The people behind Kazaa and Joltid have taken their P2P expertise to the telephony world with Skype, a P2P telephony application.
One of the biggest problems with IP telephony for home users is that most endpoint protocols, e.g. H.323 and SIP, require that you open ports in your firewall, assuming the home user is smart enough to be using a firewall. NAT is yet another problem for the traditional approaches. I tried to set up the Pingtel instant expressa SIP based softphone about a year ago to use with TellMe Studio, but gave up after an hour or so of trying because of all the hassles involved in reconfiguring my router. Skype gets around the firewall in the same way P2P programs do; they tunnel over port 80.
Conversations are encrypted via 256-bit AES and the Skype FAQ claims that 3-16 kbytes per second of bandwidth are used during a call. For comparison, the bandwidth for a regular PSTN call is about 8 kbytes per second. A good IP telephony system, however, can compress the signal to about 1 kbyte per second. Encryption via IPSec would double the bandwidth needed, but there are better solutions. Anyway, the bandwidth needed for Skype calls is a little high, relatively speaking. We’ll have to see how that affects their system, especially for people on shared or lower bandwidth connections.
Of course, bandwidth is not really a problem for individuals making an IP phone call over a non-shared broadband connection. The big problem tends to be latency induced by slow networks or bursty network traffic and errors that require digitized voice packets to be resent. Also, the echoes due to the latency have a tendency to make IP calls sometimes sound like you are listening to someone talking through a long, narrow pipe. You need to write some pretty good code to keep the jitter buffer small enough so that a caller doesn’t feel like he is on a phone call with an astronaut somewhere near Mars. With a good quality IP phone switch on a well engineered network, though, an IP call can be indistinguishable from a PSTN call.
I’ve now installed Skype and am looking for someone else to try it out with. I have a pretty good quality headset (Plantronics DSP 500), so I’m hoping the sound quality will be reasonable. Either Skype me at username wombatnation, or send me an email to set up a session. I’m obviously not always sitting at my PC, and even when I am home and using la machina, I usually have it booted into Linux. Unfortunately, Skype is Windows only for now.
Skype is the best!
Skype for Linux – see http://www.publicmind.com/enduser/category.jsp?node=409
i want to know where i can find the real setup for the skype program.
Halo-Halo – Try this one out, it needs only 8 kbit/s (1 byte/sec). One problem this is in Polish, but user interface is intuitive.
PSTN call is actually 64kbps. (8bitsx8khz)
Most VOIP calls about 8-12kbps depending on codec and stops and starts in conversation.
8 kbytes per second = 64 kbps
I should have written the number in bits to be clearer, but the value is the same. I wrote it in bytes instead of bits only because the Skype documentation also listed the bandwidth in bytes.
While VoIP calls with a G.729 codec are about 8 kbps, calls with a G.711 codec are 64 kbps. It turns out that when you are doing speech recognition, you can get much better results with a G.711 codec, especially for short utterances. But, for ordinary VoIP calls between two people, G.729 works quite well.
I’m still puzzled by why Skype eats up so much bandwidth, unless less it is true that they are using the extra bandwidth for their own purposes. If you don’t think that is possible, I suggest you carefully read their Terms of Service.
Can u tell how much bytes is used while using skype phone to call from p2p. I mean as if someone has limited bytes to be used……………….
I’m not sure if there is any such feature built into Skype, but you may be able to use Ethereal to do this. It might be a lot of work to configure, though.
The biggest problem is that Skype tunnels a proprietary protocol over port 80. You would have to filter out the Skype traffic from the HTTP traffic going over that port (assuming you are trying to measure Skype bandwidth usage independently from all of your other network usage).