A few days back there was a heightened level of excitement among VoIPers as news spread that Google Voice accepted calls on a SIP interface. This meant that SIP end points can originate SIP-based sessions by identifying the Google Voice users by their SIP URIs. But the excitement was a “delta function” as Google Voice disabled that interface subsequently. As if to tease us all Google Voice played this peekaboo game a couple of more times. It was all great fun because nothing meaningful is to be gained with SIP URI dialing and so nothing was lost. But I would like to take this opportunity to point out some fundamental things VoIP providers have to do in addition to carrying SIP URI in the signaling messages before we enjoy the benefits of “SIP URI dialing”.
The first and main point to note is that SIP URI dialing makes it easy to have SIP signaling end to end because the originating entity (User Agent or a Proxy Server) can readily identify and contact the destination entity (User Agent or Proxy Server). Of course this can be done even with E.164 dialing. For this there has to be a universal agreement on a directory service that maps E.164 numbers to location of the corresponding destination entities. But unfortunately ENUM, a technology that could facilitate this has not been widely adopted due to business and political reasons and not technical difficulties. End to end SIP signaling is not a great thing by itself; it affords us to realize many useful features and services.
Given all these varied benefits of SIP URI based dialing why am I being less enthusiastic? A short answer is that this putting the cart before the horse. Let me give a more detailed answer below.
Till recently most of the VoIP deployments have used VoIP ATA with the standard telephone set. So the SIP signaling is not truly end to end. In other places, we all celebrate the end to end nature of Internet and everybody will quote the famous paper by Saltzer et. al. But what is not usually observed is that one of the points that was vigorously debated was where to terminate error correction and acknowledgment procedures of the transport protocol. Those days the Network Interface card was outboard and some were content to terminate these procedures in these cards. But ultimately the end to end proponents prevailed. If you think about it these ATAs are analogous to the outboards of those days and we have been content with terminating SIP signaling in these boxes. Now finally the proliferation of mobile VoIP and enterprise applications have made it possible to have real end to end SIP signaling. But that is a small step.
Even those systems that have native SIP clients do not have the needed UI for users to populate the Subject header in the SIP INVITE message. I do not know of very many SIP VoIP providers who allow its users to populate their Rich Presence information or make them available to their potential callers. Finally I wonder how many SIP VoIP providers dabble in multi modal sessions or gradual escalation of modes. Let us face the fact that for most of the VoIP providers and their subscribers, the focus has been PSTN termination, pure and simple.
If we have a horse and no cart, at least some of us can ride the horse. But if we have only the cart, what good is it?
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