Merge "res_pjsip: Add rtp_keepalive endpoint option."

include/asterisk/res_pjsip.h

@@ -502,6 +502,8 @@ 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 by remote side */
 	unsigned int remotely_held:1;
 	/*! \brief Stream is on hold by local side */

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
  * @{
  */