asterisk/res/res_pjsip/pjsip_message_ip_updater.c

387 lines
13 KiB
C
Raw Normal View History

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2014-2016, Digium, Inc.
*
* Joshua Colp <jcolp@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#include "asterisk.h"
#include <pjsip.h>
#include <pjsip_ua.h>
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "include/res_pjsip_private.h"
#define MOD_DATA_RESTRICTIONS "restrictions"
static pj_status_t multihomed_on_tx_message(pjsip_tx_data *tdata);
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
2017-03-07 14:33:26 +00:00
static pj_bool_t multihomed_on_rx_message(pjsip_rx_data *rdata);
/*! \brief Outgoing message modification restrictions */
struct multihomed_message_restrictions {
/*! \brief Disallow modification of the From domain */
unsigned int disallow_from_domain_modification;
};
static pjsip_module multihomed_module = {
.name = { "Multihomed Routing", 18 },
.id = -1,
.priority = PJSIP_MOD_PRIORITY_TSX_LAYER - 1,
.on_tx_request = multihomed_on_tx_message,
.on_tx_response = multihomed_on_tx_message,
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
2017-03-07 14:33:26 +00:00
.on_rx_request = multihomed_on_rx_message,
};
/*! \brief Helper function to get (or allocate if not already present) restrictions on a message */
static struct multihomed_message_restrictions *multihomed_get_restrictions(pjsip_tx_data *tdata)
{
struct multihomed_message_restrictions *restrictions;
restrictions = ast_sip_mod_data_get(tdata->mod_data, multihomed_module.id, MOD_DATA_RESTRICTIONS);
if (restrictions) {
return restrictions;
}
restrictions = PJ_POOL_ALLOC_T(tdata->pool, struct multihomed_message_restrictions);
ast_sip_mod_data_set(tdata->pool, tdata->mod_data, multihomed_module.id, MOD_DATA_RESTRICTIONS, restrictions);
return restrictions;
}
/*! \brief Callback invoked on non-session outgoing messages */
static void multihomed_outgoing_message(struct ast_sip_endpoint *endpoint, struct ast_sip_contact *contact, struct pjsip_tx_data *tdata)
{
struct multihomed_message_restrictions *restrictions = multihomed_get_restrictions(tdata);
restrictions->disallow_from_domain_modification = !ast_strlen_zero(endpoint->fromdomain);
}
/*! \brief PJSIP Supplement for tagging messages with restrictions */
static struct ast_sip_supplement multihomed_supplement = {
.priority = AST_SIP_SUPPLEMENT_PRIORITY_FIRST,
.outgoing_request = multihomed_outgoing_message,
.outgoing_response = multihomed_outgoing_message,
};
/*! \brief Callback invoked on session outgoing messages */
static void multihomed_session_outgoing_message(struct ast_sip_session *session, struct pjsip_tx_data *tdata)
{
struct multihomed_message_restrictions *restrictions = multihomed_get_restrictions(tdata);
restrictions->disallow_from_domain_modification = !ast_strlen_zero(session->endpoint->fromdomain);
}
/*! \brief PJSIP Session Supplement for tagging messages with restrictions */
static struct ast_sip_session_supplement multihomed_session_supplement = {
.priority = 1,
.outgoing_request = multihomed_session_outgoing_message,
.outgoing_response = multihomed_session_outgoing_message,
};
/*! \brief Helper function which returns a UDP transport bound to the given address and port */
static pjsip_transport *multihomed_get_udp_transport(pj_str_t *address, int port)
{
struct ao2_container *transport_states = ast_sip_get_transport_states();
struct ast_sip_transport_state *transport_state;
struct ao2_iterator iter;
pjsip_transport *sip_transport = NULL;
if (!transport_states) {
return NULL;
}
for (iter = ao2_iterator_init(transport_states, 0); (transport_state = ao2_iterator_next(&iter)); ao2_ref(transport_state, -1)) {
if (transport_state && ((transport_state->type != AST_TRANSPORT_UDP) ||
(pj_strcmp(&transport_state->transport->local_name.host, address)) ||
(transport_state->transport->local_name.port != port))) {
continue;
}
sip_transport = transport_state->transport;
break;
}
ao2_iterator_destroy(&iter);
ao2_ref(transport_states, -1);
return sip_transport;
}
/*! \brief Helper function which determines if a transport is bound to any */
static int multihomed_bound_any(pjsip_transport *transport)
{
pj_uint32_t loop6[4] = {0, 0, 0, 0};
if ((transport->local_addr.addr.sa_family == pj_AF_INET() &&
transport->local_addr.ipv4.sin_addr.s_addr == PJ_INADDR_ANY) ||
(transport->local_addr.addr.sa_family == pj_AF_INET6() &&
!pj_memcmp(&transport->local_addr.ipv6.sin6_addr, loop6, sizeof(loop6)))) {
return 1;
}
return 0;
}
/*! \brief Helper function which determines if the address within SDP should be rewritten */
static int multihomed_rewrite_sdp(struct pjmedia_sdp_session *sdp)
{
if (!sdp->conn) {
return 0;
}
/* If the host address is used in the SDP replace it with the address of what this is going out on */
if ((!pj_strcmp2(&sdp->conn->addr_type, "IP4") && !pj_strcmp2(&sdp->conn->addr,
ast_sip_get_host_ip_string(pj_AF_INET()))) ||
(!pj_strcmp2(&sdp->conn->addr_type, "IP6") && !pj_strcmp2(&sdp->conn->addr,
ast_sip_get_host_ip_string(pj_AF_INET6())))) {
return 1;
}
return 0;
}
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
2017-03-07 14:33:26 +00:00
static void sanitize_tdata(pjsip_tx_data *tdata)
{
static const pj_str_t x_name = { AST_SIP_X_AST_TXP, AST_SIP_X_AST_TXP_LEN };
pjsip_param *x_transport;
pjsip_sip_uri *uri;
pjsip_fromto_hdr *fromto;
pjsip_contact_hdr *contact;
pjsip_hdr *hdr;
if (tdata->msg->type == PJSIP_REQUEST_MSG) {
uri = pjsip_uri_get_uri(tdata->msg->line.req.uri);
x_transport = pjsip_param_find(&uri->other_param, &x_name);
if (x_transport) {
pj_list_erase(x_transport);
}
}
for (hdr = tdata->msg->hdr.next; hdr != &tdata->msg->hdr; hdr = hdr->next) {
if (hdr->type == PJSIP_H_TO || hdr->type == PJSIP_H_FROM) {
fromto = (pjsip_fromto_hdr *) hdr;
uri = pjsip_uri_get_uri(fromto->uri);
x_transport = pjsip_param_find(&uri->other_param, &x_name);
if (x_transport) {
pj_list_erase(x_transport);
}
} else if (hdr->type == PJSIP_H_CONTACT) {
contact = (pjsip_contact_hdr *) hdr;
uri = pjsip_uri_get_uri(contact->uri);
x_transport = pjsip_param_find(&uri->other_param, &x_name);
if (x_transport) {
pj_list_erase(x_transport);
}
}
}
pjsip_tx_data_invalidate_msg(tdata);
}
static pj_status_t multihomed_on_tx_message(pjsip_tx_data *tdata)
{
struct multihomed_message_restrictions *restrictions = ast_sip_mod_data_get(tdata->mod_data, multihomed_module.id, MOD_DATA_RESTRICTIONS);
pjsip_tpmgr_fla2_param prm;
pjsip_cseq_hdr *cseq;
pjsip_via_hdr *via;
pjsip_fromto_hdr *from;
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
2017-03-07 14:33:26 +00:00
sanitize_tdata(tdata);
/* Use the destination information to determine what local interface this message will go out on */
pjsip_tpmgr_fla2_param_default(&prm);
prm.tp_type = tdata->tp_info.transport->key.type;
pj_strset2(&prm.dst_host, tdata->tp_info.dst_name);
prm.local_if = PJ_TRUE;
/* If we can't get the local address use best effort and let it pass */
if (pjsip_tpmgr_find_local_addr2(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), tdata->pool, &prm) != PJ_SUCCESS) {
return PJ_SUCCESS;
}
/* For UDP we can have multiple transports so the port needs to be maintained */
if (tdata->tp_info.transport->key.type == PJSIP_TRANSPORT_UDP ||
tdata->tp_info.transport->key.type == PJSIP_TRANSPORT_UDP6) {
prm.ret_port = tdata->tp_info.transport->local_name.port;
}
/* If the IP source differs from the existing transport see if we need to update it */
if (pj_strcmp(&prm.ret_addr, &tdata->tp_info.transport->local_name.host)) {
/* If the transport it is going out on is different reflect it in the message */
if (tdata->tp_info.transport->key.type == PJSIP_TRANSPORT_UDP ||
tdata->tp_info.transport->key.type == PJSIP_TRANSPORT_UDP6) {
pjsip_transport *transport;
transport = multihomed_get_udp_transport(&prm.ret_addr, prm.ret_port);
if (transport) {
tdata->tp_info.transport = transport;
}
}
/* If the chosen transport is not bound to any we can't use the source address as it won't get back to us */
if (!multihomed_bound_any(tdata->tp_info.transport)) {
pj_strassign(&prm.ret_addr, &tdata->tp_info.transport->local_name.host);
}
} else {
/* The transport chosen will deliver this but ensure it is updated with the right information */
pj_strassign(&prm.ret_addr, &tdata->tp_info.transport->local_name.host);
}
/* If the message needs to be updated with new address do so */
if (tdata->msg->type == PJSIP_REQUEST_MSG || !(cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL)) ||
pj_strcmp2(&cseq->method.name, "REGISTER")) {
pjsip_contact_hdr *contact = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CONTACT, NULL);
ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. *New Feature* A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. *Bug fixes* In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Review: https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close Reported by: Private Name ASTERISK-24703 #close Reported by: Matt Jordan ........ Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-12 20:34:37 +00:00
if (contact && (PJSIP_URI_SCHEME_IS_SIP(contact->uri) || PJSIP_URI_SCHEME_IS_SIPS(contact->uri))
&& !(tdata->msg->type == PJSIP_RESPONSE_MSG && tdata->msg->line.status.code / 100 == 3)) {
pjsip_sip_uri *uri = pjsip_uri_get_uri(contact->uri);
/* prm.ret_addr is allocated from the tdata pool OR the transport so it is perfectly fine to just do an assignment like this */
pj_strassign(&uri->host, &prm.ret_addr);
uri->port = prm.ret_port;
ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. *New Feature* A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. *Bug fixes* In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Review: https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close Reported by: Private Name ASTERISK-24703 #close Reported by: Matt Jordan ........ Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-12 20:34:37 +00:00
ast_debug(4, "Re-wrote Contact URI host/port to %.*s:%d\n",
(int)pj_strlen(&uri->host), pj_strbuf(&uri->host), uri->port);
if (tdata->tp_info.transport->key.type == PJSIP_TRANSPORT_UDP ||
tdata->tp_info.transport->key.type == PJSIP_TRANSPORT_UDP6) {
uri->transport_param.slen = 0;
} else {
pj_strdup2(tdata->pool, &uri->transport_param, pjsip_transport_get_type_name(tdata->tp_info.transport->key.type));
}
pjsip_tx_data_invalidate_msg(tdata);
}
}
if (tdata->msg->type == PJSIP_REQUEST_MSG && (via = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_VIA, NULL))) {
pj_strassign(&via->sent_by.host, &prm.ret_addr);
via->sent_by.port = prm.ret_port;
pjsip_tx_data_invalidate_msg(tdata);
}
if (tdata->msg->type == PJSIP_REQUEST_MSG && (from = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_FROM, NULL)) &&
(restrictions && !restrictions->disallow_from_domain_modification)) {
pjsip_name_addr *id_name_addr = (pjsip_name_addr *)from->uri;
pjsip_sip_uri *uri = pjsip_uri_get_uri(id_name_addr);
pj_sockaddr ip;
if (pj_strcmp2(&uri->host, "localhost") && pj_sockaddr_parse(pj_AF_UNSPEC(), 0, &uri->host, &ip) == PJ_SUCCESS) {
pj_strassign(&uri->host, &prm.ret_addr);
pjsip_tx_data_invalidate_msg(tdata);
}
}
/* Update the SDP if it is present */
if (tdata->msg->body && ast_sip_is_content_type(&tdata->msg->body->content_type, "application", "sdp") &&
multihomed_rewrite_sdp(tdata->msg->body->data)) {
struct pjmedia_sdp_session *sdp = tdata->msg->body->data;
static const pj_str_t STR_IP4 = { "IP4", 3 };
static const pj_str_t STR_IP6 = { "IP6", 3 };
pj_str_t STR_IP;
int stream;
STR_IP = tdata->tp_info.transport->key.type & PJSIP_TRANSPORT_IPV6 ? STR_IP6 : STR_IP4;
pj_strassign(&sdp->origin.addr, &prm.ret_addr);
sdp->origin.addr_type = STR_IP;
pj_strassign(&sdp->conn->addr, &prm.ret_addr);
sdp->conn->addr_type = STR_IP;
for (stream = 0; stream < sdp->media_count; ++stream) {
if (sdp->media[stream]->conn) {
pj_strassign(&sdp->media[stream]->conn->addr, &prm.ret_addr);
sdp->media[stream]->conn->addr_type = STR_IP;
}
}
pjsip_tx_data_invalidate_msg(tdata);
}
return PJ_SUCCESS;
}
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
2017-03-07 14:33:26 +00:00
static pj_bool_t multihomed_on_rx_message(pjsip_rx_data *rdata)
{
pjsip_contact_hdr *contact;
pjsip_sip_uri *uri;
const char *transport_id;
struct ast_sip_transport *transport;
pjsip_param *x_transport;
if (rdata->msg_info.msg->type != PJSIP_REQUEST_MSG) {
return PJ_FALSE;
}
contact = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
if (!(contact && contact->uri
&& ast_begins_with(rdata->tp_info.transport->info, AST_SIP_X_AST_TXP ":"))) {
return PJ_FALSE;
}
uri = pjsip_uri_get_uri(contact->uri);
transport_id = rdata->tp_info.transport->info + AST_SIP_X_AST_TXP_LEN + 1;
transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_id);
if (!(transport && transport->symmetric_transport)) {
return PJ_FALSE;
}
x_transport = PJ_POOL_ALLOC_T(rdata->tp_info.pool, pjsip_param);
x_transport->name = pj_strdup3(rdata->tp_info.pool, AST_SIP_X_AST_TXP);
x_transport->value = pj_strdup3(rdata->tp_info.pool, transport_id);
pj_list_insert_before(&uri->other_param, x_transport);
ast_debug(1, "Set transport '%s' on %.*s from %.*s:%d\n", transport_id,
(int)rdata->msg_info.msg->line.req.method.name.slen,
rdata->msg_info.msg->line.req.method.name.ptr,
(int)uri->host.slen, uri->host.ptr, uri->port);
return PJ_FALSE;
}
void ast_res_pjsip_cleanup_message_ip_updater(void)
{
ast_sip_unregister_service(&multihomed_module);
ast_sip_unregister_supplement(&multihomed_supplement);
ast_sip_session_unregister_supplement(&multihomed_session_supplement);
}
int ast_res_pjsip_init_message_ip_updater(void)
{
if (ast_sip_session_register_supplement(&multihomed_session_supplement)) {
ast_log(LOG_ERROR, "Could not register multihomed session supplement for outgoing requests\n");
return -1;
}
if (ast_sip_register_supplement(&multihomed_supplement)) {
ast_log(LOG_ERROR, "Could not register multihomed supplement for outgoing requests\n");
ast_res_pjsip_cleanup_message_ip_updater();
return -1;
}
if (ast_sip_register_service(&multihomed_module)) {
ast_log(LOG_ERROR, "Could not register multihomed module for incoming and outgoing requests\n");
ast_res_pjsip_cleanup_message_ip_updater();
return -1;
}
return 0;
}