Hipcricket, Inc.

I2telecom International, Inc.

Geos Communications IP Holdings, Inc., a Wholly Owned Subsidiary of Augme Technologies, Inc.

Ali Awais

Douglas F Bender

Systems and methods of Voice over Internet Protocol (VoIP) communication are provided. A particular system includes a VoIP endpoint. The VoIP endpoint is adapted to monitor a packet loss measure between the VoIP endpoint and a second VoIP endpoint during a VoIP communication using a first CODEC. The VoIP endpoint is also adapted to negotiate with the second VoIP endpoint to select a substitute CODEC having a nominal data rate that is lower than an in-use nominal data rate of the first CODEC.

Method and system of renegotiating end-to-end voice over internet protocol CODECs

A real-time bandwidth monitor (RTBM) for VoIP applications are disclosed having a media streaming function to sense the available bandwidth between two endpoints of a VoIP communication (herein, a 'call path') utilizing the media streaming function and to adapt in real-time the transmission rate of the media stream to utilize that bandwidth. The media stream may include voice content, video content, or other media content.

System and method of media over an internet protocol communication

Systems and methods of Voice over Internet Protocol (VOIP) communication are provided. A particular system includes a VoIP endpoint. The VoIP endpoint is adapted to monitor a packet loss measure between the VoIP endpoint and a second VoIP endpoint during a VoIP communication using a first CODEC. The VoIP endpoint is also adapted to negotiate with the second VoIP endpoint to select a substitute CODEC having a nominal data rate that is lower than an in-use nominal data rate of the first CODEC.

A system and method for dynamically adapting the transmission rate of packets in real-time voice over IP communications to the available bandwidth. A real-time bandwidth monitor (RTBM) for VoIP applications senses the call path bandwidth between two endpoints of a VoIP communication and adapts in real-time the packet transmission rate to utilize that bandwidth. If sufficient bandwidth is available, the RTBM selects a low compression, low latency CODEC to offer best possible voice quality to the user. If the bandwidth is constrained, the RTBM degrades gracefully by switching to a high compression CODEC. On further bandwidth reduction, the RTBM increases the media frames per packet. Because the bandwidth reduction may be transitory, the RTBM constantly monitors the end-to-end available bandwidth so as to invoke the CODEC/frame per packet combination that provides the best quality of service (QoS) achievable over the current end-to-end available bandwidth.

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Santa Clara, CA, United States of America

Ali Awais