[asterisk-dev] [Code Review]: Asterisk WebRTC Wiki Page

[Joshua Colp]

> On Aug. 22, 2012, 7:35 a.m., lathama wrote:
> > Can you expand the SRTP content or link to https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
> 
> lathama wrote:
>     Also note that manager needs enabled as does webenabled.  Maybe the webserver should work without the manager at some point.
> 
> Joshua Colp wrote:
>     Expand it with what? Really all you have to do is ensure that SRTP support is built and set encryption=yes. The vast majority of what is mentioned on the secure calling tutorial is for SIP TLS setup.
> 
> Joshua Colp wrote:
>     Your correct about manager needing to be enabled is not actually correct. I've done all of my testing without manager enabled.

And by "your correct" I meant "your statement". Naturally. >_>


- Joshua


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2080/#review6935
-----------------------------------------------------------


On Aug. 22, 2012, 6:34 a.m., Joshua Colp wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2080/
> -----------------------------------------------------------
> 
> (Updated Aug. 22, 2012, 6:34 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> I've created a wiki page at https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support which describes the WebRTC a bit, along with the configuration items and potential issues that may occur. Any feedback is welcome!
> 
> 
> Diffs
> -----
> 
> 
> Diff: https://reviewboard.asterisk.org/r/2080/diff
> 
> 
> Testing
> -------
> 
> 
> Thanks,
> 
> Joshua
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20120822/77a585c1/attachment.htm>