B. Mann Consulting

I think I just found a feature that's going to be implemented in Google Talk / Gmail real soon now: voicemail.

Try creating a label with the name "voicemail". I was cleaning up my personal account, where I get my voicemail delivered by email. I had set up a label months ago, and was trying to apply the label to one of the voicemails and got the error "The label name Voicemail is invalid". Hmmmm. So, I deleted the label, then tried to re-create it with the same name. No go...same error. Richard quickly verified that he was getting the same error.

Of course, trying to create a label with the name "Chats" gives you a different message -- "System specific names are not allowed. Please try another name." -- but it likely just means that "Voicemail" isn't truly live in the system yet.

Oh, and by the way, Bryght's switch to Gmail for domains is working really well. 

Some quick notes for an upcoming interview.

Benefits

• free voice calls between Skype users
• easy multi-person conferencing (up to 50, depending on bandwidth)
• multi-platform -- works on Windows, Mac, and Linux
• first IM client for many users who use voice feature first
• easy file sharing
• can have "real" phone numbers in many locations
• can also use video (various solutions, Windows only for now)
• Skype on your phone -- Microsoft PocketPC has a Skype client, EQO is new and works on Symbian Series 60 and other Java capable phones (client currently requires Windows on desktop)

Disclaimer: EQO is a local Vancouver company that I've met with, and my company Bryght powers their online community forums. They're also the only mobile app other than ShoZu that made my jaw drop (well, until I found out about the desktop client requirement...boo!).

Drawbacks

• have to have Skype to communicate (or pay for SkypeOut)
• built on a closed network, closed protocols (as opposed to open standards like SIP or JINGLE)
• computers on open networks might be used as a SuperNode, consuming lots of bandwidth
• Skype (now bought by Ebay) is the only provider of the service

Other forward looking options include click-to-buy or click-to-access solutions -- use your Skype ID and the client to pay for online goods and services, including restricted access to online content -- enter your Skype ID into a web form, get messaged with a special code, complete a transaction, receive an access code -- voila! instant online access. Note: this is of course possible with any IM service or even SMS, and Jabber is doing this today. In many ways, Skype is JAIS -- Just Another Identity Store -- that happens to have VoIP/IM/etc. built on top of it as an application.

There are a bunch of us (Scott, James, and Chris, too, once we told him the specs) that are lusting after the Nokia N91, but it is supposedly delayed until the first quarter of 2006.

Some background on the specs: this phone is a pretty high end unit, including a 4GB hard drive (store tons of music, photos, etc.), standard headset jack (use your killer headphones to listen to the music), 2MP camera (almost decent enough quality to leave your low-end digital camera at home) AND, the big one, built in Wi-Fi of the 802.11G flavour.

It's this last that is interesting. Nokia has already said that is working on Voice over WLAN: see Time Europe -- "2006 will be a big year for [mobile] wi-fi" and vnunet.com -- "replacing mobile and desktop phones with series 60 Nokia smartphones could make IT management easier". So, my guess is that they are specifically holding back the release of the N91 so it can be one of the flagship phones to include this functionality.

Phoneboy is where I first saw mention of Gizmo:

Hey look, it's Michael Robertson's attempt at cloning Skype: Project Gizmo. Since it basically is using the existing SIPphone infrastructure to pull it off, it's definitely SIP, it's definitely got connectivity with other SIP networks, and they've got inbound and outbound PSTN connectivity.
PhoneBoy's Blog: Project Gizmo?

It's essentially a slick client for SIPphone, which I mentioned briefly trying in this post about Skype. I'm going to download it and try it out, but I initially don't think it has the clout to take out the Skype juggernaut. Am I going to get more than a handful of people to try this out when Skype works today? I agree with Doc Searls, it is a locked silo, which I'm not happy about...but it's here today, it works, and it far surpasses the "just works" test.

What does Gizmo need to do to win my loyalty? Use Jabber for your instant messaging, and run gateways that let me connect to all the other systems with your one client. Solve my too many IM clients problem! (which is actually an identity / buddylist problem in part)

(Actually, yes, I would like to see SIP support in Adium! Where do I send my money?)

Later: OK, gets stuck at "Starting agent..." then "Failed to register with server, an unknown error occurred.", although I did get username "borismann". And yes, there are some "neat" features, like recording of calls, integration of maps and users, and even picking your own "on hold music". Somebody call me when it works.

Benjamin Kowarsch emailed me to give me the thumbs up on mentioning this. Ben's company, Sunrise Tel, have been putting together graphical front ends and configuration wizards for the Asterisk IP-PBX running on Mac OS X.

There is now a placeholder website at Astmasters.net -- "where the Macintosh Asterisk community meets". Of particular interest, if you request an account on the Macintosh Asterisk Mailing List, you can get a free VoIP account:

Regular members of the Macintosh Asterisk Mailing List may apply for a free account on the Astmasters VoIP service. This service supports both SIP and IAX and is predominantly intended for Macintosh Asterisk users to test connectivity and to be able to call other list members free of charge. However, it is also possible to make and receive calls to and from other networks.

Account holders will be provided with an Astmasters.net SIP URI which can be called from any SIP compliant IP phone. Regular PSTN telephone numbers can be called, provided that the numbers are E164 directory listed. Both VoIP calls and VoIP to PSTN calls are free of charge. User controlled Do-not-disturb (DND) is provided by default and incoming voicemail will be sent by email.

Astmasters VoIP Service

Ben is going to be completing peering with Free World Dialup as well. Oh, if only I had a few weeks to focus on nothing but Mac VoIP for a while...

Unrelated from Asterisk on Mac, I just found out about Jon's Phone Tool, an immensely scriptable application that lets you do all sorts of phone-related things from your Mac. Including, dial via Skype from your Address Book. Check it out.

From anecdotal evidence, my own usage patterns, and Tris and Jeremy, it seems like Skype is becoming not only the VoIP client of choice, but also the IM client of choice.

So why are people switching? Well, the average non-technical person at best has one IM account and one client -- and it's usually one of Yahoo, MSN, or AOL. But more technical users that communicate with a lot of people over IM are forced into using multi-protocol clients. While these clients can talk to all the systems, they often don't work well with basic functions like file transfer.

And this is where Skype comes in. It's a good IM client, it has wide-spread usage among the people that are likely to be using IM, and it has fully encrypted communications. The file transfers work really well (I've never had one fail, even behind firewalls), and you can even transfer contacts to someone else.

That was fast. SkypeIn, Skype's connection of "real" phone numbers to your Skype account, is live.

They are offering numbers from Hong Kong, the UK (well, sold out, apparently), France, and the US. I've seen people say they want to wait for Canadian numbers -- I'll probably just try a US number and get a Canadian one later.

So, looking at the list of US area codes, I was trying to figure out what area I would want a number from. I copied out all the area codes and looked them up here. Below is the list of SkypeIn area codes and what area of the US they are connected with:

While Duane and Greg have been contributing to the discussion about how we are going to see a lot in the standards-based SIP world soon, Skype keeps rolling out features.

Andy Abramson got alerted to some interesting sections in Skype's help docs, which seem to suggest that Skype will be offering virtual numbers very soon:

Skype's Help section can be a plethora of information. A loyal reader in the UK sent along a note about SkypeIN. While I have yet to verify that In is Out, certainly the details are emerging. Do a little noodling around Skype's help and see what I mean.
VoIP Watch: Skype In, Out?

Erin Dalzell sent me a link to a recent Cringely article ("The Best days of Voice-over-IP Telephone Service May Already Have Passed"). Cringely seems to think that VoIP is doomed because it will just be sneakily blocked (or rather, not tagged with QoS.

I think Cringely gives the telcos and big network operators too much credit. This would be non-trivial to implement across entire networks.

Aside from the difficulty or not of tagging all network traffic is the encrypting/tunneling option: much like P2P traffic that will likely start to travel encrypted over common ports (e.g. pretend it's regular HTTP traffic or VPN traffic or...), VoIP has the same option.

If the network doesn't know you're running VoIP, it can't block it.

Suffice it to say, I'm not too worried about the future of VoIP. We haven't seen anything truly interesting yet (well, Skype comes close, but it's closed, proprietary nature makes me not want to like it), and there is lots of room for innovation yet.

Alec Saunders is experimenting with SIPatH, a SIP proxy that runs on a $50 Linksys WRT-G router. Here's what his rules for the future of telecom are:

1. Your PSTN connectivity is outsourced to the lowest bidder. The only calls you pay for are calls that terminate on the PSTN.
2. Your "telephone number" is simply a SIP address terminating on your local SIP proxy. To call me, you use your SIP client to reach alec@saunders.com - the SIP proxy I have at home.
3. The telephone network, as we know it, ceases to exist. Telephony is nothing more than an embedded application running on a common transport.

Alec Saunders .LOG

He also points out Mr. Blog, talking about running Asterisk on the same router (where Asterisk is a full-fledged IP-PBX).

I suppose I should stop being amazed, but we're not quite there yet. I had a quick look at both projects, and they would both take more fiddling than I'm willing to deal with at this point. However, since they are running on commodity home networking gear, it means we're not more than a year or so away (less?) from having a shippable product that is pre-installed and relatively plug and play. In the meantime, I'm still looking for a good PSTN termination provider in Canada.