It is universally accepted that Skype is virally marketed with very little expenditure. But savvy claims that go unchallenged also plays a role in establishing the aura of genius. A recent interview (via Skype Journal) given by Niklas Zennström, CEO of Skype to EE Times is but the latest example of this.
EE Times wondered how Skype can offer better voice quality compare to other VoIP service providers and PSTN even. The answer is noteworthy. Here is a snippet:
“Skype is the only one that uses peer-to-peer technology for voice services. Other VoIP services are using a client-server technology… It can take advantage of full broadband bandwidth, so you don't need to compress the voice as much to get a richer sound experience.
First, it is erroneous to state that Skype does not use client-server technology; most assuredly it uses client-server architecture. If the supernodes are not servers, what are they?
Second, let us assume that for the sake of argument, that Skype is peer-to-peer. But the question is about voice quality. So what matters is the flow of media packets. By now it is well known (by Skype’s own admission) it uses UDP hole punching algorithm, just as many Session Border Controllers used by other providers. So the media flow will be directly between the end-points in either case. If there is indeed a difference in voice quality, the only explanation must be the codec.
But from the response to the question, “Have you invented your own codec for this?” we learn that they use “industry-standard codecs like G.729 and G.711, iLBC and iSAC.” We are told that these are only some of the codecs they use. Probably there are some secret ones that give them a leg up. Otherwise can’t others use these same industry-standard codecs and remove any possible differences in voice quality? I recall in one of the very early news items on Skype, it was claimed (see note below) that a team of audio experts had worked in addressing the voice quality problem and have come up with a better solution. Probably they have come up with a proprietary codec. (You own a bridge as well?)
The second statement that is worth noting is this one: “It [peer-to-peer] takes the shortest route between end users.” By implication, this is applicable to Skype as well. But another very early story (see note below) suggested that Skype finds multiple routes between the two end-points and picks one route that gives the best voice quality. It was further implied, that when the voice quality deteriorates an alternate route will be picked. So which one is true - Skype doing routing at a higher layer or the routing is done by the IP layer independent of Skype?
The third interesting statement is what we learn about the amount of compression Skype does. We are told that, “A normal telephone signal in the digital telephone network is compressed to 56 kilobits per second, which is pretty much the same as the modem line. But we can pretty much take advantage of the broadband network and use a richer sound that doesn't have to be compressed as much.” Really? The folklore says that Skype uses primarily iSAC codec from Global IP Sound. The datasheet on iSAC states that the maximum transmission rate is only 32 kbps. Looks like a lot of compression to me, given that it is a wideband codec with a higher sampling rate. (I will not take a cheap shot and point out that the uplink speed in modem tops out at 33.6 kbps.)
It sure looks like part of the viral marketing is to create an aura of mystery and a notion of uniqueness.
Note: I am not able to give reference to the two news stories, even though I have read them. I will appreciate it if you know the references and provide them to me.Posted by aswath at June 12, 2005 07:11 PM
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