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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3248/">https://reviewboard.asterisk.org/r/3248/</a>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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?</pre>
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<p>- Matt Jordan</p>
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<p>On March 2nd, 2014, 7:19 p.m. CST, Moises Silva wrote:</p>
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<div>Review request for Asterisk Developers and rnewton.</div>
<div>By Moises Silva.</div>
<p style="color: grey;"><i>Updated March 2, 2014, 7:19 p.m.</i></p>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21930">ASTERISK-21930</a>,
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23099">ASTERISK-23099</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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)</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/11/res/res_http_websocket.c <span style="color: grey">(409360)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3248/diff/" style="margin-left: 3em;">View Diff</a></p>
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