res_pjsip: Symmetric transports

A new transport parameter 'symmetric_transport' has been added.

When a request from a dynamic contact comes in on a transport with
this option set to 'yes', the transport name will be saved and used
for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE.
It's saved as a contact uri parameter named 'x-ast-txp' and will
display with the contact uri in CLI, AMI, and ARI output.  On the
outgoing request, if a transport wasn't explicitly set on the
endpoint AND the request URI is not a hostname, the saved transport
will be used and the 'x-ast-txp' parameter stripped from the
outgoing packet.

* config_transport was modified to accept and store the new parameter.

* config_transport/transport_apply was updated to store the transport
  name in the pjsip_transport->info field using the pjsip_transport->pool
  on UDP transports.

* A 'multihomed_on_rx_message' function was added to
  pjsip_message_ip_updater that, for incoming requests, retrieves the
  transport name from pjsip_transport->info and retrieves the transport.
  If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter
  containing the transport name is added to the incoming Contact header.

* An 'ast_sip_get_transport_name' function was added to res_pjsip.
  It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a
  transport name if endpoint->transport is set or if there's an
  'x-ast-txp' parameter on the uri and the uri host is an ipv4 or
  ipv6 address.  Otherwise it returns NULL.

* An 'ast_sip_dlg_set_transport' function was added to res_pjsip
  which takes an ast_sip_endpoint, a pjsip_dialog, and an optional
  pjsip_tpselector.  It calls ast_sip_get_transport_name() and if
  a non-NULL is returned, sets the selector and sets the transport
  on the dialog.  If a selector was passed in, it's updated.

* res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas
  were modified to call ast_sip_dlg_set_transport() instead of their
  original logic.

* res_pjsip/create_out_of_dialog_request was modified to call
  ast_sip_get_transport_name() and pjsip_tx_data_set_transport()
  instead of its original logic.

* Existing transport logic was removed from endpt_send_request
  since that can only be called after a create_out_of_dialog_request.

* res_pjsip/ast_sip_create_rdata was converted to a wrapper around
  a new 'ast_sip_create_rdata_with_contact' function which allows
  a contact_uri to be specified in addition to the existing
  parameters.  (See below)

* res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated
  since all it did was transport selection and that is now done in
  ast_sip_create_dialog_uac and ast_sip_create_dialog_uas.

* 'contact_uri' was added to subscription_persistence.  This was
  necessary because although the parsed rdata contact header has the
  x-ast-txp parameter added (if appropriate),
  subscription_persistence_update stores the raw packet which
  doesn't have it.  subscription_persistence_recreate was then
  updated to call ast_sip_create_rdata_with_contact with the
  persisted contact_uri so the recreated subscription has the
  correct transport info to send the NOTIFYs.

* res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since
  all it did was transport selection and that is now done in
  ast_sip_create_dialog_uac.

* pjsip_message_ip_updater/multihomed_on_tx_message was updated
  to remove all traces of the x-ast-txp parameter from the
  outgoing headers.

NOTE:  This change does NOT modify the behavior of permanent
contacts specified on an aor.  To do so would require that the
permanent contact's contact uri be updated with the x-ast-txp
parameter and the aor sorcery object updated.  If we need to
persue this, we need to think about cloning permanent contacts into
the same store as the dynamic ones on an aor load so they can be
updated without disturbing the originally configured value.

You CAN add the x-ast-txp parameter to a permanent contact's uri
but it would be much simpler to just set endpoint->transport.

Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f

res/res_pjsip.c

@@ -1187,6 +1187,22 @@
 						in-progress calls.</para>
 					</description>
 				</configOption>
+				<configOption name="symmetric_transport" default="no">
+					<synopsis>Use the same transport for outgoing reqests as incoming ones.</synopsis>
+					<description>
+						<para>When a request from a dynamic contact
+							comes in on a transport with this option set to 'yes',
+							the transport name will be saved and used for subsequent
+							outgoing requests like OPTIONS, NOTIFY and INVITE.  It's
+							saved as a contact uri parameter named 'x-ast-txp' and will
+							display with the contact uri in CLI, AMI, and ARI output.
+							On the outgoing request, if a transport wasn't explicitly
+							set on the endpoint AND the request URI is not a hostname,
+							the saved transport will be used and the 'x-ast-txp'
+							parameter stripped from the outgoing packet.
+						</para>
+					</description>
+				</configOption>
 			</configObject>
 			<configObject name="contact">
 				<synopsis>A way of creating an aliased name to a SIP URI</synopsis>
@@ -2760,7 +2776,54 @@ pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
 	return ast_pjsip_endpoint;
 }
 
-static int sip_dialog_create_from(pj_pool_t *pool, pj_str_t *from, const char *user, const char *domain, const pj_str_t *target, pjsip_tpselector *selector)
+int ast_sip_get_transport_name(const struct ast_sip_endpoint *endpoint,
+	pjsip_sip_uri *sip_uri, char *buf, size_t buf_len)
+{
+	char *host = NULL;
+	static const pj_str_t x_name = { AST_SIP_X_AST_TXP, AST_SIP_X_AST_TXP_LEN };
+	pjsip_param *x_transport;
+
+	if (!ast_strlen_zero(endpoint->transport)) {
+		ast_copy_string(buf, endpoint->transport, buf_len);
+		return 0;
+	}
+
+	x_transport = pjsip_param_find(&sip_uri->other_param, &x_name);
+	if (!x_transport) {
+		return -1;
+	}
+
+	/* Only use x_transport if the uri host is an ip (4 or 6) address */
+	host = ast_alloca(sip_uri->host.slen + 1);
+	ast_copy_pj_str(host, &sip_uri->host, sip_uri->host.slen + 1);
+	if (!ast_sockaddr_parse(NULL, host, PARSE_PORT_FORBID)) {
+		return -1;
+	}
+
+	ast_copy_pj_str(buf, &x_transport->value, buf_len);
+
+	return 0;
+}
+
+int ast_sip_dlg_set_transport(const struct ast_sip_endpoint *endpoint, pjsip_dialog *dlg,
+	pjsip_tpselector *selector)
+{
+	pjsip_sip_uri *uri;
+	pjsip_tpselector sel = { .type = PJSIP_TPSELECTOR_NONE, };
+
+	uri = pjsip_uri_get_uri(dlg->target);
+	if (!selector) {
+		selector = &sel;
+	}
+
+	ast_sip_set_tpselector_from_ep_or_uri(endpoint, uri, selector);
+	pjsip_dlg_set_transport(dlg, selector);
+
+	return 0;
+}
+
+static int sip_dialog_create_from(pj_pool_t *pool, pj_str_t *from, const char *user,
+	const char *domain, const pj_str_t *target, pjsip_tpselector *selector)
 {
 	pj_str_t tmp, local_addr;
 	pjsip_uri *uri;
@@ -2890,15 +2953,16 @@ int ast_sip_set_tpselector_from_transport_name(const char *transport_name, pjsip
 	return ast_sip_set_tpselector_from_transport(transport, selector);
 }
 
-static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
+int ast_sip_set_tpselector_from_ep_or_uri(const struct ast_sip_endpoint *endpoint,
+	pjsip_sip_uri *sip_uri, pjsip_tpselector *selector)
 {
-	const char *transport_name = endpoint->transport;
+	char transport_name[128];
 
-	if (ast_strlen_zero(transport_name)) {
+	if (ast_sip_get_transport_name(endpoint, sip_uri, transport_name, sizeof(transport_name))) {
 		return 0;
 	}
 
-	return ast_sip_set_tpselector_from_transport_name(endpoint->transport, selector);
+	return ast_sip_set_tpselector_from_transport_name(transport_name, selector);
 }
 
 void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t *pool, pjsip_uri *uri)
@@ -2906,8 +2970,8 @@ void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t
 	pjsip_sip_uri *sip_uri;
 	int i = 0;
 	pjsip_param *param;
-	const pj_str_t STR_USER = { "user", 4 };
-	const pj_str_t STR_PHONE = { "phone", 5 };
+	static const pj_str_t STR_USER = { "user", 4 };
+	static const pj_str_t STR_PHONE = { "phone", 5 };
 
 	if (!endpoint || !endpoint->usereqphone || (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
 		return;
@@ -2940,7 +3004,8 @@ void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t
 	pj_list_insert_before(&sip_uri->other_param, param);
 }
 
-pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
+pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint,
+	const char *uri, const char *request_user)
 {
 	char enclosed_uri[PJSIP_MAX_URL_SIZE];
 	pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
@@ -2965,12 +3030,13 @@ pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint,
 		return NULL;
 	}
 
-	if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
-		pjsip_dlg_terminate(dlg);
-		return NULL;
-	}
+	/* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
+	dlg->sess_count++;
+
+	ast_sip_dlg_set_transport(endpoint, dlg, &selector);
 
 	if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
+		dlg->sess_count--;
 		pjsip_dlg_terminate(dlg);
 		return NULL;
 	}
@@ -3006,11 +3072,6 @@ pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint,
 	ast_sip_add_usereqphone(endpoint, dlg->pool, dlg->target);
 	ast_sip_add_usereqphone(endpoint, dlg->pool, dlg->remote.info->uri);
 
-	/* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
-	dlg->sess_count++;
-
-	pjsip_dlg_set_transport(dlg, &selector);
-
 	if (!ast_strlen_zero(outbound_proxy)) {
 		pjsip_route_hdr route_set, *route;
 		static const pj_str_t ROUTE_HNAME = { "Route", 5 };
@@ -3079,10 +3140,13 @@ pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint,
 	pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
 	pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
 	pjsip_transport *transport;
+	pjsip_contact_hdr *contact_hdr;
 
 	ast_assert(status != NULL);
 
-	if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
+	contact_hdr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
+	if (ast_sip_set_tpselector_from_ep_or_uri(endpoint, pjsip_uri_get_uri(contact_hdr->uri),
+		&selector)) {
 		return NULL;
 	}
 
@@ -3128,8 +3192,8 @@ pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint,
 	return dlg;
 }
 
-int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port,
-	char *transport_type, const char *local_name, int local_port)
+int ast_sip_create_rdata_with_contact(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port,
+	char *transport_type, const char *local_name, int local_port, const char *contact)
 {
 	pj_str_t tmp;
 
@@ -3153,6 +3217,16 @@ int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_nam
 		return -1;
 	}
 
+	if (!ast_strlen_zero(contact)) {
+		pjsip_contact_hdr *contact_hdr;
+
+		contact_hdr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
+		if (contact_hdr) {
+			contact_hdr->uri = pjsip_parse_uri(rdata->tp_info.pool, (char *)contact,
+				strlen(contact), PJSIP_PARSE_URI_AS_NAMEADDR);
+		}
+	}
+
 	pj_strdup2(rdata->tp_info.pool, &rdata->msg_info.via->recvd_param, rdata->pkt_info.src_name);
 	rdata->msg_info.via->rport_param = -1;
 
@@ -3164,6 +3238,13 @@ int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_nam
 	return 0;
 }
 
+int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port,
+	char *transport_type, const char *local_name, int local_port)
+{
+	return ast_sip_create_rdata_with_contact(rdata, packet, src_name, src_port, transport_type,
+		local_name, local_port, NULL);
+}
+
 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
@@ -3245,14 +3326,6 @@ static int create_out_of_dialog_request(const pjsip_method *method, struct ast_s
 		pj_cstr(&remote_uri, uri);
 	}
 
-	if (endpoint) {
-		if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
-			ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
-				ast_sorcery_object_get_id(endpoint));
-			return -1;
-		}
-	}
-
 	pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
 
 	if (!pool) {
@@ -3270,6 +3343,8 @@ static int create_out_of_dialog_request(const pjsip_method *method, struct ast_s
 		return -1;
 	}
 
+	ast_sip_set_tpselector_from_ep_or_uri(endpoint, pjsip_uri_get_uri(sip_uri), &selector);
+
 	fromuser = endpoint ? (!ast_strlen_zero(endpoint->fromuser) ? endpoint->fromuser : ast_sorcery_object_get_id(endpoint)) : NULL;
 	if (sip_dialog_create_from(pool, &from, fromuser,
 				endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
@@ -3289,6 +3364,8 @@ static int create_out_of_dialog_request(const pjsip_method *method, struct ast_s
 		return -1;
 	}
 
+	pjsip_tx_data_set_transport(*tdata, &selector);
+
 	if (endpoint && !ast_strlen_zero(endpoint->contact_user)){
 		pjsip_contact_hdr *contact_hdr;
 		pjsip_sip_uri *contact_uri;
@@ -3330,6 +3407,8 @@ int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
 {
 	const pjsip_method *pmethod = get_pjsip_method(method);
 
+	ast_assert(endpoint != NULL);
+
 	if (!pmethod) {
 		ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
 		return -1;
@@ -3594,7 +3673,6 @@ static pj_status_t endpt_send_request(struct ast_sip_endpoint *endpoint,
 	struct send_request_wrapper *req_wrapper;
 	pj_status_t ret_val;
 	pjsip_endpoint *endpt = ast_sip_get_pjsip_endpoint();
-	pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
 
 	if (!cb && token) {
 		/* Silly.  Without a callback we cannot do anything with token. */
@@ -3619,11 +3697,6 @@ static pj_status_t endpt_send_request(struct ast_sip_endpoint *endpoint,
 	/* Add a reference to tdata.  The wrapper destructor cleans it up. */
 	pjsip_tx_data_add_ref(tdata);
 
-	if (endpoint) {
-		sip_get_tpselector_from_endpoint(endpoint, &selector);
-		pjsip_tx_data_set_transport(tdata, &selector);
-	}
-
 	if (timeout > 0) {
 		pj_time_val timeout_timer_val = { timeout / 1000, timeout % 1000 };