380 results about "Time Protocol" patented technology

Systems and methods are provided for receiving and transmitting an MPEG2 transport stream (TS) in a real-time protocol (RTP) / user datagram protocol (UDP) / Internet protocol (IP) packet. The receiving method comprises: receiving an IP packet via an IP network, having a variable transmission delay; accessing a timestamp carried in a RTP packet; linking the timestamp with a program clock reference (PCR) MPEG2TS carried in the RTP packet payload; and, using the timestamp to eliminate variable transmission delay jitter, associated with the PCR MPEG2TS. In one aspect of the method, the timestamp has a resolution of greater than 500 nanoseconds (ns), so that the variable transmission delay jitter, associated with the PCR MPEG2TS can be reduced to a jitter of less than 500 ns.

A real-time protocol for real-time network programming applications is disclosed. The protocol is executable on a computer network having a plurality of nodes that include applications, a network stack, an operating system, and middleware capable of executing real-time operations. The protocol can use group objects to model physical and logical devices connected to the network wherein each group object comprises at least one variable. Changes in these variables are encapsulated in variable change messages which are propagated in the network using communication objects. Also described is a real-time messaging format useful for implementing the protocol.

An apparatus and methods for Real Time Protocol (RTP) based bridging to create calls to and From any communication devices connected to the Public Switched Telephone Network (PSTN) or any public or private Internet Protocol network. The methods are represented in controls residing either in an RTP Bridge apparatus or as middleware on a service creation platform, softswitch, or SIP proxy services incorporating SIP bridging and SIP call relay technologies, with an interface to a TDM (Time Division Multiplexing) switch operated by a Local Exchange Carrier (LEC) or Competitive LEC (CLEC), or hosted by an Internet Telephony Service Provider (ITSP) with connectivity to the PSTN through media gateways. The method permits a party to request an on demand conference call by either dialing into the apparatus from the PSTN or from any type of RTP communication device such as an IP phone; or using a form of signaling from RTP translation device. The moderator initiates the request, enters the participant(s) to be included in the call, and the launch sequence is initiated and the apparatus makes contact with the phone / end point and uses RTP bridging to interlace the packet streams and deliver one stream back to all the devices. The advantage is an on demand conference call to one or many participants, on similar or different communications platforms.

A networked system is provided for transporting digital media packets, such as audio and video. The network includes network devices interconnected to send and receive packets. Each network device can receive and transmit media signals from media devices. A master clock generates a system time signal that the network devices use, together with a network time protocol to generate a local clock signal synchronised to the system time signal for both rate and offset. The local clock signal governs both the rate and offset of the received or transmitted media signals. The system, which can be implemented using conventional network equipment enables media signals to be transported to meet quality and timing requirements for high quality audio and video reproduction.

A method and apparatus to provide encryption and authentication of a mini-packet in a multiplexed real time protocol (RTP) payload. Mini-packets are assembled into a payload wherein each mini-packet includes an associated mini-header for ensuring proper processing of each mini-packet. Padding is added to mini-packets when the mini-packets are encrypted to insure each mini-packet is an integral multiple of a predetermined block size. Padding for each mini-packet is determined according to p=n−k*floor((n−1) / k), wherein p is the amount of padding added to each mini-packet, n is the actual data size, and k is the block size. The padding added to the data for each packet comprises p−1 units of padding and a final padding unit for indicating the amount of padding. An authenticator may also be added to each mini-packet. A length indicator is set in each mini-header for indicating a total length of the mini-packet including the authenticator. The authenticator may then be removed based upon knowing a type of authentication used for generating the authenticator.

A system and method of ensuring a requested Quality of Service (QoS) for a media flow that is transported through multiple transport networks. An interface is established between a Call State Control Function (P-CSCF) and a Bandwidth Broker (BB) for the passing of a session description and Binding Information from the P-CSCF to the BB. The interface uses the Common Open Policy Service (COPS) protocol and the Bandwidth Broker (BB) protocol. The Binding Information may be the source IP address plus Real Time Protocol (RTP) port and the destination IP address plus RTP port. The Binding Information is also passed back toward the Originating BB (BB-O) from a Serving BB (BB-S) in the terminating (serving) network through BBs in adjacent networks using the BB protocol.

A UMTS (Universal Mobile Telecommunications System) core network supports a compression framework that provides for header compression of General Packet Radio Service Tunneling Protocol (GTP)-Encapsulated Packets. In particular; the GTP / UDP(User Datagram Protocol) / IP(Internet Protocol) header is compressed. In addition, the UMTS core network also supports RTP(Real Time Protocol) / UDP / IP header compression independent of the GTP / UDP / IP header compression.

The present embodiments provide apparatuses, methods and systems for use in notifying a remote communication device when a local wireless device loses its wireless link. In some embodiments, a method detects when a wireless connection is lost with a first communication device, identifies a second communication device that is communicating with the first device, and notifies the second device of the lost link. In some implementations, the notification of the lost link is supplied while the method provides batch communications. Some embodiments provide voice communications, where a remote device is notified when a local device loses its wireless connection. The second device can be determined by querying a server. The notifying can include forwarding an in-band tone, and / or generating a real time protocol packet. In some embodiments, the loss of the link is detected through a radio access network, a packet control function, and / or a mobile switching center.

Techniques are described for aggregating multiple media packets to improve end-to-end bandwidth efficiency. The techniques include using an RTP aggregation protocol that is not sensitive to packet loss to aggregate multiple media packets under a single header. According to the RTP aggregation protocol, the single header for an aggregated media packet comprises a version field, a zero field, a sequence number field and a trunk ID field. The single header encapsulates the aggregated payload, which is an aggregation of Real-Time Protocol (RTP) segments. An RTP segment either has a compressed format or an uncompressed format. The uncompressed RTP segment includes the complete uncompressed RTP packet copied from the original User Datagram Protocol (UDP) packet. The compressed RTP segment includes the payload of the original RTP rather than the complete original RTP packet.

Enabling media (audio / video) scenarios across firewalls typically requires opening up multiple UDP ports in an external firewall. This is so because RTP (Real Time Protocol, RFC 1889), which is the protocol used to carry media packets over IP network, requires a separate UDP receive port for each media source. Opening up multiple media ports on the external firewall is something that administrators are not comfortable doing as they consider it security vulnerability. The system and method according to the invention provides an alternate mechanism which changes RTP protocol a little and achieves a goal of traversing firewalls for media packets using a fixed number, namely two, of UDP ports.

A technique for establishing a logical path between two servers in a coordinated timing network of a processing environment is provided. The technique includes the exchange of command and response message pairs by a server and an attached server, via a physical link. The server transmits a command message to an attached server to establish a server-time-protocol (STP) logical path and receives a response from the attached server. The technique also includes the server receiving a request transmitted by the attached server to establish an STP logical path to the server and transmitting a response to the attached server's request. A logical path between the server and the attached server is established if the attached server's response indicates that the server's request was accepted by the attached server and if the server's response indicates that the attached server's request was accepted by the server.

This invention relates to methods and devices for frequency distribution based on, for example, the IEEE 1588 Precision Time Protocol (PTP). Packet delay variation (PDV) is a direct contributor to the noise in the recovered clock and various techniques have been proposed to mitigate its effects. Embodiments of the invention provide a mechanism to directly measure and remove PDV effects in the clock recovery mechanism at a slave clock. One particular embodiment provides a clock recovery mechanism including a phase-locked loop (PLL) with a PDV compensation feature built-in. An aim of the invention is to enable a slave clock to recover the master clock to a higher quality as if the communication path between master and slave is free of PDV. This technique may allow a packet network to provide clock synchronization services to the same level as time division multiplexing (TDM) networks and Global Positioning System (GPS).

This invention relates to methods and devices for clock synchronization. The invention makes particular use of IEEE 1588 with offset and skew correction. In embodiments of the invention, the IEEE 1588 Precision Time Protocol is used to exchange time stamps between a time server and a client from which the client can estimate the clock offset and skew. In embodiments of the invention a free running clock at the client is provided with an estimation technique based on the time stamps from the IEEE 1588 PTP message exchange between the server and client clocks. The offset and skew from the estimation process can be combined with the local free running clock to give a synchronized local clock which is an accurate image of the master clock.

Apparatus for making legacy network elements transparent to IEEE 1588 Precision Time Protocol operation. Network elements are wrapped by device(s) capable of providing either transparent clock or boundary clock operation. In one embodiment, smart interface converters are used to provide transparent clock or boundary clock operation. The smart interface converters work cooperatively.

An Internet Protocol (IP)-based radio telecommunications network and method of providing data services to a mobile terminal. An Internet Protocol Shared Interworking Function (IP-SIWF) is implemented in the network, and is controlled from a Mobile Switching Center (MSC) utilizing the Media Gateway Control Protocol (MGCP) device control protocol. When a data call carrying a data payload is originated by a mobile terminal, the data call is transported utilizing radio access and radio link protocols to the IP-SIWF. The radio access and radio link protocols are terminated in the IP-SIWF, and the data payload is transported thereafter utilizing a real time protocol over a User Datagram Protocol (UDP) and an IP protocol layer.

A system for enabling music to be provided to callers placed on hold in a LAN network used for voice communications. Bandwidth consumption is minimized by ensuring that music-on-hold is provided in a manner that enables most necessary tasks to be executed by a client instead of a server. In specific embodiment, the music-on-hold system is configured to transfer a small application program or applet, having a music file and a media player, from the server to the client if necessary. Concurrently, a real time protocol (RTP) stream carrying music is used to provide music to the client from the server during the applet transfer to ensure that the held caller receives music at all times. The RTP stream is initially used to provide music to held callers.

A ticket-based secure time protocol is used to provide client devices, or users, with secure time signals. In a preferred embodiment, the secure time signals are provided by a secure time server so that multiple clients can be time-synchronized. Ticket-based authentication uses digital certificates and public key cryptography, such as Elliptic Curve Cryptography (ECC) to reduce key administration overhead and decryption processing. Standard authentication architectures and approaches, such as Kerberos, can be used for some aspects of the invention. A preferred embodiment uses Request and Reply messages that provide added security and functionality, such as authentication, sequence-checking and verification of target destination.

An apparatus and method calculate a round-trip delay (RTD) between first and second endpoints within a network, where an intermediate point is selected between the first and second endpoints within the network. A network analyzer determines a first data transmission time from the intermediate point to the first endpoint and back to the intermediate point, determines a second data transmission time from the intermediate point to the second endpoint and back to the intermediate point, and adds the first and second data transmission times to determine therefrom the round trip delay between the first and second endpoints within the network.

The invention provides a processing method and a device for time synchronization. The method comprises the following steps: according to the receiving time of a precise time protocol message at the import of the equipment and the arrival time of the precise time protocol message at the export of the equipment, adjusting the preset field value in the precise time protocol message at the export of the equipment, thereby using the preset field to record the transmission time of the precise time protocol message between the import and the export of the equipment. The invention effectively acquires the residence time of the message in the equipment.

A system and method, i.e. Voice Proxy Server (VPS), is invented for managing Voice over IP (VoIP) services between a NAT enabled private network and a public network such as the Internet. The VPS acts like the Call Agent to the nodes in the private network and it represents an embedded voice client to the real Call Agent in the public network. Each logical line of the VPS corresponds to a real voice line in the private node. VoIP signaling messages may be carried in the payload of the IP packets. VPS will properly translate IP address and / or port number information in the payload to correctly route the signaling data between networks. Additionally, VPS will forward the incoming Real Time Protocol (RTP) packets to the real destination in the private network.

A hardware accelerated streaming arrangement, especially for RTP real time protocol streaming, directs data packets for one or more streams between sources and destinations, using addressing and handling criteria that are determined in part from control packets and are used to alter or supplement headers associated with the stream content packets. A programmed control processor responds to control packets in RTCP or RTSP format, whereby the handling or direction of RTP packets can be changed. The control processor stores data for the new addressing and handling criteria in a memory accessible to a hardware accelerator, arranged to store the criteria for multiple ongoing streams at the same time. When a content packet is received, its addressing and handling criteria are found in the memory and applied, by action of the network accelerator, without the need for computation by the control processor. The network accelerator operates repetitively to continue to apply the criteria to the packets for a given stream as the stream continues, and can operate as a high date rate pipeline. The processor can be programmed to revise the criteria in a versatile manner, including using extensive computation if necessary, because the processor is relieved of repetitive processing duties accomplished by the network accelerator.

A hardware accelerated streaming arrangement, especially for RTP real time protocol streaming, employs a directing file determining the pointers, header lengths and offsets of a block of one or more data packets to be sent out through a network accelerated streaming system. The directing file is established by a control processor, for example working in the background, and is stored to provide information making it possible to determine certain information including header sizes and pointers to RTP payload and other data, without the need during egress of the data for analysis related to the type of media or protocol concerned. This relieves the control processor of functions that would otherwise require attention, and permits the egress process to proceed in a repetitive manner, preferably relying insofar as possible on hardware elements for speed and reserving the control processors computational capacity for control functions that may be more complex but are infrequent and / or not time sensitive for streaming in real time.

The invention discloses an Ethernet-based train communication network implementation method in the technical field of train communication. The Ethernet-based train communication network implementation method comprises the following steps of: setting a real-time virtual layer on a transport layer of a TCP / IP (Transmission Control Protocol / Internet Protocol) stack of Ethernet, and setting a real-time protocol layer on the real-time virtual layer, wherein the real-time virtual layer is used for realizing train real-time data communication required by IEC 61375-1 train communication network standards on the basis of COTS (Commercial Off The Shelf) Ethernet hardware, drive and the TCP / IP stack; establishing an Ethernet train communication network which comprises an Ethernet-based multifunctional train bus structure and an Ethernet-based wired train bus structure; and carrying out data communication according to a set communication organization manner. The Ethernet-based train communication network implementation method has the capability of preventing the hardware, drive and TCP / IP stack of the Ethernet from being corrected, meeting the hard real-time property and adapting to the requirement on future train control or train operation control.

A Real-Time Protocol (RTP) source node of a network operates to send a first data packet of a first size to a destination node over a path of the network that includes a plurality of intermediate nodes, at least one of the intermediate nodes having a maximum transmission unit (MTU) size smaller than the first size such that fragmentation of the first data packet occurs. The destination node sends back to the source node a RTCP report that includes a number of fragments received and a largest minimum data packet size. In response, the source node sends subsequent data packets having a second size less than or equal to the largest data packet size of the fragments. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or meaning of the claims. 37 CFR 1.72(b).

A contactless integrated circuit (IC) card can include: an analog interface block operable to demodulate a received radio frequency (RF) signal into multiple versions thereof according to a first plurality of communication protocols, respectively; a controller operable to select from among a second plurality of communication protocols; and a universal asynchronous receiver / transmitter (UART) operable to select one of the demodulated versions of the RF signal according to the selected protocol

A method is provided in one example embodiment and includes providing a time protocol assistant associated with a time-synchronized domain (TSD). The TSD includes a set of nodes that are synchronized to a same time source. The TSD has defined egress and ingress edge points where bidirectional measurements can be made and the egress and ingress edge points are coupled to the time protocol assistant. The method also includes synchronizing one or more packets flowing through a network that includes the TSD through the same time source. In more specific embodiments, the nodes are synchronized to the same time source via the network and the same time source is a grandmaster clock that synchronizes one or more transparent clocks. In yet other embodiments, the transparent clocks manipulate precision time protocol (PTP) packets sent by the grandmaster clock.

A system can include an application function which is configured to indicate per each multiplexed flow to a rules function that a flow includes multiplexed real time protocol data packets and control packets flow and to indicate the status of each component of the multiplexed flow. The rules function can be configured to inform an enforcement function which is configured to detect the multiplexed flows by the IP addresses and port numbers and to detect the multiplexed subcomponents. The enforcement function can be configured to apply separate gating to data packets and control packets according to the status of each subcomponent.

Apparatus and a method for transmitting a plurality of streams of data packets through a computer network assigns queued packets from each of the plurality of data streams to respective slots in scheduling windows such that queued packets from respective ones of the plurality of data streams occupy different slots in respectively different ones of the scheduling windows. The packets are transmitted through the network in the order defined by the scheduling windows.