[asterisk-commits] oej: branch oej/pinefrog-rtcp-1.8 r382706 - in /team/oej/pinefrog-rtcp-1.8: c...

[SVN commits to the Asterisk project]

Author: oej
Date: Fri Mar  8 10:00:17 2013
New Revision: 382706

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=382706
Log:
Right. Wrong. It's Friday afternoon. Who cares. Just drop the RTP frames in the garbage bin before you go.

Modified:
    team/oej/pinefrog-rtcp-1.8/channels/chan_sip.c
    team/oej/pinefrog-rtcp-1.8/channels/sip/rtcp.c
    team/oej/pinefrog-rtcp-1.8/res/res_rtp_asterisk.c

Modified: team/oej/pinefrog-rtcp-1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-1.8/channels/chan_sip.c?view=diff&rev=382706&r1=382705&r2=382706
==============================================================================
--- team/oej/pinefrog-rtcp-1.8/channels/chan_sip.c (original)
+++ team/oej/pinefrog-rtcp-1.8/channels/chan_sip.c Fri Mar  8 10:00:17 2013
@@ -8187,13 +8187,15 @@
 		build_callid_pvt(p);
 	else
 		ast_string_field_set(p, callid, callid);
-	/* Assign default music on hold class */
+
+	/* Set cnames for the RTCP SDES */
 	if (p->rtp) {
 		ast_rtp_instance_setcname(p->rtp, p->callid, strlen(p->callid));
 	}
 	if (p->vrtp) {
 		ast_rtp_instance_setcname(p->vrtp, p->callid, strlen(p->callid));
 	}
+	/* Assign default music on hold class */
 	ast_string_field_set(p, mohinterpret, default_mohinterpret);
 	ast_string_field_set(p, mohsuggest, default_mohsuggest);
 	p->capability = sip_cfg.capability;
@@ -20964,6 +20966,7 @@
 		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 		xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
 		check_pendings(p);
+		start_rtcp_events(p, sched);
 		break;
 
 	case 407: /* Proxy authentication */

Modified: team/oej/pinefrog-rtcp-1.8/channels/sip/rtcp.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-1.8/channels/sip/rtcp.c?view=diff&rev=382706&r1=382705&r2=382706
==============================================================================
--- team/oej/pinefrog-rtcp-1.8/channels/sip/rtcp.c (original)
+++ team/oej/pinefrog-rtcp-1.8/channels/sip/rtcp.c Fri Mar  8 10:00:17 2013
@@ -58,7 +58,7 @@
 		if (bridgepeer) {
 			/* Store the bridged peer data while we have it */
 			ast_rtp_instance_set_bridged_chan(instance, dialog->owner->name, dialog->owner->uniqueid, S_OR(bridgepeer->name,""), S_OR(bridgepeer->uniqueid,""));
-			ast_log(LOG_DEBUG, "---- Setting bridged peer name to %s\n", bridgepeer->name);
+			ast_debug(1, "---- Setting bridged peer name to %s\n", bridgepeer->name);
 		} else {
 			ast_rtp_instance_set_bridged_chan(instance, dialog->owner->name, dialog->owner->uniqueid, NULL, NULL);
 		}
@@ -83,10 +83,10 @@
 		
 			if (option_debug > 1) {
  				if (readtrans && dialog->owner->readtrans->t) {
- 					ast_log(LOG_DEBUG, "--- Audio Read translator: %s Cost %d\n", dialog->owner->readtrans->t->name, dialog->owner->readtrans->t->cost);
+ 					ast_debug(1, "--- Audio Read translator: %s Cost %d\n", dialog->owner->readtrans->t->name, dialog->owner->readtrans->t->cost);
  				}
  				if (writetrans && dialog->owner->writetrans->t) {
- 					ast_log(LOG_DEBUG, "--- Audio Write translator: %s Cost %d\n", dialog->owner->writetrans->t->name, dialog->owner->writetrans->t->cost);
+ 					ast_debug(1, "--- Audio Write translator: %s Cost %d\n", dialog->owner->writetrans->t->name, dialog->owner->writetrans->t->cost);
  				}
 			}
 		}

Modified: team/oej/pinefrog-rtcp-1.8/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-1.8/res/res_rtp_asterisk.c?view=diff&rev=382706&r1=382705&r2=382706
==============================================================================
--- team/oej/pinefrog-rtcp-1.8/res/res_rtp_asterisk.c (original)
+++ team/oej/pinefrog-rtcp-1.8/res/res_rtp_asterisk.c Fri Mar  8 10:00:17 2013
@@ -190,8 +190,6 @@
 	struct timeval dtmfmute;
 	struct timeval holdstart;       /*!< When the stream was put on hold */
 	struct ast_smoother *smoother;
-	int *ioid;
-	int *ioidrtcp;
 	unsigned short seqno;		/*!< Sequence number, RFC 3550, page 13. */
 	unsigned short rxseqno;
 	struct sched_context *sched;
@@ -3161,28 +3159,36 @@
 	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
 	if (!rtp) {		/* For some reason, there's no RTP */
+		ast_debug(1, "??????????????? NO RTP \n");
 		return;
 	}
 	if (!rtp->rtcp) {	/* No RTCP? Strange */
+		ast_debug(1, "??????????????? NO RTCP \n");
 		return;
 	}
 	/* If we already have data, don't replace it. 
 		NOTE: Should we replace it at a masquerade or something? Hmm.
 	*/
 	if (channel && !rtp->rtcp->channel[0]) {
+		ast_debug(1, "!!!!!! Setting channel name \n");
 		ast_copy_string(rtp->rtcp->channel, channel, sizeof(rtp->rtcp->channel));
 	}
 	if (uniqueid && !rtp->rtcp->uniqueid[0]) {
+		ast_debug(1, "!!!!!! Setting unique id \n");
 		ast_copy_string(rtp->rtcp->uniqueid, uniqueid, sizeof(rtp->rtcp->uniqueid));
 	}
 	if (bridgedchan) {
+		ast_debug(1, "!!!!!! Setting bridged channel name \n");
 		ast_copy_string(rtp->rtcp->bridgedchan, bridgedchan, sizeof(rtp->rtcp->bridgedchan));
 	} else {
+		ast_debug(1, "!!!!!! REmoving bridged channel name \n");
 		rtp->rtcp->bridgedchan[0] = '\0';
 	}
 	if (bridgeduniqueid) {
+		ast_debug(1, "!!!!!! Setting bridged unique id \n");
 		ast_copy_string(rtp->rtcp->bridgeduniqueid, bridgeduniqueid, sizeof(rtp->rtcp->bridgeduniqueid));
 	} else {
+		ast_debug(1, "!!!!!! Removing bridged unique id \n");
 		rtp->rtcp->bridgeduniqueid[0] = '\0';
 	}
 }