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

[Moises Silva]

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https://reviewboard.asterisk.org/r/3248/
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(Updated Oct. 26, 2014, 9:45 p.m.)


Status
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This change has been marked as submitted.


Review request for Asterisk Developers and rnewton.


Changes
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Committed in revision 426209


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
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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
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  /branches/11/res/res_http_websocket.c 409360 

Diff: https://reviewboard.asterisk.org/r/3248/diff/


Testing
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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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