void SIPClient::timerDHandler(void* clientData) { SIPClient* client = (SIPClient*)clientData; if (client->fVerbosityLevel >= 1) { client->envir() << "TIMER D EXPIRED\n"; } client->doInviteStateMachine(timerDFires); }
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. }
void SIPClient::timerBHandler(void* clientData) { SIPClient* client = (SIPClient*)clientData; if (client->fVerbosityLevel >= 1) { client->envir() << "RETRANSMISSION TIMEOUT, after " << 64*client->fT1/1000000.0 << " seconds\n"; fflush(stderr); } client->doInviteStateMachine(timerBFires); }
void SIPClient::timerAHandler(void* clientData) { SIPClient* client = (SIPClient*)clientData; if (client->fVerbosityLevel >= 1) { client->envir() << "RETRANSMISSION " << ++client->fTimerACount << ", after " << client->fTimerALen/1000000.0 << " additional seconds\n"; } client->doInviteStateMachine(timerAFires); }
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; }
char* getOptionsResponse(Medium* client, char const* url, char* username, char* password) { SIPClient* sipClient = (SIPClient*)client; sipClient->envir().setResultMsg("NOT SUPPORTED IN CLIENT");//##### return NULL;//##### }
Boolean clientTearDownSession(Medium* client, MediaSession* /*session*/) { if (client == NULL) return False; SIPClient* sipClient = (SIPClient*)client; return sipClient->sendBYE(); }
extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) { struct MPOpts *opts = demuxer->opts; Boolean success = False; do { TaskScheduler* scheduler = BasicTaskScheduler::createNew(); if (scheduler == NULL) break; UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler); 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 // Look at the stream's 'priv' field to see if we were initiated // via a SDP description: 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 or SIP URL: char const* protocol = demuxer->stream->streaming_ctrl->url->protocol; char const* url = demuxer->stream->streaming_ctrl->url->url; extern int verbose; 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 (sdpDescription == NULL) { fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n", url, env->getResultMsg()); break; } } // Now that we have a SDP description, create a MediaSession from it: MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription); if (mediaSession == NULL) break; // Create a 'RTPState' structure containing the state that we just created, // and store it in the demuxer's 'priv' field, for future reference: RTPState* rtpState = new RTPState; rtpState->sdpDescription = sdpDescription; rtpState->rtspClient = rtspClient; rtpState->sipClient = sipClient; rtpState->mediaSession = mediaSession; rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL; rtpState->flags = 0; rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0; demuxer->priv = rtpState; int audiofound = 0, videofound = 0; // Create RTP receivers (sources) for each subsession: MediaSubsessionIterator iter(*mediaSession); MediaSubsession* subsession; unsigned desiredReceiveBufferSize; while ((subsession = iter.next()) != NULL) { // Ignore any subsession that's not audio or video: if (strcmp(subsession->mediumName(), "audio") == 0) { if (audiofound) { fprintf(stderr, "Additional subsession \"audio/%s\" skipped\n", subsession->codecName()); continue; } desiredReceiveBufferSize = 100000; } else if (strcmp(subsession->mediumName(), "video") == 0) { if (videofound) { fprintf(stderr, "Additional subsession \"video/%s\" skipped\n", subsession->codecName()); continue; } desiredReceiveBufferSize = 2000000; } else { continue; } if (rtsp_port) subsession->setClientPortNum (rtsp_port); if (!subsession->initiate()) { fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg()); } else { fprintf(stderr, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession->mediumName(), subsession->codecName(), subsession->clientPortNum()); // Set the OS's socket receive buffer sufficiently large to avoid // incoming packets getting dropped between successive reads from this // subsession's demuxer. Depending on the bitrate(s) that you expect, // you may wish to tweak the "desiredReceiveBufferSize" values above. int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum(); int receiveBufferSize = increaseReceiveBufferTo(*env, rtpSocketNum, desiredReceiveBufferSize); if (verbose > 0) { fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n", subsession->mediumName(), receiveBufferSize); } if (rtspClient != NULL) { // Issue a RTSP "SETUP" command on the chosen subsession: if (!rtspClient->setupMediaSubsession(*subsession, False, rtsp_transport_tcp)) break; if (!strcmp(subsession->mediumName(), "audio")) audiofound = 1; if (!strcmp(subsession->mediumName(), "video")) videofound = 1; } } } 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 // MPlayer codec-specific initialization on each subsession: iter.reset(); while ((subsession = iter.next()) != NULL) { if (subsession->readSource() == NULL) continue; // not reading this unsigned flags = 0; if (strcmp(subsession->mediumName(), "audio") == 0) { rtpState->audioBufferQueue = new ReadBufferQueue(subsession, demuxer, "audio"); rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue); rtpCodecInitialize_audio(demuxer, subsession, flags); } else if (strcmp(subsession->mediumName(), "video") == 0) { rtpState->videoBufferQueue = new ReadBufferQueue(subsession, demuxer, "video"); rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue); rtpCodecInitialize_video(demuxer, subsession, flags); } rtpState->flags |= flags; } success = True; } while (0); if (!success) return NULL; // an error occurred // Hack: If audio and video are demuxed together on a single RTP stream, // then create a new "demuxer_t" structure to allow the higher-level // code to recognize this: if (demux_is_multiplexed_rtp_stream(demuxer)) { stream_t* s = new_ds_stream(demuxer->video); demuxer_t* od = demux_open(opts, s, DEMUXER_TYPE_UNKNOWN, opts->audio_id, opts->video_id, opts->sub_id, NULL); demuxer = new_demuxers_demuxer(od, od, od); } return demuxer; }
void SIPClient::inviteResponseHandler(void* clientData, int /*mask*/) { SIPClient* client = (SIPClient*)clientData; unsigned responseCode = client->getResponseCode(); client->doInviteStateMachine(responseCode); }