chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent.

The res_pjsip modules were manually checking both name and number
presentation values when there is a function that determines the combined
presentation for a party ID struct.  The function takes into account if
the name or number components are valid while the manual code rarely
checked if the data was even valid.

* Made use ast_party_id_presentation() rather than manually checking party
ID presentation values.

* Ensure that set_id_from_pai() and set_id_from_rpid() will not return
presentation values other than what is pulled out of the SIP headers.  It
is best if the code doesn't assume that AST_PRES_ALLOWED and
AST_PRES_USER_NUMBER_UNSCREENED are zero.

* Fixed copy paste error in add_privacy_params() dealing with RPID
privacy.

* Pulled the id->number.valid test from add_privacy_header() and
add_privacy_params() up into the parent function add_id_headers() to skip
adding PAI/RPID headers earlier.

* Made update_connected_line_information() not send out connected line
updates if the connected line number is invalid.  Lower level code would
not add the party ID information and thus the sent message would be
unnecessary.

* Eliminated RAII_VAR usage in send_direct_media_request().

Review: https://reviewboard.asterisk.org/r/4472/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432892 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 61 insertions, 34 deletions

diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 18c9ed30c..675a9f9c4 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -219,10 +219,13 @@ static void chan_pjsip_get_codec(struct ast_channel *chan, struct ast_format_cap
 
 static int send_direct_media_request(void *data)
 {
-	RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
+	struct ast_sip_session *session = data;
+	int res;
 
-	return ast_sip_session_refresh(session, NULL, NULL, NULL,
-			session->endpoint->media.direct_media.method, 1);
+	res = ast_sip_session_refresh(session, NULL, NULL, NULL,
+		session->endpoint->media.direct_media.method, 1);
+	ao2_ref(session, -1);
+	return res;
 }
 
 /*! \brief Destructor function for \ref transport_info_data */
@@ -1057,17 +1060,66 @@ static int transmit_info_with_vidupdate(void *data)
 	return 0;
 }
 
+/*!
+ * \internal
+ * \brief TRUE if a COLP update can be sent to the peer.
+ * \since 13.3.0
+ *
+ * \param session The session to see if the COLP update is allowed.
+ *
+ * \retval 0 Update is not allowed.
+ * \retval 1 Update is allowed.
+ */
+static int is_colp_update_allowed(struct ast_sip_session *session)
+{
+	struct ast_party_id connected_id;
+	int update_allowed = 0;
+
+	if (!session->endpoint->id.send_pai && !session->endpoint->id.send_rpid) {
+		return 0;
+	}
+
+	/*
+	 * Check if privacy allows the update.  Check while the channel
+	 * is locked so we can work with the shallow connected_id copy.
+	 */
+	ast_channel_lock(session->channel);
+	connected_id = ast_channel_connected_effective_id(session->channel);
+	if (connected_id.number.valid
+		&& (session->endpoint->id.trust_outbound
+			|| (ast_party_id_presentation(&connected_id) & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED)) {
+		update_allowed = 1;
+	}
+	ast_channel_unlock(session->channel);
+
+	return update_allowed;
+}
+
 /*! \brief Update connected line information */
 static int update_connected_line_information(void *data)
 {
-	RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
+	struct ast_sip_session *session = data;
 
-	if ((ast_channel_state(session->channel) != AST_STATE_UP) && (session->inv_session->role == PJSIP_UAS_ROLE)) {
-		int response_code = 0;
+	if (ast_channel_state(session->channel) == AST_STATE_UP
+		|| session->inv_session->role == PJSIP_ROLE_UAC) {
+		if (is_colp_update_allowed(session)) {
+			enum ast_sip_session_refresh_method method;
+			int generate_new_sdp;
 
-		if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
-			return 0;
+			method = session->endpoint->id.refresh_method;
+			if (session->inv_session->invite_tsx
+				&& (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE)) {
+				method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
+			}
+
+			/* Only the INVITE method actually needs SDP, UPDATE can do without */
+			generate_new_sdp = (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE);
+
+			ast_sip_session_refresh(session, NULL, NULL, NULL, method, generate_new_sdp);
 		}
+	} else if (session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED
+		&& is_colp_update_allowed(session)) {
+		int response_code = 0;
 
 		if (ast_channel_state(session->channel) == AST_STATE_RING) {
 			response_code = !session->endpoint->inband_progress ? 180 : 183;
@@ -1082,34 +1134,9 @@ static int update_connected_line_information(void *data)
 				ast_sip_session_send_response(session, packet);
 			}
 		}
-	} else {
-		enum ast_sip_session_refresh_method method = session->endpoint->id.refresh_method;
-		int generate_new_sdp;
-		struct ast_party_id connected_id;
-
-		if (session->inv_session->invite_tsx && (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE)) {
-			method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
-		}
-
-		/* Only the INVITE method actually needs SDP, UPDATE can do without */
-		generate_new_sdp = (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE);
-
-		/*
-		 * We can get away with a shallow copy here because we are
-		 * not looking at strings.
-		 */
-		ast_channel_lock(session->channel);
-		connected_id = ast_channel_connected_effective_id(session->channel);
-		ast_channel_unlock(session->channel);
-
-		if ((session->endpoint->id.send_pai || session->endpoint->id.send_rpid) &&
-		    (session->endpoint->id.trust_outbound ||
-		     ((connected_id.name.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED &&
-		      (connected_id.number.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED))) {
-			ast_sip_session_refresh(session, NULL, NULL, NULL, method, generate_new_sdp);
-		}
 	}
 
+	ao2_ref(session, -1);
 	return 0;
 }