diff options
| author | Joshua Colp <jcolp@digium.com> | 2015-01-27 17:34:37 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2015-01-27 17:34:37 +0000 |
| commit | a43d24a9d3d61274e6b7b8cf594af0d802c100b7 (patch) | |
| tree | 15b7e72afbfaf7bb1e6769944e3c083a0d2779c7 /res/res_pjsip_sdp_rtp.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_sdp_rtp.c')
| -rw-r--r-- | res/res_pjsip_sdp_rtp.c | 7 |
1 files changed, 7 insertions, 0 deletions
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index 4077caad3..a3ccd19d4 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -1180,6 +1180,10 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct a /* audio stream handles music on hold */ if (media_type != AST_MEDIA_TYPE_AUDIO) { + if ((pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE) + && (session->inv_session->state == PJSIP_INV_STATE_CONFIRMED)) { + ast_queue_control(session->channel, AST_CONTROL_UPDATE_RTP_PEER); + } return 1; } @@ -1199,6 +1203,9 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct a ast_queue_unhold(session->channel); ast_queue_frame(session->channel, &ast_null_frame); session_media->remotely_held = 0; + } else if ((pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE) + && (session->inv_session->state == PJSIP_INV_STATE_CONFIRMED)) { + ast_queue_control(session->channel, AST_CONTROL_UPDATE_RTP_PEER); } /* This purposely resets the encryption to the configured in case it gets added later */ |