bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct...
authorJoshua Colp <jcolp@digium.com>
Tue, 27 Jan 2015 17:34:37 +0000 (17:34 +0000)
committerJoshua Colp <jcolp@digium.com>
Tue, 27 Jan 2015 17:34:37 +0000 (17:34 +0000)
This change fixes two issues:

1. During a swap operation bridging added the new channel before having the swap channel
leave. This was not handled in bridge_native_rtp and could result in a channel not getting
reinvited back to Asterisk. After this change the swap channel will leave first and the
new channel will then join.

2. If a re-invite was received after a session had been established any upstream elements
(such as bridge_native_rtp) were not notified that they may want to re-evaluate things.
After this change an UPDATE_RTP_PEER control frame is queued when this situation occurs
and upstream can react.

AST-1524 #close

Review: https://reviewboard.asterisk.org/r/4378/
........

Merged revisions 431157 from http://svn.asterisk.org/svn/asterisk/branches/13

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431158 65c4cc65-6c06-0410-ace0-fbb531ad65f3


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