Abstract: Device, system, and method of Voice over Internet Protocol (VoIP) communications, and particularly of Real Time Protocol (RTP) communication. In order to improve quality-of-service or quality-of-experience for a group of VoIP calls that are served by a VoIP router, each VoIP transmitter implements and adds a pseudo-random waiting-period prior to transmitting each outgoing RTP packet, or otherwise re-orders or mixes or shuffles the order of channels of RTP packets that are buffered or queued for transmission. Accordingly, no particular VoIP channel suffers from repeated drops of its RTP packets at the VoIP router. Additionally, VoIP network analyzers operate to measure the overall VoIP network overuse, or the average RTP packet loss rate of multiple VoIP channels, based on measuring RTP packet loss rate of a single VoIP channel which enforces a random pre-transmission waiting-period.

Abstract: Device, system, and method of Voice over Internet Protocol (VoIP) communications, and particularly of Real Time Protocol (RTP) communication. In order to improve quality-of-service or quality-of-experience for a group of VoIP calls that are served by a VoIP router, each VoIP transmitter implements and adds a pseudo-random waiting-period prior to transmitting each outgoing RTP packet, or otherwise re-orders or mixes or shuffles the order of channels of RTP packets that are buffered or queued for transmission. Accordingly, no particular VoIP channel suffers from repeated drops of its RTP packets at the VoIP router. Additionally, VoIP network analyzers operate to measure the overall VoIP network overuse, or the average RTP packet loss rate of multiple VoIP channels, based on measuring RTP packet loss rate of a single VoIP channel which enforces a random pre-transmission waiting-period.

Abstract: Device, system, and method of Voice over Internet Protocol (VoIP) communications, and particularly of Real Time Protocol (RTP) communication. In order to improve quality-of-service or quality-of-experience for a group of VoIP calls that are served by a VoIP router, each VoIP transmitter implements and adds a pseudo-random waiting-period prior to transmitting each outgoing RTP packet, or otherwise re-orders or mixes or shuffles the order of channels of RTP packets that are buffered or queued for transmission. Accordingly, no particular VoIP channel suffers from repeated drops of its RTP packets at the VoIP router. Additionally, VoIP network analyzers operate to measure the overall VoIP network overuse, or the average RTP packet loss rate of multiple VoIP channels, based on measuring RTP packet loss rate of a single VoIP channel which enforces a random pre-transmission waiting-period.

Abstract: Device, system, and method of Voice over Internet Protocol (VoIP) communications, and particularly of Real Time Protocol (RTP) communication. In order to improve quality-of-service or quality-of-experience for a group of VoIP calls that are served by a VoIP router, each VoIP transmitter implements and adds a pseudo-random waiting-period prior to transmitting each outgoing RTP packet, or otherwise re-orders or mixes or shuffles the order of channels of RTP packets that are buffered or queued for transmission. Accordingly, no particular VoIP channel suffers from repeated drops of its RTP packets at the VoIP router. Additionally, VoIP network analyzers operate to measure the overall VoIP network overuse, or the average RTP packet loss rate of multiple VoIP channels, based on measuring RTP packet loss rate of a single VoIP channel which enforces a random pre-transmission waiting-period.

Abstract: Device, system, and method of Voice over Internet Protocol (VoIP) communications, and particularly of Real Time Protocol (RTP) communication. In order to improve quality-of-service or quality-of-experience for a group of VoIP calls that are served by a VoIP router, each VoIP transmitter implements and adds a pseudo-random waiting-period prior to transmitting each outgoing RTP packet, or otherwise re-orders or mixes or shuffles the order of channels of RTP packets that are buffered or queued for transmission. Accordingly, no particular VoIP channel suffers from repeated drops of its RTP packets at the VoIP router. Additionally, VoIP network analyzers operate to measure the overall VoIP network overuse, or the average RTP packet loss rate of multiple VoIP channels, based on measuring RTP packet loss rate of a single VoIP channel which enforces a random pre-transmission waiting-period.

Abstract: Device, system, and method of Voice over Internet Protocol (VoIP) communications, and particularly of Real Time Protocol (RTP) communication. In order to improve quality-of-service or quality-of-experience for a group of VoIP calls that are served by a VoIP router, each VoIP transmitter implements and adds a pseudo-random waiting-period prior to transmitting each outgoing RTP packet, or otherwise re-orders or mixes or shuffles the order of channels of RTP packets that are buffered or queued for transmission. Accordingly, no particular VoIP channel suffers from repeated drops of its RTP packets at the VoIP router. Additionally, VoIP network analyzers operate to measure the overall VoIP network overuse, or the average RTP packet loss rate of multiple VoIP channels, based on measuring RTP packet loss rate of a single VoIP channel which enforces a random pre-transmission waiting-period.

Abstract: The present invention includes devices, systems, and methods of Voice-over-Internet Protocol (VoIP) communication. For example, a method includes: receiving a data stream comprising a set of VoIP packets; and modifying a Real Time Protocol (RTP) header of at least one of said VoIP packets to modify a jitter buffer delay of said data stream. Optionally, the method includes decreasing the jitter buffer delay by: dropping at least one packet from said data stream; and decreasing a sequence number and a timestamp value in an RTP header of at least one additional packet subsequent to said at least one packet. Optionally, the method includes increasing the jitter buffer delay by: identifying a pair of consecutive packets in the incoming data stream, the pair of consecutive packets having consecutive sequence numbers; and increasing a sequence number in an RTP header of at least a latter packet in said pair of consecutive packets.

Abstract: The present invention includes devices, systems, and methods of Voice-over-Internet Protocol (VoIP) communication. For example, a method includes: receiving a data stream comprising a set of VoIP packets; and modifying a Real Time Protocol (RTP) header of at least one of said VoIP packets to modify a jitter buffer delay of said data stream. Optionally, the method includes decreasing the jitter buffer delay by: dropping at least one packet from said data stream; and decreasing a sequence number and a timestamp value in an RTP header of at least one additional packet subsequent to said at least one packet. Optionally, the method includes increasing the jitter buffer delay by: identifying a pair of consecutive packets in the incoming data stream, the pair of consecutive packets having consecutive sequence numbers; and increasing a sequence number in an RTP header of at least a latter packet in said pair of consecutive packets.

Abstract: The present invention includes devices, systems, and methods of Voice-over-Internet Protocol (VoIP) communication. For example, a method includes: receiving a data stream comprising a set of VoIP packets; and modifying a Real Time Protocol (RTP) header of at least one of said VoIP packets to modify a jitter buffer delay of said data stream. Optionally, the method includes decreasing the jitter buffer delay by: dropping at least one packet from said data stream; and decreasing a sequence number and a timestamp value in an RTP header of at least one additional packet subsequent to said at least one packet. Optionally, the method includes increasing the jitter buffer delay by: identifying a pair of consecutive packets in the incoming data stream, the pair of consecutive packets having consecutive sequence numbers; and increasing a sequence number in an RTP header of at least a latter packet in said pair of consecutive packets.

Abstract: The present invention includes devices, systems, and methods of Voice-over-Internet Protocol (VoIP) communication. For example, a method includes: receiving a data stream comprising a set of VoIP packets; and modifying a Real Time Protocol (RTP) header of at least one of said VoIP packets to modify a jitter buffer delay of said data stream. Optionally, the method includes decreasing the jitter buffer delay by: dropping at least one packet from said data stream; and decreasing a sequence number and a timestamp value in an RTP header of at least one additional packet subsequent to said at least one packet. Optionally, the method includes increasing the jitter buffer delay by: identifying a pair of consecutive packets in the incoming data stream, the pair of consecutive packets having consecutive sequence numbers; and increasing a sequence number in an RTP header of at least a latter packet in said pair of consecutive packets.