Making Use of Wideband Voice Right Now!: IdeaSIP

It’s been a while since I revisited this series on Make Use Of Wideband Voice Right Now! Over the past six months a number of things have changed. Google bought Gizmo5 and has for some time shut down new account sign-ups while they work out how to incorporate Gizmo5 into their voice applications.

With the demise of FWD as a SIP service earlier in 2009 Gizmo5 was probably the most high-profile, publicly available free SIP registrar. Their loss has to be felt as something of a setback to those who want to use HDVoice Right Now! Perhaps over time Google will make up for the current inconvenience.

However, there are less obvious SIP service providers out there who still offer basic service for free. Some of these remain a good way to get started using HDVoice. This time around I’d like to turn my gaze to Neil Fusillo’s IdeaSIP.

On the surface IdeaSIP service resembles FWD of old. They allow free registration to their SIP proxy, even using the term “Talk For Free” on their home page banner. I was recommended to IdeaSIP by Zeeek about the same time that I started listening to VUC conference calls. I can recall Neil participating in VUC calls from time to time.

The IdeaSIP web site is simple and easily navigated. The support section has some genuinely useful info including a forum section. The forum seems lightly trafficked, but I found some useful stuff there, like a link to one members written explanation of how to setup Asterisk  to access your IdeaSIP account.

IdeaSIP allows free basic accounts, charging for premium services, including enhanced voicemail and outbound termination to the PSTN (aka IdeasOUT). They had also been offering DIDs in various countries through a program called IdeasIN, but that service was recently discontinued.

Their outbound termination is offered at nominally competitive retail rates. For example, calls to the continental US are 0.018/min while calls to the UK are 0.02/min.

However, for the purposes of this article I imply don’t care about any of that!

Now on to what you really need to know….

IdeaSIP accounts are all numeric, with a format of 1101xxxxxxx. Each is exposed as a SIP URI as 1101xxxxxxx@proxy.ideasip.com. You can also get to/from IdeaSIP using SIPBroker access numbers in many countries around the world. Most of the time I’ve had my IdeaSIP account forwarded to another SIP URI, which is a handy service that they provide within the scope of the free account.

The exciting and useful part of IdeaSIP is that they allow dialing by SIP URI, and they don’t unnecessarily proxy the media stream. The end result is that from your IdeaSIP account you can make HDVoice calls to any other IdeaSIP user with an HDVoice capable phone. You can also dial by SIP URI to other HDVoice capable phones at other ITSPs. For example, I can call from my IdeaSIP account to my OnSIP accounts via SIP URI.in HDVoice

All of that is possible with the free basic account. No monthly account fee. No minimums. Just pay-as-you-go for services used. Not too shabby, eh?

Incidentally, IdeaSIP has only made limited use of twitter thus far, but one of their tweets noted that the VUC call is available via a convenient local extension.

Users can now dial 882 (VUC) to reach the ZipDX VoIP Users’ Conference bridge directly from IdeaSIP. Non- users dial 882@proxy.ideasip.com.

In an earlier tweet they announced iNum dialing from IdeaSIP accounts

Just added iNum dialout support from IdeaSIP. If you want to dian an iNum number, just dial 883 XXX XXX XXX XXX from your IdeaSIP phone.

To prove all of this HDVoice capability I started by setting up my Eyebeam soft phone to log into my IdeaSIP account. Once registered I enabled only the G.722 wideband codec. Then I called my Polycom IP650 by way of my work account, a SIP URI hosted by Junction Networks / OnSIP.

When the call was answered the Polycom phone showed the little HD icon indicating positively that the call media was wideband. Of course, I could hear that, too!

Making a further test I called the ZipDX wideband demo service at sip:wbdemo@conf.zipdx.com. This service  tells you if your phone doesn’t allow G.722. It announces that the demonstration requires wideband…but then plays it for you in low-fi anyway.

Calling from my IdeaSIP account I could hear the ZipDX demo in full wideband splendor. While on the call if you tap the # key you toggle between G.722 and G.711 media streams. This dramatically emphasizes the difference between narrowband and wideband.

One nice thing about the ZipDX wideband demo is that it provides a mechanism for checking to see if your SIP service provider handles SIP re-invites correctly. Once you get into the sample recording if you press 2 the ZipDX bridge will re-invite the call media to another path. On some services, including OnSIP, the audio will stop at that point, which indicates that the re-invite was not correctly handled. Happily, IdeaSIP handles this transparently, so all you hear is a recording indicating that the re-invite occurred.

These three simple tests prove conclusively that IdeaSIP supports the use of HDVoice, even with their free accounts.Their premium services are priced reasonably and overall the service seems to be very reliable. What more can you ask?

If you have the wideband capable phones why not enjoy the benefits of real wideband calling? IdeaSIP offers yet another great way that you can start taking advantage of open standards based HDVoice Right Now!

  • Greg

    FYI.. if you use a stun server with Onsip (I know, this goes against their recommendation) re-invites work properly with the wbdemo. I thought I was the only one having that issue.. what a relief!

    • For me this re-invite issue was only annoying with OnSIP. It was the reason that I’d never get audio when ZipDX called me to bring me into the call. However, I do plan to open a trouble ticket with OnSIP to get this investigated, as it ought not to happen. David Frankel of ZipDX is very helpful is getting such matters documented and addressed.

  • Greg

    I hope you are successful. I have opened three tickets with Onsip regarding re-invites with no meaningful response or confirmation that there is a problem. Last weekend I sent them data from traces using wireshark that shows the issue and a possible reason for the issue. The “uasnat=yes” flag is set in the Record Route line of the SIP header when a Re-invite occurs. Captures with other providers (that respond to re-invites correctly) have no such flag set or do not have the Record Route line at all. Otherwise the captures are identical.

  • Michael,

    I just noticed your write-up. Glad to hear it’s all working well for you! As always, feel free to let me know if there are things which don’t work properly, or which could use a little more finesse, or even if there are just things you’d like to see that we might be able to offer. We’re working to get our IdeasIN International DID service back via a different provider at the moment, and we have a few other odds and ends in the pipeline, but we’re always eager and willing to hear suggestions and comments from our users.

    Neil Fusillo