DeepDarkSee

Read at TechCrunch … this is great! In a move similar to the release of the GMaps API, Google is now messing with the VoIP world. Think how easy this is going to make it to integrate chat with all kinds of web and non-web apps (text and voice calls). Think of the hybrids – click on the icon on the map to call the marked location is obvious, then think social networking sites, then games and community. We’re going to see tons of creative applications of this API.

Important to note is that this release includes a peer-to-peer technology layer as well. Another big door opener. That, and the fact that the API is under a Berkeley style license – it can be used for commercial as well as free apps – and woo hoo!

Google has also just changed the rules for Skype and other VoIP providers, much like they did in the mapping world not so long ago.

VoIP (Voice over IP) is here. Over half the corporate telephone sales in 2005 will be VoIP phones. Michael Stanford, Intel’s director of VoIP strategy speaks to this as just the beachhead for Services over IP. Think ‘presence’, video conferencing, shared work environments and collaboration at new levels.

I’ve written before about my view that what’s most exciting about VoIP is our ability it integrate it with other aspects of the IP infrastructure, specifically coordinated voice and web interaction (and Web is A LOT more than HTML pages). I’ve been experimenting this year with “poor man’s presence” – leaving the phone line and/or iChat connection open between coworkers in different states. This is just the beginning.

I’ve been experimenting with VoIP for almost a year using several VoIP service providers and the Asterisk open source PBX software. I’ve setup Asterisk on a Bones in Motion server and created a virtual company PBX, with IVR and voice mail, and auto-dial out to employee desks and cell phones to connect calls. I’ve also setup a server at home and done experiments with conferences and other test and development work in the Asterisk environment.

Along the way I’ve experienced line quality and echo issues from DID providers (companies that provide you with a local number that forwards to your server, as well as outgoing calls to the normal phone network). I’ve seen long delays in getting inbound numbers working, lines stop working intermittently, echo on the line come and go, router/nat issues, DTMF transmission problems, impact from internet connectivity, and more. I’ve used SIP and IAX from DID providers, soft (PC based) phones, and dedicated VoIP phones. I’ve also had one Vonage user experience inconsistent DTMF transmission to Asterisk IVR (menus).

I find this area fascinating, and the ability to integrate incoming voice and web in a Linux/PHP environment opens incredible opportunities. These opportunities involve both the web and voice ends, and interaction between the two, driven from either or both ends. I also find it complex, fraught with complications, and more an emerging opportunity than a mature platform. I find myself driven to experiment in this space, have 101 ideas, and am poking around a lot lately.

So in this context, I decided to subscribe to Vonage and see how they are doing with things. I’ve heard rumors that some of Vonage’s back end is Asterisk based (along with Avaya), although they aren’t verifying this publicly. It’s still early on, I’m really impressed.

Early last week I signed up for the Vonage service. Got may email confirmation and account info. On Friday my premise equipment showed up. It was a Linksys router with two phone ports on the back. The fold-out quick install guide showed two configurations…’be the router’…and ‘connect to your existing router’. I used the latter, just plugged it in and connected the phone per the picture. I fully expected it to NOT work. I’ve got port 5060 forwarded to another system, and firewalling turned on. I did, however, recently update my existing router with the latest firmware image, which included in the notes a reference to fixing some VoIP issues.

So anyway, it worked! I just plugged it in and it worked. It has to be this way for Vonage to be successful on a broad commercial scale, and it’s clear that with things like DUN it’s all possible, but… it worked! I’m blown away. Maybe I’m just jaded from doing ‘do it yourself’ VoIP, but I’m still very impressed. Inbound and outbound calls, and all I had to do was plug in the cables per the picture. Vonage may just pull this off. Oh, and I was also really impressed at the low delays introduced by Vonage, and virtually immediate connects of inbound calls. Sound quality was good too.

Update: I am seeing randomly dropped DTMF characters in the IVR menu handling.