Mac Os X Rtsp Player

on OSX , I tried VLC , Quicktime , MplayerX and I can't get it to workI tried playing with settings in VLC to make it work but it didn't please help it's really anonying to have to go to windows everytime just to play the the RTSP link

Mac Os X Rtsp Player

Download File: https://urlcod.com/2vFhdq

The RTSP Video Player is a popular format for efficiently streaming real-time audio-visual media. By playing it on the VLC player on your Mac, you can enjoy the high-quality entertainment. But how well does VLC support RTSP, and are there other alternatives?

An RTSP Video Player or RTSP Stream is short for Real-Time Streaming Protocol. RTSP is a way to facilitate real-time control of streaming audio-visual (AV) data from media servers. However, the player does not stream the multimedia file itself but communicates with the server that streams it.

VLC, short for VideoLAN Client, is a free and popular media-playing software and streaming server. The media player is available on desktop operating systems like Windows and macOS and mobile platforms like Android and iOS. In addition, VLC supports AV compression methods and online streaming playback of RTSP.

Note: VLC also supports other online streaming formats like HTTP, RTMP, MMS, FTP, etc. You can open a network stream on VLC simply by entering the URL of the stream in the URL field, as you see in VLC at the top of the player. To open an RTP or UDP stream, you can press the button at the bottom of the screen.

Since RTSP facilitates real-time (live) streaming of audio-visual media on Mac, we can also access the live stream from IP (Network) Cameras and play them on different supported media players. Of course, this live media playback is supported on Windows as well. We have suggested one of these media players above: IP Camera Viewer 2.

We hope this article provided necessary insights on RTSP streaming and its related aspects. The VLC Media Player is a fan-favorite choice for RTSP streaming, but there are other alternatives that you may like better. Therefore, we recommend that you explore all the media player options available for RTSP streams and choose the most suitable VLC alternative for your Mac.

To play RTSP files on Windows 10 and macOS, you will need 5KPlayer - a professional RTSP stream player that supports RTSP streaming protocol as well as HLS IPTV streaming, and provides smooth playback of H.264, HEVC 4K 8K online movie streaming. Free download it now and enjoy the RTSP file without hiccups.

BTW, here we choose some rstp links for you to test your RSTP playerrtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.movrtsp://masds03.htc.com.tw/Bird_On_Aware_QCIF_5fps_mpeg4_23k_amr-nb_12k.mp4rtsp://masds03.htc.com.tw/99min_H264.3gprtsp://masds03.htc.com.tw/sit/SMPHQ/pcslab/VGA/VGA%20MP4%20-%20200kbps%203gp.3gp

Here we strongly recommend you 5 RTSP protocol compatible players: 5KPlayer, VLC media player, These media player big names all have support for RTSP streaming and RTP streaming. Lightweight, cross-platforms, user-friendly and robust! And each of them features differently.

5KPlayer is designed as an 4K UHD player that not only plays multimedia formats like DVD, MP4, MKV, FLV, HEVC, H.264, etc but also does well in handling live streams like RTSP, M3U8/M3U IPTV and online radios. Just copy the live stream URL to 5KPlayer, it plays immediately and smoothly. Besides, the built-in AirPlay and DLNA function are also another good options to stream video music accross platforms. With ability to access to over 300+ online videos sites including YouTube, Twitter, SoundCloud and Twitch, 5KPlayer also gives you a convenient way to download online videos for free as many as possible.

VLC media player must be the leader in the streaming player world. In addition to a RTSP player, VLC is also capable of analyzing and playing HTTP, RTP, MMS and HLS streams. For offline media streaming, VLC is similar to 5KPlayer to provide a DLNA function for multi-screen sharing. Besides, VLC AirPlay function is also under active development to Apple users. Speaking of local media files, VLC media player is also an ideal choice since it has a lot of tweaks and settings for users to customize.

FFmpeg is not a media player, actually, it is a powerful, free and open-source project for handling video, audio, and other multimedia files and streams. Some of the core algorithms of VLC, YouTube, iTunes, etc. are based on the workflow of this program. It is fast to access RTSP, RTMP, HTTP, HLS, etc stream but it is very complex to set up and use.

Megacubo is a dedicated streaming player which can automatically capture the content from transmission link/URL. In addition to RTSP, it can also play M3U8, RTMP, and IPTV playlist lists. But this player is only capable with Windows and Linux system.

Kodi is a free and open-source media player software application developed by the XBMC Foundation that can be used on Windows and macOS computer. The most important thing to make Kodi one of the best choices to play RTSP stream is that you can customize your own player (easier than VLC), not only including the skin of the player, but also the features. Besides, you can install different add-ons to make it as simple as a light-weight video player, or as powerful as a video toolbox. If you want to play RTSP streams with Kodi, go to install the RTMP Input add-on from its official site.

Step 1: Launch this RTSP/RTP player and click on its "Live" section.Step 2: Copy and paste the RTSP URL of the content into the URL box. Step 3: Click "Play" and 5KPlayer will respond soon. Note: The RTSP URL once can be sniffed out from online using " video-id" to get XML and then we can fetch RTSP URL. However, this method no longer works since YouTube stopped supporting v2.0.

To play RTSP/RTP video stream of IP camera on Windows/macOS, you will need to abide by RTSP URL formats and know IP address of your IP camera server. Regularly RTSP sample URL would be - rtsp://server.example.org:8080/test.sdp

Step 1: For instance, you are using 4XEM E103C IP camera, and you will find its RTSP URL format like this - "rtsp://ip_address/live.sdp"Step 2: Replace "ip_address" with your real IP address and paste the whole RTSP URL into 5KPlayer "Live" URL box.

Almost IP surveillance cameras support RTSP video stream, that means user can use media player to watch the live video from anywhere. RTSP is the abbreviation of real time streaming protocol, it's a network control protocol designed for use in entertainment and communications systems to control streaming media servers. The protocol is used for establishing and controlling media sessions between end points. This article describes how to play RTSP video stream of IP cameras on VLC player, QuickTime player, and a mobile phone with popular IP camera viewer App.

VLC player and QuickTime player are free media players that support cross-platforms (Windows OS, Mac OS), these two media players have capability to play most multimedia files and various streaming protocols.

rtsp://IP Camera address:8557/h264 for Primary streamrtsp://IP Camera address:554/mpeg4 for Primary streamrtsp://IP Camera address:8555/mjpeg for Primary stream rtsp://IP Camera address:8556/h264 for Secondary stream rtsp://IP Camera address:8554/mpeg4 for Secondary stream rtsp://IP Camera address:8558/mjpeg for Third stream

"openRTSP" is a command-line program that can be used to open, stream,receive, and (optionally) record media streams that are specified by aRTSPURL - i.e., an URL that begins withrtsp://(A related program- "playSIP"- can be used to play/record aSIPsession.) Basic operation Playing without receiving Playing-time options Streaming access-controlled sessions Outputting a ".mov", ".mp4", or ".avi"-format file Periodic file output 'Trick play' options Other options A note about RealAudio and RealVideo sessions Source codeSupport and customizationSummary of command-line options Basic operationThe simplest way to run this program is: openRTSP where is a RTSP URL to open(i.e., beginning with "rtsp://").The program will open the given URL (using RTSP's "DESCRIBE" command),retrieve the session's SDP description,and then, for each audio/video subsession whose RTP payload format itunderstands, "SETUP" and "PLAY" the subsession.The received data for each subsession is written into a separate outputfile, named according to its MIME type.For example, if the session contains a MPEG-1 or 2 audio subsession(RTP payload type 14) - e.g., MP3 - and a MPEG-1 or 2 videosubsession (RTP payload type 32),then each subsession's datawill be extracted from the incoming RTP packets andwritten to files named "audio-MPA-1" and "video-MPV-2" (respectively).(You will probably then need to rename these files- by giving them an appropriate filename extension(e.g., ".mp3" and ".mpg") - in orderto be able to play them using common media player tools.)You can use the"-F " option toadd a prefix to the file name that is written for each subsession.(This can be useful if you are running "openRTSP" several times,in the same directory,to read data from different RTSP sessions.)Extracting a single streamTo record only the audio stream from a session, use the"-a" command-line option.(Similarly, to record only the video stream, use the"-v" option. To record only the 'application' (e.g., 'metadata') stream, use the"-L" option.)In this case, the output audio (or video, or 'application') stream willbe written to 'stdout', rather than to a file(unless the"-P "option (see below) is given).Less verbose diagnostic outputBy default, the program will print out (to 'stderr') each completeRTSP request and response.For less verbose output, use the"-V" (upper-case) option.Playing without receivingIf you want the program to play the RTSP session,but not actually receive it, you can do so by giving theprogram the"-r" ('don't receive') option.This is useful if you have a separate application (running on the samehost) that actually receives the incoming media session(s).(Note that this separate receiving application should also send backRTCP Reception Reports, to ensure that the session doesn't time out.)If you use the "-r" option to play a unicast session,you'll probably also want to use the"-p " option.This option tells the program which client port numbers to use in theRTSP "SETUP" commands - i.e., which RTP/RTCPports the server should send to.(Without the "-r" option, the program receives the streams itself,and uses its own ephemeral port numbers for this.) must be an even number.For example, if the RTSP session consists of an audio and a videosubsession (listed in that order in the returned SDP description), then"-p 6666" will cause ports 6666 and 6667 to be used for theaudio subsession (6666 for RTP; 6667 for RTCP),and ports 6668 and 6669 to be used forthe video subsession (6668 for RTP; 6669 for RTCP).(If you use the "-r" option to play a multicast session,then you probably won't also need to use the"-p " option, because the SDPdescription for multicast sessions usually includes a port number to use.)Playing-time optionsIf the SDP description (from the RTSP server) contains a"a=range:npt= ..." attribute specifying a duration for the stream,then the program will close down the session and exit shortly after(by default, 5 seconds after) this duration elapses.You can change this duration using the"-d " option.Ifis positive, it is the total number ofseconds of the stream to be played before closing down the session and exiting. Ifis negative, then-gives the number of extra seconds to delay after the time specifiedin the SDP "a=range" attribute.(As noted above, the default value for this extra time is 5 seconds.)For example, if the SDP description contains"a=range:npt=0-25", then"-d 10"means "play the stream(s) for 10 seconds, then exit", and"-d -10"means "play the stream(s) for 35 seconds, then exit".You can also use the"-D " optionto ask that the program shut down if no new incoming RTP (i.e., data)packets are received within a period of at leastseconds.This option is useful if you are running the program automatically(e.g., from within a script), and wish to allow for the possibilityof servers that dieunexpectedly.(Note that "-d" and "-D" are different options, and may both be used.)Note, however, that ifthe program receives a RTCP "BYE" packet from the source - for every streamin the session - then the program will close down the session andexit immediately, regardless of the use of the "-d" and/or "-D" options.You can also use the"-c"option to play the mediasessions continuously.I.e., after the end time has elapsed, the program starts all over again,by issuing another set of "PLAY" requests.(Note that if you're receiving data (i.e., you don't use the "-r" option),then this means you'll get multiple copies of the data inthe output file(s).)Note that you can combine "-c" with"-d "and/or"-D ".So, for example,"-c -d 10"means "play the stream(s) for 10 seconds, then go back and playthem again for another 10 seconds, etc., etc."Streaming access-controlled sessionsSome RTSP servers require user authentication (via a name and password)before a session can be streamed. To stream such a session, use the"-u " option.(To specify an empty password, use"" for .)The program authenticates using RTSP "digest authentication"; the passwordwill not get sent in the clear over the net.Alternatively, you could try including the user name and password insidethe URL, as:"rtsp://:@:".(In this case, though, the password will be sent in the clearover the net. Also, not all servers will accept this type of URL.)Outputting a ".mov", ".mp4", or ".avi"-format fileUse the"-q"option to output the received data to 'stdout' in the form of anAppleQuickTime '.mov'-format file.Each received subsession will be have its own track in the output file.Similarly, the"-4"option produces a '.mp4'-format (i.e., MPEG-4) file.At present these options are fully supported for only a limited number of codecs.For those codecs that are not fully supported, the program willstill store all of its received data into a movie track, but willuse a dummy Media Data Atom (named '????') in the Sample Description.(This track will also be disabled.)Before you can play such a track, you will need toedit the file.If the session contains a video subsession, you should also use the"-w ","-h "and"-f "optionsto specify the width and height (in pixels), and frame rate(per-second) of the correspondingvideo track.(If these options are omitted, then the values width=240 pixels;height=180 pixels; frame-rate=15 are used.)These values are important; if they are not correct, your file might not play at all!Alternatively, if the session's SDP description contains themedia-level attribute"a=x-dimensions: ,",then these values will be used instead (in which case you won't needto use the "-w" and "-h" options).Similarly, if the session's SDP description contains themedia-level attribute"a=x-framerate: ",then this value will be used instead (in which case you won't needto use the "-f" option).If the resulting QuickTime movie file contains audio and video tracks thatare out-of-sync, then you can use the "-y"option to try to generate synchronized audio/video tracks.(This option works by listening for RTCP "Sender Report" packets- containing time synchronization information - for each stream.Some initial, unsynchronized data may end up being discarded.)The"-H"option will also generate a QuickTime 'hint track' for each audio or videotrack.This is useful if you later wish to stream the resulting ".mov" or ".mp4" file.Note:"openRTSP"s support for creating QuickTime format files israther limited.At present, only PCM (u-law and a-law),MPEG-4, GSMand QCELP (aka. 'PureVoice') audio is supported,and only MPEG-4, H.263/H.263+, and H.264 video is supported.(Also, for creating hintedQuickTime format files, QCELP audio is not currently supported.)The"-i"option produces a '.avi'-format file.(This functionality is not fully-supported.MPEG-1, 2 or 4, JPEG and H.263 video is supported,along with raw PCM or u-law audio.However, MPEG and other audio codecs are not yet supported.)Important note:If you are outputting a ".mov", ".mp4", or ".avi"-format file, you must let"openRTSP" run to completion,or else terminate it cleanly, by signalling it with SIGHUP or SIGUSR1.You must not terminate it using -C, otherwise the output file will not get written properly.Periodic file outputIf the"-P "option is given,than "openRTSP" will output a new file everyseconds.(Each output file's name will include the time range (in seconds) that it represents.)This option can also be used with the"-q", "-4", and "-i"options (for outputting '.mov', '.mp4', and '.avi' files, respectively).In this case, the output files will be written separately, rather than to 'stdout'. 'Trick play' optionsRTSP servers may optionally support 'trick play' operations on a stream - specifically, the ability toseek to a specific time within the stream,and/or the ability to play the stream in fast-forward, slow, or reverse play.Use the "-s "option to request that the stream be started at the second mark (default: 0.0).Use the "-z "option to request fast-forward play( > 1.0),slow play(0 -n"optionif you wish to be 'notified'(with a console message)when the first data (RTP) packets beginarriving.Receiving streamed data via TCP instead of UDPIf you're not receiving any data packets(you can test this using "-n"), then you may be behind a firewallthat (stupidly) blocks UDP packets.If this is the case, you can use the"-t"optionto request that the RTSP server stream RTP and RTCP data packets overits TCP connection, instead of using UDP packets.(Note that not all RTSP servers support TCP streaming, and thatTCP cannot be used to receive multicast streams.)You should use this option only if you are unable to receive UDP packets,or if you are recording the stream for later playback, and need to do sowithout packet loss.Streaming over TCP can cause incoming data to be excessively delayed,which is inappropriate if the data is being processed in real time.Alternatively, you can use the"-T "option to request that the stream be sent (using TCP) over a"RTSP-over-HTTP tunnel", using the specified HTTP port number.RTSP-over-HTTP tunneling can be useful if you are behind a HTTP-only firewall.(Note, however, that not all RTSP servers support this.)Receiving unsupported RTP payload formatsNote: An "RTP payload format" for a codec is a set of rules that define howthe codec's media frames are packed within RTP packets.This is usually defined by an IETF RFC(or, for newer payload formats, an IETF Internet-Draft).By default, the program will ignore any subsession whose RTP payload formatit doesn't understand (because, if it doesn't know the RTP payload format,it doesn't know how to extract data from the incoming RTP stream).However, if an input stream uses a RTP payload formatthat the program does not support, then you may stillbe able toreceive this data, by usingthe"-S "option.This option tells the program to assume that any such unsupported streamuses a very 'simple' RTP payload format, in which the stream's datais packed contiguously into RTP packets, following the RTP header.(In particular, the payload format must not use interleaving.) specifies the size (in bytes) ofany special header that follows the standard RTP header.(This special header is skipped over, and is not interpreted at all.)For example, if the program didn't already handle PCM u-law audio("audio/PCMU"; RTP payload format code 0), then you could receive it usingthe option"-S 0".If the program didn't already handle MPEG audio("audio/MPEG"; RTP payload format code 14), then you could receive itusing the option"-S 4"(because the RTP payload format for MPEG audio, defined in RFC 2250,specifies a (basically useless) 4-byte header at the start of the RTP payload).Outputting QOS statisticsUse the "-Q" option to output QOS("quality of service") statistics about the data stream(when the program exits).These statistics include the (minimum, average, maximum)bitrate, packet loss rate, and inter-packet gap.The "-Q" option takes an optionalparameter, which specifies the length of thetime intervals - in multiplesof 100ms - over which the "minimum, average, maximum"statistics are computed.The default value of this parameter is "10", meaningthat these statistics are measured every 1 second(i.e., 10x100ms). Outputting server optionsBy default, the program sends an "OPTIONS" command before sending "DESCRIBE".The purpose of the "OPTIONS" command is ask the server to respond with thelist of commands that it supports.If the "-o" option is given, then the program sends the"OPTIONS" command only.If the "-o" option is given, then all other command-line options- except "-V" (less verbose output) - are ignored.The "-O" option has the opposite effect: It tells theprogram to not send an "OPTIONS" command prior to sending"DESCRIBE".Outputting each frame into a separate fileIf the "-m" option is given, each incoming 'frame'will be written into a separate output file.(Note that 'frame' in this case is a discrete unit of data that comesfrom a 'RTPSource'. For some RTP payload formats (such as motion-JPEG),each file will contain a complete image.For other RTP payload formats (such as MPEG video), each file will contain a smaller unit of data, such as a video header structure,or a frame 'slice'.)To distinguish the output files, each 'frame's presentation time is usedin the suffix of the corresponding output file.Changing the output file buffer sizeIf you see an error message"The input frame data was too large for our buffer size",then this indicates that incoming RTP data formed a frame thatwas too large for this program's output file buffer.By default, a 100,000 buffer is used, so this situation usually does notoccur.(It occurs only for codecs that can have very large frames.)If, however,you see this error message, you can increase the output file buffer sizeusing the "-b " option.Changing the input network socket buffer sizeYou can also use the"-B " optionto change the size of the input buffer that the underlyingoperating system uses for network sockets.(You probably won't need to use this option, because the operating system'sdefault buffer size is usually sufficient.)Sending 'keep-alives' to satisfy buggy serversRTSP servers should listen for incoming RTSP "RR" packets (which are sent by allstandards-compliant RTSP clients)as an indication that the client is still alive(and therefore, that the session should not be timed-out).Some buggy servers, however, do not do this; they can time out (and close) the sessionprematurely.To overcome this, you can use the"-K"option,which tells the program to periodically send RTSP "OPTIONS" commands to the server,to try to keep it alive.Playing a stream without knowing its "rtsp://" URLInstead of giving the program a "rtsp://" URL on the command line, you can use the"-R"or"-R "option.This option tells the program to wait - on the specified port number - for an incoming "REGISTER" RTSP commandthat specifies the "rtsp://" URL of a stream to play.(If no is given, then the program will choose a random available port number, and announce it(to 'stderr').)Once the program receives an appropriate request - specifying a "rtsp://" URL - then it will open and stream from that URL,as usual.As a special feature - if the "reuse_connection" parameter was set in the "REGISTER" command's "Transport:" header -the program will reuse the TCP connection on which it received the "REGISTER" command.This can be useful if the stream's server is behind a firewall or NAT, but this program is run on the public Internet.(In this case, you may also wish to use the -t option, to request that the server also send its RTP/RTCP packetson this same TCP connection.)"REGISTER" is a custom RTSP commands, developed by Live Networks, Inc.It is currently non-standard, but is described in anIETF Internet-Draft.If you use the "-R" option, then you should also use the"-k " option,which specifies a username and password that should be used to authenticate the incoming "REGISTER" command.A note about RealAudio and RealVideo sessionsNote that this program cannot be used to receive RealAudio and/orRealVideo sessions - even those described by a "rtsp://" URL - becausethese sessions do not use RTP for transport.(Instead, these sessions use RealNetworks' proprietary"RDT" protocol.)Source codeThis program uses the "RTSPClient", "MediaSession","FileSink", "QuickTimeFileSink",and several "*RTPSource" modules from the "liveMedia" library,which is distributed as part of the"LIVE555 Streaming Media"source code package.The source code for the program itself is also bundled with this package,as the files "openRTSP.cpp"and "playCommon.cpp",in the "testProgs" directory.See the"LIVE555 Streaming Media"documentationfor instructions on how to build this program from source.Note:If you are looking for an example of how to use the"LIVE555 Streaming Media" code to build your own RTSP/RTP media player client,then the "openRTSP" source code is not the best example to use, becauseit includes lots of extra options, most of which you probably won't need.(Also, the "openRTSP" code was designed to be a standalone application - rather than being embedded within some other application.)Instead, you should use the"testRTSPClient" application code (also in the "testProgs" directory) as a model.Support and customizationIf you are interested in seeing new features added to the program(e.g., support for additional RTP payload formatsor QuickTime Media Types),or are interested in customizing this program's functionalityand/or embedding it within your own application,please emailsupport(at)live555.comSummary of command-line options(for "openRTSP" and "playSIP")-4output a '.mp4'-format file(to 'stdout', unless the "-P " option is also given)-aplay only the audio stream(to 'stdout', unless the "-P " option is also given)-A specify the staticRTP payload format number of the audio codecto request from the server("playSIP" only)-b change the output file buffer size-B change the input network socket buffer size-cplay continuously-CExplicitly ask for a multicast stream even if the server's "DESCRIBE" response doesn't specift a multicast address.(Note that not all servers will support this.)("openRTSP" only)-d specify an explicit duration-D specify a maximum period of inactivity to wait before exiting-E request that the server end streaming at the specified absolute time (format: "YYYYMMDDTHHMMSSZ" or "YYYYMMDDTHHMMSS.Z") (used only with -U-f specify the video frame rate (used only with "-q", "-4", or "-i")-F specify a prefix for each output file name-g specify a user agent name to use in outgoing requests-h specify the video image height (used only with "-q", "-4", or "-i")-Houtput a QuickTime 'hint track' for each audio/video track (used only with "-q" or "-4")-ioutput a '.avi'-format file (to 'stdout', unless the "-P " option is also given)-I specify a particular network interface on which to receive data (currently, IPv4 only)-k specify a user name and password that's required to authenticate an incoming "REGISTER" command (used with "-R" only)-KPeriodically send a RTSP "OPTIONS" command, to keep the connection alive. (This is useful with buggy servers that don't listen to our periodic RTCP "RR" packets instead.)-ltry to compensate for packet losses (used only with "-q", "-4", or "-i")-Lplay only the 'application' (e.g., 'metadata') stream (to 'stdout', unless the "-P " option is also given)-moutput each incoming frame into a separate file-M specify the MIME subtype of a dynamic RTP payload format for the audio codecto request from the server("playSIP" only)-nbe notified when RTP data packets start arriving-orequest the server's command options, without sending "DESCRIBE"("openRTSP" only)-Odon't request the server's command options; just send "DESCRIBE"("openRTSP" only)-p specify the client port number(s)-P write new output files every seconds-qoutput a QuickTime '.mov'-format file (to 'stdout', unless the "-P " option is also given)-Qoutput 'QOS' statistics about the data stream (when the program exits)-rplay the RTP streams, but don't receive them ourself-R (or -R )Waits for an incoming "REGISTER" command, specifying a "rtsp://" URL to play. This option is used instead of a "rtsp://" URL on the command line.("openRTSP" only)-s request that the server seek to the specified time (in seconds) before streaming-S assume a simple RTP payload format (skipping over a special header of the specified size)-tstream RTP/RTCP data over TCP,rather than (the usual) UDP.("openRTSP" only)-T like "-t", except using RTSP-over-HTTP tunneling.("openRTSP" only)-u specify a user name and password for digest authentication-U request that the server seek to the specified absolute time (format: "YYYYMMDDTHHMMSSZ" or "YYYYMMDDTHHMMSS.Z") before streaming-vplay only the video stream(to 'stdout', unless the "-P " option is also given)-Vprint less verbose diagnostic output-w specify the video image width (used only with "-q", "-4", or "-i")-ytry to synchronize the audio and video tracks (used only with "-q" or "-4")-z request that the server scale the stream (fast-forward, slow, or reverse play)"LIVE555", "openRTSP", "playSIP",and the Live Networks logo are trademarks ofLive Networks, Inc. 2ff7e9595c