diff options
author | Mark Michelson <mmichelson@digium.com> | 2015-06-23 17:43:31 -0500 |
---|---|---|
committer | Joshua Colp <jcolp@digium.com> | 2015-06-26 09:53:26 -0500 |
commit | 700606a6598344dc80e5719048bee956d4199fb2 (patch) | |
tree | e1d83196f330368a1c67458286f5e27694ed2767 /res/res_pjsip_nat.c | |
parent | 39c79cd6fb661269b8505247a2dea1ae7b7aad81 (diff) |
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
Diffstat (limited to 'res/res_pjsip_nat.c')
-rw-r--r-- | res/res_pjsip_nat.c | 90 |
1 files changed, 72 insertions, 18 deletions
diff --git a/res/res_pjsip_nat.c b/res/res_pjsip_nat.c index c717ba21c..fadefd86a 100644 --- a/res/res_pjsip_nat.c +++ b/res/res_pjsip_nat.c @@ -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) { |