[svn-commits] branch 1.2-netsec - r7927 in /branches/1.2-netsec: ./
channels/ include/aster...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Mon Jan 9 21:10:35 CST 2006
Author: kpfleming
Date: Mon Jan 9 21:10:34 2006
New Revision: 7927
URL: http://svn.digium.com/view/asterisk?rev=7927&view=rev
Log:
initial import of Asterisk SIP support for network security devices
Modified:
branches/1.2-netsec/Makefile
branches/1.2-netsec/channels/Makefile
branches/1.2-netsec/channels/chan_sip.c
branches/1.2-netsec/include/asterisk/rtp.h
branches/1.2-netsec/rtp.c
Modified: branches/1.2-netsec/Makefile
URL: http://svn.digium.com/view/asterisk/branches/1.2-netsec/Makefile?rev=7927&r1=7926&r2=7927&view=diff
==============================================================================
--- branches/1.2-netsec/Makefile (original)
+++ branches/1.2-netsec/Makefile Mon Jan 9 21:10:34 2006
@@ -104,6 +104,9 @@
# Inforce the detection of busy singal (get rid of false hangups)
# Don't use together with -DBUSYDETECT_TONEONLY
BUSYDETECT+= #-DBUSYDETECT_COMPARE_TONE_AND_SILENCE
+
+# Comment this if you want to disable MIDCOM
+MIDCOM = -DMIDCOM
ifneq ($(OSARCH),SunOS)
ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk
@@ -331,6 +334,7 @@
ASTCFLAGS+= $(TRACE_FRAMES)
ASTCFLAGS+= $(MALLOC_DEBUG)
ASTCFLAGS+= $(BUSYDETECT)
+ASTCFLAGS+= $(MIDCOM)
ASTCFLAGS+= $(OPTIONS)
ASTCFLAGS+= -fomit-frame-pointer
SUBDIRS=res channels pbx apps codecs formats agi cdr funcs utils stdtime
Modified: branches/1.2-netsec/channels/Makefile
URL: http://svn.digium.com/view/asterisk/branches/1.2-netsec/channels/Makefile?rev=7927&r1=7926&r2=7927&view=diff
==============================================================================
--- branches/1.2-netsec/channels/Makefile (original)
+++ branches/1.2-netsec/channels/Makefile Mon Jan 9 21:10:34 2006
@@ -95,6 +95,11 @@
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/lib/libpri.so.1)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/lib/libpri.so.1),)
CFLAGS+=-DZAPATA_PRI
ZAPPRI=-lpri
+endif
+
+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/lib/asterisk/modules/res_netsec.so)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/lib/asterisk/modules/res_netsec.so),)
+ CFLAGS+=-DSIP_MIDCOM
+# NETSEC=-lnetsec -lssl -lcrypto -lstdc++
endif
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/lib/libmfcr2.so.1)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/lib/libmfcr2.so.1),)
@@ -207,7 +212,7 @@
$(CC) $(SOLINK) -o $@ $< $(ZAPPRI) $(ZAPR2) -ltonezone
chan_sip.so: chan_sip.o
- $(CC) $(SOLINK) -o $@ ${CYGSOLINK} chan_sip.o ${CYGSOLIB}
+ $(CC) $(SOLINK) -o $@ ${CYGSOLINK} chan_sip.o ${CYGSOLIB} ${NETSEC}
chan_agent.so: chan_agent.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} chan_agent.o ${CYGSOLIB} ${CYG_CHAN_AGENT}
Modified: branches/1.2-netsec/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.2-netsec/channels/chan_sip.c?rev=7927&r1=7926&r2=7927&view=diff
==============================================================================
--- branches/1.2-netsec/channels/chan_sip.c (original)
+++ branches/1.2-netsec/channels/chan_sip.c Mon Jan 9 21:10:34 2006
@@ -84,6 +84,10 @@
#ifdef OSP_SUPPORT
#include "asterisk/astosp.h"
+#endif
+
+#ifdef SIP_MIDCOM
+#include "asterisk/res_netsec.h"
#endif
#ifndef DEFAULT_USERAGENT
@@ -679,6 +683,10 @@
struct ast_dsp *vad; /*!< Voice Activation Detection dsp */
+#ifdef SIP_MIDCOM
+ void *r;
+#endif
+
struct sip_peer *peerpoke; /*!< If this calls is to poke a peer, which one */
struct sip_registry *registry; /*!< If this is a REGISTER call, to which registry */
struct ast_rtp *rtp; /*!< RTP Session */
@@ -922,6 +930,25 @@
static int transmit_state_notify(struct sip_pvt *p, int state, int full, int substate);
static char *gettag(struct sip_request *req, char *header, char *tagbuf, int tagbufsize);
+#ifdef SIP_MIDCOM
+static void sip_rtp_get_peer_audio_helper(void *p, struct sockaddr_in *them);
+static void sip_rtp_get_peer_video_helper(void *p, struct sockaddr_in *them);
+static void sip_rtp_get_us_audio_helper(void *p, struct sockaddr_in *sin);
+static void sip_rtp_get_us_video_helper(void *p, struct sockaddr_in *vsin);
+static void sip_map_hook_struct(void *p, void *r);
+static void *sip_get_hook_struct(void *p);
+static int sip_get_flag_novideo(void *p);
+static int sip_cmp_sa_addr(void *p, struct sockaddr_in *addr);
+static void sip_get_recv_addr(void *p, struct in_addr *addr);
+static char *sip_get_username(void *p);
+static struct ast_channel *sip_channel_helper(void *p);
+static struct ast_channel *sip_bridged_channel_helper(void *p);
+static int sip_get_capability_helper(void *p);
+static void sip_softhangup_helper(void *p);
+
+extern struct ast_sip_hook_cb *m_cb;
+#endif
+
/*! \brief Definition of this channel for PBX channel registration */
static const struct ast_channel_tech sip_tech = {
.type = channeltype,
@@ -2094,6 +2121,11 @@
if (sip_debug_test_pvt(p))
ast_verbose("Destroying call '%s'\n", p->callid);
+
+#ifdef SIP_MIDCOM
+ if (m_cb)
+ m_cb->__sip_destroy_hook(p);
+#endif
if (dumphistory)
sip_dump_history(p);
@@ -2419,6 +2451,12 @@
if (ast->_state != AST_STATE_UP)
needcancel = 1;
+#ifdef SIP_MIDCOM
+ /* For callee to shutdown, send "BYE" instead of "CANCEL"
+ -- this needs to be verified */
+ if (m_cb && ast_test_flag(p, SIP_OUTGOING)) needcancel = 0;
+#endif
+
/* Disconnect */
p = ast->tech_pvt;
if (p->vad) {
@@ -4339,8 +4377,22 @@
ast_rtp_get_us(p->vrtp, &vsin);
if (p->redirip.sin_addr.s_addr) {
+#ifdef SIP_MIDCOM
+ if (m_cb && p->r) {
+ struct sockaddr_in redirip_hook;
+ char iabuf2[INET_ADDRSTRLEN];
+ m_cb->ast_get_redirip_audio_hook(p->r, &redirip_hook);
+ ast_log(LOG_DEBUG, "Replacing %s:%d by %s:%d in SDP before sending to %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->redirip.sin_addr), ntohs(p->redirip.sin_port), ast_inet_ntoa(iabuf2, sizeof(iabuf2), redirip_hook.sin_addr), ntohs(redirip_hook.sin_port), p->username);
+ dest.sin_port = redirip_hook.sin_port;
+ dest.sin_addr = redirip_hook.sin_addr;
+ } else {
dest.sin_port = p->redirip.sin_port;
dest.sin_addr = p->redirip.sin_addr;
+ }
+#else
+ dest.sin_port = p->redirip.sin_port;
+ dest.sin_addr = p->redirip.sin_addr;
+#endif
if (p->redircodecs)
capability = p->redircodecs;
} else {
@@ -4351,8 +4403,22 @@
/* Determine video destination */
if (p->vrtp) {
if (p->vredirip.sin_addr.s_addr) {
+#ifdef SIP_MIDCOM
+ if (m_cb && p->r) {
+ struct sockaddr_in vredirip_hook;
+ char iabuf2[INET_ADDRSTRLEN];
+ m_cb->ast_get_vredirip_video_hook(p->r, &vredirip_hook);
+ ast_log(LOG_DEBUG, "Replacing %s:%d by %s:%d in video SDP before sending to %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->vredirip.sin_addr), ntohs(p->vredirip.sin_port), ast_inet_ntoa(iabuf2, sizeof(iabuf2), vredirip_hook.sin_addr), ntohs(vredirip_hook.sin_port), p->username);
+ vdest.sin_port = vredirip_hook.sin_port;
+ vdest.sin_addr = vredirip_hook.sin_addr;
+ } else {
vdest.sin_port = p->vredirip.sin_port;
vdest.sin_addr = p->vredirip.sin_addr;
+ }
+#else
+ vdest.sin_port = p->vredirip.sin_port;
+ vdest.sin_addr = p->vredirip.sin_addr;
+#endif
} else {
vdest.sin_addr = p->ourip;
vdest.sin_port = vsin.sin_port;
@@ -4509,6 +4575,14 @@
} else {
ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
}
+#ifdef SIP_MIDCOM
+ if (m_cb) {
+ if (!m_cb->transmit_response_with_sdp_hook(p)) {
+ ast_log(LOG_NOTICE, "Failed transmit_response_with_sdp_hook()\n");
+ return -1;
+ }
+ }
+#endif
return send_response(p, &resp, retrans, seqno);
}
@@ -4570,6 +4644,19 @@
static int transmit_reinvite_with_sdp(struct sip_pvt *p)
{
struct sip_request req;
+
+#ifdef SIP_MIDCOM
+ if (m_cb) {
+ if (!m_cb->transmit_reinvite_with_sdp_hook(p)) {
+ ast_log(LOG_NOTICE, "Failed transmit_reinvite_with_sdp_hook()\n");
+ if (p->owner)
+ ast_queue_hangup(p->owner);
+ else
+ ast_set_flag(p, SIP_NEEDDESTROY);
+ }
+ }
+#endif
+
if (ast_test_flag(p, SIP_REINVITE_UPDATE))
reqprep(&req, p, SIP_UPDATE, 0, 1);
else
@@ -9467,6 +9554,16 @@
p->authtries = 0;
if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) {
process_sdp(p, req);
+#ifdef SIP_MIDCOM
+ if (m_cb) {
+ if (!m_cb->handle_response_invite_hook(p)) {
+ if (p->owner)
+ ast_queue_hangup(p->owner);
+ else
+ ast_set_flag(p, SIP_NEEDDESTROY);
+ }
+ }
+#endif
}
/* Parse contact header for continued conversation */
@@ -10308,6 +10405,19 @@
ast_set_flag(p, SIP_NEEDDESTROY);
return -1;
}
+#ifdef SIP_MIDCOM
+ if (m_cb) {
+ if (!m_cb->handle_request_invite_hook((void *)p)) {
+ ast_log(LOG_NOTICE, "Failed to NAT for (%s)\n", get_header(req, "From"));
+ if (ignore)
+ transmit_response(p, "403 Forbidden", req);
+ else
+ transmit_response_reliable(p, "403 Forbidden", req, 1);
+ ast_set_flag(p, SIP_NEEDDESTROY);
+ return 0;
+ }
+ }
+#endif
} else {
p->jointcapability = p->capability;
ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
@@ -12655,8 +12765,13 @@
if (!p)
return NULL;
ast_mutex_lock(&p->lock);
- if (p->rtp && ast_test_flag(p, SIP_CAN_REINVITE))
+ if (p->rtp && ast_test_flag(p, SIP_CAN_REINVITE)) {
rtp = p->rtp;
+#ifdef SIP_MIDCOM
+ if (m_cb)
+ m_cb->ast_rtp_nat_us_audio_hook(rtp, p->r); /* change the ip port in rtp */
+#endif
+ }
ast_mutex_unlock(&p->lock);
return rtp;
}
@@ -12671,8 +12786,13 @@
return NULL;
ast_mutex_lock(&p->lock);
- if (p->vrtp && ast_test_flag(p, SIP_CAN_REINVITE))
+ if (p->vrtp && ast_test_flag(p, SIP_CAN_REINVITE)) {
rtp = p->vrtp;
+#ifdef SIP_MIDCOM
+ if (m_cb)
+ m_cb->ast_rtp_nat_us_video_hook(rtp, p->r); /* change the ip port in rtp */
+#endif
+ }
ast_mutex_unlock(&p->lock);
return rtp;
}
@@ -12686,12 +12806,22 @@
if (!p)
return -1;
ast_mutex_lock(&p->lock);
- if (rtp)
+ if (rtp) {
ast_rtp_get_peer(rtp, &p->redirip);
+#ifdef SIP_MIDCOM
+ if (m_cb)
+ m_cb->ast_rtp_get_their_nat_audio_hook(rtp, p->r);
+#endif
+ }
else
memset(&p->redirip, 0, sizeof(p->redirip));
- if (vrtp)
+ if (vrtp) {
ast_rtp_get_peer(vrtp, &p->vredirip);
+#ifdef SIP_MIDCOM
+ if (m_cb)
+ m_cb->ast_rtp_get_their_nat_video_hook(vrtp, p->r);
+#endif
+ }
else
memset(&p->vredirip, 0, sizeof(p->vredirip));
p->redircodecs = codecs;
@@ -12958,6 +13088,26 @@
set_rtp_peer: sip_set_rtp_peer,
get_codec: sip_get_codec,
};
+
+#ifdef SIP_MIDCOM
+/*! \brief sip_helper: Interface structure with callbacks used to connect to midcom module --*/
+static struct ast_sip_helper_cb sip_helper = {
+ ast_rtp_get_peer_audio_helper: sip_rtp_get_peer_audio_helper,
+ ast_rtp_get_peer_video_helper: sip_rtp_get_peer_video_helper,
+ ast_rtp_get_us_audio_helper: sip_rtp_get_us_audio_helper,
+ ast_rtp_get_us_video_helper: sip_rtp_get_us_video_helper,
+ ast_map_hook_struct: sip_map_hook_struct,
+ ast_get_hook_struct: sip_get_hook_struct,
+ ast_get_flag_novideo: sip_get_flag_novideo,
+ ast_cmp_sa_addr: sip_cmp_sa_addr,
+ ast_get_recv_addr: sip_get_recv_addr,
+ ast_get_username: sip_get_username,
+ ast_channel_helper: sip_channel_helper,
+ ast_bridged_channel_helper: sip_bridged_channel_helper,
+ ast_get_capability_helper: sip_get_capability_helper,
+ ast_softhangup_helper: sip_softhangup_helper,
+};
+#endif
/*! \brief sip_poke_all_peers: Send a poke to all known peers */
static void sip_poke_all_peers(void)
@@ -13094,6 +13244,12 @@
/* Tell the RTP subdriver that we're here */
ast_rtp_proto_register(&sip_rtp);
+#ifdef SIP_MIDCOM
+ /* Register the sip helper functions */
+ if (m_cb)
+ m_cb->ast_sip_helper_register(&sip_helper);
+#endif
+
/* Register dialplan applications */
ast_register_application(app_dtmfmode, sip_dtmfmode, synopsis_dtmfmode, descrip_dtmfmode);
@@ -13141,6 +13297,12 @@
ast_cli_unregister_multiple(my_clis, sizeof(my_clis) / sizeof(my_clis[0]));
ast_rtp_proto_unregister(&sip_rtp);
+
+#ifdef SIP_MIDCOM
+ /* Unregister the sip helper functions */
+ if (m_cb)
+ m_cb->ast_sip_helper_unregister();
+#endif
ast_manager_unregister("SIPpeers");
ast_manager_unregister("SIPshowpeer");
@@ -13226,4 +13388,78 @@
return (char *) desc;
}
-
+#ifdef SIP_MIDCOM
+static void sip_rtp_get_peer_audio_helper(void *p, struct sockaddr_in *them)
+{
+ ast_rtp_get_peer(((struct sip_pvt*)p)->rtp, them);
+}
+
+static void sip_rtp_get_peer_video_helper(void *p, struct sockaddr_in *them)
+{
+ ast_rtp_get_peer(((struct sip_pvt*)p)->vrtp, them);
+}
+
+static void sip_rtp_get_us_audio_helper(void *p, struct sockaddr_in *sin)
+{
+ ast_rtp_get_us(((struct sip_pvt*)p)->rtp, sin);
+ sin->sin_addr = ((struct sip_pvt*)p)->ourip;
+}
+
+static void sip_rtp_get_us_video_helper(void *p, struct sockaddr_in *vsin)
+{
+ ast_rtp_get_us(((struct sip_pvt*)p)->vrtp, vsin);
+ vsin->sin_addr = ((struct sip_pvt*)p)->ourip;
+}
+
+static void sip_map_hook_struct(void *p, void *r)
+{
+ ((struct sip_pvt*)p)->r = r;
+}
+
+static void *sip_get_hook_struct(void *p)
+{
+ return ((struct sip_pvt*)p)->r;
+}
+
+static int sip_get_flag_novideo(void *p)
+{
+ return ast_test_flag((struct sip_pvt*)p, SIP_NOVIDEO);
+}
+
+static int sip_cmp_sa_addr(void *p, struct sockaddr_in *addr)
+{
+ return (((struct sip_pvt*)p)->sa.sin_addr.s_addr == addr->sin_addr.s_addr);
+}
+
+static void sip_get_recv_addr(void *p, struct in_addr *addr)
+{
+ memcpy(addr, &((struct sip_pvt *)p)->recv.sin_addr, sizeof(struct in_addr));
+}
+
+static char *sip_get_username(void *p)
+{
+ return ((struct sip_pvt*)p)->username;
+}
+
+static struct ast_channel *sip_channel_helper(void *p)
+{
+ return ((struct sip_pvt*)p)->owner;
+}
+
+static struct ast_channel *sip_bridged_channel_helper(void *p)
+{
+ return ast_bridged_channel(((struct sip_pvt*)p)->owner);
+}
+
+static int sip_get_capability_helper(void *p)
+{
+ return ((struct sip_pvt*)p)->jointcapability;
+}
+
+static void sip_softhangup_helper(void *p)
+{
+ if (p && ((struct sip_pvt *)p)->owner)
+ ast_softhangup(((struct sip_pvt *)p)->owner, AST_SOFTHANGUP_APPUNLOAD);
+}
+#endif
+
Modified: branches/1.2-netsec/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/branches/1.2-netsec/include/asterisk/rtp.h?rev=7927&r1=7926&r2=7927&view=diff
==============================================================================
--- branches/1.2-netsec/include/asterisk/rtp.h (original)
+++ branches/1.2-netsec/include/asterisk/rtp.h Mon Jan 9 21:10:34 2006
@@ -100,6 +100,11 @@
void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
+#ifdef MIDCOM
+void ast_rtp_nat_us(struct ast_rtp *rtp, struct sockaddr_in *our_nat);
+void ast_rtp_get_their_nat(struct ast_rtp *rtp, struct sockaddr_in *their_nat);
+#endif
+
void ast_rtp_destroy(struct ast_rtp *rtp);
void ast_rtp_reset(struct ast_rtp *rtp);
Modified: branches/1.2-netsec/rtp.c
URL: http://svn.digium.com/view/asterisk/branches/1.2-netsec/rtp.c?rev=7927&r1=7926&r2=7927&view=diff
==============================================================================
--- branches/1.2-netsec/rtp.c (original)
+++ branches/1.2-netsec/rtp.c Mon Jan 9 21:10:34 2006
@@ -125,6 +125,10 @@
int rtp_lookup_code_cache_code;
int rtp_lookup_code_cache_result;
int rtp_offered_from_local;
+
+#ifdef MIDCOM
+ struct sockaddr_in them_midcom_nat;
+#endif
struct ast_rtcp *rtcp;
};
@@ -1041,6 +1045,20 @@
memcpy(us, &rtp->us, sizeof(rtp->us));
}
+#ifdef MIDCOM /* RANCH */
+void ast_rtp_nat_us(struct ast_rtp *rtp, struct sockaddr_in *our_nat)
+{
+ memcpy(&rtp->them_midcom_nat, our_nat, sizeof(rtp->them_midcom_nat));
+}
+
+void ast_rtp_get_their_nat(struct ast_rtp *rtp, struct sockaddr_in *their_nat)
+{
+ their_nat->sin_family = AF_INET;
+ their_nat->sin_port = rtp->them_midcom_nat.sin_port;
+ their_nat->sin_addr = rtp->them_midcom_nat.sin_addr;
+}
+#endif
+
void ast_rtp_stop(struct ast_rtp *rtp)
{
memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
@@ -1515,7 +1533,6 @@
struct sockaddr_in t0, t1;
struct sockaddr_in vt0, vt1;
char iabuf[INET_ADDRSTRLEN];
-
void *pvt0, *pvt1;
int codec0,codec1, oldcodec0, oldcodec1;
@@ -1670,8 +1687,12 @@
if (option_debug) {
ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), t0.sin_addr), ntohs(t0.sin_port), codec0);
+ ast_log(LOG_DEBUG, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n",
+ c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vt0.sin_addr), ntohs(vt0.sin_port), codec0);
ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0);
+ ast_log(LOG_DEBUG, "Oooh, '%s' wasv %s:%d/(format %d)\n",
+ c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vac0.sin_addr), ntohs(vac0.sin_port), oldcodec0);
}
if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
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