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

[Thava Iyer]

> On Feb. 27, 2014, 4:02 p.m., Matt Jordan wrote:
> > /branches/11/res/res_http_websocket.c, lines 324-350
> > <https://reviewboard.asterisk.org/r/3248/diff/1/?file=54350#file54350line324>
> >
> >     So, I always get nervous every time I see a 'sanity' check polling loop :-)
> >     
> >     I know Thava took a similar approach on the patch on ASTERISK-23099 without the sanity check:
> >     
> >     +		   if (ast_wait_for_input(session->fd, 100) > 0) {
> >     +			while ((readlen = fread(&(buf[readnow]), 1, MAXIMUM_FRAME_SIZE, session->f)) < 1) {
> >     +			  int  ferr = ferror(session->f);
> >     +			  int feoffile = feof(session->f);
> >     +			  ast_debug(3,"ast_websocket_read() fread error  ferr=%d, feoffile=%d, returnval=%"PRIu32"\n", ferr,feoffile,readlen);
> >     +			}  
> >     +		  }  
> >     
> >     I think your approach of checking for EAGAIN is better - was that to work through the case that you mention in the comments, where the fd says it is ready to be read, but in reality no data is available?
> 
> Moises Silva wrote:
>     Yes the EAGAIN check is exactly because of that AFAIR

I guess here EAGAIN may be necessary because , here (in ws_safe_read), we try to read before checking the fd (ast_wait_for_input)  in some instances within the ast_websocket_read. This patch is clean. 
But, I've a question: what's the purpose of calling fread() with partial lens . Why not use MAX_FRAME_SIZE, so that, the data in the fd or (ssl_buff) can be read in one shot. If fragmented, then call again with (MAX_FRAME_SIZE - readlen ). This way we may avoid too many unnecessary calls to fread() and also avoid,  calling fread() before checking the fd (ast_wait_for_input)..


- Thava


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On March 2, 2014, 7:19 p.m., Moises Silva wrote:
> 
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> 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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