Softphone

Abstract: A high availability VoIP system (fig. 1, 100) interfacing with a PSTN (104) or other TDM network to provide higher availability and better failure recovery wherein the high availability VoIP system includes a plurality of gateways (108, 109) coupled to at least one hub (115) and a proxy table and a call restoration table configured in each of the plurality gateways. Further, the present invention is a method of providing a high availability VoIP system wherein the method includes configuring a plurality of gateways between a PSTN and at least one hub of the system, implementing a proxy table and a call restoration table in each of the plurality of gateways, wherein when a call is received by a gateway in the plurality of gateways from the PSTN, the call is divided into a session initiation protocol (SIP) portion and a real time protocol (RTP) portion, and further wherein the SIP portion is sent to a proxy server and the RTP portion is sent to a media server, both being located in the at least one hub and further routed to an endpoint such as a SIP controlled softphone. A further method of the present invention includes routing SIP voice calls through the plurality of gateways using a proxy server priority table.

Abstract: A system and method for providing location information to a public safety answering point from a softphone may include receiving, at a network access point, an emergency 911 call from the softphone. The emergency 911 call may be communicated to a public safety answering point. In response to a call connection message being received, an address location of the network access point to which the softphone is in communication in placing the emergency 911 call to the public safety answering point may be communicated in a type II caller ID data packet. The softphone may generate the type II caller ID data packet with the address location in a data field, such as a data field typically used for name information of a caller.

Abstract: Given the growing importance of quantitative relationships between user-perceived Quality of Experience (QoE) and network Quality of Service (QoS), this paper investigates the IQX hypothesis for two voice codecs, iLBC and G.711. This hypothesis expresses QoE as an exponential function of QoS degradation. The experiments are carried out in a controlled environment using the softphone SJPhone, the network emulator NIST Net, and a tool calculating the PESQ (Perceptual Evaluation of Speech Quality) from sent and received audio files. The IQX hypothesis is confirmed exactly for disturbances perceived on applications level, packet loss and packet reordering, which clearly correlate to the main sensitivities of the used softphone to packet-level disturbances such as loss, jitter and reordering. So, besides of providing a unified relationship between QoE and QoS, the IQX also proved to be capable of identifying the QoS parameters of relevance for QoE degradations. The study also points out interesting tracks for future work in terms of QoS degradations and related QoE evaluations.

Abstract: The present invention is directed to systems for and methods of using dual mode handsets or softphone client for voice, sms, and data services. In one embodiment of the present invention, a mobile handset uses a SIP User Agent to register on a visiting network. The mobile handset generates SIP REGISTER messages. The SIP REGISTER messages are translated into corresponding MAP registration (or RADIUS message) and authentication commands, allowing system to contact the HPLMN HLR (or home AAA) associated with the mobile device to authenticate the mobile device and register it on a VLR of a visiting network. MAP responses (or RADIUS response) are translated to corresponding SIP commands that are forwarded to the mobile device, thereby completing the connection set up.