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:
Mark Michelson
2015-06-23 17:43:31 -05:00
committed by Joshua Colp
parent db0521f905
commit 028fa54620
2 changed files with 78 additions and 24 deletions

View File

@@ -32,34 +32,88 @@
#include "asterisk/module.h"
#include "asterisk/acl.h"
static pj_bool_t handle_rx_message(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
static void rewrite_uri(pjsip_rx_data *rdata, pjsip_sip_uri *uri)
{
pj_cstr(&uri->host, rdata->pkt_info.src_name);
if (strcasecmp("udp", rdata->tp_info.transport->type_name)) {
uri->transport_param = pj_str(rdata->tp_info.transport->type_name);
} else {
uri->transport_param.slen = 0;
}
uri->port = rdata->pkt_info.src_port;
}
static int rewrite_route_set(pjsip_rx_data *rdata, pjsip_dialog *dlg)
{
pjsip_rr_hdr *rr = NULL;
pjsip_sip_uri *uri;
if (rdata->msg_info.msg->type == PJSIP_RESPONSE_MSG) {
pjsip_hdr *iter;
for (iter = rdata->msg_info.msg->hdr.prev; iter != &rdata->msg_info.msg->hdr; iter = iter->prev) {
if (iter->type == PJSIP_H_RECORD_ROUTE) {
rr = (pjsip_rr_hdr *)iter;
break;
}
}
} else {
rr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_RECORD_ROUTE, NULL);
}
if (rr) {
uri = pjsip_uri_get_uri(&rr->name_addr);
rewrite_uri(rdata, uri);
if (dlg && dlg->route_set.next && !dlg->route_set_frozen) {
pjsip_routing_hdr *route = dlg->route_set.next;
uri = pjsip_uri_get_uri(&route->name_addr);
rewrite_uri(rdata, uri);
}
return 0;
}
return -1;
}
static int rewrite_contact(pjsip_rx_data *rdata, pjsip_dialog *dlg)
{
pjsip_contact_hdr *contact;
contact = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
if (contact && !contact->star && (PJSIP_URI_SCHEME_IS_SIP(contact->uri) || PJSIP_URI_SCHEME_IS_SIPS(contact->uri))) {
pjsip_sip_uri *uri = pjsip_uri_get_uri(contact->uri);
rewrite_uri(rdata, uri);
if (dlg && !dlg->route_set_frozen && (!dlg->remote.contact
|| pjsip_uri_cmp(PJSIP_URI_IN_REQ_URI, dlg->remote.contact->uri, contact->uri))) {
dlg->remote.contact = (pjsip_contact_hdr*)pjsip_hdr_clone(dlg->pool, contact);
dlg->target = dlg->remote.contact->uri;
}
return 0;
}
return -1;
}
static pj_bool_t handle_rx_message(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
{
pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);
if (!endpoint) {
return PJ_FALSE;
}
if (endpoint->nat.rewrite_contact && (contact = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL)) &&
!contact->star && (PJSIP_URI_SCHEME_IS_SIP(contact->uri) || PJSIP_URI_SCHEME_IS_SIPS(contact->uri))) {
pjsip_sip_uri *uri = pjsip_uri_get_uri(contact->uri);
pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);
pj_cstr(&uri->host, rdata->pkt_info.src_name);
if (strcasecmp("udp", rdata->tp_info.transport->type_name)) {
uri->transport_param = pj_str(rdata->tp_info.transport->type_name);
} else {
uri->transport_param.slen = 0;
}
uri->port = rdata->pkt_info.src_port;
ast_debug(4, "Re-wrote Contact URI host/port to %.*s:%d\n",
(int)pj_strlen(&uri->host), pj_strbuf(&uri->host), uri->port);
/* rewrite the session target since it may have already been pulled from the contact header */
if (dlg && (!dlg->remote.contact
|| pjsip_uri_cmp(PJSIP_URI_IN_REQ_URI, dlg->remote.contact->uri, contact->uri))) {
dlg->remote.contact = (pjsip_contact_hdr*)pjsip_hdr_clone(dlg->pool, contact);
dlg->target = dlg->remote.contact->uri;
if (endpoint->nat.rewrite_contact) {
/* rewrite_contact is intended to ensure we send requests/responses to
* a routeable address when NAT is involved. The URI that dictates where
* we send requests/responses can be determined either by Record-Route
* headers or by the Contact header if no Record-Route headers are present.
* We therefore will attempt to rewrite a Record-Route header first, and if
* none are present, we fall back to rewriting the Contact header instead.
*/
if (rewrite_route_set(rdata, dlg)) {
rewrite_contact(rdata, dlg);
}
}