Abstract: Providing origin insight via Session Traversal Utilities for Network Address Translation (STUN) messages for web applications, and related methods, systems, and computer-readable media are disclosed herein. In one embodiment, a method for providing origin insight for web applications via STUN messages comprises determining an origin identifier for a web application attempting to send a STUN message to a STUN server; generating the STUN message incorporating the origin identifier for the web application; and transmitting the STUN message to the STUN server. In another embodiment, a method for providing STUN services to web applications comprises receiving a STUN message from a web client, and extracting an origin identifier for a web application. Provision of a STUN service to the web application is determined based on the origin identifier. If the STUN service should not be provided, the STUN message is rejected or disregarded; otherwise the STUN message is processed.

Abstract: Verification of privacy of Web Real-Time Communications (WebRTC) media channels via corresponding WebRTC data channels, and related methods, systems, and computer-readable media are disclosed. In this regard, in one embodiment, a method for verifying privacy of a WebRTC media channel comprises establishing the WebRTC media channel between first and second WebRTC clients using a keying material. The method further comprises establishing a corresponding WebRTC data channel between the first and second WebRTC clients using the keying material, and negotiating, in the WebRTC data channel, a cryptographic key exchange. The method also comprises generating a first and a second Short Authentication String (SAS) based on the cryptographic key exchange in the WebRTC data channel. The method further comprises displaying the first SAS and the second SAS, such that a mismatch between the first SAS and the second SAS indicates an existence of a man-in-the-middle (MitM) attacker.

Abstract: Virtual Web Real-Time Communications (WebRTC) gateways, and related methods, systems, and computer-readable media are disclosed herein. In one embodiment, a method for providing a virtual WebRTC gateway comprises instantiating a virtual WebRTC agent corresponding to a WebRTC client, and instantiating a virtual non-WebRTC agent corresponding to a non-WebRTC client. The method further comprises establishing a WebRTC interactive flow between the virtual WebRTC agent and the WebRTC client, and establishing a non-WebRTC interactive flow between the virtual non-WebRTC agent and the non-WebRTC client. The method also comprises directing a content of the WebRTC interactive flow to the non-WebRTC interactive flow, and a content of the non-WebRTC interactive flow to the WebRTC interactive flow, via the virtual WebRTC agent and the virtual non-WebRTC agent. In this manner, the virtual WebRTC gateway may provide interoperability between otherwise-incompatible WebRTC clients and non-WebRTC clients.

Abstract: Back-to-back Web Real-Time Communication (WebRTC) virtual agents, and related methods, systems, and computer-readable media are disclosed herein. In one embodiment, a method for providing back-to-back virtual WebRTC agents comprises receiving, by a WebRTC server executing on a computing device, a WebRTC offer/answer exchange between first and second WebRTC clients. The method further comprises instantiating one or more virtual WebRTC agents. The method also comprises establishing a first WebRTC interactive flow between the first WebRTC client and one of the one or more virtual WebRTC agents, and a second WebRTC interactive flow between the second WebRTC client and one of the one or more virtual WebRTC agents. The method additionally comprises directing a content of the first WebRTC interactive flow to the second WebRTC interactive flow, and a content of the second WebRTC interactive flow to the first WebRTC interactive flow, via the one or more virtual WebRTC agents.

Abstract: Scalable Web Real-Time Communication (WebRTC) media engines, and related methods, systems, and computer-readable media, are disclosed herein. In one embodiment, a method for providing a scalable WebRTC media engine comprises instantiating one or more virtual WebRTC agents, each corresponding to one or more of a plurality of WebRTC clients. The method further comprises establishing a plurality of WebRTC interactive flows, each connecting one of the one or more virtual WebRTC agents with the corresponding one or more of the plurality of WebRTC clients. The method also comprises receiving contents of the plurality of WebRTC interactive flows as input from the one or more virtual WebRTC agents, and synthesizing the contents of the plurality of WebRTC interactive flows. The method additionally comprises directing the synthesized contents as output to one of more of the plurality of WebRTC interactive flows via the corresponding one or more virtual WebRTC agents.

Abstract: A method and system of enabling communication between peers in a peer-to-peer (P2P) network in the presence of network address translators (NAT). The method includes: assigning a unique peer identifier to each one of the plurality of peers in the P2P network; establishing an overlay topology connecting each peer to a subset of the plurality of peers in a structured connection pattern based on the unique peer identifier of each peer, each one of the plurality of peers having more connections to close peers and fewer connections to distant peers; and routing messages between peers through the established overlay topology.

Abstract: A method for performing mode-agile communications during a communications session is disclosed. A communications terminal device supporting multiple modes of access is configured to determine when, during a session involving a first mode of access communications, a second mode of access is available and is to be used for subsequent communications for the session. The communications terminal device initiates or controls switchover from one mode of access to another.

Abstract: Distributed application of enterprise policies to WebRTC interactive sessions, and related methods, systems, and computer-readable media are disclosed. In this regard, in one embodiment, a method for applying an enterprise policy to a WebRTC interactive session comprises receiving, by a distributed policy enforcement agent of a recipient device, a WebRTC session description object directed to the recipient device originating from a sender device via a secure network connection. The method further comprises determining, by the distributed policy enforcement agent, one or more enterprise policies based on the WebRTC session description object. The method additionally comprises applying the one or more enterprise policies to the WebRTC session description object. In this manner, an enterprise may permit establishment of a WebRTC interactive session that crosses an enterprise network boundary, while at the same time ensuring that the WebRTC interactive session complies with the one or more enterprise policies.

Abstract: An approach provides interdomain traversal to support packetized voice transmissions. A centralized data store, maintained by a service provider, stores one or more user identifiers and an associated directory number. The centralized data store also stores routing information including one or more communication paths corresponding to the user identifier, wherein the routing information includes a data path or a circuit-switched path for establishing a call to the user. In response to a request for establishing the call to the directory number, the data store retrieves the routing information for use to establish the call.

Abstract: An approach provides interdomain traversal to support packetized voice transmissions. A request for establishing a voice call is received from a source endpoint behind a first network address translator of a first domain, wherein the request specifies a directory number of a destination endpoint within a second domain. A network address is determined for communicating with the destination endpoint based on the directory number. Additionally, existence of a second network address translator within the second domain is determined. If the network address can be determined, a media path is established between the source endpoint and the destination endpoint based on the network address to support the voice call.

Abstract: An approach for providing a quality indicator in support of a communication session between a near end station and a far end station over a data network. The quality of the communication session in a direction of the near end station sending to the far end station is determined. A message containing statistics associated with the communication session is transmitted according to a prescribed protocol to the near end station to notify the near end station of the quality of the communication session. The prescribed protocol supports real-time data exchange. The present invention has particular applicability to SIP (Session Initiation Protocol) IP (Internet Protocol) telephony services.

Abstract: A joining peer is bootstrapped into in a peer-to-peer (P2P) network in the presence of network address translators (NAT). A bootstrap peer is contact by the joining peer to establish a connection between the joining peer and an admitting peer in the peer-to-peer network. The joining peer receives an indication from the bootstrap peer that the connection between the joining peer and the admitting peer has been established with a message path through the bootstrap peer. The indication includes contact information for the admitting peer. The connection is them modified to remove all proxies therefrom.

Abstract: An approach provides interdomain traversal to support packetized voice transmissions. A request for establishing a voice call is received from a source endpoint behind a first network address translator of a first domain, wherein the request specifies a directory number of a destination endpoint within a second domain. A network address is determined for communicating with the destination endpoint based on the directory number. Additionally, existence of a second network address translator within the second domain is determined. If the network address can be determined, a media path is established between the source endpoint and the destination endpoint based on the network address to support the voice call.

Abstract: An approach provides interdomain traversal to support packetized voice transmissions. A request for establishing a voice call is received from a source endpoint behind a first network address translator of a first domain, wherein the request specifies a directory number of a destination endpoint within a second domain. A network address is determined for communicating with the destination endpoint based on the directory number. Additionally, existence of a second network address translator within the second domain is determined. If the network address can be determined, a media path is established between the source endpoint and the destination endpoint based on the network address to support the voice call.

Abstract: An approach for providing a quality indicator in support of a communication session between a near end station and a far end station over a data network. The quality of the communication session in a direction of the near end station sending to the far end station is determined. A message containing statistics associated with the communication session is transmitted according to a prescribed protocol to the near end station to notify the near end station of the quality of the communication session. The prescribed protocol supports real-time data exchange. The present invention has particular applicability to SIP (Session Initiation Protocol) IP (Internet Protocol) telephony services.

Abstract: A method for performing mode-agile communications during a communications session is disclosed. A communications terminal device supporting multiple modes of access is configured to determine when, during a session involving a first mode of access communications, a second mode of access is available and is to be used for subsequent communications for the session. The communications terminal device initiates or controls switchover from one mode of access to another.

Abstract: A method for performing mode-agile communications during a communications session is disclosed. A communications terminal device supporting multiple modes of access is configured to determine when, during a session involving a first mode of access communications, a second mode of access is available and is to be used for subsequent communications for the session. The communications terminal device initiates or controls switchover from one mode of access to another.

Abstract: An approach for providing voice processing over a packet switched network is disclosed. A communication system includes a special forking proxy server that receives a message for establishing a session between a calling user agent and a called user agent. The message has a first payload corresponding to a first call leg. The system also includes a bridging user agent that communicates with the first network element and generates a response that contains the first payload and a second payload that corresponds to a second call leg. The bridging user agent is configured to bridge the first call leg and the second call leg to establish a session between the calling user agent and the called user agent. The special forking proxy server is configured to fork an acknowledgement message from the calling party to acknowledge the session establishment by the called user agent. The present invention has particular applicability to SIP (Session Initiation Protocol) IP (Internet Protocol) telephony services.

Abstract: A method and system of enabling communication between peers in a peer-to-peer (P2P) network in the presence of network address translators (NAT). The method includes: assigning a unique peer identifier to each one of the plurality of peers in the P2P network; establishing an overlay topology connecting each peer to a subset of the plurality of peers in a structured connection pattern based on the unique peer identifier of each peer, each one of the plurality of peers having more connections to close peers and fewer connections to distant peers; and routing messages between peers through the established overlay topology.

Abstract: A micro-electromechanical systems (MEMS) disc drive includes high-precision and integrated components to allow for increased functionality, robustness and reduced size as compared to currently produced disc drives. Integrating multiple subcomponents of the disc drive using batch processing provides low manufacturing costs. Furthermore, using MEMS techniques, new features can be added to disc drives. For example, an environmental control component, an accelerometer and/or a thermometer may be integrated into the housing of the disc drive.