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Added SIP (IP telephony) client support. (This was already supported in the

Browse Source 
LIVE.COM libraries, so updating the MPlayer code to support it required
only relatively minor changes.)


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@10055 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
rsf 2003-05-03 06:13:11 +00:00
parent fb8258b925
commit a3b6526ac6
1 changed files with 59 additions and 26 deletions
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85
libmpdemux/demux_rtp.cpp
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@ -87,6 +87,7 @@ private:
typedef struct RTPState {
char const* sdpDescription;
RTSPClient* rtspClient;
SIPClient* sipClient;
MediaSession* mediaSession;
ReadBufferQueue* audioBufferQueue;
ReadBufferQueue* videoBufferQueue;
@ -96,6 +97,26 @@ typedef struct RTPState {
extern "C" char* network_username;
extern "C" char* network_password;
static char* openURL_rtsp(RTSPClient* client, char const* url) {
// If we were given a user name (and optional password), then use them:
if (network_username != NULL) {
char const* password = network_password == NULL ? "" : network_password;
return client->describeWithPassword(url, network_username, password);
} else {
return client->describeURL(url);
}
}
static char* openURL_sip(SIPClient* client, char const* url) {
// If we were given a user name (and optional password), then use them:
if (network_username != NULL) {
char const* password = network_password == NULL ? "" : network_password;
return client->inviteWithPassword(url, network_username, password);
} else {
return client->invite(url);
}
}
int rtspStreamOverTCP = 0;
extern "C" void demux_open_rtp(demuxer_t* demuxer) {
@ -106,6 +127,7 @@ extern "C" void demux_open_rtp(demuxer_t* demuxer) {
if (env == NULL) break;
RTSPClient* rtspClient = NULL;
SIPClient* sipClient = NULL;
if (demuxer == NULL || demuxer->stream == NULL) break; // shouldn't happen
demuxer->stream->eof = 0; // just in case
@ -115,26 +137,31 @@ extern "C" void demux_open_rtp(demuxer_t* demuxer) {
char* sdpDescription = (char*)(demuxer->stream->priv);
if (sdpDescription == NULL) {
// We weren't given a SDP description directly, so assume that
// we were given a RTSP URL:
// we were given a RTSP or SIP URL:
char const* protocol = demuxer->stream->streaming_ctrl->url->protocol;
char const* url = demuxer->stream->streaming_ctrl->url->url;
extern int verbose;
rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer");
if (rtspClient == NULL) {
fprintf(stderr, "Failed to create RTSP client: %s\n",
env->getResultMsg());
break;
if (strcmp(protocol, "rtsp") == 0) {
rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer");
if (rtspClient == NULL) {
fprintf(stderr, "Failed to create RTSP client: %s\n",
env->getResultMsg());
break;
}
sdpDescription = openURL_rtsp(rtspClient, url);
} else { // SIP
unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM)
sipClient = SIPClient::createNew(*env, desiredAudioType, NULL,
verbose, "MPlayer");
if (sipClient == NULL) {
fprintf(stderr, "Failed to create SIP client: %s\n",
env->getResultMsg());
break;
}
sipClient->setClientStartPortNum(8000);
sdpDescription = openURL_sip(sipClient, url);
}
// If we were given a user name (and optional password), then use them:
if (network_username != NULL) {
char const* password
= network_password == NULL ? "" : network_password;
sdpDescription
= rtspClient->describeWithPassword(url, network_username, password);
} else {
sdpDescription = rtspClient->describeURL(url);
}
if (sdpDescription == NULL) {
fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",
url, env->getResultMsg());
@ -152,6 +179,7 @@ extern "C" void demux_open_rtp(demuxer_t* demuxer) {
RTPState* rtpState = new RTPState;
rtpState->sdpDescription = sdpDescription;
rtpState->rtspClient = rtspClient;
rtpState->sipClient = sipClient;
rtpState->mediaSession = mediaSession;
rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL;
rtpState->flags = 0;
@ -201,6 +229,8 @@ extern "C" void demux_open_rtp(demuxer_t* demuxer) {
if (rtspClient != NULL) {
// Issue a RTSP aggregate "PLAY" command on the whole session:
if (!rtspClient->playMediaSession(*mediaSession)) break;
} else if (sipClient != NULL) {
sipClient->sendACK(); // to start the stream flowing
}
// Now that the session is ready to be read, do additional
@ -319,7 +349,7 @@ Boolean insertRTPData(demuxer_t* demuxer, demux_stream_t* ds,
return True;
}
static void teardownRTSPSession(RTPState* rtpState); // forward
static void teardownRTSPorSIPSession(RTPState* rtpState); // forward
extern "C" void demux_close_rtp(demuxer_t* demuxer) {
// Reclaim all RTP-related state:
@ -328,7 +358,7 @@ extern "C" void demux_close_rtp(demuxer_t* demuxer) {
RTPState* rtpState = (RTPState*)(demuxer->priv);
if (rtpState == NULL) return;
teardownRTSPSession(rtpState);
teardownRTSPorSIPSession(rtpState);
UsageEnvironment* env = NULL;
TaskScheduler* scheduler = NULL;
@ -338,6 +368,7 @@ extern "C" void demux_close_rtp(demuxer_t* demuxer) {
}
Medium::close(rtpState->mediaSession);
Medium::close(rtpState->rtspClient);
Medium::close(rtpState->sipClient);
delete rtpState->audioBufferQueue;
delete rtpState->videoBufferQueue;
delete rtpState->sdpDescription;
@ -479,16 +510,18 @@ static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
return dp;
}
static void teardownRTSPSession(RTPState* rtpState) {
RTSPClient* rtspClient = rtpState->rtspClient;
static void teardownRTSPorSIPSession(RTPState* rtpState) {
MediaSession* mediaSession = rtpState->mediaSession;
if (rtspClient == NULL || mediaSession == NULL) return;
if (mediaSession == NULL) return;
if (rtpState->rtspClient != NULL) {
MediaSubsessionIterator iter(*mediaSession);
MediaSubsession* subsession;
MediaSubsessionIterator iter(*mediaSession);
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
rtspClient->teardownMediaSubsession(*subsession);
while ((subsession = iter.next()) != NULL) {
rtpState->rtspClient->teardownMediaSubsession(*subsession);
}
} else if (rtpState->sipClient != NULL) {
rtpState->sipClient->sendBYE();
}
}