[asterisk-commits] res pjsip nat: Rewrite route set when required. (asterisk[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jun 26 10:59:35 CDT 2015
Matt Jordan has submitted this change and it was merged.
Change subject: res_pjsip_nat: Rewrite route set when required.
......................................................................
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
---
M res/res_pjsip.c
M res/res_pjsip_nat.c
2 files changed, 78 insertions(+), 24 deletions(-)
Approvals:
Matt Jordan: Looks good to me, approved; Verified
Joshua Colp: Looks good to me, but someone else must approve
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 8d5adf6..6a769f0 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -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">
diff --git a/res/res_pjsip_nat.c b/res/res_pjsip_nat.c
index c717ba2..fadefd8 100644
--- a/res/res_pjsip_nat.c
+++ b/res/res_pjsip_nat.c
@@ -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);
}
}
--
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Gerrit-MessageType: merged
Gerrit-Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f
Gerrit-PatchSet: 6
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
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