Cisco Technology, Inc.

Manjunath Sreedhara Bangalore

Parameswaran Kumarasamy

Kavithadevi Parameswaran

Shantanu Sarkar

Vineet Goel

Dhaval N Shah

Randall Bernard Baird

Xiaode Xu

Steven G Fry

Dhaval Bhupendrabhai Shah

Kan Wu

Sravan Vadlakonda

Prasad Miriyala

Scott S Firestone

Aseem Asthana

Amit Surendra Singh

Thiyagesan Ramalingam

Binh Don Ha

Soumya Kumar Kalahasti

Raman Shankara Krishnan

Prabandham Madan Gopal

Vikram Khurana

Ryan Knotts

Madhavi Kadiyala

Son D Nguyen

Connie Tang

Sanjay Sinha

Various embodiments provide methods and systems operable to provide access to real time control protocol (RTCP) information for improved media quality control. An example embodiment includes a message processor to receive a message, the message including information indicative of an RTP port identifier, and to add to the received message information indicative of an auxiliary RTCP port identifier; and a message communication component to communicate the information indicative of an auxiliary RTCP port identifier to a node.

Method and apparatus for providing access to real time control protocol information for improved media quality control

In one embodiment, a method includes receiving at a protocol translator a first message from a first endpoint to a second endpoint. The first message includes a list of media capabilities of the first endpoint. The method also includes transmitting from the protocol translator a second message to the second endpoint including a media capabilities request header and receiving at the protocol translator a response message from the second endpoint. The response message includes a response header including a list of the media capabilities of the second endpoint.

Terminal capabilities set exchange between heterogeneous endpoints

In one embodiment, a session is refreshed using a message that is simplified for refresh purposes. The refresh message that is used is a message used for another purpose in a protocol (e.g., SIP). For example, a reINVITE message may be used as a refresh message. If the reINVITE message is used for its intended purpose, the reINVITE message may be used to change the characteristics of a session. This reINVITE message includes an SDP and full header such that the parameters can be properly changed. However, the reINVITE message may be used to refresh a session. This may be referred to as a simple refresh message. But, the simple refresh message does not include an SDP or any other message body. The simple refresh message includes only a reduced header. This reduces time spent building and parsing the simple refresh message.

Refreshing a session initiation protocol (SIP) session

A unified access control component (UACC) can maintain information relating to network access information and physical location information associated with respective users who may access a network that can include network resources (e.g., applications, information). The UACC can cross reference the network access information (e.g., user network access events, credentials, and policy) and physical location information (e.g., user physical access events, credentials, and policy) and can generate and enforce a unified network access policy based on network access information and physical location information associated with a particular user. After network access privileges have been granted to a user, the UACC can continue to monitor the user. The UACC can include a dynamic authentication component that can request a user re-authenticate if a change in the physical location and/or network access associated with the user is detected, such that a re-computation of network access privileges is desired.

Physical security triggered dynamic network authentication and authorization

In one embodiment, a method is provided, including: initiating an interactive voice response (IVR) session between a first endpoint and an IVR system over an IP network; establishing a first media path between the first endpoint and a second endpoint; establishing a second media path between the first endpoint and the IVR system; and receiving at the IVR system a notice of a telephony event from the first endpoint. In another embodiment, a method is provided, including: initiating an interactive voice response (IVR) session between a first endpoint and an IVR system over an IP network; establishing a secure media path between the first endpoint and a second endpoint; and receiving an unsecured notice of a telephony event from the first endpoint.

Interactive voice response and secure media system integration

A conferencing system and method includes, during the conference session, invoking an interactive voice response (IVR) routine that provides names of one or more conference participants to a user of an audio-only endpoint device responsive to a request from the user to create a sidebar session. An invitation to join the sidebar session is then communicated to each of one or more participants selected by the user, the invitation being communicated via a private media channel separate from a media stream associated with the conference session. 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.

Method and apparatus for inviting non-rich media endpoints to join a conference sidebar session

Embodiments of the present invention provide techniques for refreshing a session. A session is refreshed using a refresh message that is enhanced for a refresh. For example, the refresh message does not include an SDP or any other message body. The message includes only a reduced header. This reduces time spent building and parsing the refresh message. Also, less bandwidth is used in sending the refresh messages.

Enhanced refresh in SIP network

A node for routing of calls in a network has an interface coupled to the network and at least one processor operable to route a packet-based call to a telephony destination in accordance with a protocol that includes a set of attributes that describe packet-network routing characteristics of one or more Internet Protocol (IP)-IP gateway devices in the network. The attributes are used by the at least one processor to specify a call route through an IP-IP gateway device for the packet-based call. The set of attributes include a first attribute that identifies a total administratively provisioned bandwidth capacity available on a given call route to accommodate application traffic, and a second attribute that identifies a current bandwidth that is available on the given call route to accommodate the application traffic at a given point in time. 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.

Routing protocol with packet network attributes for improved route selection

In one embodiment, a method includes dropping an initial series of video frames received by a video conferencing system from a first endpoint that has entered a video conference session as a first participant. One or more fast video update (FVU) messages received by the video conferencing system from the first endpoint during an enabling time period are then discarded. A single FVU message is then to the first endpoint after the video conferencing system has become enabled with respect to the first endpoint. An I-frame received from the first endpoint in response to the single FVU message is then sent back to the first endpoint. 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.

Minimizing fast video update requests in a video conferencing system

In one embodiment, a method includes correlating, for each intermediate device associated with a Real-time Transport Protocol (RTP) application that connects endpoints through a plurality of intermediate devices across a network, one or more first RTP segments input into the intermediate device with one or more second RTP segments output from the intermediate device. The correlation results are then graphically displayed results of the correlating to obtain a full, end-to-end picture of RTP quality from a source endpoint to a destination endpoint of the RTP application. 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.

Diagnostic tool for troubleshooting multimedia streaming applications

In one embodiment, a system includes a video mixer coupled with an audio mixer for exchange of information that includes a first set of delay values respecting input audio streams received by the audio mixer from a plurality of source endpoints, and output audio streams sent from the audio mixer to a plurality of destination endpoints. The information further including a second set of delay values respecting the corresponding input video streams. The audio mixer calculates end-to-end video delays, and the video mixer calculates end-to-end audio delays. The audio mixer delays the output audio streams to equalize the end-to-end audio and video delays in the event that the end-to-end audio delays are less than the end-to-end video delays, and the video mixer delays the output video streams to equalize the end-to-end audio and video delays in the event that the end-to-end video delays are less than the end-to-end audio delays.

Lip synchronization for audio/video transmissions over a network

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.

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Sunnyvale, CA, United States of America

Manjunath Sreedhara Bangalore