Why do people pay for choppy and unreliable cell phone service? The reason is simple: They want to be connected. With added value and extra features such as silence deleted recording, two-way recording and play back, VoIP can be the means for us to communicate with each other while on the road in the near future. However, several key issues must be addressed.
First and foremost, a new standard is needed to provide guidelines for evaluating "sub-system audio quality". Toll quality VoIP communication demands the best technology. Yet current toll quality subsystems do not necessarily make VoIP phone calls. And since quality distortion is additive, a sub-toll-quality sub-system takes away the possibility altogether. It is also important for optical fiber networks, which provide tremendous bandwidth, to be able to provide low jitter and low mean delay communication with different toll grades.
A common way to assess speech/voice quality is to use a quantitative measurement unit (QDU) and to accumulate the quality degradation along the voice/speech path. Better communication across the voice processing elements eliminates unnecessary processing and less quality degradation. Better communication between the network processes, network management program and the sub-system can greatly enhance the overall audio quality. For example, if call forwarding and conferencing reconfiguration occur, the activation of reconvergence should assist the echo canceller to adapt to optimal reflections with shorter latency.
Other issues in relation to voice quality need to be addressed. Public switched telephone network (PSTN) calls deliver consistent quality because proven tone detectors and generators are deployed. The same thing needs to happen on key VoIP components. While extra delay and transcoding may be unavoidable with VoIP, a proposed VoIP echo canceller should demonstrate the ability to handle both multiple reflections and combat transcoding and DC level shifts. An echo canceller that meets this requirement will be able to deal with the typical CLASS features such as Three Way Conferencing (TWC). The VoIP gateway equipped with better power management and power backup systems, as well as special tone detectors, will be able to provide lifeline support.
A new ITU standard, G.799.1, is currently under development to address these specific issues. The phone-VoIP gateway bridges between time division multiplexed signals and IP traffic serves to define functions to ensure quality speech performance. In other words, the G.799.1 standard is intended to provide a distortion figure for speech or other voice band signals that pass through the gateway. Besides the typical compression/decompression and echo canceller implementation/performance details, G.799.1 also provides the means to avoid tandem speech and the control and configuration of interfaces, while providing for H.248 and H.323 functionality.
For VoIP to become the norm, or, at the very least, a viable alternative for mainstream telecommunication, the first step is to ensure that it works in a corporate network. A standard testing lab, which can be used as for compatibility testing for wireless LAN products, needs to be established. The IP Gateway standard test center needs to test between the TDM and the IP interface and treat the in-between components as a voice processing black box. The various speech communication parameters such as voice quality and distortion need to be assessed.
This test procedure must ensure collective voice quality and could be similar to the GSM type approval test as defined by the ETSi standard as it evolves. The end-to-end delay and delay variation in the form of jitter can be addressed with the tremendous bandwidth provided with optic fiber trunks. With the bandwidth taken care of, top quality, high bandwidth voice communication becomes feasible.
Extra features such as silence deleted recording, answering machine tone detection, two-way recording and play back, help to demonstrate the strength of digital networks. The "follow-me" feature, to utilize the IP protocol such as IPV6 provides mobility of the phone number, in effect making the handset portable in that the VoIP user can call through any Internet gateway. This adds significant additional value to a VoIP phone call.
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New recommendations to standardize next generation IP gateways are on the way. The infrastructure to provide manageable delay to control jitter is deployed. In the near future, the 3G and 4G wireless network is moving toward full IP implementation. The process can speed up with the forming of a VoIP quality verification group. In the long term, the development of rich and unique features for VoIP phones will be the driving force behind their market acceptance, not just the fact that using it is essentially free.
