[asterisk-commits] twilson: branch 1.8 r323370 - in /branches/1.8: channels/ include/asterisk/ m...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Jun 14 11:34:00 CDT 2011


Author: twilson
Date: Tue Jun 14 11:33:55 2011
New Revision: 323370

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=323370
Log:
Add rtpkeepalives back to 1.8

The RTP-engine conversion left out support for handling rtpkeepalives.
This patch adds them back.

(closes ASTERISK-17304)
Review: https://reviewboard.asterisk.org/r/1226/

Modified:
    branches/1.8/channels/chan_sip.c
    branches/1.8/include/asterisk/rtp_engine.h
    branches/1.8/main/rtp_engine.c
    branches/1.8/res/res_rtp_asterisk.c

Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=323370&r1=323369&r2=323370
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Tue Jun 14 11:33:55 2011
@@ -5020,6 +5020,7 @@
 		}
 		ast_rtp_instance_set_timeout(dialog->vrtp, global_rtptimeout);
 		ast_rtp_instance_set_hold_timeout(dialog->vrtp, global_rtpholdtimeout);
+		ast_rtp_instance_set_keepalive(dialog->vrtp, global_rtpholdtimeout);
 
 		ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
 	}
@@ -5030,12 +5031,14 @@
 		}
 		ast_rtp_instance_set_timeout(dialog->trtp, global_rtptimeout);
 		ast_rtp_instance_set_hold_timeout(dialog->trtp, global_rtpholdtimeout);
+		ast_rtp_instance_set_keepalive(dialog->trtp, global_rtpholdtimeout);
 
 		ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1);
 	}
 
 	ast_rtp_instance_set_timeout(dialog->rtp, global_rtptimeout);
 	ast_rtp_instance_set_hold_timeout(dialog->rtp, global_rtpholdtimeout);
+	ast_rtp_instance_set_keepalive(dialog->rtp, global_rtpkeepalive);
 
 	ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, 1);
 	ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
@@ -5103,6 +5106,7 @@
 		ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
 		ast_rtp_instance_set_timeout(dialog->rtp, peer->rtptimeout);
 		ast_rtp_instance_set_hold_timeout(dialog->rtp, peer->rtpholdtimeout);
+		ast_rtp_instance_set_keepalive(dialog->rtp, peer->rtpkeepalive);
 		/* Set Frame packetization */
 		ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs);
 		dialog->autoframing = peer->autoframing;
@@ -5110,10 +5114,12 @@
 	if (dialog->vrtp) { /* Video */
 		ast_rtp_instance_set_timeout(dialog->vrtp, peer->rtptimeout);
 		ast_rtp_instance_set_hold_timeout(dialog->vrtp, peer->rtpholdtimeout);
+		ast_rtp_instance_set_keepalive(dialog->vrtp, peer->rtpkeepalive);
 	}
 	if (dialog->trtp) { /* Realtime text */
 		ast_rtp_instance_set_timeout(dialog->trtp, peer->rtptimeout);
 		ast_rtp_instance_set_hold_timeout(dialog->trtp, peer->rtpholdtimeout);
+		ast_rtp_instance_set_keepalive(dialog->trtp, peer->rtpkeepalive);
 	}
 
 	/* XXX TODO: get fields directly from peer only as they are needed using dialog->relatedpeer */
@@ -24771,8 +24777,16 @@
 		return;
 
 	/* If we have no timers set, return now */
-	if (!ast_rtp_instance_get_timeout(dialog->rtp) && !ast_rtp_instance_get_hold_timeout(dialog->rtp)) {
+	if (!ast_rtp_instance_get_keepalive(dialog->rtp) && !ast_rtp_instance_get_timeout(dialog->rtp) && !ast_rtp_instance_get_hold_timeout(dialog->rtp)) {
 		return;
+	}
+
+	/* Check AUDIO RTP keepalives */
+	if (dialog->lastrtptx && ast_rtp_instance_get_keepalive(dialog->rtp) &&
+		    (t > dialog->lastrtptx + ast_rtp_instance_get_keepalive(dialog->rtp))) {
+		/* Need to send an empty RTP packet */
+		dialog->lastrtptx = time(NULL);
+		ast_rtp_instance_sendcng(dialog->rtp, 0);
 	}
 
 	/*! \todo Check video RTP keepalives

Modified: branches/1.8/include/asterisk/rtp_engine.h
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/include/asterisk/rtp_engine.h?view=diff&rev=323370&r1=323369&r2=323370
==============================================================================
--- branches/1.8/include/asterisk/rtp_engine.h (original)
+++ branches/1.8/include/asterisk/rtp_engine.h Tue Jun 14 11:33:55 2011
@@ -373,6 +373,8 @@
 	void (*stun_request)(struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username);
 	/*! Callback to get the transcodeable formats supported */
 	int (*available_formats)(struct ast_rtp_instance *instance, format_t to_endpoint, format_t to_asterisk);
+	/*! Callback to send CNG */
+	int (*sendcng)(struct ast_rtp_instance *instance, int level);
 	/*! Linked list information */
 	AST_RWLIST_ENTRY(ast_rtp_engine) entry;
 };
@@ -1685,6 +1687,24 @@
 void ast_rtp_instance_set_hold_timeout(struct ast_rtp_instance *instance, int timeout);
 
 /*!
+ * \brief Set the RTP keepalive interval
+ *
+ * \param instance The RTP instance
+ * \param period Value to set the keepalive interval to
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_keepalive(instance, 5000);
+ * \endcode
+ *
+ * This sets the RTP keepalive interval on 'instance' to be 5000.
+ *
+ * \since 1.8
+ */
+void ast_rtp_instance_set_keepalive(struct ast_rtp_instance *instance, int timeout);
+
+/*!
  * \brief Get the RTP timeout value
  *
  * \param instance The RTP instance
@@ -1723,6 +1743,25 @@
 int ast_rtp_instance_get_hold_timeout(struct ast_rtp_instance *instance);
 
 /*!
+ * \brief Get the RTP keepalive interval
+ *
+ * \param instance The RTP instance
+ *
+ * \retval period Keepalive interval value
+ *
+ * Example usage:
+ *
+ * \code
+ * int interval = ast_rtp_instance_get_keepalive(instance);
+ * \endcode
+ *
+ * This gets the RTP keepalive interval value for the RTP instance pointed to by 'instance'.
+ *
+ * \since 1.8
+ */
+int ast_rtp_instance_get_keepalive(struct ast_rtp_instance *instance);
+
+/*!
  * \brief Get the RTP engine in use on an RTP instance
  *
  * \param instance The RTP instance
@@ -1780,6 +1819,17 @@
  * \since 1.8
  */
 struct ast_channel *ast_rtp_instance_get_chan(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Send a comfort noise packet to the RTP instance
+ *
+ * \param instance The RTP instance
+ * \param level Magnitude of the noise level
+ *
+ * \retval 0 Success
+ * \retval non-zero Failure
+ */
+int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level);
 
 int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *policy);
 struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance);

Modified: branches/1.8/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/main/rtp_engine.c?view=diff&rev=323370&r1=323369&r2=323370
==============================================================================
--- branches/1.8/main/rtp_engine.c (original)
+++ branches/1.8/main/rtp_engine.c Tue Jun 14 11:33:55 2011
@@ -65,6 +65,8 @@
 	int timeout;
 	/*! RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
 	int holdtimeout;
+	/*! RTP keepalive interval */
+	int keepalive;
 	/*! DTMF mode in use */
 	enum ast_rtp_dtmf_mode dtmf_mode;
 	/*! Glue currently in use */
@@ -1710,6 +1712,11 @@
 	instance->holdtimeout = timeout;
 }
 
+void ast_rtp_instance_set_keepalive(struct ast_rtp_instance *instance, int interval)
+{
+	instance->keepalive = interval;
+}
+
 int ast_rtp_instance_get_timeout(struct ast_rtp_instance *instance)
 {
 	return instance->timeout;
@@ -1718,6 +1725,11 @@
 int ast_rtp_instance_get_hold_timeout(struct ast_rtp_instance *instance)
 {
 	return instance->holdtimeout;
+}
+
+int ast_rtp_instance_get_keepalive(struct ast_rtp_instance *instance)
+{
+	return instance->keepalive;
 }
 
 struct ast_rtp_engine *ast_rtp_instance_get_engine(struct ast_rtp_instance *instance)
@@ -1778,3 +1790,12 @@
 {
 	return instance->srtp;
 }
+
+int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level)
+{
+	if (instance->engine->sendcng) {
+		return instance->engine->sendcng(instance, level);
+	}
+
+	return -1;
+}

Modified: branches/1.8/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/res/res_rtp_asterisk.c?view=diff&rev=323370&r1=323369&r2=323370
==============================================================================
--- branches/1.8/res/res_rtp_asterisk.c (original)
+++ branches/1.8/res/res_rtp_asterisk.c Tue Jun 14 11:33:55 2011
@@ -272,6 +272,7 @@
 static void ast_rtp_stun_request(struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username);
 static void ast_rtp_stop(struct ast_rtp_instance *instance);
 static int ast_rtp_qos_set(struct ast_rtp_instance *instance, int tos, int cos, const char* desc);
+static int ast_rtp_sendcng(struct ast_rtp_instance *instance, int level);
 
 /* RTP Engine Declaration */
 static struct ast_rtp_engine asterisk_rtp_engine = {
@@ -297,6 +298,7 @@
 	.stun_request = ast_rtp_stun_request,
 	.stop = ast_rtp_stop,
 	.qos = ast_rtp_qos_set,
+	.sendcng = ast_rtp_sendcng,
 };
 
 static inline int rtp_debug_test_addr(struct ast_sockaddr *addr)
@@ -2591,6 +2593,49 @@
 	return ast_set_qos(rtp->s, tos, cos, desc);
 }
 
+/*! \brief generate comfort noice (CNG) */
+static int ast_rtp_sendcng(struct ast_rtp_instance *instance, int level)
+{
+	unsigned int *rtpheader;
+	int hdrlen = 12;
+	int res;
+	struct ast_rtp_payload_type payload;
+	char data[256];
+	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+	struct ast_sockaddr remote_address = { {0,} };
+
+	ast_rtp_instance_get_remote_address(instance, &remote_address);
+
+	if (ast_sockaddr_isnull(&remote_address)) {
+		return -1;
+	}
+
+	payload = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(instance), AST_RTP_CN);
+
+	level = 127 - (level & 0x7f);
+	
+	rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
+
+	/* Get a pointer to the header */
+	rtpheader = (unsigned int *)data;
+	rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload.code << 16) | (rtp->seqno++));
+	rtpheader[1] = htonl(rtp->lastts);
+	rtpheader[2] = htonl(rtp->ssrc); 
+	data[12] = level;
+
+	res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 1, 0, &remote_address);
+
+	if (res < 0) {
+		ast_log(LOG_ERROR, "RTP Comfort Noise Transmission error to %s: %s\n", ast_sockaddr_stringify(&remote_address), strerror(errno));
+	} else if (rtp_debug_test_addr(&remote_address)) {
+		ast_verbose("Sent Comfort Noise RTP packet to %s (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
+				ast_sockaddr_stringify(&remote_address),
+				AST_RTP_CN, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+	}
+
+	return res;
+}
+
 static char *rtp_do_debug_ip(struct ast_cli_args *a)
 {
 	char *arg = ast_strdupa(a->argv[4]);




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