Abstract: The embodiments of this application provide a media packet forwarding method and a forwarding server. The method includes determining, by the forwarding server after receiving a to-be-forwarded RTP media packet, a type of the to-be-forwarded RTP media packet, then determining a priority of the to-be-forwarded RTP media packet according to the type of the to-be-forwarded RTP media packet, and forwarding an RTP media packet having a higher priority, selecting between the received to-be-forwarded RTP media packet and an RTP media packet being currently forwarded.

Abstract: The invention relates to wireless repeater systems and methods. In embodiments, such systems and methods involve receiving a wireless transmission signal; and processing the wireless transmission signal using a digital signal processing facility (DSP); wherein the DSP is adapted to filter at least one sub-band of the wireless transmission signal using a digital bandpass filter.

Abstract: Embodiments relate to a configurable convolution engine that receives configuration information to perform convolution and other deep machine learning operations on streaming input data of various formats. The convolution engine may include two convolution circuits that each generate a stream of values by applying convolution kernels to input data. The stream of values may each define one or more channels of image data. A channel merge circuit combines the streams of values from each convolution circuit in accordance with a selected mode of operation. In one mode, the first and second streams from the convolution circuits are merged into an output stream having the combined channels of the first and second streams in an interleaved manner. In another mode, the first stream from the first convolution circuit is fed into the input of the second convolution circuit.

Abstract: A method of initializing, provisioning, and managing a cable modem and a customer premise equipment device includes sending a customized configuration file to the cable modem. The configuration file contains service provisioning information and further includes information indicative of a device-to-device communication configuration. A message is passed from the cable modem to the customer premise equipment device indicative of the device-to-device communication configuration. The customer premise equipment device is initialized for device-to-device communication with any other customer premise equipment devices in accordance with the device-to-device communication configuration indicated in the message, which relates to physical layer and data link layer communications among customer premise equipment devices. In this way, the customer premise equipment device knows how to communicate with other customer premise equipment devices, for example, at the same subscriber location.

Abstract: Disclosed is an apparatus, method and computer program which frequency shifts a received signal by a first frequency value and low-pass filters the frequency shifted received signal at a first cutoff frequency to filter out a plurality of adjacent carrier frequencies. Parallel baseband processing is then performed on two or more adjacent carrier frequencies among the plurality of adjacent carrier frequencies based upon an adjacent carrier power criterion. In certain examples, the parallel baseband processing is adapted for detecting a frequency correction burst in two or more adjacent Global System for Mobile Communications carrier frequencies or an Enhanced Data Rates for Global System for Mobile Communications Evolution carrier frequencies.

Abstract: A handover with carrier aggregation is provided.

Abstract: A media transmission method and apparatus enable direct media transmission between terminals in a network by providing each terminal with information about the other terminal during signaling for media transmission, so that data transformation in an exchange system is not needed. As a result, media transmission quality is improved, and transmission delay is reduced.

Abstract: The invention relates to a system for connecting a first client entity (10) to a second client entity (11) in a communications network. Said first client entity has subscribed to a connection service. A routing module routes via a connecting device (50) a first call made by the second client entity to the first client entity and redirected to a third client entity (12). Said connecting device inserts a linking module (40) for the interception of said first call. A first branch is established between the second client entity and the linking module and a second branch is established between the linking module and the third client entity. The connecting device then detects a second call associated to the connection service made by the first client entity via the linking module, and then connects the first and second client entities by connecting the first branch and this second call by means of the linking module.

Abstract: This disclosure provides a method for performing remote network management on a voice home gateway terminal, and the method includes that: a remote network management server transmits voice signaling to the voice home gateway terminal; the voice home gateway terminal determines whether the voice signaling contains a non-voice triggering identifier, processes a voice service according to the voice signaling when no non-voice triggering identifier is contained, requests non-voice configuration information from the remote network management server when a non-voice triggering identifier is contained, and processes a non-voice service according to the received non-voice configuration information. This disclosure further provides a system for performing remote network management on a voice home gateway terminal. Application of the method and system enables the remote network management server to manage simultaneously a voice service and other non-voice services.

Abstract: A method for adaptive communication in an HF frequency band includes at least one of the following steps: the determination of a payload subband S that is available for a transmission in the HF band, the selection in the payload subband S of a set of n frequency channels of identical width according to a frequency allocation plan and the quality of the link of each of said channels, or the simultaneous transmission on the n frequency channels of a signal complying with an HF waveform.

Abstract: A special rendering mode for the first few seconds of play out of multimedia data minimizes the delay caused by pre-buffering of data packets in multimedia streaming applications. Instead of pre-buffering all incoming data packets until a certain threshold is reached, the streaming application starts playing out some of the data packets immediately after the arrival of the first data packet. Immediate play out of the first data packet, for example, results in minimum delay between channel selection and perception, thereby allowing a user to quickly scan through all available channels to quickly get a notion of the content. The immediate play out is done at a reduced speed.

Abstract: A tone monitoring apparatus comprises a tone detector arranged to detect, when in use, a tone having a first duration equal to or greater than a predetermined duration. The tone detector is further arranged to select, when in use, a first signal path between a channel input and a channel output, the selection being in response to the first duration of the tone being equal to or greater than the predetermined duration.

Abstract: The present invention concerns a receiver and a method for retrieving data from Ultra wideband radio transmission signals received by a receiver and transmitted in packets containing at least a preamble known to the receiver and a payload containing data unknown to the receiver. The payload data is formed of at least one burst containing at least one pulse. The method comprises the steps of receiving a signal corresponding to the preamble of a packet by a receiver; determining the energy of the received signal; determining weighting coefficients from the energy of the received signal; receiving a signal corresponding to the payload of a packet by a receiver; determining the energy of the received signal; calculating a threshold depending on the weighting coefficients; and applying a decision rule based on the sampled energy, weighted by said weighting coefficients and depending on said calculated threshold.

Abstract: A system for adapting an analog system to a packet network may include an electronic signal adapter in communication with the analog system and packet network. The electronic signal adapter may be configured to convert information received in data packets from the packet network into dual-tone multiple frequency (DTMF) signals and single tone analog signals and to communicate the DTMF and single tone analog signals to the analog system. The single tones may have a signal strength below approximately ?25 dBm. The electronic signal adapter may further be configured to receive and convert analog signals into digital data for communication in data packets over a packet network.

Abstract: It is made possible for a DTMF tone signal associated with a button depressed on an IP telephone terminal to arrive at the opposite party even if an IP telephone terminal does not have a function of sending a DTMF tone signal. If a DTMF tone signal transmission request signal detection unit in a proxy detects a DTMF tone signal transmission request signal inserted in an RTP packet or a SIP packet received from an IP telephone terminal or detects a DTMF digit signal contained in the DTMF tone signal transmission request signal, a DTMF tone signal generation unit generates a DTMF tone signal associated with the DTMF digit signal. A voice encoder encodes the DTMF tone signal in the same way as a voice signal. An RTP payload replacement unit replaces a voice signal in a payload of the RTP packet with the encoded DTMF tone signal.

Abstract: An apparatus for switching a user between an IP network and a PSTN is disclosed in the present invention and includes a user port, a Subscriber Line Interface Circuit, a detection circuit, a PSTN port, an IP network interface circuit and an action unit. A method for switching a user between the IP network and the PSTN is also disclosed. According to the method, the action unit is controlled by the control signal from a CPU, thereby implementing the user switching between the IP network and the PSTN on the apparatus. The apparatus and the method enable one or more users to switch between the IP network and the PSTN, implements flexible user selection between the two networks and offers user communication backup when failure occurs. Accordingly, the present invention increases the satisfaction of users and the core competence of enterprises.

Abstract: An apparatus and method are provided which preferably increase reliability of data sensitive to packet loss. According to principles of the invention, a quality of service value may be dynamically assigned to network packets based on the contents of the network packets.

Abstract: A channel estimator and a related method for smoothing channel responses of a multi-carrier system. The channel estimator includes a channel response computing circuit for computing at least one of a target channel estimation and a reference channel estimation by utilizing a computing algorithm, wherein the target channel estimation and the reference channel estimation corresponds to a sub-carrier channel respectively.

Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

Abstract: The method provides a method and device for detecting in a single or multi frequency signal, one or more frequencies from a predetermined set of frequencies. The Signal is subjected to a complex filter substantially passing all of the frequencies in the predetermined set of frequencies. For each of one or more pairs of members of the complex filtered signal, a complex number Yd is determined having a phase indicative of a phase difference between the two members of the pair. The one or more frequencies are then determined based upon the one or more complex numbers. The method and device of the invention may be used in an apparatus such as a call indicator (CI) detector or a DTMF detector.

Abstract: Tone detection includes receiving a series of frames having a signal containing a tone and locating a first frame in the series of frames in which there is at least a partial, tone. The signal energy of the first frame may then be compared to the signal energy of a full tone frame or compared to the signal energy of a frame. A start time of the tone in the signal may then be determined. An end time of the tone in the signal may be determined by locating a last frame in the series of frames. The duration of the tone signal may be determined based on the determined start time and end time of the tone in the signal.

Abstract: A spread-spectrum preamble synchronization peak detection system performs multiple statistical tests based on instant and time-averaged channel condition measurements to identify the synchronization peak. In a normalized peak-to-average test, a peak-to-average ratio measurement is normalized by a signal-to-noise ratio measurement to form a new statistical measure which effectively eliminates the impact of the wide dynamic range of the signal-to-noise ratio of the received samples. A transition SNR test is used to eliminate potential false alarms caused by spurious PARN peaks during the transition period at the onset of preamble arrival. Code-phase aligned time-averaging is used to estimate the signal and noise levels over a sliding window. The code-phase alignment of samples effectively separates signal and noise samples in the averaging process, and resulting in more accurate signal and noise measurements.

Abstract: The present invention is a system and method that improves upon voice activity detection by packetizing actual noise signals, typically background noise. In accordance with the present invention an access network receives an input voice signal (including noise) and converts the input voice signal into a packetized voice signal. The packetized voice signal is transmitted via a network to an egress network. The egress network receives the packetized voice signal, converts the packetized voice signal into an output voice signal, and outputs the output voice signal. The egress network also extracts and stores noise packets from the received packetized voice signal and converts the packetized noise signal into an output noise signal. When the access network ceases to receive the input voice signal while the call is still ongoing, the access network instructs the egress network to continually output the output noise signal.

Abstract: OFDM signals each including a plurality of subcarriers (including data carriers and pilot carriers) orthogonal to each other are received by a plurality of receiver antennas. The received OFDM signals are each OFDM-demodulated into a plurality of receive subcarriers. Based on the plurality of receive subcarriers, the propagation coefficients of a plurality of space paths are estimated. An inverse matrix of a propagation matrix whose elements are the estimated propagation coefficients is calculated. Interference cancellation is performed for the plurality of receive subcarriers by using the inverse matrix, and transmit subcarriers multiplexed in space are estimated. A reliability of the inverse matrix is calculated. A pilot carrier is extracted from the estimated transmit subcarriers, and the extracted pilot carrier is weighted according to the reliability. An error included in the plurality of receive subcarriers is corrected based on the weighted pilot carrier.

Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

Abstract: A method and system for providing window shaping for multiline transmission in a communications system are described. A last portion of a prefix symbol is extracted. A prefix with a prefix window is shaped to create a shaped prefix. A DMT symbol is shaped with a DMT window to create a shaped DMT symbol. The shaped DMT symbol is combined with the shaped prefix.

Abstract: A system for processing a data packet is disclosed and may include at least one processor that enables receiving of a data packet at a station on a network, the data packet having a preamble which includes a destination tag and a training sequence. The at least one processor may enable obtaining a channel model using the training sequence, and encoding each of one or more addresses that the station receives with the channel model to produce a result. The at least one processor may also enable comparing the result with the destination tag. The at least one processor may enable convolving of each of the one or more addresses that the station receives with the channel model to produce the result.

Abstract: A communication device, such as a Voice over Internet Protocol (VoIP) gateway, determines a duration for Dual Tone Multi-Frequency (DTMF) tone portions of telephony signal. If the duration is less than a pre-determined amount, a minimum duration is enforced during DTMF playback at a remote end of a network connection connected to a destination gateway. Minimum playback duration can be enforced at the terminating gateway—however, the originatinggateway can also encode a DTMF packet with a minimum duration value. At the terminating receiver it is not always possible to playback exactly what happened at the originating point in the same time frame. One solution to is to, at the terminating gateway, drop the first portion of voice packets that overlap with the end portion of played back DTMF tones.

Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

Abstract: Carrier frequency selection and handoff initiation methods in wireless communications systems employing multiple carrier frequencies are described. Although the receiver is tuned to a single band, based on the relative energy of one or more beacon signal components corresponding to the currently used carrier and one or more beacon signal components corresponding to an alternative carrier, a carrier selection and a handoff determination is made. Mobile nodes can use a single RF chain with a controllable RF filter to receive and process a signal within a first selected carrier band including components from different transmitters, e.g., a first signal component identified with the first currently selected band and a second signal component identified with a second alternative band. The signal components, e.g., beacon signal components from different transmitters may be obtained from a signal which corresponds to multiple symbol transmission time periods.

Abstract: A network interface is presented that receives packet data from a shared medium and accomplishes the signal processing required to convert the data packet to host computer formatted data separately from receiving the data packet. The network interface receives the data packet, converts the analog signal to a digitized signal, and stores the resulting sample packet in a storage queue. An off-line processor, which may be the host computer itself, performs the signal processing required to interpret the sample packet. In transmission, the off-line process converts host-formatted data to a digitized version of a transmission data packet and stores that in a transmission queue. A transmitter converts the transmission data packet format and transmits the data to the shared medium.

Abstract: In optical two-way communication using a single optical fiber, part of the light transmitted from an apparatus is received as stray light by the apparatus itself. This occasionally hampers accurate detection of disconnection from the partner apparatus and of the start of data communication by the partner apparatus after the establishment of connection. To overcome this, the timing with which an apparatus transmits a tone signal after the establishment of connection is compared with the timing with which the apparatus receives a tone signal from the partner apparatus, and the two apparatuses use different sets of data transmission properties (for example, different wavelengths) to accurately detect disconnection. Moreover, based on the properties of the received light and the characteristics of the receiving amplifier, repetition of the same code for longer than a predetermined period is detected to accurately detect disconnection.

Abstract: A method and apparatus to manage tone clamping are described.

Abstract: A signalling tone detector for use in a communications network for the detection of dual frequency multi-tone (DTMF) pairs of signal tones in a time division multiplex (TDM) signal. The detector comprises first and second arrays of adaptive notch filters corresponding to the rows and columns of a subscriber's pushbutton key set. Each said notch filter responds to a respective signal tone by the generation of a corresponding output signal. A logic circuit identifies from a combination of output signals from the adaptive notch filters the presence or absence of a pair of signal tones and, when the presence of a pair of signal tones has been identified, for decodes that pair of tones to corresponding digit information. The detector is used in an arrangement for compressing and packetising TDM traffic for transport over a connectionless network. To prevent distortion, the DTMF tones are decoded to digit information prior to transport over the connectionless network.

Abstract: A method and apparatus to manage tone clamping are described wherein atone is clamped using a buffer with a first size and the performance of the clamping is analyzed. The buffer is set to a second size in accordance with the analysis. The clamping comprises receiving audio information, storing the audio information in the buffer, detecting tone information in the audio information, and removing the tone information from the audio information. Other embodiments are described and claimed.

Abstract: A noise calculation mechanism is incorporated into a cadence tone detector which upon a miss in either a energy or silence interval allows for retention of those periods corresponding to noise and augments the appropriate energy and silence interval.

Abstract: The invention solves the problem of DTMF delay by shifting the delay and in-band signal processing to the receiving packet gateway. The transmitting gateway continues to process and transmit voice packets while also detecting DTMF signals. The receiving gateway's jitter buffer holds voice packets for the worst-case DTMF detection period. As the receiving gateway is about to play out a voice packet it checks to see if a packet has arrived indicating DTMF was present. If not, the voice is played out as usual. If DTMF is present, the voice is muted and a DTMF generator invoked by the receiving gateway to recreate the DTMF signaling. The audio remains muted until no more time periods are marked as containing DTMF. In this way, the delay in voice playout due to the possible presence of DTMF is completely subsumed in the normal jitter buffer delay.

Abstract: Methods and apparatus for performing channel estimate updates in frequency division multiplexed, e.g., (OFDM), systems are described. After generation of initial channel estimates from received pilots, channel estimates corresponding to individual tones are updated using any one of a plurality of update techniques including, e.g., a constant modulus based method and a reduced constellation decision directed update method. The channel estimate update technique to be used with for an individual tone is selected based on a comparison of a signal noise measurement to one or more thresholds. The channel estimate update technique applied to different tones of the OFDM signal at the same time may vary. Over time, as the level of noise is reduced, the channel estimate update technique will switch from an interpolated pilot method, to a constant modulus algorithm based method, to a reduced constellation decision directed method, to a full constellation decision directed update method.

Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

Abstract: The invented method involves one or more gateways sniffing the voice channel during the voice mode. If it is determined that V.8bis signals are being initiated, then the gateway breaks these exchanges by suppressing such signaling, thereby avoiding the above-described detrimental effects. Modem relay communications then are allowed to proceed. Briefly, the method and apparatus involve monitoring a call during a voice mode phase for an initiating signal representative of the defined signaling, the monitoring being performed by a local gateway and, if such an initiating signal is detected during the monitoring, then suppressing such detected signaling in such manner that the signaling does not reach the remote gateway. Preferably, the monitoring is for an initiating signal characterized by a dual tone of defined frequency and duration of approximately 1375 Hz and 2002 Hz for a duration of approximately 400 ms or 285 ms, in agreement with the ITU-T Recommendation V.

Abstract: An apparatus and a method for detecting the energy of a tone signal. A pre-process unit takes absolute values of transmitted samples and outputs the values to a delay unit for delaying the transmitted samples according to a predetermined delay interval. An operation unit subtracts the value calculated by multiplying the magnitudes of the samples transmitted before and after the burst detection from the square value of the magnitude of the sample at the time of burst detection among the delayed transmitted samples. A detection unit detects the burst by confirming whether peaks corresponding to the number of the samples transmitted before the sample of the burst detection are detected by a signal of an value input from the operation unit. Therefore, the present invention can reduce a probability of a false alarm caused by an impulse error and burst errors and thus perform stable burst detection.

Abstract: Two related voiceband compression techniques are employed in order to enable an RF telecommunications base station to accommodate data signals of high speed voiceband modems and FAX machines. A High Speed Codec enables the base station to pass voiceband modem and FAX transmissions at up to 9.6 kb/s. An Ultra-High Speed Codec supports voiceband modem and FAX transmissions up to 14.4 kb/s. The High Speed Codec operates using three 16-phase RF slots or four 8-phase RF slots, and the Ultra-High Speed Codec operates using four 16-phase RF slots. Because these codecs transmit information over several RF slots which can be contiguous, the slots within RF communication channels are dynamically allocated. The Dynamic Time slot/Bandwidth Allocation feature detects and monitors the data transmission and forms a data channel from the necessary number of slots.

Abstract: A method and system for detecting a dual tone in a composite signal having first and second components including means for separating the composite signal into its first and second components, means for estimating power of the separated first and second components, means for determining frequency of the separated first and second components, and means for comparing the power and frequency of the first and second components to at least one of a plurality of respective power and frequency thresholds to determine whether the composite signal comprises a dual tone.

Abstract: In a gateway router the length of the play-out buffer in VBD mode is not set at a previously established value. Instead the length of the play-out buffer in VBD mode is determined by the length of the buffer in the previous audio mode. When the system switches from audio mode to VDB mode, the length of the play-out buffer in audio mode is noted. The length of the play-out buffer in VBD mode is then set equal to its length in the prior audio mode plus a dilation factor. An audio concealment algorithm (which fills in gaps caused by missed packets) is used during the transition from audio to VDB to conceal any transition glitch during the answer tone.

Abstract: A system and method for detecting and removing tones from an incoming communications channel that may also carry other signals such as, for example, voice is disclosed which includes pre-detecting tones in a communication signal, processing the communication signal to invalidate the tones in response to the tone pre-detection, forwarding the processed communication signal across a network, validating the tone and forwarding tone-on signals across the network in response to the validation.

Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

Abstract: A system and method is provided for interfacing a protocol component with a physical layer component. Initially, a parameter setting message is sent to the physical layer indicating at least the mode of the processor. If the mode is nonframing, a handshaking tone or pattern detection request message is sent to the physical layer component enabling detection of specific handshaking tones or patterns. Next, a handshaking tone or pattern detection indicate message is received from the physical layer component indicating that a recognized tone or bit pattern has been detected in response to a detection request message. A signal request message is sent from the handshaking component to the physical layer component indicating that a handshaking message is to be transmitted as well as the content of that message. In addition, the signal request message also preferably includes parameters relating to the duration of the signal to be transmitted, such as a maximum and minimum symbol number.

Abstract: A network system comprises a network of data sources interconnecting plural clients by way plural network devices. A data server achieves flexibility in its use of network resources by altering the attributes of one or more network devices along the communication channels between the data server and the plural clients. This is accomplished during the normal course of data transmissions to the plural clients, thus providing real time control over the network device attribute. The invention is particularly useful in data servers, such as a video server system, in which data transmissions are characterized by a long sequence of data packets.

Abstract: A method, apparatus, and system for detecting a tone. In some embodiments, a signal is communicated by applying a first quantity of samples per period and a second quantity of samples per period. The samples may then be translated into information. For example, the first and second quantity of samples may be translated into information and identified. According to some embodiments, a signal communicates information at at least two different frequencies simultaneously to determine a common carrier for transmitting and receiving nodes.

Abstract: A system and method to detect a call trigger during a call without requiring a service platform to remain connected to the call. A first communication device transmits a call having call signaling and user communications. A signaling processor receives the call signaling from the first communication device and selects a connection to a second communication device. An interworking unit receives the user communications from the first communication device and converts the user communications from a time division multiplex format to asynchronous transfer mode cells that identify a selected connection to the second communication device and transports the user communications to the second communication device. The interworking unit detects a call trigger in the user communications. The interworking unit transmits call trigger data to the signaling processor.