There is very nearly a revolution in the works, but existing codecs are not exactly supporting the coup. High-definition, VoIP-driven voice service is certainly upon the business community. It offers the highest quality audio transmission technology can muster, and it offers audio cheaply and in a range of formats. It is poised to entirely take over the old voice standard, PSTN, but that aging standard is still firmly planted because the codecs that support VoIP are not yet working in concert with one another.
A recent Network World article discusses the nature of the divide between the two methods of voice transmission. PSTN is inherently connected to its past, a history of analog transmissions. This limited the type and amount of voice information users could send through the infrastructure -- in the beginning, it consisted of telephone lines. Now the PSTN system is mostly digital, and sound is converted into data packets. The switch from analog to digital did not greatly improve clarity, though, because the standards of the time focused on improving transfer efficiency, not sound quality.
VoIP, on the other hand, allows communications providers to use more bandwidth at rates that are still considered cheap. So, they can focus on sound quality in addition to efficiency, and that change in focus has resulted in the broad range of high-definition audio that providers can now make possible. The proliferation of VoIP services invited new standards that include ITU-T G.722, G.729.1, and Microsoft Real-time audio (RTA). And the glut of standards may simultaneously represent both good and bad aspects of next-generation audio services.
There are so many standards that it will likely require a third party, an intermediary, to act as an intermediary and force each codec to work alongside its counterparts. It is unlikely that the entire telecommunications industry will focus on a single standard. Each new standard has it merits, and therefore each new standard will have its supporters. To make the jump away from PSTN and toward an all-VoIP system, all codecs will need to find a way to play nicely with one another. This includes cable, narrowband, wideband, wired and wireless and video codecs. They all need to pass through the same system, and it will require a future generation of companies or divisions -- the intermediaries -- to make VoIP possible for all and force PSTN to take the back seat.
With WebRTC and emerging Web APIs, a new choice is rapidly emerging: Use a complete vendor service for a traditional vendor like Avaya, Cisco or Genesys, or develop a tailored implementation using these advanced components. While the telecom group may see an extension of the existing telecom solution as the right path, the Web team may see using these technologies, directly integrated to websites and eCommerce, as a faster and more powerful solution. This new challenge is something that all enterprises must understand and decide on.
This area will be a major focus of the Enterprise Track at WebRTC Conference & Expo in Atlanta, June 17-19, an event that any business with a strong Web and contact center presence needs to attend.
Edited by
Rachel Ramsey
