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/**********
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.,
59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
**********/
// Copyright (c) 1996-2004, Live Networks, Inc. All rights reserved
// A SIP client test program that opens a SIP URL argument,
// and extracts the data from each incoming RTP stream.
#include "playCommon.hh"
#include "SIPClient.hh"
Medium* createClient(UsageEnvironment& env,
int verbosityLevel, char const* applicationName) {
// First, trim any directory prefixes from "applicationName":
char const* suffix = &applicationName[strlen(applicationName)];
while (suffix != applicationName) {
if (*suffix == '/' || *suffix == '\\') {
applicationName = ++suffix;
break;
}
--suffix;
}
extern unsigned char desiredAudioRTPPayloadFormat;
extern char* mimeSubtype;
return SIPClient::createNew(env,
desiredAudioRTPPayloadFormat, mimeSubtype,
verbosityLevel, applicationName);
}
char* getOptionsResponse(Medium* client, char const* url) {
SIPClient* sipClient = (SIPClient*)client;
sipClient->envir().setResultMsg("NOT SUPPORTED IN CLIENT");//#####
return NULL;//#####
}
char* getSDPDescriptionFromURL(Medium* client, char const* url,
char const* username, char const* password,
char const* proxyServerName,
unsigned short proxyServerPortNum,
unsigned short clientStartPortNum) {
SIPClient* sipClient = (SIPClient*)client;
if (proxyServerName != NULL) {
// Tell the SIP client about the proxy:
NetAddressList addresses(proxyServerName);
if (addresses.numAddresses() == 0) {
client->envir() << "Failed to find network address for \""
<< proxyServerName << "\"\n";
} else {
NetAddress address = *(addresses.firstAddress());
unsigned proxyServerAddress // later, allow for IPv6 #####
= *(unsigned*)(address.data());
if (proxyServerPortNum == 0) proxyServerPortNum = 5060; // default
sipClient->setProxyServer(proxyServerAddress, proxyServerPortNum);
}
}
if (clientStartPortNum == 0) clientStartPortNum = 8000; // default
sipClient->setClientStartPortNum(clientStartPortNum);
char* result;
if (username != NULL && password != NULL) {
result = sipClient->inviteWithPassword(url, username, password);
} else {
result = sipClient->invite(url);
}
extern unsigned statusCode;
statusCode = sipClient->inviteStatus();
return result;
}
Boolean clientSetupSubsession(Medium* client, MediaSubsession* subsession,
Boolean streamUsingTCP) {
subsession->sessionId = "mumble"; // anything that's non-NULL will work
return True;
}
Boolean clientStartPlayingSession(Medium* client,
MediaSession* /*session*/) {
SIPClient* sipClient = (SIPClient*)client;
return sipClient->sendACK();
//##### This isn't quite right, because we should really be allowing
//##### for the possibility of this ACK getting lost, by retransmitting
//##### it *each time* we get a 2xx response from the server.
}
Boolean clientTearDownSession(Medium* client,
MediaSession* /*session*/) {
if (client == NULL) return False;
SIPClient* sipClient = (SIPClient*)client;
return sipClient->sendBYE();
}
Boolean allowProxyServers = True;
Boolean controlConnectionUsesTCP = False;
Boolean supportCodecSelection = True;
char const* clientProtocolName = "SIP";
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