res_pjsip WebRTC/websockets: Fix usage of WS vs WSS.

According to the RFC[1] WSS should only be used in the Via header
for secure Websockets.

* Use WSS in Via for secure transport.

* Only register one transport with the WS name because it would be
ambiguous.  Outgoing requests may try to find the transport by name and
pjproject only finds the first one registered.  This may mess up unsecure
websockets but the impact should be minimal.  Firefox and Chrome do not
support anything other than secure websockets anymore.

* Added and updated some debug messages concerning websockets.

* security_events.c: Relax case restriction when determining security
transport type.

* The res_pjsip_nat module has been updated to not touch the transport
on Websocket originating messages.

[1] https://tools.ietf.org/html/rfc7118

ASTERISK-26796 #close

Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12
This commit is contained in:
Jørgen H 2017-02-16 10:22:47 +00:00 committed by Joshua Colp
parent 26bf1846e2
commit 7922f26cb0
4 changed files with 54 additions and 15 deletions

View File

@ -101,6 +101,14 @@ app_voicemail
* The 'Comedian Mail' prompts can now be overriden using the 'vm-login' and
'vm-newuser' configuration options in voicemail.conf.
res_pjsip_transport_websocket
------------------
* Removed non-secure websocket support. Firefox and Chrome have not allowed
non-secure websockets for quite some time so this shouldn't be an issue
for people. Attempting to use a non-secure websocket may or may not work
when Asterisk attempts to send SIP requests to do something like initiate
call hangup.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.2.0 to Asterisk 14.3.0 ------------
------------------------------------------------------------------------------

View File

@ -42,9 +42,9 @@ static enum ast_transport security_event_get_transport(pjsip_rx_data *rdata)
} else if (rdata->tp_info.transport->key.type == PJSIP_TRANSPORT_TLS ||
rdata->tp_info.transport->key.type == PJSIP_TRANSPORT_TLS6) {
return AST_TRANSPORT_TLS;
} else if (!strcmp(rdata->tp_info.transport->type_name, "WS")) {
} else if (!strcasecmp(rdata->tp_info.transport->type_name, "WS")) {
return AST_TRANSPORT_WS;
} else if (!strcmp(rdata->tp_info.transport->type_name, "WSS")) {
} else if (!strcasecmp(rdata->tp_info.transport->type_name, "WSS")) {
return AST_TRANSPORT_WSS;
} else {
return 0;

View File

@ -35,7 +35,9 @@
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)) {
if (!strcasecmp("WSS", rdata->tp_info.transport->type_name)) {
/* WSS is special, we don't want to overwrite the URI at all as it needs to be ws */
} else 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;

View File

@ -38,7 +38,6 @@
#include "asterisk/res_pjsip_session.h"
#include "asterisk/taskprocessor.h"
static int transport_type_ws;
static int transport_type_wss;
/*!
@ -149,6 +148,7 @@ static int transport_create(void *data)
pjsip_endpoint *endpt = ast_sip_get_pjsip_endpoint();
struct pjsip_tpmgr *tpmgr = pjsip_endpt_get_tpmgr(endpt);
char *ws_addr_str;
pj_pool_t *pool;
pj_str_t buf;
pj_status_t status;
@ -183,9 +183,23 @@ static int transport_create(void *data)
goto on_error;
}
pj_sockaddr_parse(pj_AF_UNSPEC(), 0, pj_cstr(&buf, ast_sockaddr_stringify(ast_websocket_remote_address(newtransport->ws_session))), &newtransport->transport.key.rem_addr);
/*
* The type_name here is mostly used by log messages eihter in
* pjproject or Asterisk. Other places are reconstituting subscriptions
* after a restart (which could never work for a websocket connection anyway),
* received MESSAGE requests to set PJSIP_TRANSPORT, and most importantly
* by pjproject when generating the Via header.
*/
newtransport->transport.type_name = ast_websocket_is_secure(newtransport->ws_session)
? "WSS" : "WS";
ws_addr_str = ast_sockaddr_stringify(ast_websocket_remote_address(newtransport->ws_session));
ast_debug(4, "Creating websocket transport for %s:%s\n",
newtransport->transport.type_name, ws_addr_str);
pj_sockaddr_parse(pj_AF_UNSPEC(), 0, pj_cstr(&buf, ws_addr_str), &newtransport->transport.key.rem_addr);
newtransport->transport.key.rem_addr.addr.sa_family = pj_AF_INET();
newtransport->transport.key.type = ast_websocket_is_secure(newtransport->ws_session) ? transport_type_wss : transport_type_ws;
newtransport->transport.key.type = transport_type_wss;
newtransport->transport.addr_len = pj_sockaddr_get_len(&newtransport->transport.key.rem_addr);
@ -196,7 +210,6 @@ static int transport_create(void *data)
newtransport->transport.local_name.host.slen = pj_ansi_strlen(newtransport->transport.local_name.host.ptr);
newtransport->transport.local_name.port = pj_sockaddr_get_port(&newtransport->transport.key.rem_addr);
newtransport->transport.type_name = (char *)pjsip_transport_get_type_name(newtransport->transport.key.type);
newtransport->transport.flag = pjsip_transport_get_flag_from_type((pjsip_transport_type_e)newtransport->transport.key.type);
newtransport->transport.info = (char *)pj_pool_alloc(newtransport->transport.pool, 64);
@ -382,19 +395,27 @@ static pj_bool_t websocket_on_rx_msg(pjsip_rx_data *rdata)
long type = rdata->tp_info.transport->key.type;
if (type != (long)transport_type_ws && type != (long)transport_type_wss) {
if (type != (long) transport_type_wss) {
return PJ_FALSE;
}
if ((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))) {
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);
const pj_str_t *txp_str = &STR_WS;
ast_debug(4, "%s re-writing Contact URI from %.*s:%d%s%.*s to %s:%d;transport=%s\n",
pjsip_rx_data_get_info(rdata),
(int)pj_strlen(&uri->host), pj_strbuf(&uri->host), uri->port,
pj_strlen(&uri->transport_param) ? ";transport=" : "",
(int)pj_strlen(&uri->transport_param), pj_strbuf(&uri->transport_param),
rdata->pkt_info.src_name ?: "", rdata->pkt_info.src_port, pj_strbuf(txp_str));
pj_cstr(&uri->host, rdata->pkt_info.src_name);
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);
pj_strdup(rdata->tp_info.pool, &uri->transport_param, &STR_WS);
pj_strdup(rdata->tp_info.pool, &uri->transport_param, txp_str);
}
rdata->msg_info.via->rport_param = 0;
@ -429,8 +450,16 @@ static int load_module(void)
{
CHECK_PJSIP_MODULE_LOADED();
pjsip_transport_register_type(PJSIP_TRANSPORT_RELIABLE, "WS", 5060, &transport_type_ws);
pjsip_transport_register_type(PJSIP_TRANSPORT_RELIABLE | PJSIP_TRANSPORT_SECURE, "WS", 5060, &transport_type_wss);
/*
* We only need one transport type defined. Firefox and Chrome
* do not support anything other than secure websockets anymore.
*
* Also we really cannot have two transports with the same name
* because it would be ambiguous. Outgoing requests may try to
* find the transport by name and pjproject only finds the first
* one registered.
*/
pjsip_transport_register_type(PJSIP_TRANSPORT_RELIABLE | PJSIP_TRANSPORT_SECURE, "ws", 5060, &transport_type_wss);
if (ast_sip_register_service(&websocket_module) != PJ_SUCCESS) {
return AST_MODULE_LOAD_DECLINE;