/usr/include/liveMedia/SIPClient.hh is in liblivemedia-dev 2011.12.23-1.
This file is owned by root:root, with mode 0o644.
The actual contents of the file can be viewed below.
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 | /**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
This library 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 Lesser General Public License for
more details.
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
// Copyright (c) 1996-2012 Live Networks, Inc. All rights reserved.
// A generic SIP client
// C++ header
#ifndef _SIP_CLIENT_HH
#define _SIP_CLIENT_HH
#ifndef _MEDIA_SESSION_HH
#include "MediaSession.hh"
#endif
#ifndef _NET_ADDRESS_HH
#include "NetAddress.hh"
#endif
#ifndef _DIGEST_AUTHENTICATION_HH
#include "DigestAuthentication.hh"
#endif
// Possible states in the "INVITE" transition diagram (RFC 3261, Figure 5)
enum inviteClientState { Calling, Proceeding, Completed, Terminated };
class SIPClient: public Medium {
public:
static SIPClient* createNew(UsageEnvironment& env,
unsigned char desiredAudioRTPPayloadFormat,
char const* mimeSubtype = NULL,
int verbosityLevel = 0,
char const* applicationName = NULL);
void setProxyServer(unsigned proxyServerAddress,
portNumBits proxyServerPortNum);
void setClientStartPortNum(portNumBits clientStartPortNum) {
fClientStartPortNum = clientStartPortNum;
}
char* invite(char const* url, Authenticator* authenticator = NULL);
// Issues a SIP "INVITE" command
// Returns the session SDP description if this command succeeds
char* inviteWithPassword(char const* url,
char const* username, char const* password);
// Uses "invite()" to do an "INVITE" - first
// without using "password", then (if we get an Unauthorized
// response) with an authentication response computed from "password"
Boolean sendACK(); // on current call
Boolean sendBYE(); // on current call
static Boolean parseSIPURL(UsageEnvironment& env, char const* url,
NetAddress& address, portNumBits& portNum);
// (ignores any "<username>[:<password>]@" in "url")
static Boolean parseSIPURLUsernamePassword(char const* url,
char*& username,
char*& password);
protected:
virtual ~SIPClient();
private:
SIPClient(UsageEnvironment& env,
unsigned char desiredAudioRTPPayloadFormat,
char const* mimeSubtype,
int verbosityLevel,
char const* applicationName);
// called only by createNew();
void reset();
// Routines used to implement invite*():
char* invite1(Authenticator* authenticator);
Boolean processURL(char const* url);
Boolean sendINVITE();
static void inviteResponseHandler(void* clientData, int mask);
void doInviteStateMachine(unsigned responseCode);
void doInviteStateTerminated(unsigned responseCode);
TaskToken fTimerA, fTimerB, fTimerD;
static void timerAHandler(void* clientData);
static void timerBHandler(void* clientData);
static void timerDHandler(void* clientData);
unsigned const fT1; // in microseconds
unsigned fTimerALen; // in microseconds; initially fT1, then doubles
unsigned fTimerACount;
// Routines used to implement all commands:
char* createAuthenticatorString(Authenticator const* authenticator,
char const* cmd, char const* url);
Boolean sendRequest(char const* requestString, unsigned requestLength);
unsigned getResponseCode();
unsigned getResponse(char*& responseBuffer, unsigned responseBufferSize);
Boolean parseResponseCode(char const* line, unsigned& responseCode);
private:
// Set for all calls:
unsigned char fDesiredAudioRTPPayloadFormat;
char* fMIMESubtype;
unsigned fMIMESubtypeSize;
int fVerbosityLevel;
unsigned fCSeq; // sequence number, used in consecutive requests
char const* fApplicationName;
unsigned fApplicationNameSize;
char const* fOurAddressStr;
unsigned fOurAddressStrSize;
portNumBits fOurPortNum;
Groupsock* fOurSocket;
char* fUserAgentHeaderStr;
unsigned fUserAgentHeaderStrSize;
// Set for each call:
char const* fURL;
unsigned fURLSize;
struct in_addr fServerAddress;
portNumBits fServerPortNum; // in host order
portNumBits fClientStartPortNum; // in host order
unsigned fCallId, fFromTag; // set by us
char const* fToTagStr; // set by the responder
unsigned fToTagStrSize;
Authenticator fValidAuthenticator;
char const* fUserName; // 'user' name used in "From:" & "Contact:" lines
unsigned fUserNameSize;
char* fInviteSDPDescription;
char* fInviteCmd;
unsigned fInviteCmdSize;
Authenticator* fWorkingAuthenticator;
inviteClientState fInviteClientState;
char fEventLoopStopFlag;
};
#endif
|