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res_pjsip_nat: Rewrite route set when required.
When performing some provider testing, the rewrite_contact option was interfering with proper construction of a route set when sending an ACK after receiving a 200 OK response to an INVITE. The initial INVITE was sent to address sip:foo. The 200 OK had a Contact header with URI sip:bar. In addition, the 200 OK had Record-Route headers for sip:baz and sip:foo, in that order. Since the Record-Route headers had the lr parameter, the result should have been: * Set R-URI of the ACK to sip:bar. * Add Route headers for sip:foo and sip:baz, in that order. However, the rewrite_contact option resulted in our rewriting the Contact header on the 200 OK to sip:foo. The result was: * R-URI remained sip:foo. * We added Route headers for sip:foo and sip:baz, in that order. The result was that sip:bar was not indicated in the ACK at all, so the far end never received our ACK. The call eventually dropped. The intention of rewrite_contact is to rewrite the most immediate destination of our SIP request to be the same address on which we received a request or response. In the case of processing a SIP response with Record-Route headers, this means that instead of rewriting the Contact header, we should instead rewrite the bottom-most Record-Route header. In the case of processing a SIP request with Record-Route headers, this means we rewrite the top-most Record-route header. Like when we rewrite the Contact header, we also ensure to update the dialog's route set if it exists. ASTERISK-25196 #close Reported by Mark Michelson Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f
This commit is contained in:
committed by
Joshua Colp
parent
39c79cd6fb
commit
700606a659
@@ -302,9 +302,9 @@
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<configOption name="rewrite_contact">
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<synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
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<description><para>
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On inbound SIP messages from this endpoint, the Contact header will be changed to have the
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source IP address and port. This option does not affect outbound messages send to this
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endpoint.
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On inbound SIP messages from this endpoint, the Contact header or an appropriate Record-Route
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header will be changed to have the source IP address and port. This option does not affect
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outbound messages sent to this endpoint.
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</para></description>
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</configOption>
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<configOption name="rtp_ipv6" default="no">
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@@ -32,34 +32,88 @@
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#include "asterisk/module.h"
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#include "asterisk/acl.h"
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static pj_bool_t handle_rx_message(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
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static void rewrite_uri(pjsip_rx_data *rdata, pjsip_sip_uri *uri)
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{
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pj_cstr(&uri->host, rdata->pkt_info.src_name);
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if (strcasecmp("udp", rdata->tp_info.transport->type_name)) {
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uri->transport_param = pj_str(rdata->tp_info.transport->type_name);
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} else {
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uri->transport_param.slen = 0;
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}
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uri->port = rdata->pkt_info.src_port;
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}
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static int rewrite_route_set(pjsip_rx_data *rdata, pjsip_dialog *dlg)
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{
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pjsip_rr_hdr *rr = NULL;
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pjsip_sip_uri *uri;
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if (rdata->msg_info.msg->type == PJSIP_RESPONSE_MSG) {
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pjsip_hdr *iter;
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for (iter = rdata->msg_info.msg->hdr.prev; iter != &rdata->msg_info.msg->hdr; iter = iter->prev) {
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if (iter->type == PJSIP_H_RECORD_ROUTE) {
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rr = (pjsip_rr_hdr *)iter;
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break;
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}
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}
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} else {
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rr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_RECORD_ROUTE, NULL);
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}
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if (rr) {
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uri = pjsip_uri_get_uri(&rr->name_addr);
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rewrite_uri(rdata, uri);
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if (dlg && dlg->route_set.next && !dlg->route_set_frozen) {
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pjsip_routing_hdr *route = dlg->route_set.next;
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uri = pjsip_uri_get_uri(&route->name_addr);
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rewrite_uri(rdata, uri);
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}
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return 0;
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}
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return -1;
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}
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static int rewrite_contact(pjsip_rx_data *rdata, pjsip_dialog *dlg)
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{
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pjsip_contact_hdr *contact;
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contact = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
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if (contact && !contact->star && (PJSIP_URI_SCHEME_IS_SIP(contact->uri) || PJSIP_URI_SCHEME_IS_SIPS(contact->uri))) {
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pjsip_sip_uri *uri = pjsip_uri_get_uri(contact->uri);
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rewrite_uri(rdata, uri);
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if (dlg && !dlg->route_set_frozen && (!dlg->remote.contact
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|| pjsip_uri_cmp(PJSIP_URI_IN_REQ_URI, dlg->remote.contact->uri, contact->uri))) {
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dlg->remote.contact = (pjsip_contact_hdr*)pjsip_hdr_clone(dlg->pool, contact);
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dlg->target = dlg->remote.contact->uri;
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}
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return 0;
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}
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return -1;
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}
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static pj_bool_t handle_rx_message(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
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{
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pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);
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if (!endpoint) {
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return PJ_FALSE;
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}
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if (endpoint->nat.rewrite_contact && (contact = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL)) &&
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!contact->star && (PJSIP_URI_SCHEME_IS_SIP(contact->uri) || PJSIP_URI_SCHEME_IS_SIPS(contact->uri))) {
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pjsip_sip_uri *uri = pjsip_uri_get_uri(contact->uri);
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pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);
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pj_cstr(&uri->host, rdata->pkt_info.src_name);
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if (strcasecmp("udp", rdata->tp_info.transport->type_name)) {
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uri->transport_param = pj_str(rdata->tp_info.transport->type_name);
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} else {
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uri->transport_param.slen = 0;
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}
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uri->port = rdata->pkt_info.src_port;
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ast_debug(4, "Re-wrote Contact URI host/port to %.*s:%d\n",
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(int)pj_strlen(&uri->host), pj_strbuf(&uri->host), uri->port);
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/* rewrite the session target since it may have already been pulled from the contact header */
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if (dlg && (!dlg->remote.contact
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|| pjsip_uri_cmp(PJSIP_URI_IN_REQ_URI, dlg->remote.contact->uri, contact->uri))) {
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dlg->remote.contact = (pjsip_contact_hdr*)pjsip_hdr_clone(dlg->pool, contact);
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dlg->target = dlg->remote.contact->uri;
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if (endpoint->nat.rewrite_contact) {
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/* rewrite_contact is intended to ensure we send requests/responses to
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* a routeable address when NAT is involved. The URI that dictates where
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* we send requests/responses can be determined either by Record-Route
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* headers or by the Contact header if no Record-Route headers are present.
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* We therefore will attempt to rewrite a Record-Route header first, and if
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* none are present, we fall back to rewriting the Contact header instead.
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*/
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if (rewrite_route_set(rdata, dlg)) {
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rewrite_contact(rdata, dlg);
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}
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}
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