On March 3, 2014, 11:18 a.m., Jonathan Rose wrote: > > Take out that ao2_ref and this is good to go.
Ugh. That was careless of me. :/ - Jonathan ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3275/#review11023 ----------------------------------------------------------- On March 3, 2014, 11:01 a.m., Jonathan Rose wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3275/ > ----------------------------------------------------------- > > (Updated March 3, 2014, 11:01 a.m.) > > > Review request for Asterisk Developers, Joshua Colp, Kevin Harwell, and Matt > Jordan. > > > Bugs: ASTERISK-22911 > https://issues.asterisk.org/jira/browse/ASTERISK-22911 > > > Repository: Asterisk > > > Description > ------- > > This patch provides a fix for the hold problem by doing the following: > > Once an ICE session is marked as started, we start adding any new remote > candidates into a separate list until we get another attempt to start the ICE > session. > Once a call to start the ice session is made, instead of immediately quitting > if the session is already started, we check for a difference in the two > candidates lists. If the lists are identical, we wipe out the new list and > keep the old one and just quit then going on with the current ICE session. If > the lists are changed, we toss the old list and adopt the new one and restart > the ICE session. > > > Diffs > ----- > > /branches/11/res/res_rtp_asterisk.c 409155 > > Diff: https://reviewboard.asterisk.org/r/3275/diff/ > > > Testing > ------- > > SIPML client to Asterisk to Desk Phone > SIPML calls desk phone > audio test, got two way audio > SIPML holds call > SIPML resumes call > audio test, got two way audio (previously this would cause one way audio from > the SIPML client to the desk phone) > > > Thanks, > > Jonathan Rose > >
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