[pjsip] SIP PBX with PJSIP

Ruddy Gbaguidi plugworld at micnes.com
Wed Dec 4 11:57:03 EST 2013

I understand the "limitations" of each one of the solutions.
But to come back to my problem, here is the context:
We are developping a platform that will be used in call center and we 
need one of the application to embed a PBX.
So, softphones can register to it, the PBX can play a file to a caller 
and the PBX can connect one endpoint to another.
So, our application will hook to those functionalities. For example, 
when a register request is received from the PBX, the application will 
do it custom magic and accept or reject the registration.

Now, the challenge is the fact that we want that application to be cross 
platform and work on linux as well as windows.

So, to simply the question, is PJSIP (since it is cross platform) 
suitable to create sudh application. If yes, where do I start ? What 
modules do I need ?
If not, how can we modify asterisk,  freeswitch and any other solution 
to do so ?
Or are there any other avenue there ?

Thanks a lot.

Le 2013-12-04 10:10, Joshua Colp a écrit :
> Shamun Toha Md wrote:
> Hello Joshua,
> Hola,
> Everybody knows today in 2013 and soon will be 2014. Where Asterisk and
> FreeSwitch is largest, but knowing it themselves they still do not have
> SIP DoS attack resolved or embedded as default.
> Nor there documentation is friendly to set this up in a reliable way, 
> as
> a result many vendors cant just go to cloud with Asterisk or 
> FreeSwitch,
> where Skype never have SIP DoS attack they never goes down almost. I do
> not understand this logic of all the community.
> Are you referring to functionality which does automatic analysis to
> determine DoS attacks and block accordingly? Speaking for Asterisk we
> provide the information required to do this, but don't write the
> functionality which does the analysis or does the blocking. We believe
> that people who focus on this as an entire project are better suited
> in doing so. I think more documentation on this could certainly be
> useful for everyone though.
> > Asterisk should by default either kill the SIP DoS attacks or embed
> this by default in new release.
> > Asterisk know they are big and smart too, but still today Asterisk
> has no better friendly way to do mplayer, vlc, ffmpeg, gstreamer rtp
> embedded so that developers with normal knowledge can kick Google techs
> with those combinations
> There are legal reasons as well as licensing reasons. We provide the
> APIs and such to allow this to happen, though.
> > Asterisk vs Google Hangout where is Asterisk for Video? Google
> Hangout started ages after pjSIP, Asterisk history but look today 
> Google
> Hangout is the number 1 breaks any kind of NAt, ICe, Stun, Turn issues,
> its like giving a kid of 2 years old with google hangout and tell break
> the firewall he can. But putting Asterisk and putting rocket science
> still this NAt issues are not yet solved nor anyone cares
> Asterisk 11 and above use pjnath to provide ICE/STUN/TURN
> functionality. It was specifically done for WebRTC but it can be used
> elsewhere.
> As for better video support this requires core changes to fully
> accomplish, which is why it has not yet been done. Higher things have
> been on the list.
> I really do not understand Asterisks developers, are they really lazy 
> or
> retired or have less knowledge compared to Google Hangout?
> We can't just drop everything and work on new features. We have an
> extremely large existing user base we have to support and we have a
> limited number of resources. As for how we determine what we'll work
> on: We discuss the future of Asterisk every year at our developer
> conference. This ensures we are providing what developers and
> deployers want. The better NAT support you mention has never been
> mentioned, and video (or better media handling in general) has come up
> but not as a huge thing. You can see the notes for this year's
> conference here:
> https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2013
> Cheers,

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