[asterisk-dev] [Code Review] 3248: Fix for WebRTC over WSS not working

[Matt Jordan]

> On March 11, 2014, 1:09 p.m., Matt Jordan wrote:
> > Hey Moy - I'm thinking of cutting release candidates this week or the next, and I'd love to get this into 11.9.0+. Any chance you can commit this sometime this week?
> 
> Moises Silva wrote:
>     Not sure if you noticed I did a merge already, except for the latest comments that came after the merge. I can have a proper look at the remaining comments Saturday morning for sure, but not before, sorry :(

Ah, I did miss that. Great!

The auto-close stuff in the tools was probably broken around that point, so I thought this was still open. If you want to leave it open for the minor cleanups that'd be fine, otherwise, feel free to close it out.


- Matt


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On March 2, 2014, 7:19 p.m., Moises Silva wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3248/
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> 
> (Updated March 2, 2014, 7:19 p.m.)
> 
> 
> Review request for Asterisk Developers and rnewton.
> 
> 
> Bugs: ASTERISK-21930 and ASTERISK-23099
>     https://issues.asterisk.org/jira/browse/ASTERISK-21930
>     https://issues.asterisk.org/jira/browse/ASTERISK-23099
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> Several fixes for the WebSockets implementation in res/res_http_websocket.c
> 
> * Flush the websocket session FILE* as fwrite() may not actually guarantee sending
>   the data to the network. If we do not flush, it seems that buffering on the SSL
>   socket for outbound messages causes issues
> 
> * Refactored ast_websocket_read to take into account that SSL file descriptors
>   may be ready to read via fread() but poll() will not actually say so because
>   the data was already read from the network buffers and is now in the libc buffers
> 
> This should fix an issue that I have experienced and other users may have reported [1][2][3], where
> secure websockets wouldn't work, messages seem to not make it into Asterisk
> 
> [1] http://lists.digium.com/pipermail/asterisk-users/2013-August/280175.html
> [2] https://issues.asterisk.org/jira/browse/ASTERISK-21930
> [3] https://issues.asterisk.org/jira/browse/ASTERISK-23099
> 
> 
> Diffs
> -----
> 
>   /branches/11/res/res_http_websocket.c 409360 
> 
> Diff: https://reviewboard.asterisk.org/r/3248/diff/
> 
> 
> Testing
> -------
> 
> See ASTERISK-21930 for details on other users testing these changes. I did both WS and WSS calls, confirmed audio works with chrome. This patch is for Asterisk 11 as the issue is reported on Asterisk 11, but I tested a few months ago and same issue existed on 12 and trunk. I created my own team branches for those too (/team/moy/webrtc-11, /team/moy/webrtc-12, /team/moy/webrtc-trunk)
> 
> Confirmed working by Sean Bright on Jan 20, 2014 on Asterisk 11 (see ASTERISK-21930 comment)
> 
> 
> Thanks,
> 
> Moises Silva
> 
>

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