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

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changes/18/5118/1
Joshua Colp 8 years ago committed by Gerrit Code Review
parent fb11f038a3 7922f26cb0
commit e9b2360d17
4 changed files with 54 additions and 15 deletions
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8
CHANGES
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@ -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 ------------
------------------------------------------------------------------------------

4
res/res_pjsip/security_events.c
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@ -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;

4
res/res_pjsip_nat.c
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@ -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;

53
res/res_pjsip_transport_websocket.c
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@ -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;

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