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Add some debug messages to make it clear what RTP bridging functionality is in use.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2013-07-23 21:14:24 +00:00
parent e366c8c0a0
commit c780fd5498
1 changed files with 7 additions and 0 deletions
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7
bridges/bridge_native_rtp.c
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@@ -323,11 +323,15 @@ static int native_rtp_bridge_join(struct ast_bridge *bridge, struct ast_bridge_c
}
ast_rtp_instance_set_bridged(instance0, instance1);
ast_rtp_instance_set_bridged(instance1, instance0);
ast_debug(2, "Locally RTP bridged '%s' and '%s' in stack",
ast_channel_name(c0->chan), ast_channel_name(c1->chan));
break;
case AST_RTP_GLUE_RESULT_REMOTE:
glue0->update_peer(c0->chan, instance1, vinstance1, tinstance1, cap1, 0);
glue1->update_peer(c1->chan, instance0, vinstance0, tinstance0, cap0, 0);
ast_debug(2, "Remotely bridged '%s' and '%s' - media will flow directly between them\n",
ast_channel_name(c0->chan), ast_channel_name(c1->chan));
break;
case AST_RTP_GLUE_RESULT_FORBID:
break;
@@ -378,6 +382,9 @@ static void native_rtp_bridge_leave(struct ast_bridge *bridge, struct ast_bridge
case AST_RTP_GLUE_RESULT_FORBID:
break;
}
ast_debug(2, "Discontinued RTP bridging of '%s' and '%s' - media will flow through Asterisk core\n",
ast_channel_name(c0->chan), ast_channel_name(c1->chan));
}
static int native_rtp_bridge_write(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)