Thursday, December 22, 2005

LIBJINGLE in IM/P2P jungle? GoogleTalk explains what it is!

In one of my previous posts VOIP IP Telephony: A week in VOIP and IP Telephony, I mentioned about LIBJINGLE and I was excited about it. Apparently, there were many others, according to googletalk blog itself. The search link it gives lists my site in page two!. That is fine. This will be a new era in IM P2P and VOIP, IP Telephony.
As much the excitement, there were some confusion as well about the LIBJINGLE library. So Mike Jazayeri, the Product Manager has taken the task up and explains the scenario's and capabilities as far as GoogleTalk is concerned. I think it is a good read and here are the topics that Mike decided to talk about;
1. Voice calls between other PC IM/VOIP clients such as Gaim, Adium, Psi, etc. And Google Talk.
2. Voice calls between mobile devices and Google Talk!
3. Peer-to-peer applications.

I am sure there are more ways than the above to use this library. I am for one playing with my Jabber server and Asterisk server. I will let you know how it goes.

Tuesday, December 20, 2005

VON 2006 by PulverMedia coming to San Jose

VON is the industry event for VoIP, now in its 10th year of delivering for C-level attendees around the globe.
Mark your calendar NOW to attend Spring 2006 VON, set to be the biggest and best yet.
Spring 2006 VON Conference & Expo will be taking place March 14-17 at the San Jose Convention Center in San Jose, CA.
You can check out the schedule here and if you want to catch the early bird registration this is the place to go.
I will be seeing you there.

Saturday, December 17, 2005

A week in VOIP and IP Telephony

Microsoft enters VOIP arena with MCI
Microsoft has announced plans to release its first beta version of Live Messenger with VoIP.
Microsoft has staked a claim in the US VoIP market by announcing its software users can soon make Internet calls to standard phones thanks to a new partnership with MCI. Although rival products from the likes of Yahoo and Skype offer extra features, given Microsoft's broad market reach elsewhere and its alliance with MCI, this should not affect the company's foray into VoIP in the longer term.
Jingle All The Way
Today, two major advances have been made in the openness of our voice capabilities. This morning, the Jabber Software Foundation (JSF) introduced two new proposed extensions to XMPP, known as Jingle and Jingle Audio. These enhancements describe how to write software compatible with Google Talk's voice features and have been introduced into the JSF's standards process where they'll be reviewed and improved by the XMPP community. To make implementing these extensions even easier, we've released a library we call "libjingle."
Libjingle is the very same code Google Talk uses to negotiate, establish, and maintain peer-to-peer voice sessions, packaged as a library for other developers to use in their own projects. By incorporating Libjingle into your project, you enable its users to voice chat with other users of the Google Talk service.

Jingle bells continued... Google opens up google talk
Google has released this source code as part of our ongoing commitment to promoting consumer choice and interoperability in Internet-based real-time-communications. The Google source code is made available under a Berkeley-style license, which means you are free to incorporate it into commercial and non-commercial software and distribute it.
You can use any or all of these components.
* base - Low-level portable utility functions.
* p2p - The p2p stack,including base p2p functionality and client hooks intoXMPP.
* session - Phone call signaling.
* third_party - Non-Google components required for some functionality.
* xmllite - XML parser.
* xmpp - XMPP engine.

All Vonage VoIP Users Now Get E911 Service

Leading VoIP provider Vonage said Wednesday that its entire customer base has access to enhanced emergency 911 services.

“Today, any Vonage customer in the U.S. who dials 911, will get help when they need it most,” read the announcement.

Providing E911 E911 capability has held up the deployment of many VoIP providers and Vonage was no exception. The Federal Communications Commission extended the deadline for compliance to Nov. 28, but then extended it beyond that date without naming a specific date for compliance.

Tuesday, December 13, 2005

To Skype or not to Skype? Debate goes on and All agree that Skype is NOT enterprise ready, as of now!

Network World has conducted some tests regarding viability and usability of Skype use in the enterprise. I have spoken about this before and received some bad press from the Skype VOIP product users. I was not against the Skype nor any other IP Telephony or VOIP application. I was against the behavior of Skype in a network. Skype uses an unbridled use of network, draining available Internet bandwidth. It might be ok for a user with broadband connection. But think of a T1 connected office network with 10 or 20 Skype users!. It might be even worse if some of them have public IP addresses. (This is rare in an office network since all of them are behind some kind of firewall).

If you are a Skype user or a network Admin worried about Skype's network usage or related security, I think Network World's debate on whether Skype is ready for use in the enterprise is a good read. I am sure all will be able to see some light in the VOIP , Network or security tunnel created by Skype and you will agree that the peer-to-per communications software is not ready, for enterprise.
Start with Tim Greene's Beware of Skype. He says "Corporate IT time is better spent now keeping Skype off corporate networks , difficult as it may be, than trying to make it safe for business use." But not if you read my previous article VOIP IP Telephony: How to kill a Skype� / remove Skype� installs from your network! the task is much easier than all these people think.
Then you can go to other article/s which I enjoyed reading. Hope you do too.
I will be writing more on this soon.

Saturday, December 10, 2005

Yahoo undercuts Skype, and others on call rates

Yahoo, which had Voice over it's messenger service (PC to PC) for a while now, did not have ability to call-out to land like skypeout service. But things seems to be changing. All the main stream media is reporting this as it is the beginning of the world.
Yahoo has it's eyes set on many directions now, It is getting tasty now with purchase.
Hope it can cope with the additions like google does. I use services from both, Yahoo and Google. Both have their goods and bads but it seems I happened to be more at Google. I really don't like to be attached to one service so I hope yahoo to be the other service to keep me away from Google. I have been dealing with MSN as well but it need to do more to pull me over to them.

Friday, December 09, 2005

Phone in a USB Stick called PhoneDrive! maybe we call it Skypedrive!!

I saw this voip gadget on the IOCELL website, a korean company specializing in USB memory products. After digging further into the information, I found that Memsen is selling the same product here in the US of A. They call it SkypeDrive but the details are not clear at the moment.
PhoneDrive is the world's first USB flash drive for VoIP transmission incorporated according to a press release from the US counterpart (I might be wrong) of the Korean IOCELL,

"This is an intelligent product and service that will change the way people view USB drives and is the key to allow people to thrive in a user-friendly and convenient digital environment. Using the USB drive equipped with Phonedrive technology enables people to carry, store and launch their own phone calls from any PC in any place through the Internet. The Memsen Phonedrive also enables users to place calls to land and cellphones for a nominal per minute fee. Most importantly, The Memsen Phonedrive has a built-in security system that automatically deletes any personal information stored in the USB drive by a remote server if lost and stolen. You can even make calls to the Phonedrive!"
"The Phonedrive is flexible and compatible with other VOIP service providers and calling card minutes providers. Memsen is currently in talks with distributors of USB drives from other countries in the world to ensure multi-vendor interoperability and compatibility. The Memsen Phonedrive will change the way individuals and business entities use and access information from USB Flash drives."

Wednesday, December 07, 2005

Cingular HSDPA Phone announced! See and talk at the same time!

HSDPA High Speed Downlink Packet Access
Cingular's HSDPA test network in Atlanta, using Lucent equipment. The
tests reached sustained data rates over 3 Mbps. Initial HSDPA devices
will suppCingular and Lucent today announced the first successful data calls onort 3.6 Mbps peak speeds. As faster devices are released, the network can ultimately deliver theoretical peak data speeds of up to 14.4 Mbps. HSDPA is an enhanced version of UMTS/WCDMA 3G technology.

The trial network was first announced in May. Cingular has already contracted with Ericsson, Lucent, and Siemens to provide HSDPA network equipment that should be launched most major markets by the end of 2006.
Read more at Lucent.
Read the Cingular News release

Monday, December 05, 2005

Google Talk have blog now!

I wrote earlier about google talk here and I found today google blog that there is a blog page dedicated to google-talk opened.
There is not much yet on the blog but you can be the first to read first article, go there from here!
What is google talk? From the google talk site;
"Google Talk is a simple and free way to talk with and send instant messages to your friends. Like Gmail, Google Talk uses Google's innovative technologies to help people communicate more effectively and efficiently. Think of it as Google's approach to communications.

Google Talk is easy and intuitive to use. All you need to make free calls is an Internet connection, a microphone, and a speaker. After you download Google Talk, sign in with your Gmail username and password. Invite your friends to download Google Talk, and once they do, you'll be able to talk or IM with them instantly."

Lots to Talk about!

Friday, December 02, 2005

OpenSER version 1.0.0 Is released

OpenSER, a fork of SER, Has released the Version 1.0.0. For those who does not know what OpenSER and SER is, Read the end of the article. Let me write about OpenSer Version 1.0.0 first.
New in OpenSER modules

acc module
[*] call leg accounting support

- proper accounting information can be stored when the server deals with multiple redirects
[*] accounting of failed transactions based on flags

- failed transactions can be stored based on specific flag
[*] usage of pseudo-variables format in parameters

- any pseudo-variable can be now stored as adjacent information
[*] the module is compiled by default with database support

avp_radius module
[*] enhancement to support loading AVPs having integer ID or value

avpops module
[*] formatted printing with pseudo-variables support - avp_printf()

- this allow string concatenation of avps, strings and pseudo-variables
[*] perl/sed-like substitution on AVP with string value - avp_subst()

- this allow substitution or extraction of parts from an AVP
[*] support for arithmetic operations with AVPs

- integer operations with avps can be done (add, sub, mul, div, mod)
[*] support for bit-wise operations with AVPs

- bitwise operations can be done with AVPs (and, or, xor, not)

- bitwise checks can be performed via avp_check (and, or, xor)
[*] more operators for avp_check()

- new operators: ne, le, ge + bitwise operators can be used
[*] cast function with avp_copy()

- convert the value from int to str and vice-versa
[*] uri parameter for db-related function can be taken from an AVP

- along with R-URI, From and To, the URI can be now loaded from an AVP
[*] new function to check if an AVP exists -- is_avp_set()

- useful function to test which AVP exists when loading all AVPs for an user
[*] hexadecimal format for integer values

- avp_write(), avp_check() and avp_op() allow hexa integer values as parameter - to ease bitwise operation handling
[*] avp_write() and avp_pushto() can access and set the value of 'dst_uri' field (outbound proxy address)
[*] avp_write() allow $hdr(name) to be coherent with avp_printf()

- old format $hdr[name] is still valid
dispatcher module
[*] possibility to change host:port in r-uri (ds_select_domain())
[*] round robin distribution per process (alg=4)

group module
[*] uri parameter for is_user_in() can be taken from an AVP

lcr module
[*] added support for gateway prefixes

maxfwd module
[*] MAX-FORWARDS cannot exceed 256 (as per RFC3261)

nathelper module
[*] possibility to replace origin IP in SDP (o= line)

- SIP devices which check o= line can be now used with nathelper
[*] nat ping with OPTIONS requests (stateless)

- possibility to interwork with NAT boxes which close the pinhole when no traffic goes from behind the nat

- made stateless to reduce memory consumption
[*] possibility to set nat pinging method per user

- you can set per user what type of natping to send (four bytes or OPTIONS)
postgres module
[*] transaction rollback for failed queries

- postgres module encapsulates every query into transaction. If the query failed, the transaction was not finished, thus a new database connection was created on the next query
rr module
[*] enhancements which opens the road for a dialog awareness support:

- RR API - exported functions:
** add_rr_param()
** check_route_param()
** is_direction()
** get_route_param()
** register_rrcb()
[*] added callbacks - can be registered callbacks to be executed when local Route is found and processed.

textops module
[*] pseudo-variables support in subst(), subst_uri() and subst_user()

- parts of sip message can be substituted by dynamic values
[*] function to check the request's method using ID

- faster comparison of request/reply's method

- you can test the method against a set of values (is_method("ACK|BYE"))
tm module
[*] usage of pseudo-variables format in parameters

- any pseudo-variable can be sent via fifo or unix socket to external applications
[*] support for delayed CANCEL

- canceled transactions will be marked to be able to cancel delayed replies
[*] new function t_check_trans()

- checks if the request belongs to a transaction
[*] new function t_was_canceled()

- returns true if the transaction was canceled from the UAC side

- request forwarding functions return false if the transaction was already canceled
[*] pending callbacks

- support to register TM callbacks prior the transaction is created
[*] t_flush_flags() - flush to Transaction (UAS side) only the global flags

uac module
[*] annonymization finalized

- full ability to replace and restore the From and To headers in subsequent requests
uac_redirect module
[*] - special module to handle redirect replies on server

- redirects via 3xx replies can be handled on server

- address filters can be set to allow/deny redirects
xlog module
[*] use pseudo-variables in xdb() and xlog()
[*] if the first parameter of xlog() is ommited, the message is printed to L_ERR level

additional tools
[*] - script to create Postgres database structure
[*] sc.dbtext - script to manage dbtext database structure

The ChangeLog file keeps track of all important changes:

They also have listed the differences between OpenSER 1.0.0 and SER 0.9.
New in OpenSER vs. SER-0.9.0

NOTE: The next list presents what OpenSER brings new than SER 0.9.0. Several are backported from CVS head and the ones marked with (NEW) are newly added.

NOTE: The structures for 'usrloc' and 'aliases' tables has changed to store the incoming interface details. This solves the issues when dealing with NAT-ed clients and offers better support for SIP replication - requests, replies and NAT pings are sent to UA using same interface where the requests from de UA were received, so the NAT will not drop them.

Now here is info on both the SIP servers;
OpenSER is a project spawned from FhG FOKUS SIP Express Router (SER). The reason for this new venture is the lack of progressing and contributions to the SER project from the other SER team members as well as the reticience to new contributions from project's community members. We want to accelerate the integration of public contributions to the SER project.

OpenSER promotes a new management policy (OPEN) -for new code acceptance and code-through propagation- and development approach -design and architecture. We have decided to bring more dynamics into SIP world by creating this new project that can benefit of TLS and so many other contributions. We welcome your contributions to the success of this project.

The project is managed by three of SER developers. You all like SER so we will do our best to maintain our exiting work and help the users of our solutions. We will therefore maintain close contacts with our former colleagues and SER project.

Thursday, December 01, 2005

Hear Me? See Me? Yes! on Skype Beta "Skype 2.0"

Skype, which is a part of eBay now, is to offer video calls with the latest version of its internet telephony (VOIP or IP Telephony) software. Well Where do go from here? Users rejoice and Network Admins? More reasons for Skypekiller?. I don't really know because I did not test it yet.
But we do report all things VOIP and IP telephony related. Since there is a volume of Skype users large enough to fill a small country, Skype is a part of VOIP and here we go! Skype conference!!
"At Skype we want to make talking over the Internet the most natural, simple thing for people to do all over the world. With the release of our new software, it's never been easier for people to talk to one another for free, and now they can see each other with video as well" said Skype chief exec Niklas ZennstrÃm.

Skype has teamed up with Logitech and Creative who make webcams and headsets needed to use the Skype service. There are other systems and phones that also can be used with Skype.Onee or two listed here.
I think Sony can say good bye to it's IVY, software released together with Glowpoint, a IP Telephony provider, Lest that too might have rootkits accompanying it. IVY league is ok, but here we consider IVY to be a *&%^$.
Here are some info from current Beta's change log;
# feature: Skype video (webcam)
# feature: contact grouping
# feature: quickfilter in contact list and history (enable from Options->Advanced)
# feature: new language - Portuguese (Portugal)
# feature: new sound events for chat user join, leave and incoming message to existing chat
# feature: chats and conference calls shown in history
# feature: expandable "My Panel" (mood, events, services integrated)
# feature: add contact directly from main window
# feature: show file transfers in history
# feature: quickfilter in history
# feature: delete single or multiple history items
# feature: compact chat participant list
# feature: save user's last auth request
# feature: API commands GET/SET PCSPEAKER
# feature: API command SET VIDEO_IN
# change: auth requests new design
# change: changed group selection hotkeys to Ctrl-PgUp/PgDn
# change: call tab visual layout changed
# change: tray icon connecting animation
# change: changed "Free Internet Telephony" to "The whole world can talk for free" in file description
# change: import contacts error dialog displayed when there is nowhere to import contacts from
# change: file transfer in DND mode open file transfer dialog in minimized mode
# change: new installer and uninstaller icons
# change: history limited to 30 days for all events
You can read more and download from, Skype Site.
Community discission here

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