> On March 11, 2014, 6:09 p.m., Matt Jordan wrote: > > 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?
Not sure if you noticed I did a merge already, except for the latest comments that came after the merge. I can have a proper look at the remaining comments Saturday morning for sure, but not before, sorry :( - Moises ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3248/#review11146 ----------------------------------------------------------- On March 3, 2014, 1:19 a.m., Moises Silva wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3248/ > ----------------------------------------------------------- > > (Updated March 3, 2014, 1:19 a.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 > >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev