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

include/asterisk/res_pjsip.h

@@ -500,6 +500,8 @@ struct ast_sip_media_rtp_configuration {
 	enum ast_sip_session_media_encryption encryption;
 	/*! Do we want to optimistically support encryption if possible? */
 	unsigned int encryption_optimistic;
+	/*! Number of seconds between RTP keepalive packets */
+	unsigned int keepalive;
 };
 
 /*!

include/asterisk/res_pjsip_session.h

@@ -77,6 +77,8 @@ struct ast_sip_session_media {
 	enum ast_sip_session_media_encryption encryption;
 	/*! \brief The media transport in use for this stream */
 	pj_str_t transport;
+	/*! \brief Scheduler ID for RTP keepalive */
+	int keepalive_sched_id;
 	/*! \brief Stream is on hold */
 	unsigned int held:1;
 	/*! \brief Stream type this session media handles */

include/asterisk/rtp_engine.h

@@ -2288,6 +2288,22 @@ void ast_rtp_publish_rtcp_message(struct ast_rtp_instance *rtp,
 		struct ast_rtp_rtcp_report *report,
 		struct ast_json *blob);
 
+/*!
+ * \brief Get the last RTP transmission time
+ *
+ * \param rtp The instance from which to get the last transmission time
+ * \return The last RTP transmission time
+ */
+time_t ast_rtp_instance_get_last_tx(const struct ast_rtp_instance *rtp);
+
+/*!
+ * \brief Set the last RTP transmission time
+ *
+ * \param rtp The instance on which to set the last transmission time
+ * \param time The last transmission time
+ */
+void ast_rtp_instance_set_last_tx(struct ast_rtp_instance *rtp, time_t time);
+
 /*! \addtogroup StasisTopicsAndMessages
  * @{
  */