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

1 files changed, 47 insertions, 0 deletions

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);
 	}