Source: snapvoip.blogspot.com

Asterisk Developer SVN server has a new branch now available for some testing, posted Asterisk Developer team member, bweschke. The new branch is at the following SVN URL:
http://svn.digium.com/asterisk/team/bweschke/originate_w_jabber/

This is based on the 1.4 branch and has had the res_jabber module that was in /trunk back-ported to it.

What does it do?

Phase 1 of the work here was to enable call progress messages to be sent via XMPP on Manager Originate actions and also within the Dial application when the appropriate channel variables are set.

What do the messages look like?

I did not get a chance to test it yet but according to the post, they are XML messages. Here is an example message so you get an idea of what the tree looks like.

progress
12345^^CALLID^^12345
PEER/1234567890: channel start

How do I make it work?

For this I have to direct you to post and the poster. It is described well enough to get you going by bweschke atJabber call progress branch.

Jabber on…

Source: snapvoip.blogspot.com

VoIP IP Telephony @ http://snapvoip.blogspot.com
- -
———-
Adbobe recently acquired Antepo, the maker of Rivoli Presence XMPP/SIP server software.
Rivoli features native support for the Session Initiation Protocol (SIP) and the Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions (SIMPLE). The company is extending that support to include VoIP and presence integration with VoIP.
Antepo extended, in Rivoli its support for Extensible Messaging and Presence Protocol, (XMPP), which is supported by Jabber and clients such as GoogleTalk.
So now Adobe is going to have a presence for it’s products, mainly acrobat reader. Here is the info you get if you try to go to Antepo.com.

"Adobe is pleased to announce that it has acquired Antepo, Inc. Antepo is a technology company that developed the Antepo Open Presence Network (OPN) System — an award-winning platform for Enterprise Instant Messaging and Presence capabilities — enabling real-time communication and collaboration while meeting critical business requirements for control, security, integration, and compliance.

The Antepo technologies and expertise acquired will support the development of Adobe’s products and solutions for knowledge workers. The addition of Antepo’s Presence and Enterprise Instant Messaging solutions will further expand the capabilities of the Adobe® Acrobat® software family for enabling knowledge workers to communicate and collaborate with confidence."

Links;
Antepo’s new home

Source: snapvoip.blogspot.com

It was a year full of achievements and events for OpenSER in 2006. The release in summer (OpenSER 1.1.0), and a continuous increase in features set, development and robustness of OpenSER. What was new in 1.1.0 could be read in the link given below.
Since then the Development community has expanded features and capabilities of the OpenSER and intend to release a new version, very soon.
Some of the intended features for the next version, OpenSER 1.2.0 and the beginning of 2007 are;
- domainpolicy - policies to connect federations
- imc - instant messaging conferencing
- mi_fifo and mi_xmlrpc - FIFO and XMLRPC transports for the new management interface (MI)
- perl - embed perl programming in configuration file
- presence - SIMPLE Presence Server implementation
- pua, pua_mi, pua_usrloc - presence user agent client implementations for user location records and management interface
- seas - connector to SIP Application Server - WeSIP - Java SIP Servlet Application Server (http://www.wesip.eu)
- snmpstats - SNMP (Simple Network Management Interface) interface to OpenSER statistics
- sst - SIP session timer support
- xmpp - transparent SIP-XMPP gateway

Read more about these in documentation section for OpenSER 1.2.0 in the link given below.

I am looking forward to see OpenSER opening more doors in VoIP IP Telephony and SIP technology in 2007

Links;
What was new in OpenSER 1.1.0
OpenSER 1.2.0 documentation

Source: snapvoip.blogspot.com

The same team that developed GTalk2VoIP has come out with Talkonaut. It is about to offer all mobile users a combination of VoIP technology and IM chats based on Jabber (XMPP) protocol. Mobile VOIP, VOIP IP Telephony technology for mobile devices are picking up everywhere, and this is one way to do it.
Talkonaut is based on revolutionary GTalk2VoIP gateway technology and allows you to make voice calls to any Google Talk[tm] and MSN/Live Messenger[tm] users, to any SIP capable device or to other mobile or landline phones for low cost. Rates are available at GTalk2VOIP, see links below.
Talkonaut application is designed to be able to run trouble free on almost any J2ME capable handset with MIDP-2.0 and CLDC-1.1 support. Though, some handsets might be difficult to configure to run Talkonaut.
Talkonaut leverages portable J2ME framework technology developed by GTalk2VoIP TEAM, which makes it possible to run on almost any mobile handset with minimal MIDP 2.0 support. A list of devices is available on site to help with handsets and PDAs with their status regarding the operation of Talkonaut. If your handset is not on the list, do not despair, it does not mean it is not supported. Just that the particular device has not been tested yet.
All IM chats made through this method is free but there is a charge for SMS and VOIP calls. Check GTalk2VoIP site or links below for information.

Links;
Talknout Home
Talkonaut feature set
List of supported devices
Rates at GTalk2VOIP

Source: snapvoip.blogspot.com

The same team that developed GTalk2VoIP has come out with Talkonaut. It is about to offer all mobile users a combination of VoIP technology and IM chats based on Jabber (XMPP) protocol. Mobile VOIP, VOIP IP Telephony technology for mobile devices are picking up everywhere, and this is one way to do it.
Talkonaut is based on revolutionary GTalk2VoIP gateway technology and allows you to make voice calls to any Google Talk[tm] and MSN/Live Messenger[tm] users, to any SIP capable device or to other mobile or landline phones for low cost. Rates are available at GTalk2VOIP, see links below.
Talkonaut application is designed to be able to run trouble free on almost any J2ME capable handset with MIDP-2.0 and CLDC-1.1 support. Though, some handsets might be difficult to configure to run Talkonaut.
Talkonaut leverages portable J2ME framework technology developed by GTalk2VoIP TEAM, which makes it possible to run on almost any mobile handset with minimal MIDP 2.0 support. A list of devices is available on site to help with handsets and PDAs with their status regarding the operation of Talkonaut. If your handset is not on the list, do not despair, it does not mean it is not supported. Just that the particular device has not been tested yet.
All IM chats made through this method is free but there is a charge for SMS and VOIP calls. Check GTalk2VoIP site or links below for information.

Links;
Talknout Home
Talkonaut feature set
List of supported devices
Rates at GTalk2VOIP

Source: snapvoip.blogspot.com

I was at news site today reading about VOIP/IM by Carl Weinschenk. He brings out ideas and thoughts brought about by an article on Light Reading by Mark Sullivan, Google: Resistance Is Futile…
Both the articles tells us how diverse is the VOIP/IM sphere right now is. Just like the early early early email. When email from one client could not go to another email client nor read it even if it was received.
Google is going about this in another angle, with google talk.
For Internet-based VOIP, Google uses XMPP, based on Jabber protocol and setting it as the standard. ÂWe said we were going to use an open standard from day one, Google Talk product manager Mike Jazayeri says. Other IM clients such as Apple’s iChat and Cerulean Studios’ Trillian also are underpinned by the XMPP standard.
Google Talk can now jabber away with users of such services as EarthLink Inc. , Gizmo Project , NetEase.com Inc. Chikka Asia Inc. , and MediaRing Ltd. , as well as with numerous other Jabber-based clients homegrown by ISPs, universities, and corporations.
When email federation happened, usage of email exploded. Usage of wireless text messaging also grew exponentially after the wireless service providers adopted the common SMS standard enabling, for example, a Cingular Wireless LLC customer to send messages to a Sprint Wireless account.
Read both the article to get a good idea of where VOIP/IM is heading!

Google is planning to go places with googletalk too! see the this googletalk blog post;
Now anyone can Talk
Google Talk is now open to everyone! Until now, users needed a Gmail account to use Google Talk. Now, anyone can use the service by creating a Google Account.

Source: snapvoip.blogspot.com

From Openser news:
New XMPP module - allow straightforward interconnection of SIP networks with XMPP networks (Google Talk, Jabber) for instant messaging.

A big step to converge various IM/VoIP networks world wide has been done with the new XMPP module, developed by Andreea Spirea. It allows exchange of instant messages with any XMPP network out there (like Google Talk or Jabber), opening the way to add presence and voice support in the near future.

There is no requirement of mapping SIP addresses to XMPP addresses via database of other persistent storage, the addressing schema allows translation on the fly. Just install the SIP-to-XMPP gateway and all your SIP users become available in the XMPP network and your users can chat with anybody in XMPP world. You can be even a SIP-to-XMPP relay for SIP networks you peer with, there is no limitation that only local users can use the gateway.

The conversation will survive to restarts, the session being recovered form the messages. As a result, there is no need of a persistent storage, the footprint is very small, embedding the gateway in small devices should be straightforward.

For more technical details see:

http://www.openser.org/docs/modules/1.2.x/xmpp.html

Source: snapvoip.blogspot.com

From Openser news:
New XMPP module - allow straightforward interconnection of SIP networks with XMPP networks (Google Talk, Jabber) for instant messaging.

A big step to converge various IM/VoIP networks world wide has been done with the new XMPP module, developed by Andreea Spirea. It allows exchange of instant messages with any XMPP network out there (like Google Talk or Jabber), opening the way to add presence and voice support in the near future.

There is no requirement of mapping SIP addresses to XMPP addresses via database of other persistent storage, the addressing schema allows translation on the fly. Just install the SIP-to-XMPP gateway and all your SIP users become available in the XMPP network and your users can chat with anybody in XMPP world. You can be even a SIP-to-XMPP relay for SIP networks you peer with, there is no limitation that only local users can use the gateway.

The conversation will survive to restarts, the session being recovered form the messages. As a result, there is no need of a persistent storage, the footprint is very small, embedding the gateway in small devices should be straightforward.

For more technical details see:

http://www.openser.org/docs/modules/1.2.x/xmpp.html

Source: voipcentral.org

The big news that we have been waiting for: Google has released their API for GoogleTalk. This is very exciting news in the VoIP industry for the implications are tremendous and applications are virtually limitless.

Google Talk uses a Jingle XMPP to establish peer-to-peer (p2p) connections as Jingle protocol is Interactive Connectivity Establishment (ICE) based and as ICE is successful in bypassing Network Address Translation (NAT), Jingle is capable of doing the same and pass through many types of NATs. No wonder biggies like Microsoft, Cisco and Google are so interested in this and supporting it.

Libjingle is a set of components provided by Google to interoperate with Google Talk’s p2p and audio capabilities. Libijingle also has source code for Google’s implementation of Jingle Signalling (JEP-0166) and Jingle-Audio (JEP-0167) which are the proposed extensions to the XMPP standard and available in experimental/trail form.

So what the GoogleTalk API is capable of doing?

For this we are borrowing Tom Keatings idea in his blog that gave a very nice example of a possible application of the Google API. He mentions about a site that mashed Google maps with Best Buy Xbox 360 inventory and tagging the Best Buy stores with pins clicking on which you can find out how many Xbox 360s are for sale. With the Google API, a possible use could be that you could use it to make a call to the Best Buy store! Also, it will give a tremendous boost in building communities that VoIP has already started. Also, as Tom rightly mentioned, it will support even online gambling. The possibilities are virtually limitless. We can rest assured that there would be highly innovative uses of this as we move into the days ahead.

We would love to hear your ideas out on this and maybe riding on your ideas, we can post an interesting blog here on Jingle-based APIs.

And what does it mean for the industry?

Well, for starters, it will put tremendous pressure on Skype to open up its API. An interesting thought comes to my mind now, really. I cant help but think the ‘Skype paradox here’. Skype actually started as an initiative to traverse what I would term as closed network connections (including NAT and firewalls) through a p2p system. Now, it suddenly sees itself as such a closed network with its proprietary p2p telephony.
Now, in the face of being isolated it would have no other option but to open up its API more than it would ever have liked. *heh*

The same is applicable to similar companies using or contemplating using proprietary p2p for voice telephony.

Source: voipcentral.org

Today, the Jabber Software Foundation (JSF) published initial documentation of Jingle. Jingle is a set of extensions to the IETF’s Extensible Messaging and Presence Protocol or XMPP for use in VoIP, video, and other p2p multimedia sessions.

The Jingle technology represents an open version of the protocols used in the popular Google Talk application released in August 2005.

Google is supporting the standardization and evolution of these protocols through the JSF’s community standards process.

The following are the specifications that were published today:

JEP-0166: Jingle Signalling The core technology for peer-to-peer session management, which enables communication through existing firewalls and can be extended to support a wide range of session types. (Authored by Scott Ludwig and Joe Beda of Google, Peter Saint-Andre of the JSF, and Joe Hildebrand of Jabber Inc.)

JEP-0167: Jingle Audio The session description format for Jingle audio sessions, enabling seamless one-to-one voice over IP (VoIP) between Jabber/XMPP users. (Authored by Scott Ludwig of Google and Peter Saint-Andre of the JSF.)

Follow-on specifications will be published in the near future for additional session types (e.g., video) as well as to document interoperability with the IETF’s Session Initiation Protocol (SIP), the ITU’s H.323 technology, and the IAX protocol used natively in the popular Asterisk open-source PBX application.

Peter Saint-Andre, Executive Director of the Jabber Software Foundation and co-author of the Jingle specifications said;

Jingle provides a powerful framework for peer-to-peer multimedia sessions.

Joe Hildebrand, CTO of Jabber Inc. and co-author of the Jingle signaling specification, added:

By laying the groundwork for real-time collaboration, Jingle is an important piece of the puzzle for our enterprise and service provider customers, and we are committed to supporting it in our products as soon as possible.

Apart from Jabber Inc. and Google, the companies and other open-source projects that have already pledged to support the Jingle protocols are

Antepo,
Cerulean Studios (Trillian),
Coversant,
Digium (Asterisk),
Gaim,
Jive Software,
Novamens,
Psi,
SAPO and
Tipic

In its press release Jabber said that support from additional vendors is expected in the near future.

Jabber Software Foundation

Source: voipcentral.org

Yahoo! is going to upgrade its dial in/ dial out feature in its instant messenger (IM). This will enable Yahoo! users to use their IM client to dial-in/out to traditional phone or mobile networks, along with new SIP-based VoIP services.

However, it must be clarified that Yahoo! is not launching an entirely new VoIP service. It is just upgrading what is currently available.

Notwithstanding the current excitement about Yahoo! jumping into the fray for a bite in the Skype market, Yahoo! is yet to make any announcement on this. However, it is expected that the announcement of the date will come soon.

When the new service is eventually launched, the estimated 82 million people worldwide Yahoo! IM users will be able to call any traditional fixed or wireless phone number in about 180 countries. They will also be able to purchase a phone number on which to receive calls. The service will come with a free voicemail box.

Now that Yahoo! has jumped in, can Google be far behind? For sometime now, Google has been seriously contemplating on gate-crashing the VoIP market with its IM Google Talk. In fact Google has come up with a most clever step in that direction was their choice of protocol for Google Talk XMPP. Jabber Software Foundation popularized XMPP and standardized it through the IETF with what they termed as ‘famous open, secure, ad-free alternative to consumer IM services like AIM, ICQ, MSN, and Yahoo!’

One of the main benefits of choosing an open platform is that it takes advantage of available client GUIs for instance. Besides, it can leverage specific and standardized extensions of the XMPP protocol or what is known as JEPs. The Jabber Software Foundation developed a set of complementary protocol extensions by allowing custom XML payloads to be developed. The one JEP which is of interest for us now is JEP-0009. This JEP defines a method for carrying XML-RPC encoded requests and responses over Jabber/XMPP.

The John Wilson developed Groovy XML-RPC module which is built upon the Smack library and its XML-RPC code to add XML-RPC support through Google talk is particularly easy to use to expose XML-RPC services through innovative use of Groovy closures. This will enable users to do remote procedure calls through Jabber.

We can rest assured that in the next 6 months to 1 year, as the heat builds up in the VoIP market, there would be a few fires flying. With big players joining in the battle for grabbing the market share VoIP has to offer, the show can be expected to be nothing short of spectacular! As passing thoughts, just leave a penny for the small players who just started.