[asterisk-dev] [Code Review]: Asterisk WebRTC Wiki Page
Joshua Colp
reviewboard at asterisk.org
Wed Aug 22 07:38:54 CDT 2012
> 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.
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
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On Aug. 22, 2012, 6:34 a.m., Joshua Colp wrote:
>
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2080/
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>
> (Updated Aug. 22, 2012, 6:34 a.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Summary
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>
> 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
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> Diff: https://reviewboard.asterisk.org/r/2080/diff
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>
> Testing
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>
> Thanks,
>
> Joshua
>
>
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