Author: file Date: Tue Jan 27 11:32:36 2015 New Revision: 431157 URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=431157 Log: 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/ Modified: branches/13/main/bridge_channel.c branches/13/res/res_pjsip_sdp_rtp.c Modified: branches/13/main/bridge_channel.c URL: http://svnview.digium.com/svn/asterisk/branches/13/main/bridge_channel.c?view=diff&rev=431157&r1=431156&r2=431157 ============================================================================== --- branches/13/main/bridge_channel.c (original) +++ branches/13/main/bridge_channel.c Tue Jan 27 11:32:36 2015 @@ -1994,6 +1994,19 @@ bridge->uniqueid, bridge_channel, ast_channel_name(bridge_channel->chan)); return -1; } + + if (swap) { + int dissolve = ast_test_flag(&bridge->feature_flags, AST_BRIDGE_FLAG_DISSOLVE_EMPTY); + + /* This flag is cleared so the act of this channel leaving does not cause it to dissolve if need be */ + ast_clear_flag(&bridge->feature_flags, AST_BRIDGE_FLAG_DISSOLVE_EMPTY); + + ast_bridge_channel_leave_bridge(swap, BRIDGE_CHANNEL_STATE_END_NO_DISSOLVE, 0); + bridge_channel_internal_pull(swap); + + ast_set2_flag(&bridge->feature_flags, dissolve, AST_BRIDGE_FLAG_DISSOLVE_EMPTY); + } + bridge_channel->in_bridge = 1; bridge_channel->just_joined = 1; AST_LIST_INSERT_TAIL(&bridge->channels, bridge_channel, entry); @@ -2015,10 +2028,6 @@ bridge->uniqueid); ast_bridge_publish_enter(bridge, bridge_channel->chan, swap ? swap->chan : NULL); - if (swap) { - ast_bridge_channel_leave_bridge(swap, BRIDGE_CHANNEL_STATE_END_NO_DISSOLVE, 0); - bridge_channel_internal_pull(swap); - } /* Clear any BLINDTRANSFER and ATTENDEDTRANSFER since the transfer has completed. */ pbx_builtin_setvar_helper(bridge_channel->chan, "BLINDTRANSFER", NULL); Modified: branches/13/res/res_pjsip_sdp_rtp.c URL: http://svnview.digium.com/svn/asterisk/branches/13/res/res_pjsip_sdp_rtp.c?view=diff&rev=431157&r1=431156&r2=431157 ============================================================================== --- branches/13/res/res_pjsip_sdp_rtp.c (original) +++ branches/13/res/res_pjsip_sdp_rtp.c Tue Jan 27 11:32:36 2015 @@ -1179,6 +1179,10 @@ /* 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; } @@ -1198,6 +1202,9 @@ ast_queue_unhold(session->channel); ast_queue_frame(session->channel, &ast_null_frame); session_media->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 */ -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- svn-commits mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/svn-commits