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
changes/23/723/2
Mark Michelson 10 years ago committed by Joshua Colp
parent db0521f905
commit 028fa54620

@ -302,9 +302,9 @@
<configOption name="rewrite_contact">
<synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
<description><para>
On inbound SIP messages from this endpoint, the Contact header will be changed to have the
source IP address and port. This option does not affect outbound messages send to this
endpoint.
On inbound SIP messages from this endpoint, the Contact header or an appropriate Record-Route
header will be changed to have the source IP address and port. This option does not affect
outbound messages sent to this endpoint.
</para></description>
</configOption>
<configOption name="rtp_ipv6" default="no">

@ -32,35 +32,89 @@
#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)
{
pjsip_contact_hdr *contact;
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;
}
if (!endpoint) {
return PJ_FALSE;
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;
}
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))) {
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);
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_uri(rdata, uri);
/* rewrite the session target since it may have already been pulled from the contact header */
if (dlg && (!dlg->remote.contact
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) {
/* 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);
}
}
if (endpoint->nat.force_rport) {

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