Hi Jesus,
one more newbie qoffeesip question :-) -- does this have to be installed
separately (I mean qoffeesip)? Or is part of your repository as well?
Thanks,
Daniel
On 5/13/13 4:58 PM, Jesús Pérez Rubio wrote:
Hi Daniel,
We have something like you're asking for in the Github repository.
First lines of this Quickstart guide show it
(https://quobis.atlassian.net/wiki/display/QoffeeSIP/Quick+start+guide).
Only two steps are needed:
- Clone the repo: /git clone https://github.com/Quobis/QoffeeSIP.git/
- Copy examples/webphone/dist/* content to your Apache server.
It's a simple webphone that we use to develop the stack. Another
simplest one is also included in examples folder to help web
developers to include it in their site.
Nothing else, we're here if somebody needs something.
Regards. :)
2013/5/13 Daniel-Constantin Mierla <mico...@gmail.com
<mailto:mico...@gmail.com>>
Hello Jesus,
On 5/13/13 11:43 AM, Jesús Pérez Rubio wrote:
Hi Daniel, we've developed a Javascript SIP stack which supports
WebRTC
have you published any out-of-the-box phone built from your stack?
Something like one can take and in few config steps it can get the
phone on their web page, without needing to code java script.
Cheers,
Daniel
and we use Debian as base OS and Kamailio as SIP proxy. Some
notes about the enviroment in case it could help you:
- Kamailio stable (4.0) version included in official repo works fine.
- Site:
http://www.kamailio.org/wiki/packages/debs#latest_kamailio_40_release
- Howto:
https://quobis.atlassian.net/wiki/display/QoffeeSIP/Server+configurations
- I've tested it with QoffeeSIP and JsSIP some days ago and
there is no problem.
- I've been playing with "resiprocate-turn-server" package but I
had problems. It could be related with our client but we should
take a look. I didn't give a try to Google TURN server but I'm
going to do it, I'll keep you updated.
PS: I'm co-mentor with you in GSoC, so we can speak about in in
IRC channel.
2013/5/13 Daniel Pocock <dan...@pocock.com.au
<mailto:dan...@pocock.com.au>>
I'd like to write a brief blog about the status of WebRTC in
Debian,
with a focus on SIP
I understand Kamailio 4.0.1 is already in unstable, is that
recommended
for potential WebSocket users? Has anybody else written any
quickstart
blog about WebRTC with that particular version, possibly with
examples
that are consistent with the Debian usage?
For client side, SIPml5 is packaged, and I've had discussions
with the
JsSIP guys about packaging.
For TURN, has anybody tried the TURN server project from
Google code?
It appears more advanced than the existing two TURN servers
in Debian
(e.g. it has database-backed authentication)
http://code.google.com/p/rfc5766-turn-server/
and there is a package in progress:
http://mentors.debian.net/package/rfc5766-turn-server
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--
Jesús Pérez
VoIP Engineer at Quobis
Fixed: +34 902 999 465
Site: http://www.quobis.com <http://www.quobis.com/>
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--
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http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda>
-http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
*http://asipto.com/u/katu *
_______________________________________________
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--
Jesús Pérez
VoIP Engineer at Quobis
Fixed: +34 902 999 465
Site: http://www.quobis.com <http://www.quobis.com/>
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
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