Theory vs practice of SIP-based VoIP

I recently attended the VoIP conference and expo [1], at Illinois Institute of Technology, organized by Prof Carol Davids, and also got a chance to speak on a couple of topics [2]. There were many interesting presentations in the conference giving perspectives from leading software and equipment vendors, carriers and service providers, government bodies, standardization forums as well as open source developers. This article presents some of my thoughts regarding the theory and practice of SIP-based VoIP.

The inaugural session showed a demonstration of IP-based 911 call by students by integrating and using the software pieces developed at other universities. It was great to see sipd [3] being used by other universities for exciting new projects, and brought back the memories when we were developing sipd.


The session initiation protocol (SIP) was invented to create and control multimedia sessions on the Internet because the previous protocols were either insufficient (e.g., HTTP, RTSP) or too complex (e.g., H.323). Unlike HTTP which typically requires a dedicated server, SIP was designed to be more peer-to-peer. Hence, your VoIP phone itself acts like both client as well as server to send and receive SIP requests. Ideally, you do not need to keep the SIP proxies in the session path, except for initial call setup. The protocol is designed to encourage subsequent requests such as ACK, BYE or re-INVITEs to be end-to-end, if possible. The protocol includes mechanism to enable an intermediate proxy to require that all subsequent requests in a session be sent via that proxy. But this was designed to be used as an exception rather than a rule. The motivation is scalability: keep the proxies lightly loaded.

In theory, a SIP-based system follows the end-to-end principle of Internet: keep the intelligence in the end, and have the network (or intermediate proxies) be dumb. The end-to-end principle has been crucial in the success of the Internet and more recently the web applications. As long as you keep the network provider independent of your application provider, you see explosion of application innovations.


In practice, your SIP-based system is typically "owned" by your network provider who has business incentive to provide you billable applications and services, and prevent you from talking to other open SIP-based systems without going through their billed "services". The largest SIP systems such as Comcast and Verizon digital voice are designed to be closed systems which use SIP in the network without allowing end-users to access it directly. More recently, Apple's Facetime is a closed service and does not inter-operate with other SIP services. With SIP-based IP multimedia subsystem (IMS) being adopted by wireless carriers, there is more incentive for businesses to convert SIP from an end-to-end protocol to a network centered and managed service.

One of the term I kept hearing during the VoIP conference was the "managed" services, and why it is useful for consumers, and what kind of new innovations are happening. When I looked at details, these services and innovations are basically what SIP already provides, e.g., service APIs, unified communications, etc. I had the opportunity to work on some of these 5-10 years ago when I was at Columbia University. It is discouraging to know that because of narrow minded business incentives of vendors and carriers, walled gardens of SIP systems are created which prevent open innovations and require significantly many fold effort to get basic features to the consumers. First, (1) the vendors and carriers use an open protocol, SIP, to build a VoIP system, then (2) invest resources in making it a walled garden, and finally (3) invest more money and resources to create federation of these walled gardens. In the end, (2) and (3) nullify each other, and just (1) was sufficient. All the money invested in (2) and (3) gets wasted in the long term.


I would like to advise vendors and carriers to just focus on providing a good IP network with some quality of service and less restricted NATs, and leave the rest of the VoIP services to the millions of application developers. As Henry Sinnreich said during the conference, the only service is "the Internet". Instead of providing "managed" network services, open up your network for end-to-end innovations. In the long term this will boost your network and bring more revenue. With Internet and web, there is more opportunity for everyone, and a walled garden approach in the network is just going to keep you away from the long term growth. 

The open innovations in VoIP are bound to happen. If you do not want to be part of that, someone else definitely will. Adrian Georgescu presented Blink [4], a fully featured easy to use SIP user agent, as a great example of SIP-based open innovation. For every VoIP developer in "managed" service organization, there are probably a thousand independent developers such as web application developers. Sooner or later, these developers will build some open platform or system which will attract hundred times more traffic than your managed services and hence many fold more revenue. At that time all your investment in "managed" services will go down the drain. Because if you don't open up your SIP systems, these developers will not wait for it, and build something else. This has happened before with Internet and web applications, and will happen again sooner or later with Internet voice/video communication or VoIP.

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