diff options
| -rw-r--r-- | Makefile | 4 | ||||
| -rw-r--r-- | channels/Makefile | 7 | ||||
| -rw-r--r-- | channels/chan_sip.c | 244 | ||||
| -rw-r--r-- | include/asterisk/rtp.h | 5 | ||||
| -rw-r--r-- | rtp.c | 23 |
5 files changed, 277 insertions, 6 deletions
@@ -105,6 +105,9 @@ BUSYDETECT+= #-DBUSYDETECT_TONEONLY # 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 ASTVARLIBDIR=$(INSTALL_PREFIX)/var/lib/asterisk @@ -331,6 +334,7 @@ ASTCFLAGS+= $(DEBUG_THREADS) ASTCFLAGS+= $(TRACE_FRAMES) ASTCFLAGS+= $(MALLOC_DEBUG) ASTCFLAGS+= $(BUSYDETECT) +ASTCFLAGS+= $(MIDCOM) ASTCFLAGS+= $(OPTIONS) ASTCFLAGS+= -fomit-frame-pointer SUBDIRS=res channels pbx apps codecs formats agi cdr funcs utils stdtime diff --git a/channels/Makefile b/channels/Makefile index 704ed505c..349343f5a 100644 --- a/channels/Makefile +++ b/channels/Makefile @@ -97,6 +97,11 @@ ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/lib/libpri.so.1)$(wildcard $(CROSS ZAPPRI=-lpri endif +ifneq ($(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),) CFLAGS+=-DZAPATA_R2 ZAPR2=-lmfcr2 @@ -207,7 +212,7 @@ chan_zap.so: chan_zap.o $(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} ${CYG_CHAN_AGENT} diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 46c094bb1..34a57e66e 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -86,6 +86,10 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/astosp.h" #endif +#ifdef SIP_MIDCOM +#include "asterisk/res_netsec.h" +#endif + #ifndef DEFAULT_USERAGENT #define DEFAULT_USERAGENT "Asterisk PBX" #endif @@ -679,6 +683,10 @@ static struct sip_pvt { 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 const struct cfsubscription_types *find_subscription_type(enum subscripti 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, @@ -2095,6 +2122,11 @@ static void __sip_destroy(struct sip_pvt *p, int lockowner) 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 @@ static int sip_hangup(struct ast_channel *ast) 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 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p) 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 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p) /* 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 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r } 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 determine_firstline_parts( struct sip_request *req ) 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 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru 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 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int 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 @@ static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan) 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 @@ static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan) 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 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc 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; @@ -12959,6 +13089,26 @@ static struct ast_rtp_protocol sip_rtp = { 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 @@ int load_module() /* 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); @@ -13142,6 +13298,12 @@ int unload_module() 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 @@ char *description() 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 diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h index bdb0391fd..f1e771e34 100644 --- a/include/asterisk/rtp.h +++ b/include/asterisk/rtp.h @@ -100,6 +100,11 @@ void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them); 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); @@ -125,6 +125,10 @@ struct ast_rtp { 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 @@ void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us) 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 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel 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 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel 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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