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, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org
