/** * ysipchan.cpp * This file is part of the YATE Project http://YATE.null.ro * * Yet Another Sip Channel * * Yet Another Telephony Engine - a fully featured software PBX and IVR * Copyright (C) 2004, 2005 Null Team * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ #include #include #include using namespace TelEngine; /* Yate Payloads for the AV profile */ static TokenDict dict_payloads[] = { { "mulaw", 0 }, { "alaw", 8 }, { "gsm", 3 }, { "lpc10", 7 }, { "slin", 11 }, { "g726", 2 }, { "g722", 9 }, { "g723", 4 }, { "g728", 15 }, { "g729", 18 }, { "ilbc", 98 }, { "ilbc20", 98 }, { "ilbc30", 98 }, { "h261", 31 }, { "h263", 34 }, { "mpv", 32 }, { 0, 0 }, }; /* SDP Payloads for the AV profile */ static TokenDict dict_rtpmap[] = { { "PCMU/8000", 0 }, { "PCMA/8000", 8 }, { "GSM/8000", 3 }, { "LPC/8000", 7 }, { "L16/8000", 11 }, { "G726-32/8000", 2 }, { "G722/8000", 9 }, { "G723/8000", 4 }, { "G728/8000", 15 }, { "G729/8000", 18 }, { "iLBC/8000", 98 }, { "H261/90000", 31 }, { "H263/90000", 34 }, { "MPV/90000", 32 }, { 0, 0 }, }; static TokenDict dict_errors[] = { { "incomplete", 484 }, { "noroute", 404 }, { "noconn", 503 }, { "noauth", 401 }, { "nomedia", 415 }, { "busy", 486 }, { "noanswer", 487 }, { "rejected", 406 }, { "forbidden", 403 }, { "offline", 404 }, { "congestion", 480 }, { "failure", 500 }, { "looping", 483 }, { 0, 0 }, }; class RtpMedia : public String { public: RtpMedia(const char* media, const char* formats, int rport = -1, int lport = -1); virtual ~RtpMedia(); inline bool isAudio() const { return m_audio; } inline const String& suffix() const { return m_suffix; } inline const String& id() const { return m_id; } inline const String& format() const { return m_format; } inline const String& formats() const { return m_formats; } inline const String& remotePort() const { return m_rPort; } inline const String& localPort() const { return m_lPort; } const char* fmtList() const; bool update(const char* formats, int rport = -1, int lport = -1); void update(const Message& msg, bool pickFormat); private: bool m_audio; // suffix used for this type String m_suffix; // list of supported format names String m_formats; // format used for sending data String m_format; // id of the local RTP channel String m_id; // remote RTP port String m_rPort; // local RTP port String m_lPort; }; class YateUDPParty : public SIPParty { public: YateUDPParty(Socket* sock, const SocketAddr& addr, int localPort, const char* localAddr = 0); ~YateUDPParty(); virtual void transmit(SIPEvent* event); virtual const char* getProtoName() const; virtual bool setParty(const URI& uri); protected: Socket* m_sock; SocketAddr m_addr; }; class YateSIPEndPoint; class YateSIPEngine : public SIPEngine { public: YateSIPEngine(YateSIPEndPoint* ep); virtual bool buildParty(SIPMessage* message); virtual bool checkUser(const String& username, const String& realm, const String& nonce, const String& method, const String& uri, const String& response, const SIPMessage* message); inline bool prack() const { return m_prack; } private: YateSIPEndPoint* m_ep; bool m_prack; }; class YateSIPLine : public String { YCLASS(YateSIPLine,String) public: YateSIPLine(const String& name); virtual ~YateSIPLine(); void setupAuth(SIPMessage* msg) const; SIPMessage* buildRegister(int expires) const; void login(); void logout(); bool process(SIPEvent* ev); void timer(const Time& when); bool update(const Message& msg); inline const String& getLocalAddr() const { return m_localAddr; } inline const String& getPartyAddr() const { return m_partyAddr; } inline int getLocalPort() const { return m_localPort; } inline int getPartyPort() const { return m_partyPort; } inline const String& getUserName() const { return m_username; } inline const String& getAuthName() const { return m_authname ? m_authname : m_username; } inline const String& domain() const { return m_domain ? m_domain : m_registrar; } inline bool valid() const { return m_valid; } inline bool marked() const { return m_marked; } inline void marked(bool mark) { m_marked = mark; } private: void clearTransaction(); bool change(String& dest, const String& src); bool change(int& dest, int src); String m_registrar; String m_username; String m_authname; String m_password; String m_outbound; String m_domain; String m_display; Time m_resend; int m_interval; SIPTransaction* m_tr; bool m_marked; bool m_valid; String m_localAddr; String m_partyAddr; int m_localPort; int m_partyPort; bool m_localDetect; }; class YateSIPEndPoint : public Thread { public: YateSIPEndPoint(); ~YateSIPEndPoint(); bool Init(void); void run(void); bool incoming(SIPEvent* e, SIPTransaction* t); void invite(SIPEvent* e, SIPTransaction* t); void regreq(SIPEvent* e, SIPTransaction* t); bool generic(SIPEvent* e, SIPTransaction* t); bool buildParty(SIPMessage* message, const char* host = 0, int port = 0, const YateSIPLine* line = 0); inline YateSIPEngine* engine() const { return m_engine; } inline int port() const { return m_port; } inline Socket* socket() const { return m_sock; } private: void addMessage(const char* buf, int len, const SocketAddr& addr, int port); int m_port; Socket* m_sock; SocketAddr m_addr; YateSIPEngine *m_engine; }; class YateSIPConnection : public Channel { YCLASS(YateSIPConnection,Channel) public: enum { Incoming = 0, Outgoing = 1, Ringing = 2, Established = 3, Cleared = 4, }; enum { MediaMissing, MediaStarted, MediaMuted }; YateSIPConnection(SIPEvent* ev, SIPTransaction* tr); YateSIPConnection(Message& msg, const String& uri, const char* target = 0); ~YateSIPConnection(); virtual void disconnected(bool final, const char *reason); virtual bool msgProgress(Message& msg); virtual bool msgRinging(Message& msg); virtual bool msgAnswered(Message& msg); virtual bool msgTone(Message& msg, const char* tone); virtual bool msgText(Message& msg, const char* text); virtual bool callRouted(Message& msg); virtual void callAccept(Message& msg); virtual void callRejected(const char* error, const char* reason, const Message* msg); void startRouter(); bool process(SIPEvent* ev); bool checkUser(SIPTransaction* t, bool refuse = true); void doBye(SIPTransaction* t); void doCancel(SIPTransaction* t); void reInvite(SIPTransaction* t); void hangup(); inline const SIPDialog& dialog() const { return m_dialog; } inline void setStatus(const char *status, int state = -1) { m_status = status; if (state >= 0) m_state = state; } inline void setReason(const char* str = "Request Terminated", int code = 487) { m_reason = str; m_reasonCode = code; } inline SIPTransaction* getTransaction() const { return m_tr; } inline const String& callid() const { return m_callid; } inline const String& user() const { return m_user; } inline const String& getHost() const { return m_host; } inline int getPort() const { return m_port; } inline const String& getRtpAddr() const { return m_externalAddr ? m_externalAddr : m_rtpLocalAddr; } private: void setMedia(ObjList* media); void clearTransaction(); SIPMessage* createDlgMsg(const char* method, const char* uri = 0); bool emitPRACK(const SIPMessage* msg); bool dispatchRtp(RtpMedia* media, const char* addr, bool start, bool pick); SDPBody* createSDP(const char* addr = 0, ObjList* mediaList = 0); SDPBody* createProvisionalSDP(Message& msg); SDPBody* createPasstroughSDP(Message& msg); SDPBody* createRtpSDP(const char* addr, const Message& msg); SDPBody* createRtpSDP(bool start = false); bool startRtp(); bool addRtpParams(Message& msg, const String& natAddr = String::empty()); SIPTransaction* m_tr; bool m_hungup; bool m_byebye; int m_state; String m_reason; int m_reasonCode; String m_callid; // SIP dialog of this call, used for re-INVITE or BYE SIPDialog m_dialog; URI m_uri; // our external IP address, possibly outside of a NAT String m_externalAddr; // if we do RTP forwarding or not bool m_rtpForward; // remote RTP address String m_rtpAddr; // local RTP address String m_rtpLocalAddr; // list of media descriptors ObjList* m_rtpMedia; // unique SDP session number int m_sdpSession; // SDP version number, incremented each time we generate a new SDP int m_sdpVersion; String m_host; String m_user; String m_line; int m_port; Message* m_route; ObjList* m_routes; bool m_authBye; int m_mediaStatus; }; class YateSIPGenerate : public GenObject { YCLASS(YateSIPGenerate,GenObject) public: YateSIPGenerate(SIPMessage* m); virtual ~YateSIPGenerate(); bool process(SIPEvent* ev); bool busy() const { return m_tr != 0; } int code() const { return m_code; } private: void clearTransaction(); SIPTransaction* m_tr; int m_code; }; class UserHandler : public MessageHandler { public: UserHandler() : MessageHandler("user.login",150) { } virtual bool received(Message &msg); }; class SipHandler : public MessageHandler { public: SipHandler() : MessageHandler("xsip.generate",110) { } virtual bool received(Message &msg); }; class SIPDriver : public Driver { public: SIPDriver(); ~SIPDriver(); virtual void initialize(); virtual bool msgExecute(Message& msg, String& dest); virtual bool received(Message &msg, int id); inline YateSIPEndPoint* ep() const { return m_endpoint; } YateSIPConnection* findCall(const String& callid); YateSIPConnection* findDialog(const SIPDialog& dialog); YateSIPLine* findLine(const String& line); YateSIPLine* findLine(const String& addr, int port, const String& user = String::empty()); bool validLine(const String& line); private: YateSIPEndPoint *m_endpoint; }; static SIPDriver plugin; static ObjList s_lines; static Configuration s_cfg; static int s_maxForwards = 20; static bool s_privacy = false; static bool s_auto_nat = true; // Parse a SDP and return a possibly filtered list of SDP media static ObjList* parseSDP(const SDPBody* sdp, String& addr, ObjList* oldMedia = 0, const char* media = 0) { const NamedString* c = sdp->getLine("c"); if (c) { String tmp(*c); if (tmp.startSkip("IN IP4")) { tmp.trimBlanks(); // Handle the case media is muted if (tmp == "0.0.0.0") tmp.clear(); addr = tmp; } } ObjList* lst = 0; c = sdp->getLine("m"); for (; c; c = sdp->getNextLine(c)) { String tmp(*c); int sep = tmp.find(' '); if (sep < 1) continue; String type = tmp.substr(0,sep); tmp >> " "; if (media && (type != media)) continue; int port = 0; tmp >> port >> " RTP/AVP"; String fmt; bool defcodecs = s_cfg.getBoolValue("codecs","default",true); int ptime = 0; while (tmp[0] == ' ') { int var = -1; tmp >> " " >> var; int mode = 0; String payload(lookup(var,dict_payloads)); const ObjList* l = sdp->lines().find(c); while (l && (l = l->skipNext())) { const NamedString* s = static_cast(l->get()); if (s->name() == "m") break; if (s->name() != "a") continue; String line(*s); if (line.startSkip("ptime:",false)) line >> ptime; else if (line.startSkip("rtpmap:",false)) { int num = -1; line >> num >> " "; if (num == var) { for (const TokenDict* map = dict_rtpmap; map->token; map++) { if (line.startsWith(map->token)) { const char* pload = lookup(map->value,dict_payloads); if (pload) payload = pload; break; } } } } else if (line.startSkip("fmtp:",false)) { int num = -1; line >> num >> " "; if (num == var) { if (line.startSkip("mode=",false)) line >> mode; } } } if (payload == "ilbc") { if ((mode == 20) || (ptime == 20)) payload = "ilbc20"; else if ((mode == 30) || (ptime == 30)) payload = "ilbc30"; } XDebug(&plugin,DebugAll,"Payload %d format '%s'",var,payload.c_str()); if (payload && s_cfg.getBoolValue("codecs",payload,defcodecs && DataTranslator::canConvert(payload))) { if (fmt) fmt << ","; fmt << payload; } } RtpMedia* rtp = 0; // try to take the media descriptor from the old list if (oldMedia) { ObjList* om = oldMedia->find(type); if (om) rtp = static_cast(om->remove(false)); } if (rtp) rtp->update(fmt,port); else rtp = new RtpMedia(type,fmt,port); if (!lst) lst = new ObjList; lst->append(rtp); if (media) return lst; } return lst; } static bool isPrivateAddr(const String& host) { if (host.startsWith("192.168.") || host.startsWith("169.254.") || host.startsWith("10.")) return true; String s(host); if (!s.startSkip("172.",false)) return false; int i = 0; s >> i; return (i >= 16) && (i <= 31) && s.startsWith("."); } // List of critical headers we don't want to handle generically static const char* rejectHeaders[] = { "via", "route", "record-route", "call-id", "cseq", "content-length", "www-authenticate", "proxy-authenticate", "authorization", "proxy-authorization", 0 }; // Copy headers from SIP message to Yate message static void copySipHeaders(Message& msg, const SIPMessage& sip) { const ObjList* l = sip.header.skipNull(); for (; l; l = l->skipNext()) { const SIPHeaderLine* t = static_cast(l->get()); String name(t->name()); name.toLower(); const char** hdr = rejectHeaders; for (; *hdr; hdr++) if (name == *hdr) break; if (*hdr) continue; String tmp(*t); const ObjList* p = t->params().skipNull(); for (; p; p = p->skipNext()) { NamedString* s = static_cast(p->get()); tmp << ";" << s->name(); if (!s->null()) tmp << "=" << *s; } msg.addParam("sip_"+name,tmp); } } // Copy headers from Yate message to SIP message static void copySipHeaders(SIPMessage& sip, const Message& msg) { unsigned int n = msg.length(); for (unsigned int i = 0; i < n; i++) { NamedString* str = msg.getParam(i); if (!str) continue; String name(str->name()); if (!name.startSkip("sip_",false)) continue; if (name.trimBlanks().null()) continue; sip.addHeader(name,*str); } } // Copy privacy related information from SIP message to Yate message static void copyPrivacy(Message& msg, const SIPMessage& sip) { bool anonip = (sip.getHeaderValue("Anonymity") &= "ipaddr"); const SIPHeaderLine* hl = sip.getHeader("Remote-Party-ID"); if (!(anonip || hl)) return; const NamedString* p = hl ? hl->getParam("screen") : 0; if (p) msg.setParam("screened",*p); String priv; if (anonip) priv.append("addr",","); p = hl ? hl->getParam("privacy") : 0; if (p) { if ((*p &= "full") || (*p &= "full-network")) priv.append("name,uri",","); else if ((*p &= "name") || (*p &= "name-network")) priv.append("name",","); else if ((*p &= "uri") || (*p &= "uri-network")) priv.append("uri",","); } if (priv) msg.setParam("privacy",priv); } // Copy privacy related information from Yate message to SIP message static void copyPrivacy(SIPMessage& sip, const Message& msg) { String screened(msg.getValue("screened")); String privacy(msg.getValue("privacy")); if (screened.null() && privacy.null()) return; bool screen = screened.toBoolean(); bool anonip = (privacy.find("addr") >= 0); bool privname = (privacy.find("name") >= 0); bool privuri = (privacy.find("uri") >= 0); if (anonip) sip.setHeader("Anonymity","ipaddr"); if (screen || privname || privuri) { const char* caller = msg.getValue("caller","anonymous"); String tmp; tmp << "\"" << msg.getValue("callername",caller) << "\""; tmp << " <" << caller << "@" << msg.getValue("domain","domain") << ">"; SIPHeaderLine* hl = new SIPHeaderLine("Remote-Party-ID",tmp); if (screen) hl->setParam("screen","yes"); if (privname && privuri) hl->setParam("privacy","full"); else if (privname) hl->setParam("privacy","name"); else if (privuri) hl->setParam("privacy","uri"); else hl->setParam("privacy","none"); sip.addHeader(hl); } } RtpMedia::RtpMedia(const char* media, const char* formats, int rport, int lport) : String(media), m_audio(true), m_formats(formats) { DDebug(&plugin,DebugAll,"RtpMedia::RtpMedia('%s','%s',%d,%d) [%p]", media,formats,rport,lport,this); if (operator!=("audio")) { m_audio = false; m_suffix << "_" << media; } int q = m_formats.find(','); m_format = m_formats.substr(0,q); if (rport >= 0) m_rPort = rport; if (lport >= 0) m_lPort = lport; } RtpMedia::~RtpMedia() { DDebug(&plugin,DebugAll,"RtpMedia::~RtpMedia() '%s' [%p]",c_str(),this); } const char* RtpMedia::fmtList() const { if (m_format) return m_format.c_str(); if (m_formats) return m_formats.c_str(); // unspecified audio assumed to support G711 if (m_audio) return "alaw,mulaw"; return 0; } // Update members with data taken from a SDP, return true if something changed bool RtpMedia::update(const char* formats, int rport, int lport) { DDebug(&plugin,DebugAll,"RtpMedia::update('%s',%d,%d) [%p]", formats,rport,lport,this); bool chg = false; String tmp(formats); if (m_formats != tmp) { chg = true; m_formats = tmp; int q = m_formats.find(','); m_format = m_formats.substr(0,q); } if (rport >= 0) { tmp = rport; if (m_rPort != tmp) { chg = true; m_rPort = tmp; } } if (lport >= 0) { tmp = lport; if (m_lPort != tmp) { chg = true; m_lPort = tmp; } } return chg; } // Update members from a dispatched "chan.rtp" message void RtpMedia::update(const Message& msg, bool pickFormat) { m_id = msg.getValue("rtpid",m_id); m_lPort = msg.getValue("localport",m_lPort); if (pickFormat) m_format = msg.getValue("format"); } YateUDPParty::YateUDPParty(Socket* sock, const SocketAddr& addr, int localPort, const char* localAddr) : m_sock(sock), m_addr(addr) { DDebug(&plugin,DebugAll,"YateUDPParty::YateUDPParty() %s:%d [%p]",localAddr,localPort,this); m_localPort = localPort; m_party = m_addr.host(); m_partyPort = m_addr.port(); if (localAddr) m_local = localAddr; else { m_local = "localhost"; Socket s(PF_INET,SOCK_DGRAM,IPPROTO_UDP); if (s.valid() && s.connect(m_addr)) { SocketAddr laddr; if (s.getSockName(laddr)) m_local = laddr.host(); } } DDebug(&plugin,DebugAll,"YateUDPParty local %s:%d party %s:%d", m_local.c_str(),m_localPort, m_party.c_str(),m_partyPort); } YateUDPParty::~YateUDPParty() { DDebug(&plugin,DebugAll,"YateUDPParty::~YateUDPParty() [%p]",this); m_sock = 0; } void YateUDPParty::transmit(SIPEvent* event) { const SIPMessage* msg = event->getMessage(); if (!msg) return; String tmp; if (msg->isAnswer()) tmp << "code " << msg->code; else tmp << "'" << msg->method << " " << msg->uri << "'"; if (plugin.debugAt(DebugInfo)) { String buf((char*)msg->getBuffer().data(),msg->getBuffer().length()); Debug(&plugin,DebugInfo,"Sending %s %p to %s:%d\n------\n%s------", tmp.c_str(),msg,m_addr.host().c_str(),m_addr.port(),buf.c_str()); } m_sock->sendTo( msg->getBuffer().data(), msg->getBuffer().length(), m_addr ); } const char* YateUDPParty::getProtoName() const { return "UDP"; } bool YateUDPParty::setParty(const URI& uri) { if (m_partyPort && m_party && s_cfg.getBoolValue("general","ignorevia",true)) return true; if (uri.getHost().null()) return false; int port = uri.getPort(); if (port <= 0) port = 5060; if (!m_addr.host(uri.getHost())) { Debug(&plugin,DebugWarn,"Could not resolve UDP party name '%s' [%p]", uri.getHost().safe(),this); return false; } m_addr.port(port); m_party = uri.getHost(); m_partyPort = port; DDebug(&plugin,DebugInfo,"New UDP party is %s:%d (%s:%d) [%p]", m_party.c_str(),m_partyPort, m_addr.host().c_str(),m_addr.port(), this); return true; } YateSIPEngine::YateSIPEngine(YateSIPEndPoint* ep) : SIPEngine(s_cfg.getValue("general","useragent")), m_ep(ep), m_prack(false) { addAllowed("INVITE"); addAllowed("BYE"); addAllowed("CANCEL"); if (s_cfg.getBoolValue("general","registrar")) addAllowed("REGISTER"); m_prack = s_cfg.getBoolValue("general","prack"); if (m_prack) addAllowed("PRACK"); NamedList *l = s_cfg.getSection("methods"); if (l) { unsigned int len = l->length(); for (unsigned int i=0; igetParam(i); if (!n) continue; String meth(n->name()); meth.toUpper(); addAllowed(meth); } } } bool YateSIPEngine::buildParty(SIPMessage* message) { return m_ep->buildParty(message); } bool YateSIPEngine::checkUser(const String& username, const String& realm, const String& nonce, const String& method, const String& uri, const String& response, const SIPMessage* message) { Message m("user.auth"); m.addParam("username",username); m.addParam("realm",realm); m.addParam("nonce",nonce); m.addParam("method",method); m.addParam("uri",uri); m.addParam("response",response); if (message) { m.addParam("ip_host",message->getParty()->getPartyAddr()); m.addParam("ip_port",String(message->getParty()->getPartyPort())); } if (!Engine::dispatch(m)) return false; // FIXME: deal with empty passwords or just disallow them if (m.retValue().null()) return true; String res; buildAuth(username,realm,m.retValue(),nonce,method,uri,res); if (res == response) return true; // if the URI included some parameters retry after stripping them off int sc = uri.find(';'); if (sc < 0) return false; buildAuth(username,realm,m.retValue(),nonce,method,uri.substr(0,sc),res); return (res == response); } YateSIPEndPoint::YateSIPEndPoint() : Thread("YSIP EndPoint"), m_sock(0), m_engine(0) { Debug(&plugin,DebugAll,"YateSIPEndPoint::YateSIPEndPoint() [%p]",this); } YateSIPEndPoint::~YateSIPEndPoint() { Debug(&plugin,DebugAll,"YateSIPEndPoint::~YateSIPEndPoint() [%p]",this); plugin.channels().clear(); s_lines.clear(); if (m_engine) { // send any pending events while (m_engine->process()) ; delete m_engine; m_engine = 0; } if (m_sock) { delete m_sock; m_sock = 0; } } bool YateSIPEndPoint::buildParty(SIPMessage* message, const char* host, int port, const YateSIPLine* line) { if (message->isAnswer()) return false; DDebug(&plugin,DebugAll,"YateSIPEndPoint::buildParty(%p,'%s',%d,%p)", message,host,port,line); URI uri(message->uri); if (line) { if (!host) host = line->getPartyAddr(); if (port <= 0) port = line->getPartyPort(); line->setupAuth(message); } if (!host) { host = uri.getHost().safe(); if (port <= 0) port = uri.getPort(); } if (port <= 0) port = 5060; SocketAddr addr(AF_INET); if (!addr.host(host)) { Debug(&plugin,DebugWarn,"Error resolving name '%s'",host); return false; } addr.port(port); DDebug(&plugin,DebugAll,"built addr: %s:%d", addr.host().c_str(),addr.port()); // reuse the variables now we finished with them host = line ? line->getLocalAddr().c_str() : 0; port = line ? line->getLocalPort() : 0; if (port <= 0) port = m_port; YateUDPParty* party = new YateUDPParty(m_sock,addr,port,host); message->setParty(party); party->deref(); return true; } bool YateSIPEndPoint::Init() { if (m_sock) { Debug(&plugin,DebugInfo,"Already initialized."); return true; } m_sock = new Socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP); if (!m_sock->valid()) { Debug(DebugGoOn,"Unable to allocate UDP socket"); return false; } SocketAddr addr(AF_INET); addr.port(s_cfg.getIntValue("general","port",5060)); addr.host(s_cfg.getValue("general","addr","0.0.0.0")); if (!m_sock->bind(addr)) { Debug(DebugWarn,"Unable to bind to preferred port - using random one instead"); addr.port(0); if (!m_sock->bind(addr)) { Debug(DebugGoOn,"Unable to bind to any port"); return false; } } if (!m_sock->getSockName(addr)) { Debug(DebugGoOn,"Unable to figure out what I'm bound to"); return false; } if (!m_sock->setBlocking(false)) { Debug(DebugGoOn,"Unable to set non-blocking mode"); return false; } Debug(DebugInfo,"SIP Started on %s:%d", addr.host().safe(), addr.port()); m_port = addr.port(); m_engine = new YateSIPEngine(this); return true; } void YateSIPEndPoint::addMessage(const char* buf, int len, const SocketAddr& addr, int port) { SIPMessage* msg = SIPMessage::fromParsing(0,buf,len); if (!msg) return; if (!msg->isAnswer()) { URI uri(msg->uri); YateSIPLine* line = plugin.findLine(addr.host(),addr.port(),uri.getUser()); const char* host = 0; if (line && line->getLocalPort()) { host = line->getLocalAddr(); port = line->getLocalPort(); } YateUDPParty* party = new YateUDPParty(m_sock,addr,port,host); msg->setParty(party); party->deref(); } m_engine->addMessage(msg); msg->deref(); } void YateSIPEndPoint::run() { struct timeval tv; char buf[1500]; /* Watch stdin (fd 0) to see when it has input. */ for (;;) { /* Wait up to 5000 microseconds. */ tv.tv_sec = 0; tv.tv_usec = 5000; bool ok = false; m_sock->select(&ok,0,0,&tv); if (ok) { // we can read the data int res = m_sock->recvFrom(buf,sizeof(buf)-1,m_addr); if (res <= 0) { if (!m_sock->canRetry()) { Debug(DebugGoOn,"SIP error on read: %d", m_sock->error()); } } else if (res >= 72) { buf[res]=0; Debug(&plugin,DebugInfo,"Received %d bytes SIP message from %s:%d\n------\n%s------", res,m_addr.host().c_str(),m_addr.port(),buf); // we got already the buffer and here we start to do "good" stuff addMessage(buf,res,m_addr,m_port); //m_engine->addMessage(new YateUDPParty(m_sock,m_addr,m_port),buf,res); } #ifdef DEBUG else Debug(&plugin,DebugInfo,"Received short SIP message of %d bytes",res); #endif } else Thread::check(); SIPEvent* e = m_engine->getEvent(); // hack: use a loop so we can use break and continue for (; e; m_engine->processEvent(e),e = 0) { if (!e->getTransaction()) continue; plugin.lock(); GenObject* obj = static_cast(e->getTransaction()->getUserData()); YateSIPConnection* conn = YOBJECT(YateSIPConnection,obj); YateSIPLine* line = YOBJECT(YateSIPLine,obj); YateSIPGenerate* gen = YOBJECT(YateSIPGenerate,obj); if (conn && (conn->refcount() > 0)) conn->ref(); else conn = 0; plugin.unlock(); if (conn) { if (conn->process(e)) { delete e; conn->deref(); break; } else { conn->deref(); continue; } } if (line) { if (line->process(e)) { delete e; break; } else continue; } if (gen) { if (gen->process(e)) { delete e; break; } else continue; } if ((e->getState() == SIPTransaction::Trying) && !e->isOutgoing() && incoming(e,e->getTransaction())) { delete e; break; } } } } bool YateSIPEndPoint::incoming(SIPEvent* e, SIPTransaction* t) { if (t->isInvite()) invite(e,t); else if (t->getMethod() == "BYE") { YateSIPConnection* conn = plugin.findCall(t->getCallID()); if (conn) conn->doBye(t); else t->setResponse(481); } else if (t->getMethod() == "CANCEL") { YateSIPConnection* conn = plugin.findCall(t->getCallID()); if (conn) conn->doCancel(t); else t->setResponse(481); } else if (t->getMethod() == "REGISTER") regreq(e,t); else return generic(e,t); return true; } void YateSIPEndPoint::invite(SIPEvent* e, SIPTransaction* t) { if (!plugin.canAccept()) { Debug(DebugWarn,"Refusing new SIP call, full or exiting"); t->setResponse(480); return; } if (e->getMessage()->getParam("To","tag")) { SIPDialog dlg(*e->getMessage()); YateSIPConnection* conn = plugin.findDialog(dlg); if (conn) conn->reInvite(t); else { Debug(DebugWarn,"Got re-INVITE for missing dialog"); t->setResponse(481); } return; } YateSIPConnection* conn = new YateSIPConnection(e,t); conn->startRouter(); } void YateSIPEndPoint::regreq(SIPEvent* e, SIPTransaction* t) { if (Engine::exiting()) { Debug(&plugin,DebugWarn,"Dropping request, engine is exiting"); t->setResponse(500, "Server Shutting Down"); return; } const SIPHeaderLine* hl = e->getMessage()->getHeader("Contact"); if (!hl) { t->setResponse(400); return; } String user; int age = t->authUser(user); DDebug(&plugin,DebugAll,"User '%s' age %d",user.c_str(),age); if ((age < 0) || (age > 10)) { t->requestAuth(s_cfg.getValue("general","realm","Yate"),"",age >= 0); return; } URI addr(*hl); Message *m = new Message("user.register"); m->addParam("username",user); m->addParam("number",addr.getUser()); m->addParam("driver","sip"); String data("sip/" + addr); if (s_auto_nat && isPrivateAddr(addr.getHost()) && !isPrivateAddr(e->getMessage()->getParty()->getPartyAddr())) { Debug(DebugInfo,"Registration NAT detected: private '%s' public '%s'", addr.getHost().c_str(),e->getMessage()->getParty()->getPartyAddr().c_str()); m->addParam("reg_nat_addr",addr.getHost()); int pos = data.find(addr.getHost()); if (pos >= 0) data = data.substr(0,pos) + e->getMessage()->getParty()->getPartyAddr() + data.substr(pos + addr.getHost().length()); } m->addParam("data",data); bool dereg = false; hl = e->getMessage()->getHeader("Expires"); if (hl) { m->addParam("expires",*hl); if (*hl == "0") { *m = "user.unregister"; dereg = true; } } hl = e->getMessage()->getHeader("User-Agent"); if (hl) m->addParam("device",*hl); // Always OK deregistration attempts if (Engine::dispatch(m) || dereg) t->setResponse(200); else t->setResponse(404); m->destruct(); } bool YateSIPEndPoint::generic(SIPEvent* e, SIPTransaction* t) { String meth(t->getMethod()); meth.toLower(); String user; if (s_cfg.getBoolValue("methods",meth,true)) { int age = t->authUser(user); DDebug(&plugin,DebugAll,"User '%s' age %d",user.c_str(),age); if ((age < 0) || (age > 10)) { t->requestAuth("realm","",age >= 0); return true; } } Message m("sip." + meth); if (e->getMessage()->getParam("To","tag")) { SIPDialog dlg(*e->getMessage()); YateSIPConnection* conn = plugin.findDialog(dlg); if (conn) { m.userData(conn); conn->complete(m); } } if (user) m.addParam("username",user); m.addParam("ip_host",e->getMessage()->getParty()->getPartyAddr()); m.addParam("ip_port",String(e->getMessage()->getParty()->getPartyPort())); m.addParam("sip_uri",t->getURI()); m.addParam("sip_callid",t->getCallID()); // establish the dialog here so user code will have the dialog tag handy t->setDialogTag(); m.addParam("xsip_dlgtag",t->getDialogTag()); copySipHeaders(m,*e->getMessage()); if (Engine::dispatch(m)) { t->setResponse(m.getIntValue("code",200)); return true; } return false; } // Incoming call constructor - just before starting the routing thread YateSIPConnection::YateSIPConnection(SIPEvent* ev, SIPTransaction* tr) : Channel(plugin,0,false), m_tr(tr), m_hungup(false), m_byebye(true), m_state(Incoming), m_rtpForward(false), m_rtpMedia(0), m_sdpSession(0), m_sdpVersion(0), m_port(0), m_route(0), m_routes(0), m_authBye(true), m_mediaStatus(MediaMissing) { Debug(this,DebugAll,"YateSIPConnection::YateSIPConnection(%p,%p) [%p]",ev,tr,this); setReason(); m_tr->ref(); m_routes = m_tr->initialMessage()->getRoutes(); m_callid = m_tr->getCallID(); m_dialog = *m_tr->initialMessage(); m_host = m_tr->initialMessage()->getParty()->getPartyAddr(); m_port = m_tr->initialMessage()->getParty()->getPartyPort(); m_address << m_host << ":" << m_port; m_uri = m_tr->initialMessage()->getHeader("From"); m_uri.parse(); m_tr->setUserData(this); URI uri(m_tr->getURI()); YateSIPLine* line = plugin.findLine(m_host,m_port,m_uri.getUser()); Message *m = message("call.route"); if (line) { // call comes from line we have registered to - trust it... m_user = line->getUserName(); m_externalAddr = line->getLocalAddr(); m_line = *line; m->addParam("username",m_user); m->addParam("in_line",m_line); } else { String user; int age = tr->authUser(user); DDebug(this,DebugAll,"User '%s' age %d",user.c_str(),age); if (age >= 0) { if (age < 10) { m_user = user; m->addParam("username",m_user); } else m->addParam("expired_user",user); m->addParam("xsip_nonce_age",String(age)); } } if (s_privacy) copyPrivacy(*m,*ev->getMessage()); m->addParam("caller",m_uri.getUser()); m->addParam("called",uri.getUser()); String tmp(ev->getMessage()->getHeaderValue("Max-Forwards")); int maxf = tmp.toInteger(s_maxForwards); if (maxf > s_maxForwards) maxf = s_maxForwards; tmp = maxf-1; m->addParam("antiloop",tmp); m->addParam("ip_host",m_host); m->addParam("ip_port",String(m_port)); m->addParam("sip_uri",uri); m->addParam("sip_from",m_uri); m->addParam("sip_callid",m_callid); m->addParam("sip_contact",ev->getMessage()->getHeaderValue("Contact")); m->addParam("sip_user-agent",ev->getMessage()->getHeaderValue("User-Agent")); if (ev->getMessage()->body && ev->getMessage()->body->isSDP()) { setMedia(parseSDP(static_cast(ev->getMessage()->body),m_rtpAddr,m_rtpMedia)); if (m_rtpMedia) { m_rtpForward = true; // guess if the call comes from behind a NAT if (s_auto_nat && isPrivateAddr(m_rtpAddr) && !isPrivateAddr(m_host)) { Debug(this,DebugInfo,"RTP NAT detected: private '%s' public '%s'", m_rtpAddr.c_str(),m_host.c_str()); m->addParam("rtp_nat_addr",m_rtpAddr); m_rtpAddr = m_host; } m->addParam("rtp_forward","possible"); m->addParam("rtp_addr",m_rtpAddr); ObjList* l = m_rtpMedia->skipNull(); for (; l; l = l->skipNext()) { RtpMedia* r = static_cast(l->get()); m->addParam("media"+r->suffix(),"yes"); m->addParam("rtp_port"+r->suffix(),r->remotePort()); m->addParam("formats"+r->suffix(),r->formats()); } } } DDebug(this,DebugAll,"RTP addr '%s' [%p]",m_rtpAddr.c_str(),this); m_route = m; Engine::enqueue(message("chan.startup")); } // Outgoing call constructor - in call.execute handler YateSIPConnection::YateSIPConnection(Message& msg, const String& uri, const char* target) : Channel(plugin,0,true), m_tr(0), m_hungup(false), m_byebye(true), m_state(Outgoing), m_rtpForward(false), m_rtpMedia(0), m_sdpSession(0), m_sdpVersion(0), m_port(0), m_route(0), m_routes(0), m_authBye(false), m_mediaStatus(MediaMissing) { Debug(this,DebugAll,"YateSIPConnection::YateSIPConnection(%p,'%s') [%p]", &msg,uri.c_str(),this); m_targetid = target; setReason(); m_rtpForward = msg.getBoolValue("rtp_forward"); m_line = msg.getValue("line"); String tmp; YateSIPLine* line = 0; if (m_line) { line = plugin.findLine(m_line); if (line && (uri.find('@') < 0)) { if (!uri.startsWith("sip:")) tmp = "sip:"; tmp << uri << "@" << line->domain(); } if (line) m_externalAddr = line->getLocalAddr(); } if (tmp.null()) tmp = uri; m_uri = tmp; m_uri.parse(); SIPMessage* m = new SIPMessage("INVITE",m_uri); plugin.ep()->buildParty(m,msg.getValue("host"),msg.getIntValue("port"),line); if (!m->getParty()) { Debug(this,DebugWarn,"Could not create party for '%s' [%p]",m_uri.c_str(),this); m->destruct(); tmp = "Invalid address: "; tmp << m_uri; msg.setParam("reason",tmp); setReason(tmp); return; } int maxf = msg.getIntValue("antiloop",s_maxForwards); m->addHeader("Max-Forwards",String(maxf)); m->complete(plugin.ep()->engine(), msg.getValue("caller"), msg.getValue("domain",(line ? line->domain().c_str() : 0))); if (plugin.ep()->engine()->prack()) m->addHeader("Supported","100rel"); m_host = m->getParty()->getPartyAddr(); m_port = m->getParty()->getPartyPort(); m_address << m_host << ":" << m_port; m_dialog = *m; if (s_privacy) copyPrivacy(*m,msg); SDPBody* sdp = createPasstroughSDP(msg); if (!sdp) sdp = createRtpSDP(m_host,msg); m->setBody(sdp); m_tr = plugin.ep()->engine()->addMessage(m); if (m_tr) { m_tr->ref(); m_callid = m_tr->getCallID(); m_tr->setUserData(this); } m->deref(); Message* s = message("chan.startup"); s->setParam("caller",msg.getValue("caller")); s->setParam("called",msg.getValue("called")); s->setParam("billid",msg.getValue("billid")); Engine::enqueue(s); } YateSIPConnection::~YateSIPConnection() { Debug(this,DebugAll,"YateSIPConnection::~YateSIPConnection() [%p]",this); hangup(); clearTransaction(); setMedia(0); if (m_route) { delete m_route; m_route = 0; } if (m_routes) { delete m_routes; m_routes = 0; } } void YateSIPConnection::setMedia(ObjList* media) { if (media == m_rtpMedia) return; ObjList* tmp = m_rtpMedia; m_rtpMedia = media; if (tmp) { ObjList* l = tmp->skipNull(); for (; l; l = l->skipNext()) { RtpMedia* m = static_cast(l->get()); clearEndpoint(*m); } tmp->destruct(); } } void YateSIPConnection::startRouter() { Message* m = m_route; m_route = 0; Channel::startRouter(m); } void YateSIPConnection::clearTransaction() { if (m_tr) { if (driver()) driver()->lock(); m_tr->setUserData(0); if (m_tr->isIncoming()) { if (m_tr->setResponse(m_reasonCode,m_reason.null() ? "Request Terminated" : m_reason.c_str())) m_byebye = false; } m_tr->deref(); m_tr = 0; if (driver()) driver()->unlock(); } } void YateSIPConnection::hangup() { if (m_hungup) return; m_hungup = true; const char* error = lookup(m_reasonCode,dict_errors); Debug(this,DebugAll,"YateSIPConnection::hangup() state=%d trans=%p error='%s' code=%d reason='%s' [%p]", m_state,m_tr,error,m_reasonCode,m_reason.c_str(),this); Message* m = message("chan.hangup"); if (m_reason) m->addParam("reason",m_reason); Engine::enqueue(m); switch (m_state) { case Cleared: clearTransaction(); return; case Incoming: if (m_tr) { clearTransaction(); return; } break; case Outgoing: case Ringing: if (m_tr) { SIPMessage* m = new SIPMessage("CANCEL",m_uri); plugin.ep()->buildParty(m,m_host,m_port,plugin.findLine(m_line)); if (!m->getParty()) Debug(this,DebugWarn,"Could not create party for '%s:%d' [%p]", m_host.c_str(),m_port,this); else { const SIPMessage* i = m_tr->initialMessage(); m->copyHeader(i,"Via"); m->copyHeader(i,"From"); m->copyHeader(i,"To"); m->copyHeader(i,"Call-ID"); String tmp; tmp << i->getCSeq() << " CANCEL"; m->addHeader("CSeq",tmp); plugin.ep()->engine()->addMessage(m); } m->deref(); } break; } clearTransaction(); m_state = Cleared; if (m_byebye) { m_byebye = false; SIPMessage* m = createDlgMsg("BYE"); if (m) { if (m_reason) { // FIXME: add SIP and Q.850 cause codes, set the proper reason SIPHeaderLine* hl = new SIPHeaderLine("Reason","SIP"); hl->setParam("text","\"" + m_reason + "\""); m->addHeader(hl); } plugin.ep()->engine()->addMessage(m); m->deref(); } } if (!error) error = m_reason.c_str(); disconnect(error); } // Creates a new message in an existing dialog SIPMessage* YateSIPConnection::createDlgMsg(const char* method, const char* uri) { if (!uri) uri = m_uri; SIPMessage* m = new SIPMessage(method,uri); m->addRoutes(m_routes); plugin.ep()->buildParty(m,m_host,m_port,plugin.findLine(m_line)); if (!m->getParty()) { Debug(this,DebugWarn,"Could not create party for '%s:%d' [%p]", m_host.c_str(),m_port,this); m->destruct(); return 0; } m->addHeader("Call-ID",m_callid); String tmp; tmp << "<" << m_dialog.localURI << ">"; SIPHeaderLine* hl = new SIPHeaderLine("From",tmp); tmp = m_dialog.localTag; if (tmp.null() && m_tr) tmp = m_tr->getDialogTag(); if (tmp) hl->setParam("tag",tmp); m->addHeader(hl); tmp.clear(); tmp << "<" << m_dialog.remoteURI << ">"; hl = new SIPHeaderLine("To",tmp); tmp = m_dialog.remoteTag; if (tmp.null() && m_tr) tmp = m_tr->getDialogTag(); if (tmp) hl->setParam("tag",tmp); m->addHeader(hl); if (m_tr && m_tr->initialMessage()) m->copyHeader(m_tr->initialMessage(),"Contact"); return m; } // Emit a PRovisional ACK if enabled in the engine bool YateSIPConnection::emitPRACK(const SIPMessage* msg) { if (!plugin.ep()->engine()->prack()) return false; if (!(msg && msg->isAnswer() && (msg->code > 100) && (msg->code < 200))) return false; const SIPHeaderLine* rs = msg->getHeader("RSeq"); const SIPHeaderLine* cs = msg->getHeader("CSeq"); if (!(rs && cs)) return false; String tmp; const SIPHeaderLine* co = msg->getHeader("Contact"); if (co) { tmp = *co; Regexp r("^[^<]*<\\([^>]*\\)>.*$"); if (tmp.matches(r)) tmp = tmp.matchString(1); } SIPMessage* m = createDlgMsg("PRACK",tmp); if (!m) return false; tmp = *rs; tmp << " " << *cs; m->addHeader("RAck",tmp); plugin.ep()->engine()->addMessage(m); m->deref(); return true; } // Creates a SDP for provisional (1xx) messages SDPBody* YateSIPConnection::createProvisionalSDP(Message& msg) { if (m_rtpForward) return createPasstroughSDP(msg); // check if our peer can source at least audio data if (!(getPeer() && getPeer()->getSource())) return 0; if (m_rtpAddr.null()) return 0; return createRtpSDP(true); } // Creates a SDP from RTP address data present in message SDPBody* YateSIPConnection::createPasstroughSDP(Message& msg) { String tmp = msg.getValue("rtp_forward"); msg.clearParam("rtp_forward"); if (!(m_rtpForward && tmp.toBoolean())) return 0; String addr(msg.getValue("rtp_addr")); if (addr.null()) return 0; ObjList* lst = 0; unsigned int n = msg.length(); for (unsigned int i = 0; i < n; i++) { const NamedString* p = msg.getParam(i); if (!p) continue; // search for rtp_port or rtp_port_MEDIANAME parameters tmp = p->name(); if (!tmp.startSkip("rtp_port",false)) continue; if (tmp && (tmp[0] != '_')) continue; // now tmp holds the suffix for the media, null for audio bool audio = tmp.null(); // check if media is supported, default only for audio if (!msg.getBoolValue("media"+tmp,audio)) continue; int port = p->toInteger(); if (!port) continue; const char* fmts = msg.getValue("formats"+tmp); if (!fmts) continue; if (audio) tmp = "audio"; else tmp >> "_"; RtpMedia* rtp = 0; // try to take the media descriptor from the old list if (m_rtpMedia) { ObjList* om = m_rtpMedia->find(tmp); if (om) rtp = static_cast(om->remove(false)); } if (rtp) rtp->update(fmts,-1,port); else rtp = new RtpMedia(tmp,fmts,-1,port); if (!lst) lst = new ObjList; lst->append(rtp); } if (!lst) return 0; m_rtpLocalAddr = addr; setMedia(lst); SDPBody* sdp = createSDP(m_rtpLocalAddr); if (sdp) msg.setParam("rtp_forward","accepted"); return sdp; } // Dispatches a RTP message for a media, optionally start RTP and pick parameters bool YateSIPConnection::dispatchRtp(RtpMedia* media, const char* addr, bool start, bool pick) { if (!(media && addr)) return false; Message m("chan.rtp"); complete(m,true); m.userData(static_cast(this)); m.addParam("media",*media); m.addParam("direction","bidir"); if (m_rtpLocalAddr) m.addParam("localip",m_rtpLocalAddr); m.addParam("remoteip",addr); if (start) { m.addParam("remoteport",media->remotePort()); m.addParam("format",media->format()); } if (!Engine::dispatch(m)) return false; if (!pick) return true; m_rtpForward = false; m_rtpLocalAddr = m.getValue("localip",m_rtpLocalAddr); m_mediaStatus = MediaStarted; media->update(m,start); return true; } // Creates a set of unstarted external RTP channels from remote addr and builds SDP from them SDPBody* YateSIPConnection::createRtpSDP(const char* addr, const Message& msg) { bool defaults = true; ObjList* lst = 0; unsigned int n = msg.length(); for (unsigned int i = 0; i < n; i++) { const NamedString* p = msg.getParam(i); if (!p) continue; // search for rtp_port or rtp_port_MEDIANAME parameters String tmp(p->name()); if (!tmp.startSkip("media",false)) continue; if (tmp && (tmp[0] != '_')) continue; // since we found at least one media declaration disable defaults defaults = false; // now tmp holds the suffix for the media, null for audio bool audio = tmp.null(); // check if media is supported, default only for audio if (!p->toBoolean(audio)) continue; const char* fmts = msg.getValue("formats"+tmp); if (audio && !fmts) fmts = "alaw,mulaw"; if (!fmts) continue; if (audio) tmp = "audio"; else tmp >> "_"; RtpMedia* rtp = 0; // try to take the media descriptor from the old list if (m_rtpMedia) { ObjList* om = m_rtpMedia->find(tmp); if (om) rtp = static_cast(om->remove(false)); } if (rtp) rtp->update(fmts); else rtp = new RtpMedia(tmp,fmts); if (!lst) lst = new ObjList; lst->append(rtp); } if (defaults && !lst) { lst = new ObjList; lst->append(new RtpMedia("audio",msg.getValue("formats","alaw,mulaw"))); } setMedia(lst); ObjList* l = m_rtpMedia->skipNull(); for (; l; l = l->skipNext()) { RtpMedia* m = static_cast(l->get()); if (!dispatchRtp(m,addr,false,true)) return 0; } return createSDP(getRtpAddr()); } // Creates a set of started external RTP channels from remote addr and builds SDP from them SDPBody* YateSIPConnection::createRtpSDP(bool start) { if (m_rtpAddr.null()) { m_mediaStatus = MediaMuted; return createSDP(0); } ObjList* l = m_rtpMedia->skipNull(); for (; l; l = l->skipNext()) { RtpMedia* m = static_cast(l->get()); if (!dispatchRtp(m,m_rtpAddr,start,true)) return 0; } return createSDP(getRtpAddr()); } // Starts an already created set of external RTP channels bool YateSIPConnection::startRtp() { if (m_mediaStatus != MediaStarted) return false; DDebug(this,DebugAll,"YateSIPConnection::startRtp() [%p]",this); bool ok = true; ObjList* l = m_rtpMedia->skipNull(); for (; l; l = l->skipNext()) { RtpMedia* m = static_cast(l->get()); ok = dispatchRtp(m,m_rtpAddr,true,false) && ok; } return ok; } // Creates a SDP body from transport address and list of media descriptors SDPBody* YateSIPConnection::createSDP(const char* addr, ObjList* mediaList) { DDebug(this,DebugAll,"YateSIPConnection::createSDP('%s',%p) [%p]", addr,mediaList,this); if (!mediaList) mediaList = m_rtpMedia; // if we got no media descriptors we simply create no SDP if (!mediaList) return 0; if (m_sdpSession) ++m_sdpVersion; else m_sdpVersion = m_sdpSession = Time::secNow(); // no address means on hold or muted String origin; origin << "yate " << m_sdpSession << " " << m_sdpVersion << " IN IP4 " << (addr ? addr : m_host.safe()); String conn; conn << "IN IP4 " << (addr ? addr : "0.0.0.0"); SDPBody* sdp = new SDPBody; sdp->addLine("v","0"); sdp->addLine("o",origin); sdp->addLine("s","SIP Call"); sdp->addLine("c",conn); sdp->addLine("t","0 0"); bool defcodecs = s_cfg.getBoolValue("codecs","default",true); for (ObjList* ml = mediaList->skipNull(); ml; ml = ml->skipNext()) { RtpMedia* m = static_cast(ml->get()); String frm(m->fmtList()); ObjList* l = frm.split(',',false); frm = *m; frm << " " << m->localPort() << " RTP/AVP"; ObjList rtpmap; int ptime = 0; ObjList* f = l; for (; f; f = f->next()) { String* s = static_cast(f->get()); if (s) { int mode = 0; if (*s == "ilbc20") ptime = mode = 20; else if (*s == "ilbc30") ptime = mode = 30; int payload = s->toInteger(dict_payloads,-1); if (payload >= 0) { const char* map = lookup(payload,dict_rtpmap); if (map && s_cfg.getBoolValue("codecs",*s,defcodecs && DataTranslator::canConvert(*s))) { frm << " " << payload; String* temp = new String("rtpmap:"); *temp << payload << " " << map; rtpmap.append(temp); if (mode) { temp = new String("fmtp:"); *temp << payload << " mode=" << mode; rtpmap.append(temp); } } } } } delete l; if (*m == "audio") { // always claim to support telephone events frm << " 101"; rtpmap.append(new String("rtpmap:101 telephone-event/8000")); } if (ptime) { String* temp = new String("ptime:"); *temp << ptime; rtpmap.append(temp); } sdp->addLine("m",frm); for (f = rtpmap.skipNull(); f; f = f->skipNext()) { String* s = static_cast(f->get()); if (s) sdp->addLine("a",*s); } } return sdp; } // Add RTP forwarding parameters to a message bool YateSIPConnection::addRtpParams(Message& msg, const String& natAddr) { if (!(m_rtpMedia && m_rtpAddr)) return false; ObjList* l = m_rtpMedia->skipNull(); for (; l; l = l->skipNext()) { RtpMedia* m = static_cast(l->get()); msg.addParam("formats"+m->suffix(),m->formats()); msg.addParam("media"+m->suffix(),"yes"); } if (!startRtp() && m_rtpForward) { if (natAddr) msg.addParam("rtp_nat_addr",natAddr); msg.addParam("rtp_forward","yes"); msg.addParam("rtp_addr",m_rtpAddr); l = m_rtpMedia->skipNull(); for (; l; l = l->skipNext()) { RtpMedia* m = static_cast(l->get()); msg.addParam("rtp_port"+m->suffix(),m->remotePort()); } return true; } return false; } // Process SIP events belonging to this connection bool YateSIPConnection::process(SIPEvent* ev) { DDebug(this,DebugInfo,"YateSIPConnection::process(%p) %s [%p]", ev,SIPTransaction::stateName(ev->getState()),this); m_dialog = *ev->getTransaction()->recentMessage(); const SIPMessage* msg = ev->getMessage(); int code = ev->getTransaction()->getResponseCode(); if (msg && !msg->isOutgoing() && msg->isAnswer() && (code >= 300)) { setReason(msg->reason,code); hangup(); } if (ev->getState() == SIPTransaction::Cleared) { if (m_tr) { DDebug(this,DebugInfo,"YateSIPConnection clearing transaction %p [%p]", m_tr,this); m_tr->setUserData(0); m_tr->deref(); m_tr = 0; } if (m_state != Established) hangup(); return false; } if (!msg || msg->isOutgoing()) return false; String natAddr; if (msg->body && msg->body->isSDP()) { DDebug(this,DebugInfo,"YateSIPConnection got SDP [%p]",this); setMedia(parseSDP(static_cast(msg->body),m_rtpAddr,m_rtpMedia)); // guess if the call comes from behind a NAT if (s_auto_nat && isPrivateAddr(m_rtpAddr) && !isPrivateAddr(m_host)) { Debug(this,DebugInfo,"RTP NAT detected: private '%s' public '%s'", m_rtpAddr.c_str(),m_host.c_str()); natAddr = m_rtpAddr; m_rtpAddr = m_host; } DDebug(this,DebugAll,"RTP addr '%s' [%p]",m_rtpAddr.c_str(),this); } if ((!m_routes) && msg->isAnswer() && (msg->code > 100) && (msg->code < 300)) m_routes = msg->getRoutes(); if (msg->isAnswer() && ((msg->code / 100) == 2)) { const SIPMessage* ack = m_tr->latestMessage(); if (ack && ack->isACK()) { m_uri = ack->uri; m_uri.parse(); } setReason("",0); setStatus("answered",Established); Message *m = message("call.answered"); addRtpParams(*m,natAddr); Engine::enqueue(m); } if ((m_state < Ringing) && msg->isAnswer()) { if (msg->code == 180) { setStatus("ringing",Ringing); Message *m = message("call.ringing"); addRtpParams(*m,natAddr); Engine::enqueue(m); } if (msg->code == 183) { setStatus("progressing"); Message *m = message("call.progress"); addRtpParams(*m,natAddr); Engine::enqueue(m); } if ((msg->code > 100) && (msg->code < 200)) emitPRACK(msg); } if (msg->isACK()) { DDebug(this,DebugInfo,"YateSIPConnection got ACK [%p]",this); startRtp(); } return false; } void YateSIPConnection::reInvite(SIPTransaction* t) { if (!checkUser(t)) return; DDebug(this,DebugAll,"YateSIPConnection::reInvite(%p) [%p]",t,this); // hack: use a while instead of if so we can return or break out of it while (t->initialMessage()->body && t->initialMessage()->body->isSDP()) { // accept re-INVITE only for local RTP, not for pass-trough if (m_rtpForward || (m_mediaStatus == MediaMissing)) break; String addr; ObjList* lst = parseSDP(static_cast(t->initialMessage()->body),addr); if (!lst) break; // guess if the call comes from behind a NAT if (s_auto_nat && isPrivateAddr(addr) && !isPrivateAddr(m_host)) { Debug(this,DebugInfo,"RTP NAT detected: private '%s' public '%s'", addr.c_str(),m_host.c_str()); addr = m_host; } m_rtpAddr = addr; setMedia(lst); Debug(this,DebugAll,"New RTP addr '%s'",m_rtpAddr.c_str()); m_mediaStatus = MediaMissing; // let RTP guess again the local interface m_rtpLocalAddr.clear(); // clear all data endpoints - createRtpSDP will build new ones clearEndpoint(); SIPMessage* m = new SIPMessage(t->initialMessage(), 200); SDPBody* sdp = createRtpSDP(true); m->setBody(sdp); t->setResponse(m); m->deref(); return; } t->setResponse(488); } bool YateSIPConnection::checkUser(SIPTransaction* t, bool refuse) { if (m_user.null()) return true; int age = t->authUser(m_user); if ((age >= 0) && (age <= 10)) return true; DDebug(this,DebugAll,"YateSIPConnection::checkUser(%p) failed, age %d [%p]",t,age,this); if (refuse) t->requestAuth("realm","",false); return false; } void YateSIPConnection::doBye(SIPTransaction* t) { if (m_authBye && !checkUser(t)) return; DDebug(this,DebugAll,"YateSIPConnection::doBye(%p) [%p]",t,this); const SIPHeaderLine* hl = t->initialMessage()->getHeader("Reason"); if (hl) { const NamedString* text = hl->getParam("text"); if (text) m_reason = *text; // FIXME: add SIP and Q.850 cause codes } t->setResponse(200); m_byebye = false; hangup(); } void YateSIPConnection::doCancel(SIPTransaction* t) { #ifdef DEBUG if (!checkUser(t,false)) Debug(DebugMild,"User authentication failed for user '%s' but CANCELing anyway [%p]", m_user.c_str(),this); #endif DDebug(this,DebugAll,"YateSIPConnection::doCancel(%p) [%p]",t,this); if (m_tr) { t->setResponse(200); m_byebye = false; clearTransaction(); disconnect("Cancelled"); } else t->setResponse(481); } void YateSIPConnection::disconnected(bool final, const char *reason) { Debug(this,DebugAll,"YateSIPConnection::disconnected() '%s' [%p]",reason,this); if (reason) { int code = lookup(reason,dict_errors); if (code) setReason(lookup(code,SIPResponses,reason),code); else setReason(reason); } Channel::disconnected(final,reason); } bool YateSIPConnection::msgProgress(Message& msg) { Channel::msgProgress(msg); if (m_tr && (m_tr->getState() == SIPTransaction::Process)) { SIPMessage* m = new SIPMessage(m_tr->initialMessage(), 183); m->setBody(createProvisionalSDP(msg)); m_tr->setResponse(m); m->deref(); } setStatus("progressing"); return true; } bool YateSIPConnection::msgRinging(Message& msg) { Channel::msgRinging(msg); if (m_tr && (m_tr->getState() == SIPTransaction::Process)) { SIPMessage* m = new SIPMessage(m_tr->initialMessage(), 180); m->setBody(createProvisionalSDP(msg)); m_tr->setResponse(m); m->deref(); } setStatus("ringing"); return true; } bool YateSIPConnection::msgAnswered(Message& msg) { if (m_tr && (m_tr->getState() == SIPTransaction::Process)) { SIPMessage* m = new SIPMessage(m_tr->initialMessage(), 200); SDPBody* sdp = createPasstroughSDP(msg); if (!sdp) { m_rtpForward = false; // don't start RTP yet, only when we get the ACK sdp = createRtpSDP(false); } m->setBody(sdp); m_tr->setResponse(m); m->deref(); } setReason("",0); setStatus("answered",Established); return true; } bool YateSIPConnection::msgTone(Message& msg, const char* tone) { if (m_rtpMedia && (m_mediaStatus == MediaStarted)) { ObjList* l = m_rtpMedia->find("audio"); const RtpMedia* m = static_cast(l ? l->get() : 0); if (m) { msg.setParam("targetid",m->id()); return false; } } // FIXME: when muted or doing RTP forwarding we should use INFO messages return false; } bool YateSIPConnection::msgText(Message& msg, const char* text) { if (null(text)) return false; SIPMessage* m = createDlgMsg("MESSAGE"); if (m) { m->setBody(new SIPStringBody("text/plain",text)); plugin.ep()->engine()->addMessage(m); m->deref(); return true; } return false; } bool YateSIPConnection::callRouted(Message& msg) { Channel::callRouted(msg); if (m_tr && (m_tr->getState() == SIPTransaction::Process)) { String s(msg.retValue()); if (s.startSkip("sip/",false) && s && msg.getBoolValue("redirect")) { Debug(this,DebugAll,"YateSIPConnection redirecting to '%s' [%p]",s.c_str(),this); SIPMessage* m = new SIPMessage(m_tr->initialMessage(),302); s = "<" + s + ">"; m->addHeader("Contact",s); m_tr->setResponse(m); m->deref(); m_byebye = false; setReason("Redirected",302); setStatus("redirected"); return false; } if (msg.getBoolValue("progress",s_cfg.getBoolValue("general","progress",true))) m_tr->setResponse(183); } return true; } void YateSIPConnection::callAccept(Message& msg) { m_user = msg.getValue("username"); if (m_authBye) m_authBye = msg.getBoolValue("xsip_auth_bye",true); if (m_rtpForward) { String tmp(msg.getValue("rtp_forward")); if (tmp != "accepted") m_rtpForward = false; } Channel::callAccept(msg); } void YateSIPConnection::callRejected(const char* error, const char* reason, const Message* msg) { Channel::callRejected(error,reason,msg); int code = lookup(error,dict_errors,500); if (code == 401) m_tr->requestAuth("realm","",false); else m_tr->setResponse(code,reason); setReason(reason,code); } YateSIPLine::YateSIPLine(const String& name) : String(name), m_resend((u_int64_t)0), m_interval(0), m_tr(0), m_marked(false), m_valid(false), m_localPort(0), m_partyPort(0), m_localDetect(false) { DDebug(&plugin,DebugInfo,"YateSIPLine::YateSIPLine('%s') [%p]",c_str(),this); s_lines.append(this); } YateSIPLine::~YateSIPLine() { DDebug(&plugin,DebugInfo,"YateSIPLine::~YateSIPLine() '%s' [%p]",c_str(),this); s_lines.remove(this,false); logout(); } void YateSIPLine::setupAuth(SIPMessage* msg) const { if (msg) msg->setAutoAuth(getAuthName(),m_password); } SIPMessage* YateSIPLine::buildRegister(int expires) const { String exp(expires); String tmp; tmp << "sip:" << m_registrar; SIPMessage* m = new SIPMessage("REGISTER",tmp); plugin.ep()->buildParty(m,0,0,this); if (!m->getParty()) { Debug(&plugin,DebugWarn,"Could not create party for '%s' [%p]", m_registrar.c_str(),this); m->destruct(); return 0; } tmp = "\""; tmp << (m_display.null() ? m_username : m_display); tmp << "\" getParty()->getLocalAddr() << ":"; tmp << m->getParty()->getLocalPort() << ">"; m->addHeader("Contact",tmp); m->addHeader("Expires",exp); tmp = ""; m->addHeader("To",tmp); m->complete(plugin.ep()->engine(),m_username,domain()); return m; } void YateSIPLine::login() { if (m_registrar.null() || m_username.null()) { logout(); m_valid = true; return; } DDebug(&plugin,DebugInfo,"YateSIPLine '%s' logging in [%p]",c_str(),this); clearTransaction(); SIPMessage* m = buildRegister(m_interval); if (!m) { m_valid = false; return; } DDebug(&plugin,DebugInfo,"YateSIPLine '%s' emiting %p [%p]", c_str(),m,this); m_tr = plugin.ep()->engine()->addMessage(m); if (m_tr) { m_tr->ref(); m_tr->setUserData(this); } m->deref(); } void YateSIPLine::logout() { m_resend = 0; bool sendLogout = m_valid && m_registrar && m_username; clearTransaction(); m_valid = false; if (sendLogout) { DDebug(&plugin,DebugInfo,"YateSIPLine '%s' logging out [%p]",c_str(),this); SIPMessage* m = buildRegister(0); m_partyAddr.clear(); m_partyPort = 0; if (!m) return; plugin.ep()->engine()->addMessage(m); m->deref(); } } bool YateSIPLine::process(SIPEvent* ev) { DDebug(&plugin,DebugInfo,"YateSIPLine::process(%p) %s [%p]", ev,SIPTransaction::stateName(ev->getState()),this); if (ev->getTransaction() != m_tr) return false; if (ev->getState() == SIPTransaction::Cleared) { clearTransaction(); m_valid = false; m_resend = m_interval*(int64_t)1000000 + Time::now(); return false; } const SIPMessage* msg = ev->getMessage(); if (!(msg && msg->isAnswer())) return false; if (ev->getState() != SIPTransaction::Process) return false; clearTransaction(); DDebug(&plugin,DebugAll,"YateSIPLine '%s' got answer %d [%p]", c_str(),msg->code,this); switch (msg->code) { case 200: if (msg->getParty()) { if (m_localDetect) { SIPHeaderLine* hl = const_cast(msg->getHeader("Via")); if (hl) { const NamedString* par = hl->getParam("received"); if (par && *par) m_localAddr = *par; par = hl->getParam("rport"); if (par) { int port = par->toInteger(0,10); if (port > 0) m_localPort = port; } } if (m_localAddr.null()) m_localAddr = msg->getParty()->getLocalAddr(); if (!m_localPort) m_localPort = msg->getParty()->getLocalPort(); DDebug(&plugin,DebugInfo,"SIP line '%s' on local address %s:%d", c_str(),m_localAddr.c_str(),m_localPort); } m_partyAddr = msg->getParty()->getPartyAddr(); m_partyPort = msg->getParty()->getPartyPort(); } m_valid = true; // re-register at 3/4 of the expire interval m_resend = m_interval*(int64_t)750000 + Time::now(); Debug(&plugin,DebugInfo,"SIP line '%s' logon success to %s:%d", c_str(),m_partyAddr.c_str(),m_partyPort); break; default: m_valid = false; Debug(&plugin,DebugWarn,"SIP line '%s' logon failure %d",c_str(),msg->code); } return false; } void YateSIPLine::timer(const Time& when) { if (!m_resend || (m_resend > when)) return; m_resend = m_interval*(int64_t)1000000 + when; login(); } void YateSIPLine::clearTransaction() { if (m_tr) { DDebug(&plugin,DebugInfo,"YateSIPLine clearing transaction %p [%p]", m_tr,this); m_tr->setUserData(0); m_tr->deref(); m_tr = 0; } } bool YateSIPLine::change(String& dest, const String& src) { if (dest == src) return false; // we need to log out before any parameter changes logout(); dest = src; return true; } bool YateSIPLine::change(int& dest, int src) { if (dest == src) return false; // we need to log out before any parameter changes logout(); dest = src; return true; } bool YateSIPLine::update(const Message& msg) { DDebug(&plugin,DebugInfo,"YateSIPLine::update() '%s' [%p]",c_str(),this); String oper(msg.getValue("operation")); if (oper == "logout") { logout(); return true; } bool chg = false; chg = change(m_registrar,msg.getValue("registrar")) || chg; chg = change(m_outbound,msg.getValue("outbound")) || chg; chg = change(m_username,msg.getValue("username")) || chg; chg = change(m_authname,msg.getValue("authname")) || chg; chg = change(m_password,msg.getValue("password")) || chg; chg = change(m_domain,msg.getValue("domain")) || chg; m_display = msg.getValue("description"); m_interval = msg.getIntValue("interval",600); String tmp(msg.getValue("localaddress")); m_localDetect = (tmp == "auto"); if (!m_localDetect) { int port = 0; if (tmp) { int sep = tmp.find(':'); if (sep > 0) { port = tmp.substr(sep+1).toInteger(5060); tmp = tmp.substr(0,sep); } else if (sep < 0) port = 5060; } chg = change(m_localAddr,tmp) || chg; chg = change(m_localPort,port) || chg; } tmp = msg.getValue("operation"); // if something changed we logged out so try to climb back if (chg) login(); return chg; } YateSIPGenerate::YateSIPGenerate(SIPMessage* m) : m_tr(0), m_code(0) { m_tr = plugin.ep()->engine()->addMessage(m); if (m_tr) { m_tr->ref(); m_tr->setUserData(this); } m->deref(); } YateSIPGenerate::~YateSIPGenerate() { clearTransaction(); } bool YateSIPGenerate::process(SIPEvent* ev) { DDebug(&plugin,DebugInfo,"YateSIPGenerate::process(%p) %s [%p]", ev,SIPTransaction::stateName(ev->getState()),this); if (ev->getTransaction() != m_tr) return false; if (ev->getState() == SIPTransaction::Cleared) { clearTransaction(); return false; } const SIPMessage* msg = ev->getMessage(); if (!(msg && msg->isAnswer())) return false; if (ev->getState() != SIPTransaction::Process) return false; clearTransaction(); Debug(&plugin,DebugAll,"YateSIPGenerate got answer %d [%p]", msg->code,this); m_code = msg->code; return false; } void YateSIPGenerate::clearTransaction() { if (m_tr) { DDebug(&plugin,DebugInfo,"YateSIPGenerate clearing transaction %p [%p]", m_tr,this); m_tr->setUserData(0); m_tr->deref(); m_tr = 0; } } bool UserHandler::received(Message &msg) { String tmp(msg.getValue("protocol")); if (tmp != "sip") return false; tmp = msg.getValue("account"); if (tmp.null()) return false; YateSIPLine* line = plugin.findLine(tmp); if (!line) line = new YateSIPLine(tmp); line->update(msg); return true; } bool SipHandler::received(Message &msg) { Debug(&plugin,DebugInfo,"SipHandler::received() [%p]",this); const char* method = msg.getValue("method"); String uri(msg.getValue("uri")); Regexp r("<\\([^>]\\+\\)>"); if (uri.matches(r)) uri = uri.matchString(1); if (!(method && uri)) return false; YateSIPLine* line = plugin.findLine(msg.getValue("line")); if (line && !line->valid()) { msg.setParam("error","offline"); return false; } SIPMessage* sip = new SIPMessage(method,uri); plugin.ep()->buildParty(sip,msg.getValue("host"),msg.getIntValue("port"),line); copySipHeaders(*sip,msg); const char* type = msg.getValue("xsip_type"); const char* body = msg.getValue("xsip_body"); if (type && body) sip->setBody(new SIPStringBody(type,body,-1)); sip->complete(plugin.ep()->engine(),msg.getValue("user"),msg.getValue("domain")); if (!msg.getBoolValue("wait")) { // no answer requested - start transaction and forget plugin.ep()->engine()->addMessage(sip); return true; } YateSIPGenerate gen(sip); while (gen.busy()) Thread::yield(); if (gen.code()) msg.setParam("code",String(gen.code())); else msg.clearParam("code"); return true; } YateSIPConnection* SIPDriver::findCall(const String& callid) { XDebug(this,DebugAll,"SIPDriver finding call '%s'",callid.c_str()); Lock mylock(this); ObjList* l = channels().skipNull(); for (; l; l = l->skipNext()) { YateSIPConnection* c = static_cast(l->get()); if (c->callid() == callid) return c; } return 0; } YateSIPConnection* SIPDriver::findDialog(const SIPDialog& dialog) { XDebug(this,DebugAll,"SIPDriver finding dialog '%s'",dialog.c_str()); Lock mylock(this); ObjList* l = channels().skipNull(); for (; l; l = l->skipNext()) { YateSIPConnection* c = static_cast(l->get()); if (c->dialog() == dialog) return c; } return 0; } // find line by name YateSIPLine* SIPDriver::findLine(const String& line) { if (line.null()) return 0; ObjList* l = s_lines.find(line); return l ? static_cast(l->get()) : 0; } // find line by party address and port YateSIPLine* SIPDriver::findLine(const String& addr, int port, const String& user) { if (!(port && addr)) return 0; Lock mylock(this); ObjList* l = s_lines.skipNull(); for (; l; l = l->skipNext()) { YateSIPLine* sl = static_cast(l->get()); if (sl->getPartyPort() && (sl->getPartyPort() == port) && (sl->getPartyAddr() == addr)) { if (user && (sl->getUserName() != user)) continue; return sl; } } return 0; } // check if a line is either empty or valid (logged in or no registrar) bool SIPDriver::validLine(const String& line) { if (line.null()) return true; YateSIPLine* l = findLine(line); return l && l->valid(); } bool SIPDriver::received(Message &msg, int id) { if (id == Timer) { ObjList* l = s_lines.skipNull(); for (; l; l = l->skipNext()) static_cast(l->get())->timer(msg.msgTime()); } else if (id == Halt) { channels().clear(); s_lines.clear(); } return Driver::received(msg,id); } bool SIPDriver::msgExecute(Message& msg, String& dest) { if (!msg.userData()) { Debug(this,DebugWarn,"SIP call found but no data channel!"); return false; } if (!validLine(msg.getValue("line"))) { // asked to use a line but it's not registered msg.setParam("error","offline"); return false; } YateSIPConnection* conn = new YateSIPConnection(msg,dest,msg.getValue("id")); if (conn->getTransaction()) { CallEndpoint* ch = static_cast(msg.userData()); if (ch && conn->connect(ch,msg.getValue("reason"))) { msg.setParam("peerid",conn->id()); msg.setParam("targetid",conn->id()); conn->deref(); return true; } } conn->destruct(); return false; } SIPDriver::SIPDriver() : Driver("sip","varchans"), m_endpoint(0) { Output("Loaded module SIP Channel"); } SIPDriver::~SIPDriver() { Output("Unloading module SIP Channel"); } void SIPDriver::initialize() { Output("Initializing module SIP Channel"); s_cfg = Engine::configFile("ysipchan"); s_cfg.load(); s_maxForwards = s_cfg.getIntValue("general","maxforwards",20); s_privacy = s_cfg.getBoolValue("general","privacy"); s_auto_nat = s_cfg.getBoolValue("general","nat",true); if (!m_endpoint) { m_endpoint = new YateSIPEndPoint(); if (!(m_endpoint->Init())) { delete m_endpoint; m_endpoint = 0; return; } m_endpoint->startup(); setup(); installRelay(Halt); installRelay(Progress); Engine::install(new UserHandler); if (s_cfg.getBoolValue("general","generate")) Engine::install(new SipHandler); } } /* vi: set ts=8 sw=4 sts=4 noet: */