This invention relates to medical diagnostic ultrasound systems and, in particular, to ultrasonic diagnostic imaging systems capable of multiplexing voice and image information over a common data network.
At many medical facilities it is common practice for patients to be scanned in an ultrasound exam by a sonographer and for the images to be read for diagnosis by a radiologist or echocardiographer in a separate reading room or at a remote workstation. In such a setting the physician reading the images can make diagnoses of multiple patients being scanned at the same time through the networking of the ultrasound systems used for the examinations with the reading workstation. When the diagnoses are being made while the patient is in the scanning room, a physician may frequently learn that additional images or different views would be helpful or necessary for a reliable diagnosis. At those times the diagnosing physician will want the additional scanning to be performed while the patient is still available in the medical facility. The conventional way this is done is for the physician to leave the reading room and go to the scanning room to try to intercept the patient and the sonographer before the patient has departed. Alternatively, the physician may try to do this by telephoning the sonographer in the scanning room. It would be desirable to be able to contact the sonographer more quickly and easily from the reading room.
US patent application publication no. 2003/0083563 (Katsman et al.) provides one solution to this situation, which is to enable the sonographer and the physician to communicate with each other through the ultrasound system. The ultrasound system and the reading workstation are both equipped with a microphone, loudspeaker, and a speech recognition and processing system. When a person speaks into the microphone the speech is converted into digital speech data and compressed. The compressed speech data is transmitted over the network connecting the two devices to the terminal. The receiving terminal decompresses the data, the speech recognition and processing system processes the digital speech data and transmits it to the loudspeaker. By this means the sonographer and the reading physician can speak to each other and the physician can give instructions to the sonographer during the ultrasound exam. However the manner in which the image and voice data share the network connection is not explained. It would be desirable to multiplex the voice and image communications so that the voice and image data would automatically share the network connection whenever a speaker decides to speak. It is further desirable to be able to extend the ability to engage in such voice and image communication to communicating with other people not on the medical facility's network.
In accordance with the principles of the present invention, a diagnostic ultrasound system and remote terminal are described which are able to exchange voice communication through packets of voice data using a TCP/IP Internet protocol. When image communication between the same two devices also uses a TCP/IP protocol, the image and voice data packets can both share the same data network, with the header information of the packets providing the correct and accurate routing of the respective data packets. The packetized voice transmissions can be routed to others outside the local area network over external carrier system such as public telephone networks. An embodiment of the present invention can thus also be used to communicate with people outside of the medical facility. A real-time protocol can be used to ensure that transmitted voice packets are received in a timely way so as to be reproduced as normal, uninterrupted speech.
In the drawings:
Referring first to
In accordance with the principles of the present invention each of the ultrasound systems, the workstation and the office PC are capable of providing voice communication between operators of the devices over the same packet switching data network 300. An embodiment of an ultrasound system with these capabilities is shown in
From an overall viewpoint, the operator's voice is digitized by the sound card into bytes of data. A nominal voice bandwidth is 4 kHz, which means that a sampling bandwidth of 8 kHz would be sufficient to digitize the typical voice frequencies. Most sound cards are capable of digitizing analog signals at a much higher rate, usually on the order of 44 kHz sampling to produce 16-bit bytes. Since the voice bandwidth does not require this high a digitization rate, a number of successive bytes can be aggregated and sent as the payload of an IP packet. In addition, the digitized voice data may be compressed before transmission using a compression protocol such as MP-MLQ or ACELP, Standard ITU-T G.723.1. The packetized voice data is then sent from the host ultrasound system over the network. This may be done directly from one endpoint to another, e.g., from the ultrasound system directly to the workstation, but generally the packet traffic is mediated by a gatekeeper such as a router which manages data traffic by performing duties such as translating IP addresses of the endpoint devices, granting or denying access, call signaling to connect the call, call authorization, bandwidth management and call management. The voice packets may be directed by multiple gatekeepers before reaching the destination device. At the receiving device the packet data is unpacked in accordance with instructions provided by the packet protocols and reassembled to its original state. The bytes of data are converted back to analog signals by a D/A converter in the sound card at the receiving endpoint and played as a voice through the loudspeaker at the receiving end.
The protocol stack 46 shown is typical for the H.323 standard for voice communication over a TCP/IP network. Other protocols such as SIP (Session Initiation Protocol) may alternatively be used. At the bottom of the stack is the physical layer which performs connection services and signal conversion for the data link layer above. The data link layer in this embodiment is an Ethernet protocol layer. The network layer is the IP protocol so that the voice packets can share the communication medium with other IP service packets including image communication between the ultrasound system and the workstation. At the next layer it is seen that the audio and registration packets use the User Datagram Protocol (UDP) while the control and signaling packets use the Transmission Control Protocol (TCP) as the transport protocol. Both the source and receiver endpoints support the H.245 and Q.931 protocols. H.245 allows usage of channels and Q.931 is needed for call signaling and setting up the call. In the illustrated stack H.225.0/Q.931 Call Signaling is used to provide the signaling for call control. For the received voice to sound natural and not broken up, it is important for the voice data to arrive at the destination substantially in real time. This is accomplished by the use of RTP, the real time transport protocol that carries the voice packets. When the call is made through a gatekeeper (e.g., a router) rather than directly from endpoint to endpoint as is possible in a single LAN (Local Area Network) with direct endpoint call signaling between the two transport addresses, the H.225 RAS (Registration, Admission, Status) channel is used to communicate between endpoints and the gatekeeper. The RAS channel performs procedures such as determining a gatekeeper with which it should register, endpoint registration of the packet's transport and alias (alternate) addresses, endpoint location, and admission, status, and disengage messages. The procedure to set up a call involves discovering a gatekeeper with which the endpoint can register; registration with the gatekeeper; entering the call setup phase; capability exchange between the endpoint and the gatekeeper; and establishing the call. In this example the voice packet is sent by way of the Ethernet connection 36, although communication may also be delivered and received by other ports such as a modem 32 or a serial port 31.
By use of this protocol stack a voice packet is passed from the source terminal, the ultrasound system in this instance, to a series of one or more gatekeepers (routers) until finally arriving at the destination terminal, the workstation in this example. At the workstation the various header layers are examined and stripped off until the voice data is delivered to the sound card, where it is converted to an analog signal and played through the loudspeaker 28 at the workstation. A codec may be used to decompress data that was compressed at the source. The workstation has the same communication hardware, software and protocol stack as does the ultrasound system so that the physician at the workstation can communicate by voice back to the ultrasound system operator.
In a constructed embodiment the operating system 34 will generally run user interface software to permit the ultrasound system or workstation operator to easily access the voice communication capability. For calling out, such software will display a selection of IP addresses or other alias addresses such as telephone numbers from which the operator can choose to initiate a call. When an incoming call is received, the software will make an audible sound through the loudspeaker 28 and/or display an incoming call icon on the display screen. The operator will touch a key on the control panel 115 or on the display screen to answer the call.
An embodiment of the present invention need not be constrained to calling only those connected to the LAN of the medical facility. The same voice packets can be transmitted by a gateway 250 which is connected to the Internet or a public switched telephone network as illustrated in
Filing Document | Filing Date | Country | Kind | 371c Date |
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PCT/IB06/51476 | 5/10/2006 | WO | 00 | 11/13/2007 |
Number | Date | Country | |
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60683508 | May 2005 | US |