Experimenting With VoIP

Longtime readers of this blog know that I’m a sucker for novel uses of technology, especially those that give me a chance to learn something new. So when I spotted a Vonage VoIP telephone on clearance at the local Best Buy, I couldn’t resist. After a quick Google search to confirm it could be unlocked and used with any provider, I took one home. I don’t have it working perfectly yet, but I’m well on the way into a new area of techie exploration: Voice-over-IP with SIP!

Introducing SIP

I was vaguely aware of the open voice-over-IP protocol, SIP, already since I’d run into it during my work with power-over-Ethernet and Cisco router configuration. I had also heard of Asterisk, the open source PBX software project, while hacking my (now decommissioned) Linksys NSLU2 home server. And I’ve been a VoIP user for years, relying on my cable company for home phone service and dabbling with Skype as an inexpensive second phone line for conference calls. But I had never looked too deeply at these things, let alone tried to put them all together!

Here’s what I know: SIP is an open protocol that sets up voice links between endpoints, much as HTTP is used to connect web browsers to servers. VoIP generally consists of a stream of UDP packets containing encoded voice traffic, and SIP manages the connection. SIP has historically had a hard time with NAT routers (like my Tomato-powered WRT54GL and most other consumer gear) but the advent of UPnP and STUN has made things work a bit better.

Although there are a wide variety of services that make use of the SIP protocol, it hasn’t really reached critical mass with end users. One reason for this is the search for revenue: Providers are reluctant to allow open traffic to flow across the network since only captive customers are profitable. This is the reason that companies like Vonage and Skype (not to mention Google’s cool new Voice offering) are walled off from the world.

The Inter-Phone-Network

Although a growing number of SIP services allow free calling and open connectivity, it is extremely difficult to get out of the SIP world and into conventional phone and Internet voice networks. It’s a lot like OpenID or XMPP: Most don’t support it at all, some are happy enough to allow you in, but no one wants to let you out!

I was interested in connecting to a few “voice networks”:

1. The conventional public switched telephone network (PSTN) world that we’re all familiar with: The domain of landlines, cell phones, conference calls, and fax machines. The PSTN doesn’t care about VoIP, so upstarts have to learn to connect to it!
2. SIP networks are a set of IP-driven services that shadow the PSTN. Many companies use internal SIP networks rather than conventional analog PBX systems, and there are public SIP providers as well. Most of these allow free unlimited use internally in hopes of attracting customers to pay for PSTN links.
3. Google Voice (formerly Grand Central) is a nifty call management service for PSTN services. Along with whiz-bang features like voicemail transcription and call screening, Google Voice has taken some tentative steps towards integration with the VoIP world. Since it (apparently) uses SIP internally, folks have been trying to connect Google Voice to SIP networks. The company currently allows just one, Gizmo, to connect natively.
4. The proprietary Skype VoIP network, which I’ve used for a while. Skype offers paid “SkypeOut” service, allowing unlimited calling to US and toll-free numbers for a low quarterly fee. I’ve been using this for a while and have grown dissatisfied with its quality, reliability, and feature set. The company also sells “SkypeIn“, which assigns a PSTN phone number to your Skype account, allowing folks to call your computer. There’s also an iPhone app and a world of Skype hardware, which is really little more than standard audio gear like headsets, microphones, and sound cards.

The Vtech IP 8100

The phone hardware I’m working with is a Vtech IP 8100-1 kit manufactured to support the (expensive) Vonage service. I picked it up on clearance (there are three more at $25 – anyone want one?) as it appears Vtech is out of this particular business. The kit consists of a base station (which is a hybrid home Ethernet router and 5.8 GHz phone base station) and a solid wireless handset with a speakerphone.

Vonage appears to be a straight SIP provider but it uses locked hardware to force customers to pony up more than $20 per month for service. Happily and predictably there is an active community working on unlocking Vonage hardware for use with any SIP provider. My new phone was (partially) running with Gizmo within 30 minutes of opening the box!

Here’s a quick rundown for the folks at home:

1. Open up the gear and charge the handset
2. Plug the base station’s WAN (lower) port into your network so it can talk to Vonage (the bottom light will turn green when it’s set)
3. Get on a Windows PC (finding one was actually the hardest part for me!)
4. Download the latest version of CYT unlock
5. Attach the PC’s ethernet port directly to the IP 8100’s LAN (upper) port
6. In a web browser, navigate to http://192.168.15.1 and log in with “VTech” as both the username and password and leave the window open
7. From the command line, run “cyt46.exe VTECH”
8. Select option 1 and wait

Now you have created a super user account with “Admin” (case sensitive!) as both the username and password. This unlocks a new “configuration” page where you can set all of the SIP parameters for your provider of choice. It also tells the base station not to look at the Vonage TFTP servers for its configuration anymore.

Seriously, unlocking the hardware was simple, but getting SIP service up and running is proving much more difficult!

SIP In, SIP Out

One key paradigm shift for VoIP users is the separation of inbound and outbound services. This seems alien to most folks, but VoIP users happily use one service to receive calls and a completely different one to place them. Google Voice is the shiny new incoming call handler that everyone wants to try, but most expect to continue to rely on unknowns like Gizmo or Nonoh for outbound service.

This is where my SIP experiments hit a rock: I was able to set up my Google Voice number (781-Ped-Xing, for all you Max Headroom freaks!) to route inbound calls to my now-Gizmo-powered SIP phone, but only when the Gizmo app was open and running. And outbound calling through Google or Gizmo didn’t work at all!

This is not, as they say, a satisfactory condition. So I’m working on it. I’ve considered running my own Asterisk server (on the Mac Mini) to take care of local SIP with my phone, but this seems overly complex. I’d really love it if I could get Gizmo to work correctly, using Google for inbound calls and Gizmo for outbound, but this isn’t working yet. I’ve also considered using another service provider altogether but found their pricing and terms to be questionable. Plus, only Gizmo connects to Google!

I’ll keep working on it, posting the results here. Subscribe to my Terabyte Home feed via RSS or email to follow this discussion, and let me know if you have any suggestions! And if you want a Vtech IP 8100 kit, drop by your local Best Buy’s clearance section or drop me a line and I’ll grab one for you!