CA2410748A1 - Audio-video-over-ip method, system and apparatus - Google Patents
Audio-video-over-ip method, system and apparatus Download PDFInfo
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- 238000000034 method Methods 0.000 title claims abstract description 49
- 230000035899 viability Effects 0.000 claims abstract description 10
- 238000004891 communication Methods 0.000 claims abstract description 7
- 238000012423 maintenance Methods 0.000 claims description 3
- 230000005540 biological transmission Effects 0.000 description 16
- 230000006870 function Effects 0.000 description 12
- 238000007906 compression Methods 0.000 description 9
- 230000006835 compression Effects 0.000 description 8
- 230000003139 buffering effect Effects 0.000 description 7
- 230000004044 response Effects 0.000 description 6
- 239000002131 composite material Substances 0.000 description 4
- 230000005236 sound signal Effects 0.000 description 4
- 230000015556 catabolic process Effects 0.000 description 3
- 238000006243 chemical reaction Methods 0.000 description 3
- 230000003247 decreasing effect Effects 0.000 description 3
- 238000006731 degradation reaction Methods 0.000 description 3
- 238000005070 sampling Methods 0.000 description 3
- 235000008694 Humulus lupulus Nutrition 0.000 description 2
- 238000013475 authorization Methods 0.000 description 2
- 230000006837 decompression Effects 0.000 description 2
- 230000001934 delay Effects 0.000 description 2
- 238000001514 detection method Methods 0.000 description 2
- 238000005516 engineering process Methods 0.000 description 2
- 230000009118 appropriate response Effects 0.000 description 1
- 230000008859 change Effects 0.000 description 1
- 239000000463 material Substances 0.000 description 1
- 239000000203 mixture Substances 0.000 description 1
- 238000005457 optimization Methods 0.000 description 1
- 238000012552 review Methods 0.000 description 1
- 230000008054 signal transmission Effects 0.000 description 1
Classifications
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1083—In-session procedures
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/60—Network streaming of media packets
- H04L65/70—Media network packetisation
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/80—Responding to QoS
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L67/00—Network arrangements or protocols for supporting network services or applications
- H04L67/01—Protocols
- H04L67/10—Protocols in which an application is distributed across nodes in the network
- H04L67/1001—Protocols in which an application is distributed across nodes in the network for accessing one among a plurality of replicated servers
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
- H04L2012/6472—Internet
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
- H04L2012/6483—Video, e.g. MPEG
Abstract
An audio/video-over-IP method system and apparatus is disclosed.
The method includes the steps of receiving and/or transmitting an audio andlor video signal, and automatically switching communication protocols to ensure signal viability.
The method includes the steps of receiving and/or transmitting an audio andlor video signal, and automatically switching communication protocols to ensure signal viability.
Description
AudioNideo-Over-IP Method, System and Apparatus Field of the Invention The present invention relates generally to audio/video transmission techniques and applications, and more particularly to "Video-over-IP"
transmission methods and devices.
Background of the Invention Internet audio/video streaming methods are generally poor in quality, and lack stability and flexibility. While methods are known that deliver video streaming over IP networks, they do not provide automated protocol switching, require large amounts of bandwidth, and tend to be easily interrupted due to inherent network architectures.
Existing methods typically require from about 3 - 45 Mbps of transmission bandwidth channel for the same signal quality. They also require separate stand-alone devices to pertorm standards conversion for both the input and the output, which usually results in degradation of quality and delays. In existing devices, the protocol switching is not embedded, and therefore has to be manually selected by an operator, which sometimes can be incorrect for the given scenario or a certain network infrastructure, causing interruptions in video/audio transmission.
What is needed is a way to provide audio/video capture, compression, transmission and decompression of a video and audio signal all in the same apparatus in an automated manner.
For the foregoing reasons, there is a need for an improved method and apparatus for "Video over IP".
transmission methods and devices.
Background of the Invention Internet audio/video streaming methods are generally poor in quality, and lack stability and flexibility. While methods are known that deliver video streaming over IP networks, they do not provide automated protocol switching, require large amounts of bandwidth, and tend to be easily interrupted due to inherent network architectures.
Existing methods typically require from about 3 - 45 Mbps of transmission bandwidth channel for the same signal quality. They also require separate stand-alone devices to pertorm standards conversion for both the input and the output, which usually results in degradation of quality and delays. In existing devices, the protocol switching is not embedded, and therefore has to be manually selected by an operator, which sometimes can be incorrect for the given scenario or a certain network infrastructure, causing interruptions in video/audio transmission.
What is needed is a way to provide audio/video capture, compression, transmission and decompression of a video and audio signal all in the same apparatus in an automated manner.
For the foregoing reasons, there is a need for an improved method and apparatus for "Video over IP".
Summary of the Invention The present invention is directed to an audio/video-over-IP method, system and apparatus. The method includes the steps of receiving and/or transmitting an audio and/or video signal, and automatically switching communication protocols to ensure signal viability.
In an aspect of the invention, the method further includes the step of automatically applying at least one signal maintenance technique to maintain signal viability/quality.
The invention requires a mere 1 Mbps while others need to have at least 3 - 45 Mbps for the same signal quality. The invention provides standards converting on the input and output together in the same device.
Automated standards conversion, automated protocol switching, and mirror checking result in low bandwidth requirements for high quality video. Both the encoder and decoder are assembled using relatively inexpensive "off the shelf' components to provide a high quality video transmission device compatible within any IP network, in addition to virtually all audio/video standards.
Other aspects and features of the present invention will become apparent to those ordinarily skilled in the art upon review of the following description of specific embodiments of the invention in conjunction with the accompanying figures.
Brief Description of the Drawings These and other features, aspects, and advantages of the present invention will become better understood with regard to the following description, appended claims, and accompanying drawings where:
Figure 1 is an overview of an audio/video-over-IP method in accordance with the present invention;
In an aspect of the invention, the method further includes the step of automatically applying at least one signal maintenance technique to maintain signal viability/quality.
The invention requires a mere 1 Mbps while others need to have at least 3 - 45 Mbps for the same signal quality. The invention provides standards converting on the input and output together in the same device.
Automated standards conversion, automated protocol switching, and mirror checking result in low bandwidth requirements for high quality video. Both the encoder and decoder are assembled using relatively inexpensive "off the shelf' components to provide a high quality video transmission device compatible within any IP network, in addition to virtually all audio/video standards.
Other aspects and features of the present invention will become apparent to those ordinarily skilled in the art upon review of the following description of specific embodiments of the invention in conjunction with the accompanying figures.
Brief Description of the Drawings These and other features, aspects, and advantages of the present invention will become better understood with regard to the following description, appended claims, and accompanying drawings where:
Figure 1 is an overview of an audio/video-over-IP method in accordance with the present invention;
Figure 2 is an overview of an audio/video-over-IP apparatus in accordance with the present invention;
Figure 3 illustrates audio/video capture and compression schematics (units 100 -170);
Figure 4 illustrates a schematic drawing of an embodiment of the present invention;
Figure 5 illustrates a signal transmission routine (units 300 - 570); and Figure 6 illustrates a signal receiving and output schematics (units 600 - 770).
Detailed Description of the Presently Preferred Embodiment The present invention is directed to an audio/video-over-IP method, system and apparatus. As illustrated in Figure 1, the method includes the steps of receiving and/or transmitting an audio and/or video signal 102, and automatically switching communication protocols to ensure signal viability 104. In an embodiment of the present invention, the method further includes the step of automatically applying at least one signal maintenance technique to maintain signal viability/quality.
As illustrated in Figure 2, the apparatus includes means for receiving and/or transmitting an audio and/or video signal 12, and means for automatically switching communication protocols to ensure signal viability 14.
The invention optimizes connection between audio/video source and receivers) by means of converting original audio/video signal into a digital compressed audio/video data stream and transmitting this data stream from source location to receivers) locations) using an IP network, such as the Internet, to achieve a quality of audio/video reception that is close to or equal to that of a satellite link.
The invention captures an analog audio/video signal from the source, converts the signal into a digital format and transmitting that signal over an IP
(Internet Protocol) network to a receiver. Audio/video compression techniques are used to reduce the storage space and bandwidth required to operate with digital audio/video data. The optimization and switching of transmission protocols ensures that the connection between a client and an audio/video source will stay on as long as required, and provide the necessary connection quality sufficient for an acceptable level of audio/video reception.
An encoding apparatus to capture of analog audio/video signal and scramble it into a coded bit stream. The apparatus performs the following procedures upon software execution. It authenticates a client by means of a username and password, checks for a valid IP address and the unique MAC
(Media Access Control) address of the client, then a load balancing procedure provides continuous uninterrupted audio/video streaming at or above the average signal quality, withstanding network lags, increased amount of hops (connecting IP network routers between server and client). In instances when a default server detects a lag (delay) between itself and a client, the database is prompted to perform a mirror search for the closest mirror server. This is provided by instructing server computers that are listed as mirrors to send an echo signal (PING) to a client, calculating the lowest echo response time, and based on the lowest calculated response time, redirecting a client to request an audio/video data stream from the closest and/or fastest mirror server.
A protocol switching procedure is executed upon the server and client establishing a successful connection. The server is instructed to send a specific data packet that will predetermine if a multicast (UDP) packet reached the clients machine and inferring that a multicast UDP protocol can be used.
In the event that no UDP packet reached the client computer, an appropriate response protocol is instructed to initialize a Unicast (UDP) method of a packet delivery. In the unlikely event of a UDP "lock" by a firewall, proxy server or other network device, when the client cannot receive any UDP
packets, the server will switch to the less efficient but more stable HTTP
protocol.
If a network slowdown or quality degradation is observed, the server can be instructed to perform various efficiency techniques. These techniques can include resizing the video to a smaller format, reducing the frame rate to sustain video quality, decreasing the audio sampling rate, and increasing buffering times, therefore managing the client's connection bandwidth, all geared towards optimizing and increasing the chances of non-stop 5 audio/video reception by the client.
Video format detection is performed upon the completion of the aforementioned steps. The video data stream is then passed to an encoding engine, which analyzes and encodes the video in its original format, such as PAL SECAM or NTSC.
According to another aspect of the present invention, there is provided a decoding apparatus, configured to only listen for incoming data streams by means of a network interface card (NIC). The decoding device is designed to receive scrambled audio/video bit streams, and can convert a stream into a viewable audio/video signal in addition to the option of storing it in digital format on a storage device, and/or storing it in analog device, such as a VCR
type device.
Figure 3 illustrates an analog or digital audio/video input device [100]
that provides encoding engine with a raw digital audio/video data. Data can be taken from a local storage device, or be streamed live into the capture device, from camera, microphone, a satellite live feed, and the like. Video source can be in any format such as NTSC, PAL, or SECAM, and any resolution. Inputs supported are Composite Video, S-Video, RGB/SCART. A method is to fully decode the incoming video signal into separate components (RGB or YUV), mix this with the scan converter components, and re-encode back to video.
The video signal is never decoded, so it remains at a very high bandwidth, particularly true when using a composite video input. Signal delays (from video input to the encoding engine [110] are minimized to approximately 20ns.
Sync pulse width and sub carrier frequency of the video input remain unchanged. Video input sync & sub carrier SC/H timings are unaffected. In brief, this input device provides encoding engine with a digital video stream.
Figure 3 illustrates audio/video capture and compression schematics (units 100 -170);
Figure 4 illustrates a schematic drawing of an embodiment of the present invention;
Figure 5 illustrates a signal transmission routine (units 300 - 570); and Figure 6 illustrates a signal receiving and output schematics (units 600 - 770).
Detailed Description of the Presently Preferred Embodiment The present invention is directed to an audio/video-over-IP method, system and apparatus. As illustrated in Figure 1, the method includes the steps of receiving and/or transmitting an audio and/or video signal 102, and automatically switching communication protocols to ensure signal viability 104. In an embodiment of the present invention, the method further includes the step of automatically applying at least one signal maintenance technique to maintain signal viability/quality.
As illustrated in Figure 2, the apparatus includes means for receiving and/or transmitting an audio and/or video signal 12, and means for automatically switching communication protocols to ensure signal viability 14.
The invention optimizes connection between audio/video source and receivers) by means of converting original audio/video signal into a digital compressed audio/video data stream and transmitting this data stream from source location to receivers) locations) using an IP network, such as the Internet, to achieve a quality of audio/video reception that is close to or equal to that of a satellite link.
The invention captures an analog audio/video signal from the source, converts the signal into a digital format and transmitting that signal over an IP
(Internet Protocol) network to a receiver. Audio/video compression techniques are used to reduce the storage space and bandwidth required to operate with digital audio/video data. The optimization and switching of transmission protocols ensures that the connection between a client and an audio/video source will stay on as long as required, and provide the necessary connection quality sufficient for an acceptable level of audio/video reception.
An encoding apparatus to capture of analog audio/video signal and scramble it into a coded bit stream. The apparatus performs the following procedures upon software execution. It authenticates a client by means of a username and password, checks for a valid IP address and the unique MAC
(Media Access Control) address of the client, then a load balancing procedure provides continuous uninterrupted audio/video streaming at or above the average signal quality, withstanding network lags, increased amount of hops (connecting IP network routers between server and client). In instances when a default server detects a lag (delay) between itself and a client, the database is prompted to perform a mirror search for the closest mirror server. This is provided by instructing server computers that are listed as mirrors to send an echo signal (PING) to a client, calculating the lowest echo response time, and based on the lowest calculated response time, redirecting a client to request an audio/video data stream from the closest and/or fastest mirror server.
A protocol switching procedure is executed upon the server and client establishing a successful connection. The server is instructed to send a specific data packet that will predetermine if a multicast (UDP) packet reached the clients machine and inferring that a multicast UDP protocol can be used.
In the event that no UDP packet reached the client computer, an appropriate response protocol is instructed to initialize a Unicast (UDP) method of a packet delivery. In the unlikely event of a UDP "lock" by a firewall, proxy server or other network device, when the client cannot receive any UDP
packets, the server will switch to the less efficient but more stable HTTP
protocol.
If a network slowdown or quality degradation is observed, the server can be instructed to perform various efficiency techniques. These techniques can include resizing the video to a smaller format, reducing the frame rate to sustain video quality, decreasing the audio sampling rate, and increasing buffering times, therefore managing the client's connection bandwidth, all geared towards optimizing and increasing the chances of non-stop 5 audio/video reception by the client.
Video format detection is performed upon the completion of the aforementioned steps. The video data stream is then passed to an encoding engine, which analyzes and encodes the video in its original format, such as PAL SECAM or NTSC.
According to another aspect of the present invention, there is provided a decoding apparatus, configured to only listen for incoming data streams by means of a network interface card (NIC). The decoding device is designed to receive scrambled audio/video bit streams, and can convert a stream into a viewable audio/video signal in addition to the option of storing it in digital format on a storage device, and/or storing it in analog device, such as a VCR
type device.
Figure 3 illustrates an analog or digital audio/video input device [100]
that provides encoding engine with a raw digital audio/video data. Data can be taken from a local storage device, or be streamed live into the capture device, from camera, microphone, a satellite live feed, and the like. Video source can be in any format such as NTSC, PAL, or SECAM, and any resolution. Inputs supported are Composite Video, S-Video, RGB/SCART. A method is to fully decode the incoming video signal into separate components (RGB or YUV), mix this with the scan converter components, and re-encode back to video.
The video signal is never decoded, so it remains at a very high bandwidth, particularly true when using a composite video input. Signal delays (from video input to the encoding engine [110] are minimized to approximately 20ns.
Sync pulse width and sub carrier frequency of the video input remain unchanged. Video input sync & sub carrier SC/H timings are unaffected. In brief, this input device provides encoding engine with a digital video stream.
Audio source can be in any format including digital WAV (CD) format, 5.1 Dolby (6-channel audio) or a regular tape recorder or a microphone.
[110J Audio/Video compression encoding engine. A software-based audio/video compression processes digital audio/video stream with a audio/video compression codec, so that the quality of audio/video stream can be preserved without significant degradation, and a reduced size of the binary audio/video stream that can be decompressed and converted to audio/video output by a decoder device employing the same compression codec as in encoder engine. The audio/video compression codec ratio can vary greatly from virtually loss-less, less than 1 % video quality loss, to a low bandwidth digital audio/video stream with a much higher loss of video and audio quality.
[115] Optional storage of compressed and uncompressed digital audio/video signal for later transmission, compression, playback or editing is performed automatically by a user pre-selecting a storage option on the encoder. Digitized and/or compressed audio/video stream is then sent to the storage device for archiving. Archived files can be edited, played back, compressed, and/or transmitted unchanged at a later time.
The transmission and broadcast engine [120]. A device comprised of a network interface card (NIC) and software network engine in charge of distribution of the digital audio/video stream over IP networks such as the Internet. The engine can distribute an audio/video signal as a point-to-point connection and broadcast like, point-to-multipoint distribution using UDP
(User Datagram Protocol) and/or HTTP protocols. The transmission and broadcast engine are able to multicast a single audio/video stream so that multiple users can receive it simultaneously. This procedure only functions where permitted by network operators, since many networks lock out the Multicast UDP protocol to avoid unnecessary network traffic.
/P network (1301. This is a network "cloud" that consists of 2 or more computers. This can be the Internet with millions of clients or merely a local area network (LAN) with just a few computers. This network should operate using the IP (Internet Protocol) format, and preferably support UDP multicast for a more efficient broadcast distribution.
A network receiver f 1401. The network receiver comprises a network interface card and software drivers that are able to communicate using IP
protocols with an encoding device and it's transmission and broadcast engine.
This device should have a connection to the same IP network as the encoder device, and be capable of receiving a digital audio/video stream in the same format as it is was sent without dropping packets, and maintaining the same rate of reception that was set by the encoder device. The received compressed digital audio/video stream is then sent to a decoder engine, and/or optionally to a storage device for archiving.
A decoder engine f 1501 is implemented as a software algorithm that is able to decode compressed digital audio/video stream received by the network receiver from an encoder by means of an IP network, or from a storage device. The compressed audio/video stream is converted into an uncompressed audio/video signal that can be used by an output device.
[160] A storage medium that provides space for the archiving of compressed digital audio/ video data streams. Stored data can be retrieved and sent to the decoder engine for a decompression procedure to convert the audio/video stream into an uncompressed playable format, or be used by editing software to perform desired editing procedures.
[170] Audio/video output device converts uncompressed digital audio/video stream from the decoder engine into a playable analog audio/video format that will be sent to a playback device such as monitor or speakers to further monitor the output.
Figure 4 illustrates [300] Initialization of algorithm functions and routines. [310] Program intertace loading, Graphic User Interface including login and password input fields. This is in form of a web page that can be viewed by any computer with Internet browser capabilities and an IP network connection.
[320] Request for login and password, as well as probing of an IP
address and sending this information to an encoding unit that checks the validity of a connected user.
[330] Authentication procedure detects if the username and password are authenticated. If authentication fails, then an error message is displayed and a log entry added with a timestamp and the IP information of a possible unauthorized user [360].
[340] An IP Check procedure verifies an authorized decoder device by comparing the client's MAC (Media Access Control) address and subnet mask with an existing record in an access database. MAC address is a unique identifier that is proprietary to every network device and no two are the same.
This function is particularly important for ensuring a secure connection in point-to-point sessions where audio/video information is only intended for one specific subscriber.
[350] By compiling results from authentication and IP check procedures, the program has two options: grant access to a user or redirect to an error message and a log entry creation procedure [330] if authentication fails.
[355] An error message function displays an event to a user depending on the result received from the authentication procedure or an IP check, and logs the client's IP address and timestamps it.
[360] If authorization is granted, the decoder is checked for a '1 on 1' priority tag meaning it will receive the audio/video data stream alone and no other device will be authorized to view the same data channel, unless it has the same priority tag. The tag can be configured to be granted only to mirror servers and/or a connection that requires higher security and session stability.
This can be used to increase security for copyrighted material, or sensitive audio/video transmissions such as audio\video conferences, as well as to increase the quality of the connection.
[365] If the connection is tagged as 1-1, then the source is set to "direct", meaning that the client will get the audio/video stream directly from the encoder. If there is no 1-1 tag present then the session will proceed to a mirror checking procedure, which will appoint an appropriate source for the audio/video data stream.
[370] Mirror Check procedure. A mirror lookup routine is initiated after a user has been checked for a 1-1 priority tag, and barring any such detection, can then access mirror servers if any are present.
[375] If no mirror server computers are present, the client will attempt to connect to the encoder server directly. However, it can only be available if no 1-1 connections are established between the encoder and other clients.
[380 -385] Check if 1-1 connection has been established with a server, if it is true that it is not a '1-1' session, it will be refused a connection and will proceed to an error message and termination signal, since all the resources and bandwidth available to the encoding server should be allocated to 1-1 sessions. This in turn indicates that it is a closed session, or mirror servers have established connections with the encoder and will provide other users with retransmissions of the audio/video stream.
[390] If mirror servers) are present then redirect the session to °Find Fastest mirror" [390], if not then redirect to "Direct" source [400]. Users iP
address is determined, after which a host lookup is performed. From the acquired client's host information, perform a WHOIS function to an Internic database to determine the registration country of a primary host name, and an entry is added to a log file. A database is then contacted to lookup a mirror site in the given country or region to ensure a faster connection and better network quality. If no results are found using the WHOIS function, redirect the user to the fastest mirror server as determined by a PING command, or a default streaming server.
[400] Set server source to "Direct" if bandwidth and CPU load permit, 5 and no next fastest mirror can be set as source.
[420] Protocol Check is initiated after a successful username/password and IP authentication. Data packets are sent in Multicast and Unicast (UDP) protocols, if the response is received within a permitted time frame, Multicast 10 or Unicast is adopted as the default streaming protocol for the given user/group. If the UDP protocol is locked for the user, then the HTTP protocol will be used as the transport protocol for the audio/video stream.
[430] Check bandwidth procedure. A bandwidth check is initiated after the Protocol Check has been accomplished by sending data packets at different buffer size values and determining the mean value of the response time. If the result confirms an appropriate time for network performance, the user is passed directly to a Mirror lookup function with default buffer time settings. If degraded network performance has been detected reapply a Bandwidth check function with lower buffer values. After the results have been analyzed, techniques can be applied such as setting the buffering time to a value higher than the default value, decreasing the audio sampling rate, decreasing the bandwidth settings, resizing the video, and/or reducing the frame rate; all to achieve sustainable audio/video quality with lower bandwidth settings.
[440] Adjust buffering, resize video and adjust audio bit rate to match the available bandwidth.
[450] Display a warning if the bandwidth is lower than is necessary for high quality audio/video reception, such as "The bandwidth speed is Low".
[460] Increase the buffering value, resize the video, reduce the frame rate, and change the sampling rate of the audio stream.
[470] Initialize video streaming procedure.
[480] Receive a termination signal after the streaming session is complete or interrupted.
[490] When fastest mirror server is found, the decoder is redirected to the fastest "Mirror" source to receive an audio/video stream from that server.
[500] A protocol check is initiated after successful username/password and IP authentications. Data packets are sent in Multicast and Unicast protocols, if the response is received within a permitted time frame, multicast or unicast is adopted as the default streaming protocol for the given user/group.
[510] Check bandwidth procedure. A Bandwidth Check is initiated after the Protocol Check has been accomplished by sending data packets at different buffer size values and determining the mean value of the response time. If the result confirms an appropriate time for network performance, the user is passed directly to a Mirror lookup function with default buffer time settings. If degraded network performance has been detected, reapply a Bandwidth check function with lower buffer values. After the results have been analyzed, set the buffering time to a value higher than the default value, decrease the bandwidth settings, resize the video, reduce the frame rate; all geared towards achieving better video quality with lower bandwidth settings.
[520] Display a warning if the bandwidth is lower than is necessary for high quality audio video reception, such as "The bandwidth speed is Low".
[530] Increase the buffering value, resize the video and reduce the frame rate.
[540] Set the buffering and resize video to match the available bandwidth.
[550] Initialize video streaming procedure.
[560] Receive termination signal after the streaming is complete or interrupted.
[570] End of program.
Figure 5 Illustrates [600] Initialization of algorithm functions and routines. [610] Program intertace loading, Graphic User Interface displaying login and password input fields.
[620] Ask user for Server IP address, username and password [630] Establish connection to server using a TCP/IP connection [640] Check authorization of username and password, IP and MAC
addresses [650] Error function, gives an error message if the client was unable to establish a successful connection to the server if username and password did not match (660] Establish a session with the Server with given credentials [670] Send an Echo signal, pertorm a handshake, and exchange headers to validate all connection properties and settings [680] Open a listening port on the Client side for incoming video bit streams.
[690] Start receiving streaming packets.
[700] Prompt user to select an output type, such as video out and/or storage.
[710] Storage selected instead of direct to video, therefore monitor output [720] Analyze video [730] Check for the existence of a video output port, such as Composite or S-Video. The lower the graphics resolution and refresh rate, the better the image quality. All scan converters store the computer image to be converted to video in their own internal memory, and in order to do so the computer image has to be "sampled" multiple times during each scan line.
Each sample stores one pixel of information in memory. The number of samples taken is proportional to the image quality, such as the more samples the better. Higher graphics resolutions take less time to display each scan-line than lower ones do, therefore there will be more samples per line for lower resolution modes, since there is more time for more samples to be taken, and hence this will provide a better image quality.
[740] Error message "No video output port detected", and video signal is directed to default VGA port.
[750] Initialize and select video output port (S-VHS, Analog composite or SCART). Absolute maximum resolution 1600x1200, Maximum 1024x768 with no line dropping in NTSC, 1280x1024 in PAL. 24 bit compatible - 23 bits stored. 24 kHz to 100 kHz horizontal scan rate. Virtually any vertical scan rate accepted, therefore the horizontal scan rate is more important. Separate TTL-level HSync & VSync positive or negative are performed.
[760] Selecting output format. The lower the graphics resolution, the better the 'vertical' image quality. Video monitors have a fixed number of lines available for displaying pictures; for PAL it is 576 and for NTSC 480.
Although some of these are typically off the top and bottom edges of the screen.
Therefore, the more scan-lines a graphics resolution has, such as an 800x600 resolution has 600 scan-lines), the more difficult is it to squeeze all these lines into the limited number available on a TV monitor. Thus, lowering the graphics resolution helps to improve image quality. Software resizes the video signal accordingly to fit all lines with the aspect ratio of the original image preserved.
[770] End.
The invention creates a video/audio link that enables a video signal of a broadcast quality NTSC 525 Lines/60Hz, PAL 625 lines/50Hz be sent and received over any IP network with bandwidth not exceeding 1 Mbps for the above resolutions. The system acts as a standard converter that encodes a video signal in any format be it NTSC, SECAM, or PAL, and decodes it in any desired format NTSC or SECAM in real time. The software provides uninterrupted video broadcasting by using mirror technology, which finds the optimal connection between client and server such as by maintaining fewer hops and avoiding congested zones. By employing a protocol switching technology, the invention enables a video/audio signal to penetrate any virtually any IP network to deliver a stable audio/video stream between client and server. For example, if a multicast protocol is blocked by a router or firewall settings, the software will switch to a slightly less efficient, yet more circumstance-appropriate unicast protocol.
As well, if the connection between "encoder" and "decoder" is found to be slower than between "mirror" and "decoder" it switches to the optimal streaming server for more reliable stream acquisition.
Video transmission method and apparatus, captures audio/video signals) from an analog source, digitizes, compresses and transmits digitized video and audio signals over IP networks. It is designed to provide video and audio links) for Point-To-Point or Point-To-Multipoint transmissions. It ensures accurate transmission of TV quality video and CD quality audio signals over IP networks.
In embodiments of the present invention, uses include content delivery networks; telecommunications; live event streaming; corporate meetings;
distance education; and telemedicine.
5 The invention requires a mere 1 Mbps while others need to have at least 3 - 45 Mbps for the same signal quality. The invention provides standards converting on the input and output together in the same device.
Automated standards conversion, automated protocol switching, and mirror checking result in low bandwidth requirements for high quality video. Both the 10 encoder and decoder are assembled using relatively inexpensive "off the shelf" components to provide a high quality video transmission device compatible within any IP network, in addition to virtually all audio/video standards.
15 Although the present invention has been described in considerable detail with reference to certain preferred embodiments thereof, other versions are possible. Therefore, the spirit and scope of the appended claims should not be limited to the description of the preferred embodiments contained herein.
[110J Audio/Video compression encoding engine. A software-based audio/video compression processes digital audio/video stream with a audio/video compression codec, so that the quality of audio/video stream can be preserved without significant degradation, and a reduced size of the binary audio/video stream that can be decompressed and converted to audio/video output by a decoder device employing the same compression codec as in encoder engine. The audio/video compression codec ratio can vary greatly from virtually loss-less, less than 1 % video quality loss, to a low bandwidth digital audio/video stream with a much higher loss of video and audio quality.
[115] Optional storage of compressed and uncompressed digital audio/video signal for later transmission, compression, playback or editing is performed automatically by a user pre-selecting a storage option on the encoder. Digitized and/or compressed audio/video stream is then sent to the storage device for archiving. Archived files can be edited, played back, compressed, and/or transmitted unchanged at a later time.
The transmission and broadcast engine [120]. A device comprised of a network interface card (NIC) and software network engine in charge of distribution of the digital audio/video stream over IP networks such as the Internet. The engine can distribute an audio/video signal as a point-to-point connection and broadcast like, point-to-multipoint distribution using UDP
(User Datagram Protocol) and/or HTTP protocols. The transmission and broadcast engine are able to multicast a single audio/video stream so that multiple users can receive it simultaneously. This procedure only functions where permitted by network operators, since many networks lock out the Multicast UDP protocol to avoid unnecessary network traffic.
/P network (1301. This is a network "cloud" that consists of 2 or more computers. This can be the Internet with millions of clients or merely a local area network (LAN) with just a few computers. This network should operate using the IP (Internet Protocol) format, and preferably support UDP multicast for a more efficient broadcast distribution.
A network receiver f 1401. The network receiver comprises a network interface card and software drivers that are able to communicate using IP
protocols with an encoding device and it's transmission and broadcast engine.
This device should have a connection to the same IP network as the encoder device, and be capable of receiving a digital audio/video stream in the same format as it is was sent without dropping packets, and maintaining the same rate of reception that was set by the encoder device. The received compressed digital audio/video stream is then sent to a decoder engine, and/or optionally to a storage device for archiving.
A decoder engine f 1501 is implemented as a software algorithm that is able to decode compressed digital audio/video stream received by the network receiver from an encoder by means of an IP network, or from a storage device. The compressed audio/video stream is converted into an uncompressed audio/video signal that can be used by an output device.
[160] A storage medium that provides space for the archiving of compressed digital audio/ video data streams. Stored data can be retrieved and sent to the decoder engine for a decompression procedure to convert the audio/video stream into an uncompressed playable format, or be used by editing software to perform desired editing procedures.
[170] Audio/video output device converts uncompressed digital audio/video stream from the decoder engine into a playable analog audio/video format that will be sent to a playback device such as monitor or speakers to further monitor the output.
Figure 4 illustrates [300] Initialization of algorithm functions and routines. [310] Program intertace loading, Graphic User Interface including login and password input fields. This is in form of a web page that can be viewed by any computer with Internet browser capabilities and an IP network connection.
[320] Request for login and password, as well as probing of an IP
address and sending this information to an encoding unit that checks the validity of a connected user.
[330] Authentication procedure detects if the username and password are authenticated. If authentication fails, then an error message is displayed and a log entry added with a timestamp and the IP information of a possible unauthorized user [360].
[340] An IP Check procedure verifies an authorized decoder device by comparing the client's MAC (Media Access Control) address and subnet mask with an existing record in an access database. MAC address is a unique identifier that is proprietary to every network device and no two are the same.
This function is particularly important for ensuring a secure connection in point-to-point sessions where audio/video information is only intended for one specific subscriber.
[350] By compiling results from authentication and IP check procedures, the program has two options: grant access to a user or redirect to an error message and a log entry creation procedure [330] if authentication fails.
[355] An error message function displays an event to a user depending on the result received from the authentication procedure or an IP check, and logs the client's IP address and timestamps it.
[360] If authorization is granted, the decoder is checked for a '1 on 1' priority tag meaning it will receive the audio/video data stream alone and no other device will be authorized to view the same data channel, unless it has the same priority tag. The tag can be configured to be granted only to mirror servers and/or a connection that requires higher security and session stability.
This can be used to increase security for copyrighted material, or sensitive audio/video transmissions such as audio\video conferences, as well as to increase the quality of the connection.
[365] If the connection is tagged as 1-1, then the source is set to "direct", meaning that the client will get the audio/video stream directly from the encoder. If there is no 1-1 tag present then the session will proceed to a mirror checking procedure, which will appoint an appropriate source for the audio/video data stream.
[370] Mirror Check procedure. A mirror lookup routine is initiated after a user has been checked for a 1-1 priority tag, and barring any such detection, can then access mirror servers if any are present.
[375] If no mirror server computers are present, the client will attempt to connect to the encoder server directly. However, it can only be available if no 1-1 connections are established between the encoder and other clients.
[380 -385] Check if 1-1 connection has been established with a server, if it is true that it is not a '1-1' session, it will be refused a connection and will proceed to an error message and termination signal, since all the resources and bandwidth available to the encoding server should be allocated to 1-1 sessions. This in turn indicates that it is a closed session, or mirror servers have established connections with the encoder and will provide other users with retransmissions of the audio/video stream.
[390] If mirror servers) are present then redirect the session to °Find Fastest mirror" [390], if not then redirect to "Direct" source [400]. Users iP
address is determined, after which a host lookup is performed. From the acquired client's host information, perform a WHOIS function to an Internic database to determine the registration country of a primary host name, and an entry is added to a log file. A database is then contacted to lookup a mirror site in the given country or region to ensure a faster connection and better network quality. If no results are found using the WHOIS function, redirect the user to the fastest mirror server as determined by a PING command, or a default streaming server.
[400] Set server source to "Direct" if bandwidth and CPU load permit, 5 and no next fastest mirror can be set as source.
[420] Protocol Check is initiated after a successful username/password and IP authentication. Data packets are sent in Multicast and Unicast (UDP) protocols, if the response is received within a permitted time frame, Multicast 10 or Unicast is adopted as the default streaming protocol for the given user/group. If the UDP protocol is locked for the user, then the HTTP protocol will be used as the transport protocol for the audio/video stream.
[430] Check bandwidth procedure. A bandwidth check is initiated after the Protocol Check has been accomplished by sending data packets at different buffer size values and determining the mean value of the response time. If the result confirms an appropriate time for network performance, the user is passed directly to a Mirror lookup function with default buffer time settings. If degraded network performance has been detected reapply a Bandwidth check function with lower buffer values. After the results have been analyzed, techniques can be applied such as setting the buffering time to a value higher than the default value, decreasing the audio sampling rate, decreasing the bandwidth settings, resizing the video, and/or reducing the frame rate; all to achieve sustainable audio/video quality with lower bandwidth settings.
[440] Adjust buffering, resize video and adjust audio bit rate to match the available bandwidth.
[450] Display a warning if the bandwidth is lower than is necessary for high quality audio/video reception, such as "The bandwidth speed is Low".
[460] Increase the buffering value, resize the video, reduce the frame rate, and change the sampling rate of the audio stream.
[470] Initialize video streaming procedure.
[480] Receive a termination signal after the streaming session is complete or interrupted.
[490] When fastest mirror server is found, the decoder is redirected to the fastest "Mirror" source to receive an audio/video stream from that server.
[500] A protocol check is initiated after successful username/password and IP authentications. Data packets are sent in Multicast and Unicast protocols, if the response is received within a permitted time frame, multicast or unicast is adopted as the default streaming protocol for the given user/group.
[510] Check bandwidth procedure. A Bandwidth Check is initiated after the Protocol Check has been accomplished by sending data packets at different buffer size values and determining the mean value of the response time. If the result confirms an appropriate time for network performance, the user is passed directly to a Mirror lookup function with default buffer time settings. If degraded network performance has been detected, reapply a Bandwidth check function with lower buffer values. After the results have been analyzed, set the buffering time to a value higher than the default value, decrease the bandwidth settings, resize the video, reduce the frame rate; all geared towards achieving better video quality with lower bandwidth settings.
[520] Display a warning if the bandwidth is lower than is necessary for high quality audio video reception, such as "The bandwidth speed is Low".
[530] Increase the buffering value, resize the video and reduce the frame rate.
[540] Set the buffering and resize video to match the available bandwidth.
[550] Initialize video streaming procedure.
[560] Receive termination signal after the streaming is complete or interrupted.
[570] End of program.
Figure 5 Illustrates [600] Initialization of algorithm functions and routines. [610] Program intertace loading, Graphic User Interface displaying login and password input fields.
[620] Ask user for Server IP address, username and password [630] Establish connection to server using a TCP/IP connection [640] Check authorization of username and password, IP and MAC
addresses [650] Error function, gives an error message if the client was unable to establish a successful connection to the server if username and password did not match (660] Establish a session with the Server with given credentials [670] Send an Echo signal, pertorm a handshake, and exchange headers to validate all connection properties and settings [680] Open a listening port on the Client side for incoming video bit streams.
[690] Start receiving streaming packets.
[700] Prompt user to select an output type, such as video out and/or storage.
[710] Storage selected instead of direct to video, therefore monitor output [720] Analyze video [730] Check for the existence of a video output port, such as Composite or S-Video. The lower the graphics resolution and refresh rate, the better the image quality. All scan converters store the computer image to be converted to video in their own internal memory, and in order to do so the computer image has to be "sampled" multiple times during each scan line.
Each sample stores one pixel of information in memory. The number of samples taken is proportional to the image quality, such as the more samples the better. Higher graphics resolutions take less time to display each scan-line than lower ones do, therefore there will be more samples per line for lower resolution modes, since there is more time for more samples to be taken, and hence this will provide a better image quality.
[740] Error message "No video output port detected", and video signal is directed to default VGA port.
[750] Initialize and select video output port (S-VHS, Analog composite or SCART). Absolute maximum resolution 1600x1200, Maximum 1024x768 with no line dropping in NTSC, 1280x1024 in PAL. 24 bit compatible - 23 bits stored. 24 kHz to 100 kHz horizontal scan rate. Virtually any vertical scan rate accepted, therefore the horizontal scan rate is more important. Separate TTL-level HSync & VSync positive or negative are performed.
[760] Selecting output format. The lower the graphics resolution, the better the 'vertical' image quality. Video monitors have a fixed number of lines available for displaying pictures; for PAL it is 576 and for NTSC 480.
Although some of these are typically off the top and bottom edges of the screen.
Therefore, the more scan-lines a graphics resolution has, such as an 800x600 resolution has 600 scan-lines), the more difficult is it to squeeze all these lines into the limited number available on a TV monitor. Thus, lowering the graphics resolution helps to improve image quality. Software resizes the video signal accordingly to fit all lines with the aspect ratio of the original image preserved.
[770] End.
The invention creates a video/audio link that enables a video signal of a broadcast quality NTSC 525 Lines/60Hz, PAL 625 lines/50Hz be sent and received over any IP network with bandwidth not exceeding 1 Mbps for the above resolutions. The system acts as a standard converter that encodes a video signal in any format be it NTSC, SECAM, or PAL, and decodes it in any desired format NTSC or SECAM in real time. The software provides uninterrupted video broadcasting by using mirror technology, which finds the optimal connection between client and server such as by maintaining fewer hops and avoiding congested zones. By employing a protocol switching technology, the invention enables a video/audio signal to penetrate any virtually any IP network to deliver a stable audio/video stream between client and server. For example, if a multicast protocol is blocked by a router or firewall settings, the software will switch to a slightly less efficient, yet more circumstance-appropriate unicast protocol.
As well, if the connection between "encoder" and "decoder" is found to be slower than between "mirror" and "decoder" it switches to the optimal streaming server for more reliable stream acquisition.
Video transmission method and apparatus, captures audio/video signals) from an analog source, digitizes, compresses and transmits digitized video and audio signals over IP networks. It is designed to provide video and audio links) for Point-To-Point or Point-To-Multipoint transmissions. It ensures accurate transmission of TV quality video and CD quality audio signals over IP networks.
In embodiments of the present invention, uses include content delivery networks; telecommunications; live event streaming; corporate meetings;
distance education; and telemedicine.
5 The invention requires a mere 1 Mbps while others need to have at least 3 - 45 Mbps for the same signal quality. The invention provides standards converting on the input and output together in the same device.
Automated standards conversion, automated protocol switching, and mirror checking result in low bandwidth requirements for high quality video. Both the 10 encoder and decoder are assembled using relatively inexpensive "off the shelf" components to provide a high quality video transmission device compatible within any IP network, in addition to virtually all audio/video standards.
15 Although the present invention has been described in considerable detail with reference to certain preferred embodiments thereof, other versions are possible. Therefore, the spirit and scope of the appended claims should not be limited to the description of the preferred embodiments contained herein.
Claims (4)
1. An audio/video-over-IP method comprising the steps of:
(i) receiving and/or transmitting an audio and/or video signal; and (ii) automatically switching communication protocols to ensure signal viability.
(i) receiving and/or transmitting an audio and/or video signal; and (ii) automatically switching communication protocols to ensure signal viability.
2. The method according to claim 1, further including the step of automatically applying at least one signal maintenance technique to maintain signal viability/quality.
3. An audio/video-over-IP apparatus comprising:
means for receiving and/or transmitting an audio and/or video signal;
and means for automatically switching communication protocols to ensure signal viability.
means for receiving and/or transmitting an audio and/or video signal;
and means for automatically switching communication protocols to ensure signal viability.
4. A storage medium readable by a computer encoding a computer process to provide an audio/video-over-IP method, the computer process comprising:
a processing portion for receiving and/or transmitting an audio andlor video signal; and a processing portion for automatically switching communication protocols to ensure signal viability.
a processing portion for receiving and/or transmitting an audio andlor video signal; and a processing portion for automatically switching communication protocols to ensure signal viability.
Priority Applications (3)
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|---|---|---|---|
| CA002410748A CA2410748A1 (en) | 2002-11-01 | 2002-11-01 | Audio-video-over-ip method, system and apparatus |
| PCT/CA2003/001685 WO2004040874A1 (en) | 2002-11-01 | 2003-11-03 | Apparatuses and method for audio/video streaming over ip |
| AU2003280255A AU2003280255A1 (en) | 2002-11-01 | 2003-11-03 | Apparatuses and method for audio/video streaming over ip |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CA002410748A CA2410748A1 (en) | 2002-11-01 | 2002-11-01 | Audio-video-over-ip method, system and apparatus |
Publications (1)
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| CA2410748A1 true CA2410748A1 (en) | 2004-05-01 |
Family
ID=32181913
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| CA002410748A Abandoned CA2410748A1 (en) | 2002-11-01 | 2002-11-01 | Audio-video-over-ip method, system and apparatus |
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| AU (1) | AU2003280255A1 (en) |
| CA (1) | CA2410748A1 (en) |
| WO (1) | WO2004040874A1 (en) |
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|---|---|---|---|---|
| CN100492439C (en) * | 2006-01-19 | 2009-05-27 | 山东大学 | TV experimental remote teaching system based on wideband interconnection network |
| US7844723B2 (en) | 2007-02-13 | 2010-11-30 | Microsoft Corporation | Live content streaming using file-centric media protocols |
| CN102170562A (en) * | 2011-03-04 | 2011-08-31 | 西安电子科技大学 | Network video service device |
| EP3032826A4 (en) | 2013-08-06 | 2016-08-31 | Ricoh Co Ltd | Information processing device, and determination result provision method |
| CN113596567A (en) * | 2021-06-10 | 2021-11-02 | 保升(中国)科技实业有限公司 | Video networking technology |
| CN113938155B (en) * | 2021-09-24 | 2023-05-05 | 广州市迪士普音响科技有限公司 | Wireless conference audio transmission method, system, device and medium |
| CN114245170B (en) * | 2022-02-24 | 2022-09-13 | 国能信息技术有限公司 | Audio and video unidirectional transmission scheduling method and system based on shunting transmission |
Family Cites Families (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| AU2001236570A1 (en) * | 2000-01-28 | 2001-08-07 | Ibeam Broadcasting Corporation | Method and apparatus for encoder-based distribution of live video and other streaming content |
| US20020089973A1 (en) * | 2000-11-17 | 2002-07-11 | Yehuda Manor | System and method for integrating voice, video, and data |
| US20020071052A1 (en) * | 2000-12-07 | 2002-06-13 | Tomoaki Itoh | Transmission rate control method |
| US20020099858A1 (en) * | 2001-08-06 | 2002-07-25 | Muse Corporation | Network communications protocol |
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2002
- 2002-11-01 CA CA002410748A patent/CA2410748A1/en not_active Abandoned
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2003
- 2003-11-03 WO PCT/CA2003/001685 patent/WO2004040874A1/en not_active Application Discontinuation
- 2003-11-03 AU AU2003280255A patent/AU2003280255A1/en not_active Abandoned
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| WO2004040874A1 (en) | 2004-05-13 |
| AU2003280255A1 (en) | 2004-05-25 |
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