[svn-commits] file: trunk r370072 - in /trunk: ./ channels/ channels/sip/ channels/sip/incl...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Jul 16 07:35:12 CDT 2012
Author: file
Date: Mon Jul 16 07:35:04 2012
New Revision: 370072
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=370072
Log:
Add support for SIP over WebSocket.
This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb.
Review: https://reviewboard.asterisk.org/r/2008
Modified:
trunk/CHANGES
trunk/channels/chan_sip.c
trunk/channels/sip/include/sip.h
trunk/channels/sip/sdp_crypto.c
trunk/channels/sip/security_events.c
trunk/configs/sip.conf.sample
trunk/include/asterisk/http_websocket.h
trunk/res/res_http_websocket.c
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=370072&r1=370071&r2=370072
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Mon Jul 16 07:35:04 2012
@@ -105,6 +105,8 @@
* Add support for lightweight NAT keepalive. If enabled a blank packet will
be sent to the remote host at a given interval to keep the NAT mapping open.
This can be enabled using the keepalive configuration option.
+ * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
+ as the transport.
Chan_local changes
------------------
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=370072&r1=370071&r2=370072
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Jul 16 07:35:04 2012
@@ -1167,6 +1167,12 @@
/*! \brief A per-thread temporary pvt structure */
AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup);
+/*! \brief A per-thread buffer for transport to string conversion */
+AST_THREADSTORAGE(sip_transport_str_buf);
+
+/*! \brief Size of the SIP transport buffer */
+#define SIP_TRANSPORT_STR_BUFSIZE 128
+
/*! \brief Authentication container for realm authentication */
static struct sip_auth_container *authl = NULL;
/*! \brief Global authentication container protection while adjusting the references. */
@@ -2525,6 +2531,54 @@
return _sip_tcp_helper_thread(tcptls_session);
}
+/*! \brief SIP WebSocket connection handler */
+static void sip_websocket_callback(struct ast_websocket *session, struct ast_variable *parameters, struct ast_variable *headers)
+{
+ int res;
+
+ if (ast_websocket_set_nonblock(session)) {
+ goto end;
+ }
+
+ while ((res = ast_wait_for_input(ast_websocket_fd(session), -1)) > 0) {
+ char *payload;
+ uint64_t payload_len;
+ enum ast_websocket_opcode opcode;
+ int fragmented;
+
+ if (ast_websocket_read(session, &payload, &payload_len, &opcode, &fragmented)) {
+ /* We err on the side of caution and terminate the session if any error occurs */
+ break;
+ }
+
+ if (opcode == AST_WEBSOCKET_OPCODE_TEXT || opcode == AST_WEBSOCKET_OPCODE_BINARY) {
+ struct sip_request req = { 0, };
+
+ if (!(req.data = ast_str_create(payload_len))) {
+ goto end;
+ }
+
+ if (ast_str_set(&req.data, -1, "%s", payload) == AST_DYNSTR_BUILD_FAILED) {
+ deinit_req(&req);
+ goto end;
+ }
+
+ req.socket.fd = ast_websocket_fd(session);
+ set_socket_transport(&req.socket, ast_websocket_is_secure(session) ? SIP_TRANSPORT_WSS : SIP_TRANSPORT_WS);
+ req.socket.ws_session = session;
+
+ handle_request_do(&req, ast_websocket_remote_address(session));
+ deinit_req(&req);
+
+ } else if (opcode == AST_WEBSOCKET_OPCODE_CLOSE) {
+ break;
+ }
+ }
+
+end:
+ ast_websocket_unref(session);
+}
+
/*! \brief Check if the authtimeout has expired.
* \param start the time when the session started
*
@@ -2800,6 +2854,7 @@
we receive is not the same - we should generate an error */
req.socket.tcptls_session = tcptls_session;
+ req.socket.ws_session = NULL;
handle_request_do(&req, &tcptls_session->remote_address);
}
@@ -3306,29 +3361,53 @@
if (!strcasecmp(transport, "tls")) {
res |= SIP_TRANSPORT_TLS;
}
+ if (!strcasecmp(transport, "ws")) {
+ res |= SIP_TRANSPORT_WS;
+ }
+ if (!strcasecmp(transport, "wss")) {
+ res |= SIP_TRANSPORT_WSS;
+ }
return res;
}
/*! \brief Return configuration of transports for a device */
-static inline const char *get_transport_list(unsigned int transports) {
- switch (transports) {
- case SIP_TRANSPORT_UDP:
- return "UDP";
- case SIP_TRANSPORT_TCP:
- return "TCP";
- case SIP_TRANSPORT_TLS:
- return "TLS";
- case SIP_TRANSPORT_UDP | SIP_TRANSPORT_TCP:
- return "TCP,UDP";
- case SIP_TRANSPORT_UDP | SIP_TRANSPORT_TLS:
- return "TLS,UDP";
- case SIP_TRANSPORT_TCP | SIP_TRANSPORT_TLS:
- return "TLS,TCP";
- default:
- return transports ?
- "TLS,TCP,UDP" : "UNKNOWN";
- }
+static inline const char *get_transport_list(unsigned int transports)
+{
+ char *buf;
+
+ if (!transports) {
+ return "UNKNOWN";
+ }
+
+ if (!(buf = ast_threadstorage_get(&sip_transport_str_buf, SIP_TRANSPORT_STR_BUFSIZE))) {
+ return "";
+ }
+
+ memset(buf, 0, SIP_TRANSPORT_STR_BUFSIZE);
+
+ if (transports & SIP_TRANSPORT_UDP) {
+ strncat(buf, "UDP,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
+ }
+ if (transports & SIP_TRANSPORT_TCP) {
+ strncat(buf, "TCP,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
+ }
+ if (transports & SIP_TRANSPORT_TLS) {
+ strncat(buf, "TLS,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
+ }
+ if (transports & SIP_TRANSPORT_WS) {
+ strncat(buf, "WS,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
+ }
+ if (transports & SIP_TRANSPORT_WSS) {
+ strncat(buf, "WSS,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
+ }
+
+ /* Remove the trailing ',' if present */
+ if (strlen(buf)) {
+ buf[strlen(buf) - 1] = 0;
+ }
+
+ return buf;
}
/*! \brief Return transport as string */
@@ -3341,6 +3420,9 @@
return "TCP";
case SIP_TRANSPORT_TLS:
return "TLS";
+ case SIP_TRANSPORT_WS:
+ case SIP_TRANSPORT_WSS:
+ return "WS";
}
return "UNKNOWN";
@@ -3352,9 +3434,13 @@
switch (t) {
case SIP_TRANSPORT_UDP:
return "udp";
+ case SIP_TRANSPORT_WS:
+ return "ws";
case SIP_TRANSPORT_TLS:
case SIP_TRANSPORT_TCP:
return "tcp";
+ case SIP_TRANSPORT_WSS:
+ return "wss";
}
return "udp";
@@ -3366,8 +3452,10 @@
switch (t) {
case SIP_TRANSPORT_TCP:
case SIP_TRANSPORT_UDP:
+ case SIP_TRANSPORT_WS:
return "sip";
case SIP_TRANSPORT_TLS:
+ case SIP_TRANSPORT_WSS:
return "sips";
}
return "sip";
@@ -3414,6 +3502,11 @@
res = ast_sendto(p->socket.fd, data->str, ast_str_strlen(data), 0, dst);
} else if (p->socket.tcptls_session) {
res = sip_tcptls_write(p->socket.tcptls_session, data->str, ast_str_strlen(data));
+ } else if (p->socket.ws_session) {
+ if (!(res = ast_websocket_write(p->socket.ws_session, AST_WEBSOCKET_OPCODE_TEXT, data->str, ast_str_strlen(data)))) {
+ /* The WebSocket API just returns 0 on success and -1 on failure, while this code expects the payload length to be returned */
+ res = ast_str_strlen(data);
+ }
} else {
ast_debug(2, "Socket type is TCP but no tcptls_session is present to write to\n");
return XMIT_ERROR;
@@ -4730,6 +4823,9 @@
if (peer->socket.tcptls_session) {
ao2_ref(peer->socket.tcptls_session, -1);
peer->socket.tcptls_session = NULL;
+ } else if (peer->socket.ws_session) {
+ ast_websocket_unref(peer->socket.ws_session);
+ peer->socket.ws_session = NULL;
}
ast_cc_config_params_destroy(peer->cc_params);
@@ -5298,10 +5394,15 @@
if (to_sock->tcptls_session) {
ao2_ref(to_sock->tcptls_session, -1);
to_sock->tcptls_session = NULL;
+ } else if (to_sock->ws_session) {
+ ast_websocket_unref(to_sock->ws_session);
+ to_sock->ws_session = NULL;
}
if (from_sock->tcptls_session) {
ao2_ref(from_sock->tcptls_session, +1);
+ } else if (from_sock->ws_session) {
+ ast_websocket_ref(from_sock->ws_session);
}
*to_sock = *from_sock;
@@ -6012,6 +6113,9 @@
if (p->socket.tcptls_session) {
ao2_ref(p->socket.tcptls_session, -1);
p->socket.tcptls_session = NULL;
+ } else if (p->socket.ws_session) {
+ ast_websocket_unref(p->socket.ws_session);
+ p->socket.ws_session = NULL;
}
if (p->peerauth) {
@@ -9334,7 +9438,7 @@
int image = FALSE;
int text = FALSE;
int processed_crypto = FALSE;
- char protocol[5] = {0,};
+ char protocol[6] = {0,};
int x;
numberofports = 0;
@@ -9354,8 +9458,8 @@
/* Check for 'audio' media offer */
if (strncmp(m, "audio ", 6) == 0) {
- if ((sscanf(m, "audio %30u/%30u RTP/%4s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
- (sscanf(m, "audio %30u RTP/%4s %n", &x, protocol, &len) == 2 && len > 0)) {
+ if ((sscanf(m, "audio %30u/%30u RTP/%5s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
+ (sscanf(m, "audio %30u RTP/%5s %n", &x, protocol, &len) == 2 && len > 0)) {
codecs = m + len;
/* produce zero-port m-line since it may be needed later
* length is "m=audio 0 RTP/" + protocol + " " + codecs + "\0" */
@@ -9377,9 +9481,21 @@
ast_log(LOG_WARNING, "%d ports offered for audio media, not supported by Asterisk. Will try anyway...\n", numberofports);
}
- if (!strcmp(protocol, "SAVP")) {
+ if (!strcmp(protocol, "SAVPF") && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
+ ast_log(LOG_WARNING, "Received SAVPF profle in audio offer but AVPF is not enabled: %s\n", m);
+ continue;
+ } else if (!strcmp(protocol, "SAVP") && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
+ ast_log(LOG_WARNING, "Received SAVP profile in audio offer but AVPF is enabled: %s\n", m);
+ continue;
+ } else if (!strcmp(protocol, "SAVP") || !strcmp(protocol, "SAVPF")) {
secure_audio = 1;
- } else if (strcmp(protocol, "AVP")) {
+ } else if (!strcmp(protocol, "AVPF") && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
+ ast_log(LOG_WARNING, "Received AVPF profile in audio offer but AVPF is not enabled: %s\n", m);
+ continue;
+ } else if (!strcmp(protocol, "AVP") && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
+ ast_log(LOG_WARNING, "Received AVP profile in audio offer but AVPF is enabled: %s\n", m);
+ continue;
+ } else if (strcmp(protocol, "AVP") && strcmp(protocol, "AVPF")) {
ast_log(LOG_WARNING, "Unknown RTP profile in audio offer: %s\n", m);
continue;
}
@@ -9414,8 +9530,8 @@
}
/* Check for 'video' media offer */
else if (strncmp(m, "video ", 6) == 0) {
- if ((sscanf(m, "video %30u/%30u RTP/%4s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
- (sscanf(m, "video %30u RTP/%4s %n", &x, protocol, &len) == 2 && len > 0)) {
+ if ((sscanf(m, "video %30u/%30u RTP/%5s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
+ (sscanf(m, "video %30u RTP/%5s %n", &x, protocol, &len) == 2 && len > 0)) {
codecs = m + len;
/* produce zero-port m-line since it may be needed later
* length is "m=video 0 RTP/" + protocol + " " + codecs + "\0" */
@@ -9437,9 +9553,21 @@
ast_log(LOG_WARNING, "%d ports offered for video stream, not supported by Asterisk. Will try anyway...\n", numberofports);
}
- if (!strcmp(protocol, "SAVP")) {
+ if (!strcmp(protocol, "SAVPF") && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
+ ast_log(LOG_WARNING, "Received SAVPF profle in video offer but AVPF is not enabled: %s\n", m);
+ continue;
+ } else if (!strcmp(protocol, "SAVP") && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
+ ast_log(LOG_WARNING, "Received SAVP profile in video offer but AVPF is enabled: %s\n", m);
+ continue;
+ } else if (!strcmp(protocol, "SAVP") || !strcmp(protocol, "SAVPF")) {
secure_video = 1;
- } else if (strcmp(protocol, "AVP")) {
+ } else if (!strcmp(protocol, "AVPF") && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
+ ast_log(LOG_WARNING, "Received AVPF profile in video offer but AVPF is not enabled: %s\n", m);
+ continue;
+ } else if (!strcmp(protocol, "AVP") && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
+ ast_log(LOG_WARNING, "Received AVP profile in video offer but AVPF is enabled: %s\n", m);
+ continue;
+ } else if (strcmp(protocol, "AVP") && strcmp(protocol, "AVPF")) {
ast_log(LOG_WARNING, "Unknown RTP profile in video offer: %s\n", m);
continue;
}
@@ -9474,18 +9602,18 @@
}
/* Check for 'text' media offer */
else if (strncmp(m, "text ", 5) == 0) {
- if ((sscanf(m, "text %30u/%30u RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
- (sscanf(m, "text %30u RTP/AVP %n", &x, &len) == 1 && len > 0)) {
+ if ((sscanf(m, "text %30u/%30u RTP/%s %n", &x, &numberofports, protocol, &len) == 2 && len > 0) ||
+ (sscanf(m, "text %30u RTP/%s %n", &x, protocol, &len) == 1 && len > 0)) {
codecs = m + len;
/* produce zero-port m-line since it may be needed later
- * length is "m=text 0 RTP/AVP " + codecs + "\0" */
- if (!(offer->decline_m_line = ast_malloc(17 + strlen(codecs) + 1))) {
+ * length is "m=text 0 RTP/" + protocol + " " + codecs + "\0" */
+ if (!(offer->decline_m_line = ast_malloc(13 + strlen(protocol) + 1 + strlen(codecs) + 1))) {
ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer declination\n");
res = -1;
goto process_sdp_cleanup;
}
/* guaranteed to be exactly the right length */
- sprintf(offer->decline_m_line, "m=text 0 RTP/AVP %s", codecs);
+ sprintf(offer->decline_m_line, "m=text 0 RTP/%s %s", protocol, codecs);
if (x == 0) {
ast_log(LOG_WARNING, "Ignoring text stream offer because port number is zero\n");
@@ -9495,6 +9623,17 @@
/* Check number of ports offered for stream */
if (numberofports > 1) {
ast_log(LOG_WARNING, "%d ports offered for text stream, not supported by Asterisk. Will try anyway...\n", numberofports);
+ }
+
+ if (!strcmp(protocol, "AVPF") && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
+ ast_log(LOG_WARNING, "Received AVPF profile in text offer but AVPF is not enabled: %s\n", m);
+ continue;
+ } else if (!strcmp(protocol, "AVP") && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
+ ast_log(LOG_WARNING, "Received AVP profile in text offer but AVPF is enabled: %s\n", m);
+ continue;
+ } else if (strcmp(protocol, "AVP") && strcmp(protocol, "AVPF")) {
+ ast_log(LOG_WARNING, "Unknown RTP profile in text offer: %s\n", m);
+ continue;
}
if (has_media_stream(p, SDP_TEXT)) {
@@ -10692,13 +10831,23 @@
*/
static void set_destination(struct sip_pvt *p, char *uri)
{
- char *h, *maddr, hostname[256];
+ char *trans, *h, *maddr, hostname[256];
int hn;
int debug=sip_debug_test_pvt(p);
int tls_on = FALSE;
if (debug)
ast_verbose("set_destination: Parsing <%s> for address/port to send to\n", uri);
+
+ if ((trans = strcasestr(uri, ";transport="))) {
+ trans += strlen(";transport=");
+
+ if (!strncasecmp(trans, "ws", 2)) {
+ if (debug)
+ ast_verbose("set_destination: URI is for WebSocket, we can't set destination\n");
+ return;
+ }
+ }
/* Find and parse hostname */
h = strchr(uri, '@');
@@ -12026,6 +12175,15 @@
}
}
+static char *get_sdp_rtp_profile(const struct sip_pvt *p, unsigned int secure)
+{
+ if (ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
+ return secure ? "SAVPF" : "AVPF";
+ } else {
+ return secure ? "SAVP" : "AVP";
+ }
+}
+
/*! \brief Add Session Description Protocol message
If oldsdp is TRUE, then the SDP version number is not incremented. This mechanism
@@ -12186,7 +12344,7 @@
if (needvideo) {
get_crypto_attrib(p, p->vsrtp, &v_a_crypto);
ast_str_append(&m_video, 0, "m=video %d RTP/%s", ast_sockaddr_port(&vdest),
- v_a_crypto ? "SAVP" : "AVP");
+ get_sdp_rtp_profile(p, a_crypto ? 1 : 0));
/* Build max bitrate string */
if (p->maxcallbitrate)
@@ -12207,7 +12365,7 @@
ast_verbose("Lets set up the text sdp\n");
get_crypto_attrib(p, p->tsrtp, &t_a_crypto);
ast_str_append(&m_text, 0, "m=text %d RTP/%s", ast_sockaddr_port(&tdest),
- t_a_crypto ? "SAVP" : "AVP");
+ get_sdp_rtp_profile(p, a_crypto ? 1 : 0));
if (debug) { /* XXX should I use tdest below ? */
ast_verbose("Text is at %s\n", ast_sockaddr_stringify(&taddr));
}
@@ -12224,7 +12382,7 @@
get_crypto_attrib(p, p->srtp, &a_crypto);
ast_str_append(&m_audio, 0, "m=audio %d RTP/%s", ast_sockaddr_port(&dest),
- a_crypto ? "SAVP" : "AVP");
+ get_sdp_rtp_profile(p, a_crypto ? 1 : 0));
/* Now, start adding audio codecs. These are added in this order:
- First what was requested by the calling channel
@@ -14639,6 +14797,9 @@
if (socket->tcptls_session) {
ao2_ref(socket->tcptls_session, -1);
socket->tcptls_session = NULL;
+ } else if (socket->ws_session) {
+ ast_websocket_unref(socket->ws_session);
+ socket->ws_session = NULL;
}
}
}
@@ -14661,6 +14822,9 @@
if (peer->socket.tcptls_session) {
ao2_ref(peer->socket.tcptls_session, -1);
peer->socket.tcptls_session = NULL;
+ } else if (peer->socket.ws_session) {
+ ast_websocket_unref(peer->socket.ws_session);
+ peer->socket.ws_session = NULL;
}
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
@@ -16840,6 +17004,11 @@
uint16_t port;
ast_copy_string(via, sip_get_header(req, "Via"), sizeof(via));
+
+ /* If this is via WebSocket we don't use the Via header contents at all */
+ if (!strncasecmp(via, "SIP/2.0/WS", 10)) {
+ return;
+ }
/* Work on the leftmost value of the topmost Via header */
c = strchr(via, ',');
@@ -20984,6 +21153,9 @@
if (p->socket.tcptls_session) {
ao2_ref(p->socket.tcptls_session, -1);
p->socket.tcptls_session = NULL;
+ } else if (p->socket.ws_session) {
+ ast_websocket_unref(p->socket.ws_session);
+ p->socket.ws_session = NULL;
}
set_socket_transport(&p->socket, transport);
@@ -27196,6 +27368,9 @@
(s->tcptls_session->fd != -1)) {
return s->tcptls_session->fd;
}
+ if ((s->type & (SIP_TRANSPORT_WS | SIP_TRANSPORT_WSS))) {
+ return s->ws_session ? ast_websocket_fd(s->ws_session) : -1;
+ }
/*! \todo Check this... This might be wrong, depending on the proxy configuration
If proxy is in "force" mode its correct.
@@ -29188,6 +29363,10 @@
if (!strncasecmp(trans, "udp", 3)) {
peer->transports |= SIP_TRANSPORT_UDP;
+ } else if (!strncasecmp(trans, "wss", 3)) {
+ peer->transports |= SIP_TRANSPORT_WSS;
+ } else if (!strncasecmp(trans, "ws", 2)) {
+ peer->transports |= SIP_TRANSPORT_WS;
} else if (sip_cfg.tcp_enabled && !strncasecmp(trans, "tcp", 3)) {
peer->transports |= SIP_TRANSPORT_TCP;
} else if (default_tls_cfg.enabled && !strncasecmp(trans, "tls", 3)) {
@@ -29538,6 +29717,8 @@
ast_set2_flag(&peer->flags[2], !strcasecmp(v->value, "32"), SIP_PAGE3_SRTP_TAG_32);
} else if (!strcasecmp(v->name, "snom_aoc_enabled")) {
ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_SNOM_AOC);
+ } else if (!strcasecmp(v->name, "avpf")) {
+ ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_USE_AVPF);
}
}
@@ -29651,7 +29832,6 @@
* 3. The socket.type is not set yet. */
if (((peer->socket.type != peer->default_outbound_transport) && (peer->expire == -1)) ||
!(peer->socket.type & peer->transports) || !(peer->socket.type)) {
-
set_socket_transport(&peer->socket, peer->default_outbound_transport);
}
@@ -30189,6 +30369,10 @@
default_transports |= SIP_TRANSPORT_TCP;
} else if (!strncasecmp(trans, "tls", 3)) {
default_transports |= SIP_TRANSPORT_TLS;
+ } else if (!strncasecmp(trans, "wss", 3)) {
+ default_transports |= SIP_TRANSPORT_WSS;
+ } else if (!strncasecmp(trans, "ws", 2)) {
+ default_transports |= SIP_TRANSPORT_WS;
} else {
ast_log(LOG_NOTICE, "'%s' is not a valid transport type. if no other is specified, udp will be used.\n", trans);
}
@@ -32598,6 +32782,8 @@
sip_register_tests();
network_change_event_subscribe();
+ ast_websocket_add_protocol("sip", sip_websocket_callback);
+
return AST_MODULE_LOAD_SUCCESS;
}
@@ -32609,6 +32795,8 @@
struct ast_context *con;
struct ao2_iterator i;
int wait_count;
+
+ ast_websocket_remove_protocol("sip", sip_websocket_callback);
network_change_event_unsubscribe();
acl_change_event_unsubscribe();
Modified: trunk/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/sip.h?view=diff&rev=370072&r1=370071&r2=370072
==============================================================================
--- trunk/channels/sip/include/sip.h (original)
+++ trunk/channels/sip/include/sip.h Mon Jul 16 07:35:04 2012
@@ -35,6 +35,7 @@
#include "asterisk/indications.h"
#include "asterisk/security_events.h"
#include "asterisk/features.h"
+#include "asterisk/http_websocket.h"
#ifndef FALSE
#define FALSE 0
@@ -369,10 +370,11 @@
#define SIP_PAGE3_NAT_AUTO_RPORT (1 << 2) /*!< DGP: Set SIP_NAT_FORCE_RPORT when NAT is detected */
#define SIP_PAGE3_NAT_AUTO_COMEDIA (1 << 3) /*!< DGP: Set SIP_PAGE2_SYMMETRICRTP when NAT is detected */
#define SIP_PAGE3_DIRECT_MEDIA_OUTGOING (1 << 4) /*!< DP: Only send direct media reinvites on outgoing calls */
+#define SIP_PAGE3_USE_AVPF (1 << 5) /*!< DGP: Support a minimal AVPF-compatible profile */
#define SIP_PAGE3_FLAGS_TO_COPY \
(SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32 | SIP_PAGE3_NAT_AUTO_RPORT | SIP_PAGE3_NAT_AUTO_COMEDIA | \
- SIP_PAGE3_DIRECT_MEDIA_OUTGOING)
+ SIP_PAGE3_DIRECT_MEDIA_OUTGOING | SIP_PAGE3_USE_AVPF)
#define CHECK_AUTH_BUF_INITLEN 256
@@ -564,6 +566,8 @@
SIP_TRANSPORT_UDP = 1, /*!< Unreliable transport for SIP, needs retransmissions */
SIP_TRANSPORT_TCP = 1 << 1, /*!< Reliable, but unsecure */
SIP_TRANSPORT_TLS = 1 << 2, /*!< TCP/TLS - reliable and secure transport for signalling */
+ SIP_TRANSPORT_WS = 1 << 3, /*!< WebSocket, unsecure */
+ SIP_TRANSPORT_WSS = 1 << 4, /*!< WebSocket, secure */
};
/*! \brief Automatic peer registration behavior
@@ -769,6 +773,7 @@
int fd; /*!< Filed descriptor, the actual socket */
uint16_t port;
struct ast_tcptls_session_instance *tcptls_session; /* If tcp or tls, a socket manager */
+ struct ast_websocket *ws_session; /*! If ws or wss, a WebSocket session */
};
/*! \brief sip_request: The data grabbed from the UDP socket
@@ -1284,7 +1289,7 @@
enum sip_transport default_outbound_transport; /*!< Peer Registration may change the default outbound transport.
If register expires, default should be reset. to this value */
/* things that don't belong in flags */
- unsigned short transports:3; /*!< Transports (enum sip_transport) that are acceptable for this peer */
+ unsigned short transports:5; /*!< Transports (enum sip_transport) that are acceptable for this peer */
unsigned short is_realtime:1; /*!< this is a 'realtime' peer */
unsigned short rt_fromcontact:1;/*!< copy fromcontact from realtime */
unsigned short host_dynamic:1; /*!< Dynamic Peers register with Asterisk */
Modified: trunk/channels/sip/sdp_crypto.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/sdp_crypto.c?view=diff&rev=370072&r1=370071&r2=370072
==============================================================================
--- trunk/channels/sip/sdp_crypto.c (original)
+++ trunk/channels/sip/sdp_crypto.c Mon Jul 16 07:35:04 2012
@@ -218,7 +218,7 @@
return -1;
}
- if (session_params) {
+ if (!ast_strlen_zero(session_params)) {
ast_log(LOG_WARNING, "Unsupported crypto parameters: %s", session_params);
return -1;
}
Modified: trunk/channels/sip/security_events.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/security_events.c?view=diff&rev=370072&r1=370071&r2=370072
==============================================================================
--- trunk/channels/sip/security_events.c (original)
+++ trunk/channels/sip/security_events.c Mon Jul 16 07:35:04 2012
@@ -45,8 +45,10 @@
case SIP_TRANSPORT_UDP:
return AST_SECURITY_EVENT_TRANSPORT_UDP;
case SIP_TRANSPORT_TCP:
+ case SIP_TRANSPORT_WS:
return AST_SECURITY_EVENT_TRANSPORT_TCP;
case SIP_TRANSPORT_TLS:
+ case SIP_TRANSPORT_WSS:
return AST_SECURITY_EVENT_TRANSPORT_TLS;
}
Modified: trunk/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=370072&r1=370071&r2=370072
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Mon Jul 16 07:35:04 2012
@@ -745,7 +745,7 @@
;
;register => tls://username:xxxxxx@sip-tls-proxy.example.org
;
-; The 'transport' part defaults to 'udp' but may also be 'tcp' or 'tls'.
+; The 'transport' part defaults to 'udp' but may also be 'tcp', 'tls', 'ws', or 'wss'.
; Using 'udp://' explicitly is also useful in case the username part
; contains a '/' ('user/name').
@@ -977,7 +977,10 @@
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.
;encryption_taglen=80 ; Set the auth tag length offered in the INVITE either 32/80 default 80
-
+;
+;avpf=yes ; Enable inter-operability with media streams using the AVPF RTP profile.
+ ; This will cause all offers and answers to use AVPF (or SAVPF). This
+ ; option may be specified at the global or peer scope.
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
Modified: trunk/include/asterisk/http_websocket.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/http_websocket.h?view=diff&rev=370072&r1=370071&r2=370072
==============================================================================
--- trunk/include/asterisk/http_websocket.h (original)
+++ trunk/include/asterisk/http_websocket.h Mon Jul 16 07:35:04 2012
@@ -176,105 +176,12 @@
*/
int ast_websocket_is_secure(struct ast_websocket *session);
-#endif /* _ASTERISK_HTTP_WEBSOCKET_H */
-/*
- * Asterisk -- An open source telephony toolkit.
+/*!
+ * \brief Set the socket of a WebSocket session to be non-blocking.
*
- * Copyright (C) 2012, Digium, Inc.
- *
- * Joshua Colp <jcolp at digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
+ * \retval 0 on success
+ * \retval -1 on failure
*/
+int ast_websocket_set_nonblock(struct ast_websocket *session);
-#ifndef _ASTERISK_HTTP_WEBSOCKET_H
-#define _ASTERISK_HTTP_WEBSOCKET_H
-
-#include "asterisk/module.h"
-
-/*!
- * \file http_websocket.h
- * \brief Support for WebSocket connections within the Asterisk HTTP server.
- *
- * \author Joshua Colp <jcolp at digium.com>
- *
- */
-
-/*! \brief WebSocket operation codes */
-enum ast_websocket_opcode {
- AST_WEBSOCKET_OPCODE_TEXT = 0x1, /*!< Text frame */
- AST_WEBSOCKET_OPCODE_BINARY = 0x2, /*!< Binary frame */
- AST_WEBSOCKET_OPCODE_PING = 0x9, /*!< Request that the other side respond with a pong */
- AST_WEBSOCKET_OPCODE_PONG = 0xA, /*!< Response to a ping */
- AST_WEBSOCKET_OPCODE_CLOSE = 0x8, /*!< Connection is being closed */
- AST_WEBSOCKET_OPCODE_CONTINUATION = 0x0, /*!< Continuation of a previous frame */
-};
-
-/*!
- * \brief Callback for when a new connection for a sub-protocol is established
- *
- * \param f Pointer to the file instance for the session
- * \param fd File descriptor for the session
- * \param remote_address The address of the remote party
- *
- * \note Once called the ownership of the session is transferred to the sub-protocol handler. It
- * is responsible for closing and cleaning up.
- *
- */
-typedef void (*ast_websocket_callback)(FILE *f, int fd, struct ast_sockaddr *remote_address);
-
-/*!
- * \brief Add a sub-protocol handler to the server
- *
- * \param name Name of the sub-protocol to register
- * \param callback Callback called when a new connection requesting the sub-protocol is established
- *
- * \retval 0 success
- * \retval -1 if sub-protocol handler could not be registered
- */
-int ast_websocket_add_protocol(char *name, ast_websocket_callback callback);
-
-/*!
- * \brief Remove a sub-protocol handler from the server
- *
- * \param name Name of the sub-protocol to unregister
- * \param callback Callback that was previously registered with the sub-protocol
- *
- * \retval 0 success
- * \retval -1 if sub-protocol was not found or if callback did not match
- */
-int ast_websocket_remove_protocol(char *name, ast_websocket_callback callback);
-
-/*!
- * \brief Read a WebSocket frame and handle it
- *
- * \param f Pointer to the file stream, used to respond to certain frames
- * \param buf Pointer to the buffer containing the frame
- * \param buflen Size of the buffer
- * \param payload_len Pointer to a uint64_t which will be populated with the length of the payload if present
- * \param opcode Pointer to an int which will be populated with the opcode of the frame
- *
- * \retval NULL if no payload is present
- * \retval non-NULL if payload is present, returned pointer points to beginning of payload
- */
-char *ast_websocket_read(FILE *f, char *buf, size_t buflen, uint64_t *payload_len, int *opcode);
-
-/*!
- * \brief Construct and transmit a WebSocket frame
- *
- * \param f Pointer to the file stream which the frame will be sent on
- * \param opcode WebSocket operation code to place in the frame
- * \param payload Optional pointer to a payload to add to the frame
- * \param actual_length Length of the payload (0 if no payload)
- */
-void ast_websocket_write(FILE *f, int op_code, char *payload, uint64_t actual_length);
-
-#endif /* _ASTERISK_HTTP_WEBSOCKET_H */
+#endif
Modified: trunk/res/res_http_websocket.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_http_websocket.c?view=diff&rev=370072&r1=370071&r2=370072
==============================================================================
--- trunk/res/res_http_websocket.c (original)
+++ trunk/res/res_http_websocket.c Mon Jul 16 07:35:04 2012
@@ -267,6 +267,23 @@
return session->secure;
}
+int ast_websocket_set_nonblock(struct ast_websocket *session)
+{
+ int flags;
+
+ if ((flags = fcntl(session->fd, F_GETFL)) == -1) {
+ return -1;
+ }
+
+ flags |= O_NONBLOCK;
+
+ if ((flags = fcntl(session->fd, F_SETFL, flags)) == -1) {
+ return -1;
+ }
+
+ return 0;
+}
+
int ast_websocket_read(struct ast_websocket *session, char **payload, uint64_t *payload_len, enum ast_websocket_opcode *opcode, int *fragmented)
{
char buf[MAXIMUM_FRAME_SIZE] = "";
More information about the svn-commits
mailing list