Patents
Literature
Hiro is an intelligent assistant for R&D personnel, combined with Patent DNA, to facilitate innovative research.
Hiro

501 results about "Real-time Transport Protocol" patented technology

The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.

Push-to-talk wireless telecommunications system utilizing a voice-over-IP network

A method and system to provide push-to-talk from one user to another in a wireless packet data telecommunications network is described. The system may include: a wireless communication network including push-to-talk (PTT) functionality, with a Session Initiation Protocol (SIP) Proxy Server; a SIP Registrar and Location Server operable to store contact addresses of active mobile devices; a Realtime Transport Protocol (RTP) Media Gateway (PTT Server) operable to function as a call endpoint for each of a plurality of mobile devices wherein the plurality of mobile devices are segmented into membership groups, the PTT Server further operable to multicast a communication from one member of the group to the other members of the group; and an Internet Protocol (IP) network interconnecting the SIP Proxy server, the SIP Registrar and Location Server, and the PTT Server.
Owner:APPLE INC

System and method for assisting in controlling real-time transport protocol flow through multiple networks via use of a cluster of session routers

A system for assisting in controlling real-time transport protocol flow through multiple networks via use of a cluster of session routers is disclosed. The system utilizes a first computer and a cluster of computers, wherein the cluster of computers comprises at least two associated computers that are connected to the first computer. Each of the associated computers comprises a second transceiver, a second memory having logic stored therein defining functions to be performed by the associated computers, and a second processor. The second processor is configured by the second memory to perform the functions of: performing an inbound screen on route information received by an associated computer, from the first computer, to determine if the received route information should be discarded; if the route information is not discarded, comparing the received and screened route information to a local policy defined within the cluster of computers; and performing an outbound screen on the received and screened information prior to transmitting the received and screened information outside the cluster of computers.
Owner:PRIMARY NETWORKS +1

System and method for assisting in controlling real-time transport protocol flow through multiple networks via media flow routing

A system and method for assisting in controlling real-time transport protocol flow through multiple networks via media flow routing is disclosed. The system utilizes a first computer and a second computer connected to the first computer, via a group of associated computers, wherein each of the first computer, second computer, and group of associated computers comprise a transceiver, a memory, and a processor. The processor is configured by the memory to perform the functions of: performing an inbound screen on route information received from the first computer, to determine if the received route information should be discarded; if the route information is not discarded, comparing the received and screened route information to a local policy defined within the second computer; performing an outbound screen on the received and screened information prior to transmitting the received and screened route information to the first computer; and selecting a primary route from the received route information and local route information in accordance with the local policy, wherein the primary route is a path from the second computer to the first computer via the group of associated computers.
Owner:ACME PACKET +1

System and method for providing rapid rerouting of real-time multi-media flows

A system and method for providing rapid rerouting of real-time transport protocol (RTP) multi-media flows is disclosed. Generally, a first endpoint is connected to a second endpoint, wherein the first endpoint comprises a transceiver, software stored within the first endpoint defining functions to be performed by the first endpoint, and a processor configured by the software. The processor is configured to perform the steps of, performing flow processing on a data packet received at a first endpoint, from a second endpoint, removing a multi-protocol label switching (MPLS) tag from the data packet, translating a source address and destination address of the data packet, and determining a forwarding destination if more than one destination address of the data packet is provided.
Owner:PRIMARY NETWORKS +1

Medium streaming distribution system

A medium streaming distribution system reduces effects of packet loss in a network before the packet reaches radio base station. A medium distribution device for packet-transmits via the base station, a medium stream to the network by a real time transmission protocol. A packet analyzer monitors the packet arriving at the radio base station and transmits feedback Information associated with loss of a packet to the medium distribution device. Based on the feedback from a relay device and a terminal device of the medium stream, the transmission rates from the medium distribution device to the relay device and from the relay device to the terminal device are obtained to provide a greater transmission rate in a surplus band for re-transmission or a forward error correction.
Owner:QUALCOMM INC

Method and apparatus for adaptive buffering

An adaptive buffering scheme allows more effective media transport and buffering. In one aspect of the adaptive buffering scheme, buffering parameters are adapted to different media characteristics, such as media play commands or the amount of encoding / transcoding required for the particular media stream. In another aspect of the adaptive buffering scheme, buffering is adapted to different transmission or memory conditions, such as transmission rate, packet jitter, or the amount of available buffer memory. In one example, the adaptive buffering is supported using Real Time Streaming Protocol (RTSP), and / or Real Time Transport Protocol (RTP) and associated Real Time Control Protocol (RTCP), and / or Session Description Protocol (SDP) messages.
Owner:SHARP KK

System and method for assisting in controlling real-time transport protocol flow through multiple networks via screening

A system for assisting in controlling real-time transport protocol flow through multiple networks via screening is disclosed. The system utilizes a transceiver, a memory having logic stored therein defining functions to be performed by the system, and a processor. The processor is configured by the memory to perform the functions of: determining if route information received by the transceiver is from an external source or an internal source; if the route information is from an external source, performing a first internal screen of the route information, wherein a destination address defined by the received route information is compared to a local destination address defined by the first internal screen; and if the route information is from an internal source, performing a second internal screen of the route information, wherein an origin address defined by the received route information is compared to a local origin address defined by the second internal screen.
Owner:PRIMARY NETWORKS +1

Dynamically selecting codecs for managing an audio message

A system, method, and apparatus are directed towards managing a Voice over IP (VOIP) messages over a network, employing the Real-time Transport Protocol (RTP) and Session Initiation Protocol (SIP) over the Transmission Control Protocol (TCP). The VOIP messages are sent by a source device to a destination device through a relay server. The relay server may throttle the VOIP messages employing buffer management. When the buffer is substantially full, the relay server will drop packets from the source device. Indication of the lost packets may be provided to the source device through a Real-time Transport Control Protocol (RTCP) report. The source device may then employ the RTCP report to modify a type of codec employed, and thereby adjust a rate of flow of VOIP packets sent towards the destination device. Additionally, the relay server may provide port translation services for RTP / RTCP packets between the source and destination devices.
Owner:R2 SOLUTIONS

Digital rights management for local recording and home network distribution

The systems disclosed here provide a complete standards-based end-to-end scalable system for storage, delivery and in-home distribution of digital content over IP networks using standard protocols such as Real-time Transport Protocol (“RTP”) or IP-encapsulated MPEG-2 Transport Stream, or traditional MPEG-2 networks. Mechanisms are provided for receiving content from one security domain, re-encrypting that content uniquely for a receiving device, persistently storing that content, and playing back that content at a later time to and within another security domain. The systems also provide the ability to stream the persistently-stored content from the initial receiving device to another device that has been authenticated as part of a, e.g., home network. This allows a media server, e.g., a dual-tuner set-top box (“STB”) with hard drive, to deliver recorded content to any TV in the house by streaming to media clients such as STBs.
Owner:GOOGLE TECH HLDG LLC

Push-to-talk wireless telecommunications system utilizing an voice-over-IP network

A method and system to provide push-to-talk from one user to another in a wireless packet data telecommunications network is described. The system may include: a wireless communication network including push-to-talk (PTT) functionality, with a Session Initiation Protocol (SIP) Proxy Server; a SIP Registrar and Location Server operable to store contact addresses of active mobile devices; a Realtime Transport Protocol (RTP) Media Gateway (PTT Server) operable to function as a call endpoint for each of a plurality of mobile devices wherein the plurality of mobile devices are segmented into membership groups, the PTT Server further operable to multicast a communication from one member of the group to the other members of the group; and an Internet Protocol (IP) network interconnecting the SIP Proxy server, the SIP Registrar and Location Server, and the PTT Server.
Owner:APPLE INC

System and method for assisting in controlling real-time transport protocol flow through multiple networks

A system for assisting in controlling real-time transport protocol flow through multiple networks is disclosed The system utilizes at least a first computer and a second computer that is connected to the first computer, wherein the second computer comprises a second transceiver, a second memory having logic stored therein defining functions to be performed by the second computer, and a second processor The second processor is configured by the second memory to perform the functions of: performing an inbound screen on route information received by the second computer, from the first computer, to determine if the received route information should be discarded; if the route information is not discarded, comparing the received and screened route information to a local policy defined with the second computer; and performing an outbound screen on the received and screened information prior to transmitting the received and screened information.
Owner:ACME PACKET

Data communication system, data transmission apparatus, data reception apparatus, data communication method, and computer program

In a data communication system, the function of automatic repeat request is provided for transmission of packets based on a data communication protocol such as the Real-Time Transport Protocol or the User Datagram Protocol. Lost packets are detected at various timings, for example, when the beginning packet of each frame is received, the final frame of each frame is received, at a time limit of processing, and at a regular interval, and retransmission requests are issued accordingly. A data reception terminal does not issue a retransmission request if associated retransmission data will not be in time for playing with consideration of processing time and roundtrip time, thereby avoiding the transmission of useless retransmission request packets and retransmission packets.
Owner:SONY CORP

Functionality split between mobile terminal and terminal equipment for internet protocol multimedia signal exchange

The present invention provides a method and apparatus for connecting terminal equipment to a wireless network with a mobile terminal, wherein the mobile terminal is assigned proxy functions that control access of the terminal equipment to an internet protocol multimedia subsystem (IMS) in the wireless network. The proxy control functions include identification or authentication functions, as well as call control functions. The terminal equipment performs protocol stream processing functions for communicating with the internet protocol multimedia subsystem (IMS). The protocol stream processing functions include real-time transport protocol (RTP) and real-time transport control protocol (RTCP) functions. The wireless network includes a universal mobile telecommunications system (UMTS) network coupled to the internet protocol multimedia subsystem (IMS). The internet protocol multimedia subsystem (IMS) includes a session initiation protocol (SIP) server for providing internet protocol multimedia information signals.
Owner:NOKIA CORP

System and Method for Provisioning Charging and Policy Control in a Network Environment

A method is provided in one example embodiment and includes requesting real-time transfer protocol (RTP) quality metrics to be generated by a network for a specific flow in the network. The method also includes recording the RTP quality metrics and reporting the metrics as accounting data associated with the flow. In more specific embodiments, the accounting data includes a correlator with application level accounting and the accounting data is used for quality monitoring and troubleshooting on a per-subscriber basis, on a per-flow basis, or on a per-call basis. An application function can request RTP quality monitoring with a trigger that identifies if a quality parameter reaches a certain threshold.
Owner:CISCO TECH INC

System and Method for Determining Flow Quality Statistics for Real-Time Transport Protocol Data Flows

Systems and methods for determining lost packets for real-time transport protocol (RTP) data flows is disclosed. Generally, a first endpoint is connected to a second endpoint, wherein the first endpoint comprises a transceiver, software stored within the first endpoint defining functions to be performed by the first endpoint, and a processor. The processor is configured by the software to perform the steps of determining a sequence number of a received RTP data packet within said RTP data flow, storing said determined sequence number, calculating whether said determined sequence number sequentially falls within a predetermined numerical order, and if said sequence number of said received RTP data packet does not sequentially fall within said numerical order, storing said sequence number as a missed RTP data packet.
Owner:ACME PACKET

Method and system for key management in voice over internet protocol

A method and system for a secure telephone protocol are disclosed, which can be implemented using current Voice over IP (VoIP) protocols, Session Initiation Protocol (SIP, as specified in the Request for Comment (RFC) 3261 from the Internet Engineering Task Force (IETF)), Real Time Transport Protocol (RTP, as specified in RFC 3550), and Secure RTP (SRTP, as specified in RFC 3711). The secure telephone protocol can include a shared secret value that is cached and then re-used later to authenticate a long series of session keys to be used for numerous separate secure phone calls over a long period of time, thereby providing cryptographic key continuity without the need for voice authentication. In an embodiment, the secure telephone protocol can utilize the Diffie-Hellman key exchange during call setup, and AES for encrypting the voice stream.
Owner:ZIMMERMANN PHILIP R

Method and apparatus for S.I.P./H. 323 interworking

An interworking function (IWF) for a first and second protocol based network, for example, an H.323 protocol based network and an SIP protocol based network comprises an interworking gateway server including a state machine for defining each call processing state and a translation table for use in translating addresses formatted in each protocol. A method of interworking for use in interworking between said first protocol based network and said second protocol based network comprises the steps of receiving at said interworking gateway server serving said first and second protocol based networks a request from an endpoint in the first or second protocol based networks, establishing a state machine in memory whereby, for each state of said state machine, a message associated with that state is categorized as one of a triggering message, a non-triggering message and an error message, establishing a translation table in said memory whereby an address formatted in said first protocol has a one-for-one correspondence with an address formatted in said second protocol, processing said request in accordance with said translation table and said state machine and permitting communication between said first and second endpoints utilizing a realtime transport protocol. In the event media is terminated at said interworking gateway server, the interworking gateway server, in one embodiment, comprises a media switching fabric for switching media terminated at the gateway to an addressed endpoint capable of receiving it.
Owner:AMERICAN TELEPHONE & TELEGRAPH CO

Push-to-talk-over-cellular (POC)

A Push-to-Talk-over Cellular (PoC) implementation for use in a wireless communications network, wherein one or more servers interface to the wireless communications network to perform the PoC call sessions. Both the servers and the mobile units that use the PoC call sessions communicate with each other using SIP / IP (Session Initiation Protocol / Internet Protocol) control messages within the wireless communications network, and one or more of the servers switches RTP / IP (Realtime Transport Protocol / Internet Protocol), RTCP / IP (Realtime Transport Control Protocol / Internet Protocol), or MBCP / IP (Media Burst Control Protocol / Internet Protocol) voice packets for the PoC call sessions between the mobile units across the wireless communications network.
Owner:KODIAK NETWORKS

System and method for assisting in controlling real-time transport protocol flow through multiple networks

A system for assisting in controlling real-time transport protocol flow through multiple networks is disclosed. The system utilizes at least a first computer and a second computer that is connected to the first computer, wherein the second computer comprises a second transceiver, a second memory having logic stored therein defining functions to be performed by the second computer, and a second processor. The second processor is configured by the second memory to perform the functions of: performing an inbound screen on route information received by the second computer, from the first computer, to determine if the received route information should be discarded; if the route information is not discarded, comparing the received and screened route information to a local policy defined with the second computer, and performing an outbound screen on the received and screened information prior to transmitting the received and screened information.
Owner:PRIMARY NETWORKS +1

System and method for determining flow quality statistics for real-time transport protocol data flows

A system and method for determining flow quality statistics for real-time transport protocol (RTP) data flows is disclosed. Generally, a first endpoint is connected to a second endpoint, wherein the first endpoint comprises a transceiver, software stored within the first endpoint defining functions to be performed by the first endpoint, and a processor. The processor is configured by the software to perform the steps of, determining latency for the RTP data flows, determining jitter for the RTP data flows, and / or determining lost packets for the RTP data flows. Latency is determined by the first endpoint transmitting a test data packet to the second endpoint; the second endpoint looping the test data packet back to the first endpoint; comparing when the test data packet was received by the first endpoint to when the test data packet was sent to the second endpoint, to determine a round trip time; and, dividing the round trip time in two, resulting in the latency. Jitter is determined by beginning a timer when a first data packet of an RTP data flow is received by the first endpoint; stopping the timer when a second data packet of the RTP data flow is received by the first endpoint; and, adding measured time from the beginning of the timer to the stopping of the timer to an aggregate to obtain the jitter for the RTP data flow. Lost packets are determined by determining a sequence number of a received RTP data packet within the RTP data flow; storing the determined sequence number; calculating whether the determined sequence number sequentially falls within a numerical order; and, if the sequence number of the received RTP data packet does not sequentially fall within the numerical order, storing the sequence number as a missed RTP data packet.
Owner:PRIMARY NETWORKS +1

Measuring speech quality over a communications network

When the PESQ and similar algorithms are used to measure speech quality, a particular voice call is set up to transmit only test voice signals over a communications network. This enables the test voice signals to be easily identified and provides a means of determining the amount of degradation that occurs as a result of transmission of the test voice signals. However, one problem is that in packet switched networks the transmission characteristics change with time. Thus the estimated MOS score obtained cannot be assumed to give an accurate speech quality measure for a voice call later made between the same two points. By adding test voice information to an ongoing voice call and transmitting that test voice information integrally with the ongoing voice call these problems are addressed. The test voice information is sent in packets during silent periods in the voice call such that the voice call is not affected by the test voice information. An identifier is used to identify or label the packets comprising test voice information. In the case that real-time transport protocol is used, the identifier is preferably a payload type value.
Owner:NORTEL NETWORKS LTD

Proxy server for relaying VOIP messages

A system, method, and apparatus are directed towards managing a Voice over IP (VOIP) message over a network, where the VOIP message may employ the Real-time Transport Protocol (RTP) and possibly Session Initiation Protocol (SIP) over the User Datagram Protocol (UDP). The invention enables a VOIP client device, such as an IP phone, and the like, to communicate a message to a local proxy residing on a local computing device. The communications between the VOIP device and the local proxy may employ SIP / RTP over UDP. Upon receipt of the communications, the local proxy converts the transport protocol to another transport protocol, such as Transmission Control Protocol (TCP). The local proxy may also perform a port translation on the message. The converted communications may then be sent to a remote server, where it may be employed in its present SIP / RTP over TCP format, or be converted back to using UDP.
Owner:R2 SOLUTIONS

Method and system for key management in voice over internet protocol

A method and system for a secure telephone protocol are disclosed, which can be implemented using current Voice over IP (VoIP) protocols, Session Initiation Protocol (SIP, as specified in the Request for Comment (RFC) 3261 from the Internet Engineering Task Force (IETF)), Real Time Transport Protocol (RTP, as specified in RFC 3550), and Secure RTP (SRTP, as specified in RFC 3711). The secure telephone protocol can include a shared secret value that is cached and then re-used later to authenticate a long series of session keys to be used for numerous separate secure phone calls over a long period of time, thereby providing cryptographic key continuity without the need for voice authentication. In an embodiment, the secure telephone protocol can utilize the Diffie-Hellman key exchange during call setup, and AES for encrypting the voice stream.
Owner:ZIMMERMANN PHILIP R

Medium streaming distribution system

A medium streaming distribution system reduces effects of packet loss in a network before the packet reaches radio base station. A medium distribution device for packet-transmits via the base station, a medium stream to the network by a real time transmission protocol. A packet analyzer monitors the packet arriving at the radio base station and transmits feedback information associated with loss of a packet to the medium distribution device. Based on the feedback from a relay device and a terminal device of the medium stream, the transmission rates from the medium distribution device to the relay device and from the relay device to the terminal device are obtained to provide a greater transmission rate in a surplus band for re-transmission or a forward error correction.
Owner:QUALCOMM INC

Application and communication method of terminal of network media phone

This invention relates to the application and communication method for network media telephone terminals. The terminal includes a phone module and IP phone module imaging the target called number into related network IP address for Volp communication, PCI bus with Ethernet interface and phone / multimedia processor controlled by CPU, the media port is connected by D / A conversion, a special inserted CPU, TCP / IP protocol stack the real time transmission protocol of the transmission stream media. The terminal and the software exchange center based on IP interact synchronously or asynchronously, and the IP calling cetral platform provides support to communication for stream media forms and speech.
Owner:SHANGHAI TT&S TELECOM TECH CO LTD

internet based TV stream data real time transmission and service apparatus and method

In internet, this invention uses the code-converting server (CS) real time to code-convert the received directly TV media (TM) data, then distributes the code-converted TM data toward all cell media server (CMS). CMS stores these data and offers the TM program ordering service and the real time (RT) directly playing service. These two services can be switched seamlessly. The transferring and service of RT TM operation is realized in Internet and its service quality can be ensured. This device consists of CS, the data receiving / storing entity and media servers. Here, the data receiving entity is the data front end of CMS. Both reside on the same physical server and connect to CS via Internet to execute RT transferring protocol.
Owner:UNIV OF SCI & TECH OF CHINA

Method and apparatus for providing trace route and timing information for media streams

An apparatus and method that use media packets, such as Real-time Transport Protocol (RTP), packets with a specially define profile to determine the route and round-trip-time information. The route is determined by transmitting by a first network endpoint one or a group of media packets having a number of hops equal to a predefined number and the network address of a second endpoint, incrementing the number of hops upon a non-media packet being received in response to the one or group by a network node because the number of hops was exceeded at that network node. After recording the identification of that network node and incrementing the number of hops, the network endpoint re-transmits the one or group of media packets. The round-trip-time information is determined by timestamp information inserted into the one or group of media packets by the first network endpoint and timestamp information inserted into another one or another group of media packets transmitted from the second network endpoint to the first network endpoint in response to receipt of the one or group of media packets.
Owner:AVAYA INC

Method for real-time transport protocol (RTP) packet authentication

ActiveUS20050265349A1Reduce maximum processor cycle consumedConstant overheadMultiple keys/algorithms usageUser identity/authority verificationTraffic capacityConfidentiality
A method for Real-time Transport Protocol (RTP) packet authentication on a packet data network. In particular, the invention relates to a method for preventing toll fraud, privacy compromise, voice quality degradation, or denial of service (DoS) on Voice over IP networks. The Real-time Transport Protocol (RTP) is susceptible to several security attacks, including thirdparty snooping of private conversations, injection of forged content, and introduction or modification of packets to degrade voice quality. The Secure Real-time Transport Protocol (SRTP) provides confidentiality, message authentication, and replay protection for RTP traffic. However, SRTP incurs an additional overhead to verify the HMAC-SHA1 message authentication code for each packet. SRTP+ significantly decrease the verification overhead compared to SRTP and thereby increases the number of faked packets required to mount a successful denial of service attack. SRTP+ provides packet authentication but not integrity. SRTP+ is compatible with SRTP.
Owner:AVAYA INC
Who we serve
  • R&D Engineer
  • R&D Manager
  • IP Professional
Why Patsnap Eureka
  • Industry Leading Data Capabilities
  • Powerful AI technology
  • Patent DNA Extraction
Social media
Patsnap Eureka Blog
Learn More
PatSnap group products