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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 3, 2014, 1:19 a.m.)
Review request for Asterisk Developers and rnewton.
Changes
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Addressed all comments
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 (updated)
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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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