Some days I think that I’m unique in my dedication to VoIP-ish pursuits. Of course that can’t possibly be true or no-one would be reading this stuff at all. But how many people do you think are so dedicated to the matter that they wear a VoIP capable headset while travelling?
There has been for many years a subtle conflict ongoing in telecom space. Various vendors have created digital encoding techniques (codecs) that target common network issues. Since various network realities exist so too do various approaches to the problems faced. So a range of codecs exist in the marketplace. Typically a high-quality solution comes with an associated cost, reflecting the very fact that the solution has merit.
The poster-child for this is the G.729a codec. Over time this patented codec has become the industry standard low-bitrate codec for voice applications. Who can argue. It works well. It squeezes reasonable voice quality down to under 30 kbps and it’s compute overhead is acceptable on available hardware.
About a week ago I read a tweet someone posted stemming from a conversation with their boss. The topic under consideration was blogging. The boss asserted that they should be posting more frequently, even if they are short posts. Their own impression had been that quality trumps quantity and longer, well-considered posts take time.
The initial tweet was met with a range of replies, myself amongst them. I instinctively agreed with the stance that quality is an imperative. However, as is often the case, after mulling it over for a while I’m not sure the answer is so simple or obvious. To borrow from my roots a Western Canadian cultural-ism, the better answer could be “that depends.”
The release a couple of months back of the Skype v4.o client for Windows was noteworthy as the introduction of their in-house developed SILK codec. Earlier today during an eComm 2009 presentation Jonathan Christensen, Skype GM Audio & Video, announced that SILK was being released under a royalty free license.
SILK was notable as being capable of narrowband (8KHz), wide band (16KHz) and super-wideband (24KHz) sample rates. Skype claims the codec dynamically adapts both sample rate and bitrate in response to variable network quality. They have published a PDF with a very general overiew of codec performance expressed in terms of bitrates, CPU requirements and MOS scores.
Earlier today John Todd let it be known that Digium's IAX2 protocol had been officially accepted as RFC5456. This news comes to me by way of The Daily Asterisk News.
A common wisdom here is that one should use a proper hardware phone rather that an extra software on the user’s PC. Why is that such a big issue?
One thing that bothers me with the current crop of hardware SIP phones is that they are hopelessly proprietary.
So what would it take to build a fully-adaptable phone?
I am 100% behind the assertion that most users want a hard phone on their desk. Soft phones, even good ones, seem to be exclusively the domain of those who travel and vertical niches like call centers.