Matthew Jordan
a5df2542c3
Don't attempt a websocket protocol removal if res_http_websocket isn't there
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This patch sets the protocols container provided by res_http_websocket to NULL
when the module gets unloaded and adds the necessary checks when adding/
removing a websocket protocol. This prevents some FRACKing on an invalid
pointer to the disposed container if a module that uses res_http_websocket is
unloaded after it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-08 15:38:34 +00:00
David M. Lee
cc01a79463
Added missing newlines to websocket ast_logs.
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Without these newlines, log messages just continue tacking onto the same
line, and do not flush immediately.
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Merged revisions 376561 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-20 22:06:05 +00:00
Joshua Colp
9f55e5e928
Make res_http_websocket an optional dependency on supported platforms for chan_sip.
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(closes issue ASTERISK-20439)
Reported by: sruffell
Patches:
0001-chan_sip-websocket-support-is-an-optional-API.patch uploaded by sruffell (license 5417)
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Merged revisions 373914 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-27 17:12:08 +00:00
Kinsey Moore
9b16c8b0f6
Clean up and ensure proper usage of alloca()
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This replaces all calls to alloca() with ast_alloca() which calls gcc's
__builtin_alloca() to avoid BSD semantics and removes all NULL checks
on memory allocated via ast_alloca() and ast_strdupa().
(closes issue ASTERISK-20125)
Review: https://reviewboard.asterisk.org/r/2032/
Patch-by: Walter Doekes (wdoekes)
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Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 370643 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-31 20:21:43 +00:00
Joshua Colp
e938737570
Add support for SIP over WebSocket.
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This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb.
Review: https://reviewboard.asterisk.org/r/2008
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16 12:35:04 +00:00
Joshua Colp
380c7c5c39
Add res_http_websocket module which implements the WebSocket protocol according to RFC 6455.
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Review: https://reviewboard.asterisk.org/r/1952/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-02 21:13:36 +00:00
