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

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.

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.

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.

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.

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.

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.

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.

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.

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.

The disclosed techniques provide for eliminating real-time transport protocol, RTP, extension bit X, marker bit M, time stamp and sequence number information and/or user data protocol, UDP, checksum information from the header of a packet for transmission.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.