res_pjsip: Add rtp_keepalive endpoint option.

This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the
chan_sip option, this specifies an interval, in seconds, at which we
will send RTP comfort noise frames. This can be useful for keeping RTP
sessions alive as well as keeping NAT associations alive during lulls.

ASTERISK-25242 #close
Reported by Mark Michelson

Change-Id: I06660ba672c0a343814af4cec838e6025cafd54b

5 files changed, 63 insertions, 1 deletions

diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 658a55e88..5fc6f0d23 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -787,6 +787,14 @@
 						have this accountcode set on it.
 					</para></description>
 				</configOption>
+				<configOption name="rtp_keepalive">
+					<synopsis>Number of seconds between RTP comfort noise keepalive packets.</synopsis>
+					<description><para>
+						At the specified interval, Asterisk will send an RTP comfort noise frame. This may
+						be useful for situations where Asterisk is behind a NAT or firewall and must keep a
+						hole open in order to allow for media to arrive at Asterisk.
+					</para></description>
+				</configOption>
 			</configObject>
 			<configObject name="auth">
 				<synopsis>Authentication type</synopsis>
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index 90a2650c0..5d85ec880 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1880,6 +1880,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "use_avpf", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.use_avpf));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "force_avp", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.force_avp));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "media_use_received_transport", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.use_received_transport));
+	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rtp_keepalive", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, media.rtp.keepalive));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "one_touch_recording", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, info.recording.enabled));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "inband_progress", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, inband_progress));
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "call_group", "", group_handler, callgroup_to_str, NULL, 0, 0);
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index fb70dd331..100224c10 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -107,6 +107,39 @@ static void format_cap_only_type(struct ast_format_cap *caps, enum ast_media_typ
 	}
 }
 
+static int send_keepalive(const void *data)
+{
+	struct ast_sip_session_media *session_media = (struct ast_sip_session_media *) data;
+	struct ast_rtp_instance *rtp = session_media->rtp;
+	int keepalive;
+	time_t interval;
+	int send_keepalive;
+
+	if (!rtp) {
+		return 0;
+	}
+
+	keepalive = ast_rtp_instance_get_keepalive(rtp);
+
+	if (!ast_sockaddr_isnull(&session_media->direct_media_addr)) {
+		ast_debug(3, "Not sending RTP keepalive on RTP instance %p since direct media is in use\n", rtp);
+		return keepalive * 1000;
+	}
+
+	interval = time(NULL) - ast_rtp_instance_get_last_tx(rtp);
+	send_keepalive = interval >= keepalive;
+
+	ast_debug(3, "It has been %d seconds since RTP was last sent on instance %p. %sending keepalive\n",
+			(int) interval, rtp, send_keepalive ? "S" : "Not s");
+
+	if (send_keepalive) {
+		ast_rtp_instance_sendcng(rtp, 0);
+		return keepalive * 1000;
+	}
+
+	return (keepalive - interval) * 1000;
+}
+
 /*! \brief Internal function which creates an RTP instance */
 static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_media *session_media, unsigned int ipv6)
 {
@@ -1227,6 +1260,17 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct a
 	/* This purposely resets the encryption to the configured in case it gets added later */
 	session_media->encryption = session->endpoint->media.rtp.encryption;
 
+	if (session->endpoint->media.rtp.keepalive > 0 &&
+			stream_to_media_type(session_media->stream_type) == AST_MEDIA_TYPE_AUDIO) {
+		ast_rtp_instance_set_keepalive(session_media->rtp, session->endpoint->media.rtp.keepalive);
+		/* Schedule the initial keepalive early in case this is being used to punch holes through
+		 * a NAT. This way there won't be an awkward delay before media starts flowing in some
+		 * scenarios.
+		 */
+		session_media->keepalive_sched_id = ast_sched_add_variable(sched, 500, send_keepalive,
+			session_media, 1);
+	}
+
 	return 1;
 }
 
@@ -1256,6 +1300,9 @@ static void change_outgoing_sdp_stream_media_address(pjsip_tx_data *tdata, struc
 static void stream_destroy(struct ast_sip_session_media *session_media)
 {
 	if (session_media->rtp) {
+		if (session_media->keepalive_sched_id != -1) {
+			AST_SCHED_DEL(sched, session_media->keepalive_sched_id);
+		}
 		ast_rtp_instance_stop(session_media->rtp);
 		ast_rtp_instance_destroy(session_media->rtp);
 	}
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index 6389ff848..9c492b3fc 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -1220,6 +1220,7 @@ static int add_session_media(void *obj, void *arg, int flags)
 		return CMP_STOP;
 	}
 	session_media->encryption = session->endpoint->media.rtp.encryption;
+	session_media->keepalive_sched_id = -1;
 	/* Safe use of strcpy */
 	strcpy(session_media->stream_type, handler_list->stream_type);
 	ao2_link(session->media, session_media);
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index adce9e7ed..5d206c1a0 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -2166,6 +2166,7 @@ static int __rtp_sendto(struct ast_rtp_instance *instance, void *buf, size_t siz
 	void *temp = buf;
 	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 	struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance);
+	int res;
 
 	*ice = 0;
 
@@ -2184,7 +2185,11 @@ static int __rtp_sendto(struct ast_rtp_instance *instance, void *buf, size_t siz
 	}
 #endif
 
-	return ast_sendto(rtcp ? rtp->rtcp->s : rtp->s, temp, len, flags, sa);
+	res = ast_sendto(rtcp ? rtp->rtcp->s : rtp->s, temp, len, flags, sa);
+	if (res > 0) {
+		ast_rtp_instance_set_last_tx(instance, time(NULL));
+	}
+	return res;
 }
 
 static int rtcp_sendto(struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int *ice)