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| author | Joshua Colp <jcolp@digium.com> | 2015-01-27 17:32:36 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2015-01-27 17:32:36 +0000 |
| commit | b64f4bb6ee54044f44ee1322f2b2c89ebcd968d1 (patch) | |
| tree | 3976cc93fda27925977635ee605bfdf96decec9b /res/res_pjsip/pjsip_global_headers.c | |
| parent | a620b287bdcbe8b4303d195fc1663aa89bca235e (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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_pjsip/pjsip_global_headers.c')
0 files changed, 0 insertions, 0 deletions