647 results about "Sip server" patented technology

Methods and systems for optimized routing of real time media session data between session initiation protocol SIP endpoints. In one embodiment of the invention, first and second SIP endpoints each provide notification of network address translation NAT topology thereof and based on the NAT topologies of the two endpoints an SIP server decides whether to route media between the two endpoints via a direct path or via a media relay.

A method for operating a server having a maximum capacity for servicing requests comprises the following steps: receiving a plurality of requests; classifying each request according to a value; determining a priority for handling the request according to the value, such that requests with higher values are assigned higher priorities; placing each request in one of multiple queues according to its priority value; and dropping the requests with the lowest priority when the plurality of requests are received at a rate that exceeds the maximum capacity. The server operates according to a session initiation protocol.

Classifying each request comprises running a classification algorithm. The classification algorithm comprising steps of: receiving a rule set, each rule comprising headers and conditions; creating a condition table by taking a union of all conditions in the rules; creating a header table by extracting a common set of headers from the condition table; extracting the relevant headers from the header table; determining a matching rule; creating a bit vector table; selecting the matching rule according to data in the bit vector table; and applying the rule to place the message in the appropriate queue.

A mechanism for achieving resiliency and load balancing for SIP application services and, in particular, in geographic distributed sites. A method performs a distribution of SIP requests among SIP servers, where at least two sites with a load balancer in each site is configured. The method includes receiving a SIP request by a first load balancer in a first site; determining whether the SIP request should be redirected to a second site; and redirecting the SIP request to an address of a second load balancer in the second site. The invention also includes a SIP proxy including a receiving unit receiving SIP requests; a load balancing unit distributing SIP requests between SIP entities; and a health monitoring unit verifying availability of the SIP entities. The SIP proxy may further be configured with a proximity measuring unit determining a proximity to a SIP entity.

A method of extending the use of SIP (Session Initiation Protocol) in a communication system wherein H.323 or SIP-compliant end-points communicate with the media traffic directly between other H.323 or SIP-compliant end-points. An extension of the SIP protocol is utilized to provide better support for charging. A plurality of equivalent INVITE messages are sent to the SIP server at regular time intervals for the duration of the session. Charging for SIP conversations can be determined in a very accurate and expedient manner by observing when the INVITE messages cease, indicating that the conversation between the two end-points is concluded.

The invention includes a method and apparatus for providing SIP message prioritization between network elements along at least a portion of an end-to-end path between a SIP client and a SIP server. The method includes determining a SIP message prioritization policy and distributing the SIP message prioritization policy toward a prioritizing network element adapted to assign message priority levels to respective received SIP messages using the SIP message prioritization policy, process the received SIP messages according to the respective assigned message priority levels, and transmit the prioritized SIP messages toward at least one network element in a manner for propagating the respective assigned message priority levels to the at least one network element. The prioritizing network elements include SIP network elements and non-SIP network elements.

Provided in a proxy server which enables a plurality of SIP servers to copy registration information after presenting, to an SIP server requesting SIP Digest authentication, properness of one who makes an access and of a REGISTER request to be transmitted. With a REGISTER request generation function provided in addition to a function that a common SIP proxy server holds, a proxy server disposed between a user agent and an SIP server generates a REGISTER request to a spare SIP server and transmits the same to the spare SIP server, thereby realizing registration information copying, and holds a user identifier and a password of the proxy server to execute Digest authentication with the spare SIP server. Moreover, after confirming a REGISTER request processing completion response (200 OK) from a working SIP server from which registration information is copied, the proxy server transmits a copied REGISTER request to the spare SIP server.

A system and method for establishing a SIP session between calling and called end-points. A SIP server receives a SIP INVITE request for establishing the SIP session. The SIP INVITE request includes standard routing information in its header. The SIP server receives the SIP INVITE request and responds by transmitting to the calling end-point a HTML link associated with a web server. In rendering the contents of the HTML link, the calling end-point transmits an HTTP request to the web server including voice cookie information. The voice cookie information may include information gathered about the caller such as the caller profile information, transaction information, caller intent information, or recent and past history associated with a web site domain. The web server forwards the voice cookies to the SIP server. The SIP server determines an address of the called end-point as a function of the routing information in the SIP INVITE request and the voice cookie information. The call is then routed to the determined address along with the voice cookies or information derived from the voice cookies. New voice cookies may be written to the web server which transmits these to the calling end-point within an HTTP response.

A packet sent in a mobile IP communication between different regions passes through a particular address translation unit, so that the communication route cannot be optimized. The packet cannot be delivered to a mobile node not having a fixed home address. To overcome the difficulties, the address translation unit includes a function to process the mobile IPv6 protocol. The mobile node includes a function to dynamically acquire a home address. A first network and a second network each identify the mobile node by a SIP (session initiation protocol) identifier. The mobile node acquires a home address on the second network and registers a corresponding relation between the identifiers and the home address thereof. A terminal on the second network acquires the home address of the mobile node from information of the mobile node registered to the SIP server according to a SIP procedure.

With the widespread adoption of SIP-based VoIP, understanding the characteristics of SIP traffic behavior is critical to problem diagnosis and security protection of VoIP services. A general methodology is provided for profiling SIP-based VoIP traffic behavior at several levels: SIP server host, server entity (e.g., registrar and call proxy) and individual user levels. Using SIP traffic traces captured in a production VoIP network, the characteristics of SIP-based VoIP traffic behavior in an operational environment is illustrated and the effectiveness of the general profiling methodology is demonstrated. In particular, the profiling methodology identifies anomalies due to performance problems and/or implementation flaws through a case study. The efficacy of the methodology in detecting potential VoIP attacks is also demonstrated through a test bed experimentation.

A SIP header reduction protocol operating on a communication network to compress the SIP header fields on SIP control messages used to initiate a SIP communication session. The new protocol is implemented on the mobile use equipment (UE) (301) and P-CSCF proxy SIP server (302). The prior SIGCOMP compression dictionary is used to compress those header field parameters defined by the SIGCOMP dictionary. A new 3G Dictionary is implemented with header field parameters for several wireless specific data elements and parameters necessary to access various wireless technologies and services which are missing from the SIP/SDP based SIGCOMP dictionary. Finally, an Encoding Assistant (EA) (310, 355) function is implemented on the UE and P-CSCF that operates a standard encoding and decoding of SIP message header fields to reduce those SIP header fields that are dynamic over time but are essentially static in terms of most user communication. The EA parameters can be updated periodically, while the 3G and SIGCOMP dictionary are static and not updated.

A boundary node router determines whether a received packet includes a SIP message and a REGISTER message, and, if so, adds its router identifier to a REGISTER message which includes a user identifier and a user terminal identifier, sent from a user terminal, and sends the message to a SIP server. The SIP server has a user management table which stores the user terminal identifier, the router identifier of a router which serves the user terminal, and a predetermined Quality of Service type, in association with each user identifier. The SIP server extracts a user identifier, a user terminal identifier, and a router identifier from a received REGISTER message, and updates the user terminal identifier and the router identifier corresponding to the extracted user identifier, to the extracted user terminal identifier and the extracted router identifier.

Methods of operating an SIP load balancer, one or more SIP server nodes and/or one or more SIP server instances are provided that efficiently manage a processing load. Configurations for an SIP load balancer, one or more SIP server nodes and/or one or more SIP server instances to enable efficient management of a processing load are also described. In certain ones of these, state data for a set of SIP processes may be sent from a SIP server instance and used by a SIP load balancer to remove inactive SIP processes from a data structure. The management of a processing load comprising a number of SIP processes following the failure of a SIP server node and/or a SIP server instance is also provided.

A communication system includes a packet-based data network coupled to various network entities. The communications system includes a community that has a call processing system and various other devices, such as an integrated voice response (IVR) system and plural agent systems. The call processing system includes a combination of logical entities that perform call processing tasks. As an example, the logical entities may include a Session Initiation Protocol (SIP) proxy, a SIP client, and a SIP server. In response to a call request from outside the community, the call processing system, under control of the server, sends back responses to the originating system. The call processing system also establishes a call with a network element inside the community, such as the IVR system. The IVR system is capable of receiving input data from the originating system. Based on the received input data, the call processing system can reconnect or forward the call to one of the agent systems.

In a load balancer and a load balancing method, if load on a server becomes higher and the server enters a high-load state, processing is transferred from the server to a second server. In a load balancer, for user terminals registered to SIP servers, a server as a register request destination is dynamically changed according to load on the user terminals, which makes it possible to transfer the processing to a low-load SIP server.

An SIP (Session Initiation Protocol) server sharing module relays SIP messages which are transmitted and received between a plurality of client groups and a SIP proxy server shared by the plurality of client groups. The SIP server sharing module includes a message source group identifying section configured to identify a specific client group to which a specific client terminal belongs which transmits a client transmission SIP message, from the client transmission SIP message which is transmitted to the SIP proxy server from each of client terminals belonging to the plurality of client groups, and a group tag insetting section configured to insert a group tag corresponding to the specific client group identified by the message source group identifying section in the client transmission SIP message. A group tag removing section removes a group tag contained in a server transmission SIP message transmitted to a destination client terminal, which belongs to the specific client group, from the SIP proxy server. A message destination group identifying section prevents the server transmission SIP message from being transmitted to a client terminal which belongs to a client group different from the specific client group corresponding to the group tag contained in the server transmission SIP message.

A method and Session Initiation Protocol (SIP) application server, the server comprising a SIP container supporting SIP communications and a SIP service running on the SIP server. When an SIP message is received at a SIP application server, the SIP container detects the message is destined to the local SIP service and determines which end-point capabilities are included in the SIP message, the end-point capabilities being, for example, indicative of the requirements for the proper processing of the message. The SIP container further determines whether or not end-point capabilities of the SIP service match the ones included in the SIP message, and upon determining that a match exist, invokes the SIP service and relays the message to the service. Otherwise, if a match is not detected, the SIP container does not invoke the SIP service, but rather issues an error message, or further relays the SIP message to another destination.

A SIP server having the function of accepting simultaneous IPv4/IPv6 registrations from an IPv4/IPv6 dual SIP terminal, recognizing a protocol communicable with a communication partner terminal, and notifying the IPv4/IPv6 dual SIP terminals of the result of the recognition if necessary, thereby to allow VOIP communication between the IPv4/IPv6 dual SIP terminal and an IPv4 terminal or an IPv6 terminal to be the communication partner.

The present invention supports a message protocol on a wireless communication network between the access gateway and the proxy session initiation server. Upon disruption of a communication session with a mobile station, an information packet indication disruption of the communication session is transmitted from the access gateway to the proxy session initiation protocol server. Upon receipt of the information packet, the proxy session initiation server initiates a resource release procedure freeing network resources supporting the communication session. The proxy session initiation server releases the resources and transmits an information packet to the serving session initiation server, which also initiates a release procedure. Answer and acknowledgement messages are also sent in response to the release request and/or the release of resources.

The invention includes programs at client end, programs at SIP server end, program for selecting peer-to-peer bodies, and programs at stream media server, which are setup at client end, SIP server, expanded media server, and stream media server in sequence. Based on SIP protocol, using P2P network, client end requests program from SIP server. The SIP server forwards the request to an expanded media server. The expanded media server selects best program source, which can be other client or a stream media server. Based on address of the program source, client end can down load program of stream media, and sends information of buffering program to the expanded media server. Client end or stream media server also can send information of changing node state to the expanded media server in order to cause refreshing database of media information. Advantages are: small time delay, small bandwidth consumption, and good audio/video effect.

A session initiation protocol (SIP) server adds billing and authentication information to conventional SIP messages used in establishing call transfers. This additional information is later verified by a SIP server and used to enable advanced billing and fraud protection features for call transfers in a SIP telecommunications network.