SIP PBX with PJSIP

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Hello Joshua,

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.

> 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

> 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

I really do not understand Asterisks developers, are they really lazy or
retired or have less knowledge compared to Google Hangout?


thank you
Best regards
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