A method for variably controlling the number of packets constituting a voice data frame according to the traffic condition to secure the real-time transmission of voice data. The method comprises the steps of: upon the receipt of a voice data frame, storing a stamp time of the voice data frame and the number of packets constituting the voice data frame; removing an RTP header included in the voice data frame and storing the RTP header-removed voice data in a receiving buffer; calculating an anticipated delay time depending on the stored packet number when there is a previously received frame; calculating an error time depending on the difference between an actual delay time and the anticipated delay time; increasing the number of packets constituting the transmission frame when the error time is greater than a threshold value; decreasing the number of packets constituting the transmission frame when the error time is less than the threshold value; updating the DSP component data with the changed packet number; setting a valid flag in a transmission buffer when the updated number of packets is written in the transmission buffer; and, reading the updated number of packets from the transmission buffer to assemble a transmission frame.
A low speed encoding method based on Internet protocol (IP) includes the steps of determining speech characteristic parameters in TN duration, determining an optimized frame length T for successive speech data processing according to the characteristic parameters, making compressed encoding of the speech data in every T, assembling a packet of the encoded bits with TCP and UDP, again assembling a packet of the assembled bits with IP, and finally outputting the channel. The method uses a single frame, variable length frame, intra-frame adaptive low speed speech encoding method, which has the advantages of reducing the bit rate and raising transmission efficiency. The method takes an optimized length encoded frame as a unit to break the IP datagram, and therefore raises encoding and decoding quality of the speech data greatly. Informal tests show that the method can raise a MOS (mean opinion score) value from 0.1 to 0.2.
An integrated VoIP unified message processing system includes a voice platform that processes data in native VoIP format. There is no use of hardware telephone interface cards (TICs) or software transcoding to transform data to PCM or other formats. Cost reductions are achieved by the elimination of expensive dedicated hardware and scalability is achieved by obviating the need for software transcoding.
A method and apparatus negotiates a maximum frame size to be used over a frame relay network. A local maximum frame size is identified by a first endpoint device of a frame relay network so that other frames sent and received using that size will not cause other frames sent by the network device to be sent a period of time exceeding an acceptable delay after the other frames are received. The acceptable delay is the lowest acceptable delay among originators and recipients of frames that use the endpoint device. The frame size identified is transmitted to other endpoint devices that can communicate with the first endpoint device. These other endpoint devices identify acceptable delays that correspond to the originators and recipients of such endpoint devices.
