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author | Joshua Colp <jcolp@digium.com> | 2015-01-27 17:34:37 +0000 |
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committer | Joshua Colp <jcolp@digium.com> | 2015-01-27 17:34:37 +0000 |
commit | a43d24a9d3d61274e6b7b8cf594af0d802c100b7 (patch) | |
tree | 15b7e72afbfaf7bb1e6769944e3c083a0d2779c7 /res/res_pjsip/pjsip_configuration.c | |
parent | fb8a2e039978b264737aa7a4ed08f9af4c329f47 (diff) |
bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media.
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
Diffstat (limited to 'res/res_pjsip/pjsip_configuration.c')
0 files changed, 0 insertions, 0 deletions