mr3.2.1

Thu Apr 3 21:43:04 EDT 2014

Hi Günter,

Thanks a lot for your feedback! You're pretty spot-on with some points
you raised. Yes, Sipwise is not like a small company anymore, so we need
to generate a significant revenue year by year to continue what we're
doing and to grow further. To do so, we sell commercial solutions, and
with that revenue, we're able to hire even more people capable of
supporting and enhancing the really great things we're doing in the open
source VoIP area.

However, Sipwise is one of the few companies actively investing money,
manpower and other resources into projects like Jitsi, Kamailio and Sems
for those guys to be able to enhance further, strengthening the overall
eco-system. When you're seeing those projects announcing fancy new
features, they are usually sponsored by somebody, which at a great
extend is ourselves. This thrives the overall ecosystem, and everybody
is profiting from that. I was just sitting together with Emil from the
Jitsi project today, discussing ways how to install their sexy video
bridging solution in the sipwise spce in a way that it's just going to
work plug-and-play for all of you guys. So you probably MIGHT
underestimate what we're actually doing in the overall open source
landscape. And that MIGHT be our fault not communicating publicly what
we're actually up to.

We're at this very moment at the Kamailo World Conference in Berlin
(which we're sponsoring to make it happen), and we're getting a lot of
great feedback from huge companies like Orange, who are using our open
source software components to serve their customers, and that makes us
really proud, although it doesn't actually get us any revenue.

Having said that, we love to experiment with our own technology, and it
doesn't necessarily mean we're going to provide 1:1 how-tos on
everything we do. But you know what? It's an open source project, and of
course we're relying on you guys to spread it by word-of-mouth, and by
writing your own tutorials and how-tos on how to do stuff with the
system. We're more than happy to help the community out if you guys are
having issues with certain scenarios, but then again, we're all sitting
in the same boat, so the more you contribute in terms of documentation
etc, the more it will spread, the more people will contribute, and the
faster it's going to evolve.

To answer some of your questions: Yes, one of our devs is currently
working on docker.io containers for spce (we're already supporting
vmware, virtualbox, vagrant an Amazon AWS AMIs). Yes, we just released
the open source rtpengine, the only highly-scalable dtls/srtp-based
media engine capable of interconnecting webrtc clients with the legacy
sip world (and basically a 100% of the people demoing webrtc at Kamailio
Word are actually using it), and it will soon be part of the standard
web interface within the CE, together with a websocket-based
chat-client. We just sponsored carbon-copy and jabber-search extensions
in Jitsi, and we're working on getting the really cool Jitsi-Videobridge
to be able to be used via plain SIP too. And then again, if you want
anything faster/earlier, get in touch with us to sponsor one or another
feature, or if you're serious with your customers, buy a PRO version
with HA and all the other fancy stuff, which directly goes back to and
helps other open source projects you might be using already.

So, if it took you a while to find our project, help us reduce this time
for other users by spreading the word, and by writing/blogging about
different use cases and experiments and whatever you're doing with the
software!

Andreas

On 04/03/2014 12:53 PM, chymian wrote:
> thanks andreas, that helped.
> 
> and yes, I noticed that the skype-like-howto is a bit aged, but for getting my hands on your great software, I was looking for a howto for a federated sip setup – since this train is accelerating fast – and this was the closest I found. 
> It would be nice, if sipwise could update this, or write an federated sip howto – there is a big demand out there (not only since M$ took over skype).
> i.e. I will integrate FS, video conferences, WebRTC, etc. in my websites (project-mgmt & collaboration/social/group-sites) since I consider it a must for an state-of-the-art collaboration-site. and not all the services will be given away free, so, there is the potential to be a future customer of yours, needing billing & prepaid services.
> 
> in that light, I loved the ”tv over webrtc" blog, but again, just a white paper, and no howto. I know, you as company have to make money and want/have to sell your solutions & expertise – on the other hand, the more you give, the more you will get back. especially in the OSS world, it’s easy to to make yourself an good name and state your quality (look at debian) and generate money out of being known widely. and the big point, activate even more community and harvest the synergy.
> 
> since a couple of weeks, since I was looking at federated sip (kicked of by danial pocock’s fed. sip blog for debian) I found all types of solutions, but sipwise didn’t appear on my radar just till the end of my search – after 4 weeks. and that, while I was looking exactly for a solution like yours… 
> jitsi was all over the place – they have sexy solutions, well documented and a big community, flooding the interested ones with a lot of howtos.
> 
> as an newcomer to the "roll your own VoIP-service” world, my impression so far: sipwise has a really great solution, but doesn’t give information and howtos to the community willingly, holding back a lot of “sexy” possible solutions (why?) and therefore not attracting a lot of people, not drawing a lot of attention towards themselves.
> in my impression, you appear as a specialized business company, who’s "only" interest is, to sell expertise & solutions and also “have a OSS” version, for the very interested ones, who are willing and can afford to spent very much time to dig in and set up even the easiest solutions. I'm aware, that the SIP/telephony-business is a complicated one, but not everybody want to be a full blown telco… and in my opinion, sipwise has the possibility to be both, a full blown telco-specialist and an easy to go, fed. sip provider for the “masses” - if you want that to be that, and not leave that marketshare to jitsi, sipexecs, a.o. (which is doing the right thing, in integrating it’s self in clearos and do have a lot of howtos at hand)
> 
> so on my wish list would be a debian based, zentyal.org style SMB-server, with integrated fed. SIP, webRTC & videoconferences, (running on docker.io containers), a little bit like /http://suite.tiki.org/[1] is doing it on centOS (w/o docker). or you even could team up with univention, for the money-market…
> 
> 
> just my 2 cent
> 
> und danke für sipwise:CE!
> 
> guenter
> 
> 
> 
> PS: the URL has an typo, and for the mail-archive:
> https://your-ip:1443/device/autoprov/static/jitsi?user=${username}&pass=${password}&uuid=${uuid}
> 
> 
> 
> 
> Am Mittwoch, 2. April 2014, 11:23:08 schrieb Andreas Granig:
>> Hi,
>>
>> The post you're referring to is pretty old and not up-to-date anymore,
>> the URL you should try is:
>>
>> https://your-ip:1443/device/autoprov/static/jitsi?user=${username}&pass=b${password}&uuid=${uuid}
>>
>> Andreas
>>
>> On 04/02/2014 10:49 AM, günter wrote:
>>> BUMP
>>>
>>> anyone? please!
>>>
>>> Am Dienstag, 1. April 2014, 01:12:28 schrieb chymian:
>>>> hey,
>>>>
>>>> fairly new to sipwise, I could need some advice here on provisioning.
>>>>
>>>> the setup is mr3.2.1 with the “skype-like-howto” /http://www.sipwise.com/news/technical/byov-skype-replacement//[1]
>>>>
>>>> A) fresh-install on a virtual machine with a cloudserver jiffybox.de
>>>> B) used the vagrant file to UP a testmachine - just as reverence.
>>>>
>>>> on both occasions, the jitsi client fails to provision itself.
>>>>
>>>> updated jitsi already from debian version 2.4.4997-1 to the latest nightly: 2.5.5168-1
>>>> on my workstation: debian jessie
>>>>
>>>>
>>>>
>>>> 1.) when using the URL: https://192.168.0.122/jitsi?user=${username}&pass=${password}&uuid=${uuid} (IP changed)
>>>> 	the jitsi config get’s the first http-page from the self-care web-interface included. :(
>>>>
>>>>
>>>>
>>>>
>>>> 2.) when I use the same URL with the provisioning port 2443, I get an err.-msg. in /var/log/ngcp/oss.log
>>>> https://192.168.0.122:2443/jitsi?user=${username}&pass=${password}&uuid=${uuid}
>>>>
>>>> Apr  1 00:54:47 spce apache2[6535]: [info] [client 192.168.44.1] Connection to child 0 established (server myserver:443)
>>>> Apr  1 00:54:47 spce apache2[6535]: [info] Seeding PRNG with 656 bytes of entropy
>>>> Apr  1 00:54:57 spce apache2[6535]: [info] Initial (No.1) HTTPS request received for child 0 (server myserver:443)
>>>> Apr  1 00:54:57 spce apache2[6535]: [error] [client 192.168.44.1] File does not exist: /var/www/jitsi
>>>> Apr  1 00:54:57 spce oss: 192.168.44.1 - - [01/Apr/2014:00:54:57 +0200] "POST /jitsi HTTP/1.1" 404 1290 "-" "Jitsi/2.5.5168"
>>>> Apr  1 00:55:02 spce apache2[6535]: [info] [client 192.168.44.1] (70007)The timeout specified has expired: SSL input filter read failed.
>>>> Apr  1 00:55:02 spce apache2[6535]: [info] [client 192.168.44.1] Connection closed to child 0 with standard shutdown (server myserver:443)
>>>>
>>>>
>>>> it’s the same game on both hosts.
>>>> no account is created an it starts with the account-wizzard.
>>>>
>>>> do I miss something?
>>>> how is that supposed to work?
>>>>
>>>> ANY help appreciated
>>>> TIA
>>>> günter
>>>>
>>>> --------
>>>> [1] http://www.sipwise.com/news/technical/byov-skype-replacement/
>>>
>>>
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>>>
>>
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> 
> 
> --------
> [1] http://suite.tiki.org
> 
> 
> 
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