2656 lines
71 KiB
C++
2656 lines
71 KiB
C++
/**
|
|
* 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 <yatephone.h>
|
|
#include <yatesip.h>
|
|
|
|
#include <string.h>
|
|
|
|
|
|
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<NamedString*>(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<RtpMedia*>(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<const SIPHeaderLine*>(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<NamedString*>(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; i<len; i++) {
|
|
NamedString *n = l->getParam(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<GenObject*>(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<SDPBody*>(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<RtpMedia*>(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;
|
|
Message* s = message("chan.startup");
|
|
s->addParam("caller",m_uri.getUser());
|
|
s->addParam("called",uri.getUser());
|
|
Engine::enqueue(s);
|
|
}
|
|
|
|
// 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();
|
|
setMaxcall(msg);
|
|
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<RtpMedia*>(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<RtpMedia*>(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<CallEndpoint *>(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<RtpMedia*>(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<RtpMedia*>(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<RtpMedia*>(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<RtpMedia*>(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<RtpMedia*>(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<String*>(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<String*>(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<RtpMedia*>(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<RtpMedia*>(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<SDPBody*>(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);
|
|
maxcall(0);
|
|
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<SDPBody*>(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<const RtpMedia*>(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 << "\" <sip:";
|
|
tmp << m_username << "@";
|
|
tmp << m->getParty()->getLocalAddr() << ":";
|
|
tmp << m->getParty()->getLocalPort() << ">";
|
|
m->addHeader("Contact",tmp);
|
|
m->addHeader("Expires",exp);
|
|
tmp = "<sip:";
|
|
tmp << m_username << "@" << domain() << ">";
|
|
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<SIPHeaderLine*>(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<YateSIPConnection*>(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<YateSIPConnection*>(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<YateSIPLine*>(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<YateSIPLine*>(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<YateSIPLine*>(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<CallEndpoint*>(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: */
|