The Good, the Bad and the Ugly

Skype have closed one more third-party this week. Nimbuzz announced that, like Fring, they were forced by Skype to stop supporting it in their Clients.
It is even more evident now, that the good signs about Skype Openness we saw beginning last year are completely flushed now, probably due their eagerness of pursuing an IPO and new management.
Although, what they did not count is the bad impact in their own user base and public image effects of such blocks. It proves once more that companies should not rely on Skype as something solid to build product features. Not only that, according to Skype, third-party MUST not build product features based on their services.
Skype also claim they have an API. Really? Not exactly, they have a Client SDK/API. And why it matters, is that if a client or device want to make use of it, they need to have exclusive granted privileges for a black box piece of software to be embedded and shipped within your product. Yes, if you want to support Skype, you need to build your product on top of that, and also make sure the product is exclusively Skype friendly. If that wasn't enough they also say that they do not guarantee for how long the piece of software will be valid.
Skype wants to do what AOL did, trying to force their users to install their Software and only use their own Browser/Client to access Internet. 
The Bad.


SIP is widely adopted worldwide specially among technical people, geeks and behind the curtains, even used as base landlines telephony like in Netherlands.

• The interoperability is something achievable theoretically, meaning that several different SIP Clients can talk to each other
• SIP release the Telecom from the "Dark Ages" to the "Modern Times", showing that Telephony could be done using IP and good software
• Vast majority of Communication equipments and platforms have support for it.
• It enabled the possibility to create huge portfolios of Value aggregated Services based on Legacy Telecom Centrals
• It is responsible for the cheap VoIP as we know it
• Most terminations that Skype buys, are provided via SIP
• Estimated numbers of SIP Providers worldwide is bigger than 50.000
• SIP Providers can sell termination minutes not only to end users, but also to other Providers

Although there are some not pretty facts about SIP:

• SIP Protocol is not as practical and simple as it should be by 2010
• SIP is not extensible, the current attempt of extending it (3GPP) is failing badly, taking more than 10 years of pure specifications, without any success to reach mass market. The current result is a HUGE stack of workarounds badly described and full of contradictions
• SIP Calling is great and almost straight-forward, but Messaging, Presence, Contacts and Extended Services are way too complex for current Market Status
• SIP is always backed by big corporations that tries to mystify it, trying as much as possible to postpone the life-cycle of the protocol.
• The current Market needs and demands extra features than only Calling. There is no such thing as a mass market SIP based service with extensive features like: Presence, Contacts, Whiteboard, Video, File Sharing etc.
• Apple Facetime is a frustrated attempt to use SIP for extended Video Call service ( Why frustrated? A service that works based on "trial-error" and does not have interoperability support in 2010, is a failure by default ). Apple end up implementing a half-SIP, half-XMPP, half-STUN solution. Which is completely closed for interoperability. It claims to implement it ALL, and supports NONE.
• The complexity of the protocol makes it hard and restricted to highly specialized developers, which have their jobs lifetime guaranteed, but decreases the potential mass adoption of the protocol. The Ugly. 

Google Call on GMail

Google Call, as I mentioned before in a previous post, was added as a Calling feature direct to the browser.
I hope this take down Skype monopoly built on top of a closed and proprietary fuzzyware.

Google now has the most powerful position on VoIP world and soon will take down Skype eagerness for a proprietary/closed network.

Google's solution is built on top of a plugin, which is embedded on new Chrome and also easy to install on other browsers. They are using Standard XMPP and Jingle at Client Level with SIP in the backend, Streaming RTP with Standard Codecs directly from the browser. ( NO CRAP FLASH TRANSCODE USED! )

Google Call offer several advantages:

• Browser Based
• Open, so they can make use of third-party clients for it ( Do you need more reasons??? )
• VoIP everywhere, browser or in your favorite Client. Solid Model.

Hope Nimbuzz adds support for it soon!

You can try it here: http://www.google.com/chat/voice/


As you asked how it works:
Google is making use of recent acquired GIPS Company Technology to add native RTP streaming directly to the browser.
For PSTN Termination, Google is using Gizmo5, together with previous Google Voice partners.
I promise to post deeper technical information like protocol details and Codec later on.

Mobile VoIP is now Illegal in Egypt

The Government of Egypt made mobile VoIP officially illegal this week.
It would even be acceptable if it was for all VoIP, but not. VoIP still acceptable if provided by the State Hold Telecom Companies of Egypt.
Main blocks includes Skype and known international SIP providers. Although I'm not the biggest fan of Skype closed protocol, I consider this completely unfair and offensive against Skype.

More than a political problem for Egypt citizens this is a crime against their freedom.

It is more than clear that the only reason is money, as State Telecoms were almost bankrupt end of last year. This is a criminal behavior aiming the State and not Citizens that should be condemned by other countries as well.

Jingle fortunately still not blocked, so if you blocked on Egypt, feel free to re-enforce mobile VoIP usage over Jingle.

Suggested Clients:

• Nimbuzz on Symbian, Windows Mobile, Android and IPhone
• Native GTalk Client for Maemo

How to use your SIP Server with Nimbuzz Client

You can use Nimbuzz Client (Mobile or PC), which is a XMPP Jingle Client to connect to your SIP Provider or Personal SIP Server(Company Asterisk or iptel.org Services for instance). This proves how powerful and inter-operable Jingle is.

There are some requirements in order to use it:
* Your SIP Server/Provider MUST be connected to the Internet in a public IP.
* Your SIP Server/Provider MUST provide or act like a RTP Proxy. (Asterisk does that by default)
* Your Nimbuzz Client MUST be connected to the Internet, through a connection that doesn't block UDP/RTP Traffic and also allows TCP connections to port 5222, which will be used by the XMPP Connection.

How to setup with your own SIP Server(General):
1) Setup your SIP Server using your domain on port 5060. Setting up a SIP Proxy is optional.
2) Enable RTP Proxy, in order to guarantee audio both ways, even when users are behind NAT.
3) Create the users/passwords.
4) Make sure to configure your firewall correctly enabling UDP in/out traffic to Internet on port 5060 and in all the port range used by your RTP Proxy (usually 15000 to 65000).
5) Log on Nimbuzz Client using your Nimbuzz Account, go to SIP Settings and use the credentials that you created previously:
Username: user01@yourdomain.com
Password: pass
Proxy: (Leave it blank if you didn't setup it)
6) Done, you now should be able to place and receive calls in your Nimbuzz Client using your own SIP Server. (Refer to troubleshooting section in case of issues.)

How to setup with using SIP Provider(iptel.org in the example):
1) Log on Nimbuzz Client using your Nimbuzz Account, go to SIP Settings and use your SIP credentials that you created previously:
Username: userABC@iptel.org
Password: pass
Proxy: sip.iptel.org
2) Done, you now should be able to place and receive calls in your Nimbuzz Client using your SIP Provider. (Refer to troubleshooting section in case of issues.)

Troubleshooting:

* If you cannot even get registered to your SIP Server/Provider:
Double Check your credentials on Nimbuzz SIP Settings and also in your SIP Server Configuration.

* If you cannot get Calls completed:
Debug the SIP Signalling and also refer to your SIP Server Call Routing Configurations. (In Asterisk check for extensions.conf).
When using a SIP provider contact your provider for details of how to place calls(Number format, Valid Destinations, etc).

* If you cannot get Audio Both Ways:
Make sure your SIP Server/Provider has RTP Proxy enable. It is very likely that if you are experiencing this situation, you or the person you are calling to, is behind a NAT.
The interoperability of this service depends on RTP Proxy availability to guarantee both ways audio for users behind NAT.
Check you SIP Server Configuration or contact your SIP provider for further details and support.
Once the RTP Proxy is SIP Provider/Server responsibility.
Also make sure you are using an Internet connection that does NOT block UDP/RTP Traffic. This is a requirement. If using 3G check with your carrier about UDP/RTP restrictions.

I hope this article clarifies a little how to integrate to your SIP Services. If you already using/used another provider, please post comments about how you did it.

Google Voice - So... What New?

Google Announced officially the release of Google Voice.
Besides all market posts and mass comments, the set of new or innovative Voice Features so far are empty. They are just trying to move backwards in time when talking about obsolete and non simple features.
I honestly was expecting much more than a service binded to a real mobile number attached to a formal operator.
In a time that everybody is questioning the meaning of a phone number in the "post email ERA", Google came and released a service based on operator number.
When it might be a short term boom, but as other examples, this might achieve some success not for an innovative concept, but based on Google fame. Google fans probably will use it just to be recognized as early-adopters.
Where is the Interoperability? The Extensibility? The Open Service API? It seems like some basic principles of new Internet where forgotten somewhere under operators shoes.

In short words: "Google Voice does basically what your phone and your operator together always did."

If you are really looking for something that your phone never did before, try Nimbuzz Mobile and feel the real sensation of doing something that your phone never have done before.

Nimbuzz will show you that a "phone number" is the last thing in the world that you need in a full communicator solution.

Nimbuzz uses XMPP and Jingle in order to provide the biggest and feature rich mobile platform ever.

My Fridge XMPP Client

No, it is not a Jetsons Episode! A few weeks ago I spoke with a guy from an appliance Industry, and guess what?
XMPP is being tested as the solution for domestic devices interactivity.
Meaning that in some years you will probably check in real time the presence of milk and ice cream in your fridge directly from your phone, wherever you are! (Via Nimbuzz?)
Interesting things you might be able to do:
* Monitor you food consumption at the end of the month.
* Check whats missing at the supermarket. Goodbye to shopping lists.
* Check if you forgot your oven on, and be able to turn it off remotely.
* Speed up the cooling of your beer before you get home.
* Get updated in real time and receive pictures in case someone break into your house. Directly in your phone.



I cannot tell the company name, but they promised to have something in the Brazilian market by 2011 as pilot.

Check some Jetsons and Inspector Gadget episodes for more ideas.