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I believe the underlying tech, Global Index, is patented and valuable. Global Index is the p2p matchmaking software that runs under Skype (and Kazaa back in the

by qeorge 15y ago

I believe the underlying tech, Global Index, is patented and valuable. Global Index is the p2p matchmaking software that runs under Skype (and Kazaa back in the day), which means Skype doesn't have to run centralized servers to route call data. Its also what lets them punch holes in firewalls. My understanding is that their competitors don't (can't?) do this. For example, SIP runs through a central server (e.g., Google's). This allows Skype to run their service more cheaply, and its cheaper to provide service for each marginal user. Its a pretty strong network effect. So, in addition to their brand and userbase, they do have some tech MS might want and can't get elsewhere.

cygwin98 15y ago
My understanding is that their competitors don't (can't?) do this. For example, SIP runs through a central server (e.g., Google's). This allows Skype to run their service more cheaply, and its cheaper to provide service for each marginal user. Its a pretty strong network effect. This maybe the case for those small VoIP startups, but for big Internet Co. such as Google, Microsoft and other portals who already have big datacenters, this should not be an issue.
tene 15y ago
You seem to be confused about SIP. SIP is the "Session Initiation Protocol", and its entire point is to coordinate initiation of a direct session between endpoints, the so-called "SIP Trapezoid". SIP is a lot like SMTP. When you make a SIP call, you talk to your local SIP server, which talks to N intermediaries (usually 0) before going out to the public internet. After passing off to the destination server, the connection can go through N intermediaries on the remote side before being delivered to a final user agent. This connection isn't for delivering audio and video, though, it's for coordinating a direct RTP connection to use for audio and video. Once the two endpoints are communicating, they send information about their real network location, work out details about NAT, and set up a direct RTP connection, possibly using STUN to deal with NAT issues. In a normal SIP environment, neither party's SIP server ever sees any of the actual media. This is all fairly similar to what skype does, except skype has additional infrastructure for routing media through other clients, in cases where two endpoints can't find a way to communicate directly, as I understand it. I'm not actually authoritatively-informed about skype, though, so don't rely on this last bit.
- qeorge 15y ago
  Thanks tene, that's really good info. I was 100% turned around on that - definitely thought SIP had to route the call data through a central server.

by qeorge 15y ago

cygwin98 15y ago

My understanding is that their competitors don't (can't?) do this. For example, SIP runs through a central server (e.g., Google's). This allows Skype to run their service more cheaply, and its cheaper to provide service for each marginal user. Its a pretty strong network effect. This maybe the case for those small VoIP startups, but for big Internet Co. such as Google, Microsoft and other portals who already have big datacenters, this should not be an issue.

tene 15y ago

You seem to be confused about SIP. SIP is the "Session Initiation Protocol", and its entire point is to coordinate initiation of a direct session between endpoints, the so-called "SIP Trapezoid". SIP is a lot like SMTP. When you make a SIP call, you talk to your local SIP server, which talks to N intermediaries (usually 0) before going out to the public internet. After passing off to the destination server, the connection can go through N intermediaries on the remote side before being delivered to a final user agent. This connection isn't for delivering audio and video, though, it's for coordinating a direct RTP connection to use for audio and video. Once the two endpoints are communicating, they send information about their real network location, work out details about NAT, and set up a direct RTP connection, possibly using STUN to deal with NAT issues. In a normal SIP environment, neither party's SIP server ever sees any of the actual media. This is all fairly similar to what skype does, except skype has additional infrastructure for routing media through other clients, in cases where two endpoints can't find a way to communicate directly, as I understand it. I'm not actually authoritatively-informed about skype, though, so don't rely on this last bit.

qeorge 15y ago
Thanks tene, that's really good info. I was 100% turned around on that - definitely thought SIP had to route the call data through a central server.