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Figure 1. VoP Software Architecture
MGCP is a centralized call-processing system in which the intelligence resides primarily at the head end. The cable modems and NIs are similar to "dumb" clients and the system relies on call agents to negotiate the call through the network. SIP is a distributed system in which the intelligence resides in the NI and the head end is mainly a gateway to the public telephone network.
The greatest benefit of MGCP implementations is simplified, efficient management and administration. Fault detection and isolation are typically limited to the head end. And there is no need to distribute software upgrades and patches to customers, meaning that there is also no concern about software-version synchronization among all NIs.
The supporters of SIP, on the other hand, argue that it is a more scalable and reliable system. The case for scalability is due to the fact that the head end, acting mainly as a gateway, is unlikely to bottleneck subscriber capacity. Traffic load would be the only potential bottleneck, not processing time. Supporters also claim that SIP is more reliable because an SIP–based network architecture does not have a single point of failure.
TI's Telogy Software™ products MGCP– and SIP–compliant architecture processes voice packets similarly using either protocol. The software is broken down into two parts, the DSP and microprocessor components. The DSP processes voice data and passes voice packets to the microprocessor with generic voice headers. The microprocessor component is responsible for moving voice packets and adapting the generic voice headers to the NCS protocol. The microprocessor also processes signaling information and converts it from a telephony signaling protocol to IP.
This partitioning of functionality between the DSP and microprocessor provides a clean interface between the generic processing functions (such as compression, echo cancellation, and voice-activity detection) and the application-specific signaling and protocol processing.
- DSP Component (or Voice Processing Module)—This software prepares voice samples for transmission over the packet network. Its components perform echo cancellation, voice compression (to conserve cable bandwidth), voice-activity detection, jitter removal, clock synchronization, and voice packetization. This unique software, along with TI's programmable DSP technology, provides a comprehensive yet flexible foundation that allows equipment providers to shave months off typical development schedules, resulting in tremendous cost savings and a critical time-to-market boost.
- Microprocessor Component—In NCS–based products, the microprocessor component handles detection of various events, reporting of events to call agents, dual-tone multifrequency (DTMF) digit collection and reporting, application of signals and forwarding of audio packets. The microprocessor component in NCS–based products is comprised of the following software modules:
- XGCP signaling module (XGCM)
- Digit collection module (DCM)
- DSP interface module (DIM)
This interface receives pulse code modulation (PCM) samples from the digital interface and forwards them to the appropriate DSP software modules for processing. It also forwards processed PCM samples received from DSP software modules to the digital interface. It performs continuous re-sampling of output samples to avoid sample slips.
Tone Generator
This generates DTMF tones and call-progress tones under command of the host (telephone, modem, PBX, or telephone switch). It supports U.S. and international tones.
Echo Canceller
This performs G.165 and G.168–compliant echo cancellation on sampled, full-duplex voice signals. It has a programmable range of tail lengths.
Voice Activation Detector
This monitors the received signal for voice activity. When no activity is detected for a specific period of time, the software informs the IP. This prevents the encoder output from being transported across the network when there is silence to save bandwidth. This software also measures the idle noise characteristics of the telephony interface. It reports this information to the IP in order to relay this information to the head end for noise generation when no voice is present.
Tone Detector
This detects the reception of DTMF tones and performs voice/fax discrimination. Detected tones are reported to the host so that the appropriate speech or fax functions are activated.
Voice Codec Software
This software compresses the voice data for transmission over the packet network. It is capable of numerous compression ratios through its modular architecture. A compression ratio of 8:1 is achievable with the G.729 voice codec.
Fax Software
This software performs a fax relay function by demodulating PCM data, extracting the relevant information, and packing the fax-line scan data into frames for transmission.
Voice Playout Unit
This buffers voice packets received from the packet network and sends them to the voice codec for playout. The following features are supported:
- FIFO buffer that stores voice code words before playout to remove timing jitter from the incoming packet sequence
- Continuous-phase resampler that removes timing-frequency offset without causing packet slips or loss of data
- Timing-jitter measurement which allows adaptive control of FIFO delay
The voice packetization protocols use a sequence number field in the transmit-packet stream to maintain temporal integrity of voice during playout. Using this approach, the transmitter inserts the contents of a free-running, modulo-16 packet counter into each transmitted packet, allowing the receiver to detect lost packets and to reproduce silent intervals during playout.
Packet Voice Protocol
This encapsulates compressed voice and fax data for end-to-end transmission over a backbone network between two ports.
Control Interface Software
This coordinates the exchange of monitor and control information between the DSP and host via a mailbox mechanism. Information exchanged includes software downline load, configuration data, and status reporting.
Real-Time Portability Environment
This provides the operating environment for the software residing on the DSP. It also provides synchronization functions, task management, memory management, and timer management.
Unsolicited Grant Service (UGS)
Cable networks are asymmetric—the downstream data received is streaming while the data transmitted upstream is either transmitted on a collision timeframe or must get a time slot or "grant." Because requesting a grant can cause significant delay, UGS ensures that cable modems will be contacted at regular intervals without having to make separate requests. The concatenation process mentioned earlier can lighten UGS requirements and increase the efficient use of bandwidth.
UGS with Activity Detection (UGS–AD)
Upon detection of voice inactivity UGS–AD enables network resources to be diverted to other cable modems and data flows, maximizing the efficiency of all data transmissions.
The DIM is responsible for the interface to the TI DSP–based Telogy Software™ products. The microprocessor communicates with the DSP through a shared memory arrangement whose mechanics are hidden by the DIM. The DIM shields the rest of the microprocessor software from the complexities of the DSP interface.
XGCP Signaling Module
This module is responsible for providing MGCP embedded client functionality. It parses and processes each message received from an MGCP call agent. It reports detected events to the call agent, generates signals requested by the call agent, reports detected DTMF digits, and sets up connections requested by the call agent. This module is also responsible for forwarding audio packets received from the DSP to the packet network interface and forwarding audio packets received from the packet network interface to the DSP.
Digit Collection Module
This module is responsible for processing dialed digits received from the XGCP module. It accumulates all the dialed digits and matches them against the digit map. It reports the results along with the accumulated digits to the XGCP module.
Network Management Module
This module is responsible for providing the management interface to configure and maintain the other modules of TI Telogy Software™ products. A sample module is provided, but the customer may replace the sample with a custom module. A proprietary voice packet MIB is supported because no standard MIB exists.
