Asterisk 10 Beta 1 From New Asterisk 10 Branch Is Ready For Downloading And Testing

http://snapvoip.blogspot.com/
The Asterisk Development Team announced the availability of the first beta release of Asterisk10, Asterisk 10.0.0-beta1. The release is available download and consumption at Asterisk download site.

As we mentioned before, Digium and Asterisk.org has dropped "1." from the Asterisk version numbers and the new Asterisk 10 branch, will continue to march forward.
Of course Asterisk team is requesting your participation in testing, we are testing it with PIAF-RED;

All interested users of Asterisk are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk. Additionally users can make use of the RPM and DEB packages now being built for all Asterisk releases. More information

Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk versions page:

A short list of included features includes:

* T.38 gateway functionality has been added to res_fax.
* Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
* New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
* Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application.
* Support for defining hints has been added to pbx_lua.
* Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
* Much, much more!

A full list of new features can be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog

OpenSIPS 1.6 eBootcamp, Get Ready For A Torrent Of SIP Technology

OpenSIPS 1.6 eBootcamp http://snapvoip.blogspot.com/

OpenSIPS is holding a eBootcamp on OpenSIPS 1.6. This bootcamp is unique in itself because it is held over the Internet for over 7 weeks. Participants will learn from downloading installing and configuring and on to administration of OpenSIPS. It will start on September 19th 2011. The ebootcamp will accept late registrations up until 12th September 2011. So you better hurry. Contact bootcamp@opensips.org for registration and more information

The users will learn how to authenticate users, install a GUI to help with daily administration, forward calls to the PSTN through Dialplan, integrate Asterisk and Voice Mail, Presence agent, Load Balancing, Traverse Nat for SIP and generate CDR records to a Radius Server. At the end, you will learn how to use troubleshooting tools to solve end user problems.

This is basically the same OpenSIPS Bootcamp but help over a broadband connection over a longer period, seven weeks to be exact.

"The live classes will be taken online by web-conference every Tuesday and Thursday 03:00PM GMT, 11:00AM EDT 08:00AM, PDT. To attend this training you will need to have broadband Internet access. You are going to receive a DVD with a virtual machine to run the labs. The virtual machine will be available in the VMWARE format. You can download the free VMWARE player to run the VM. We suggest that you have one separate desktop or server for your VM and at least one IP Phone/ATA in your private labs to complete the training. A LMS (Learning management system) will be available with forums, quizzes and support materials."

The course of learning will cover key objectives;

Install OpenSIPS on a Linux Machine

Routing basics and the default configuration

OpenSIPS authentication using MySQL and Memcache

Install OpenSIPS control Panel.

Connect to the PSTN using Dialplan and Dynamic Routing

Voicemail integration using Call Forward and AVPs

Implement a presence agent

Understand important aspects of load balancing and high availability

Implement SIP NAT traversal using RTPProxy

Account Calls to MySQL

How to use test and monitoring tools to check your configuration

OpenSIPS

SIP Forum Releases SIPit Interoperability Test Results from SIPit 28 Held In in Huntsville, Alabama

SIPit 28 SIP Interoperability http://snapvoip.blogspot.com/

SIP Forum has released the test results from the latest SIPit event SIPit 28, the SIP interoperability event, was hosted by Digium on April 11–15, 2011 and held at the Jackson Center in Huntsville, Alabama's Research Park. SIPit is organized by the SIP Forum's Test Event Working Group (TEWG) and serves as a "plugfest" for participating companies to perform SIP interoperability testing with other participants in a live network environment. There were 19 companies from 10 countries who participated in the SIPit 28 to test and confirm SIP interoperability this spring. High definition IP video was a key focus at the event and IPv6 support in the industry was also on the rise, the forum noted.

"SIPit28 was one of the smoothest events to date featuring a lot of positive energy and hard work from all involved, We received great feedback from participating companies who were able to glean a good deal of valuable data which provided them a very high return.", said Robert Sparks, chair of the SIP Forum's Test Event Working Group and organizer of SIPit 28.said Robert Sparks, chair of the SIP Forum's Test Event Working Group and organizer of SIPit 28.

Follow the links after the jump.

Following is the press release by SIP Forum.

NORTH ANDOVER, MA (June 27, 2011) - The SIP Forum has announced the results of its latest SIP Interoperability Testing event, SIPit 28, the world's premier interoperability testing event for the Session Initiation Protocol (SIP) for application developers, service providers, and IP communications equipment manufacturers, held earlier this spring at Jackson Center in Huntsville, Alabama's Research Park. Hosted by Digium, the five-day testing event once again demonstrated the critical role that Session Initiation Protocol (SIP) plays in today's telecommunications industry while also highlighting the growth, evolution and adoption of SIP services in fixed and mobile IP networks worldwide.

SIPit 28 provided 19 attending companies with valuable "real-world" deployment data gathered from the 40 distinct SIP implementations conducted during the week as participants tested SIP interworking in a variety of simulated, live IP network conditions and settings. Testing exercised the full range of defined transport protocols for SIP, from UDP to TLS over SCTP.

As part of the week-long testing event, which took place April 11-15, participants showed a growing interest in deployment of high definition (HD) video services across global IP networks. Participants executed a number of different telepresence scenarios with intricate SIP signaling, testing for quality of service (QoS), network resiliency and interoperability involving new types of rich media endpoints with IP application servers, SIP proxy servers and session border controllers.

"Rolling out IP video in both enterprise and service provider networks and making it work is a key objective for many operators, equipment makers and application developers in today's telecom world and is something attending companies really focused on at this year's SIPit" said Robert Sparks, chair of the SIP Forum's Test Event Working Group and organizer of SIPit 28.

"We were able to successfully execute a number of testing scenarios that examined different network topologies under a variety of simulated conditions. Participants were able to verify correct behavior of advanced applications in an environment that is very difficult to duplicate without so many different implementations in the same room at the same time."

In addition, IPv6 was also a hot topic in preparation for the industry-wide rollout of this next generation Internet protocol, with many participants uncovering new, positive information surrounding interoperability of IPv6 equipment and services, particularly when implementations use a combination of IPv6 and IPv4 for media and signaling. Across the testing, 68 percent of the implementations at the event supported IPv6 which represented a significant increase from SIPit 27 in 2010 which saw 53 percent of attendees achieve IPv6 support.

A full summary report of SIPit 28 results is available at the SIP Forum website or by clicking here (https://www.sipit.net/SIPit28_summary).

"The SIPit events continue to be critically important and popular events for the IP communications industry because they give participants an opportunity to vigorously test implementations of their own IP communications product or service with other members of the telecom community in a non-commercial, real-world testing environment," said Marc Robins, SIP Forum President and Managing Director. "Every time we have a SIPit event, we receive scores of positive feedback about how valuable the testing was to the development of products and services, as well as how useful it is to standards bodies such as the IETF for gaining insight into the growth of SIP and the specific SIP implementations."

SIPit is organized by the SIP Forum's Test Event Working Group (TEWG) and serves as a "plugfest" for participating companies to perform SIP interoperability testing with other participants in a live network environment. Conducted twice a year, with events rotating in the United States, Europe, and Asia, the SIP Forum has hosted 28 events around the globe. In the coming months, the SIP Forum will release details about its next event - SIPit 29 - which will be hosted by ETSI and take place this fall in France. For information about past SIPits, please visit www.sipit.net.

"SIPit28 was one of the smoothest events to date featuring a lot of positive energy and hard work from all involved," continued Sparks. "We received great feedback from participating companies who were able to glean a good deal of valuable data which provided them a very high return."

SIPit via SIP Forum

SIP/IP Trunking Market Report & Forecast 2010 For UK

http://snapvoip.blogspot.com/
Research & Markets has published “UK IP/SIP Trunking Market Report & Forecast 2011”, a report and forecast on SIP, SIP trunking etc. SIP Trunking is experiencing a sharp growth in UK. The 20 page report carries a wealth of information to interested parties. Below is the table of contents and the press release;

1 Executive Summary

2 Market Definitions

3 Market Forecast

4 Market Dynamics
4.1 Strategies
4.1.1 Hosted Enablement
4.1.2 PBX Manufacture teaming
4.1.3 ISDN Replacement
4.1.4 Cloud Communications/Computing
4.1.5 Business Continuity
4.1.6 PBX Enhancement
4.2 Vendor Propositions
4.2.1 The Changing Proposition
4.2.2 IP-PBX Interoperability
4.3 Relationship to PSTN/ISDN dynamics
4.4 Pricing
4.4.1 Pricing Comparisons

5 UK SIP Trunking Service Providers

6 Challenges to Growth

7 Technology Factors
7.1 Access & QoS
7.2 Gateways
7.2.1 Interop with ITSPs
7.2.2 Survivability
7.2.3 The Future of Gateways
7.2.4 Gateway Vendors

DUBLIN--(BUSINESS WIRE)--Research and Markets (http://www.researchandmarkets.com/research/658681/uk_ipsip_trunking) has announced the addition of the "UK IP/SIP Trunking Market Report & Forecast 2011" report to their offering.

“UK IP/SIP Trunking Market Report & Forecast 2011”

As the market for IP trunking breaks 231,000, and the UK trunking market enters a period of sharp growth. illume launches its 2011 UK IP/SIP Trunking Market report and forecast. As usual illume provides a unique insight into the market, based on its regular market interviews and quarterly industry survey. The report is authored by Matthew Townend and highlights include:

• Market Forecast (2010 - 2013)
• Market Dynamics (Strategies, Vendor Propositions and Pricing)
• Identification of the UK's SIP Trunking Providers
• Challenges to growth
• Technology factors (QoS, Gateways, Broadband etc)

This 20 page report provides insight into the current market numbers and likely growth over the next 3 years. illume has also undertaken a detailed review of the current service provider propositions and strategies. As customers and providers look at SIP Trunking as a possible recession proof service the purchase of this report will enable its readers to gain a better understanding of the market opportunity that really exists.

To purchase the report click the "Buy Now button" and proceed to check out where you can select your preferred method of payment. After payment has been made you will be emailed your UK IP/SIP Trunking Market Report & Forecast 2011 as a PDF file within 24 hours.

For more information visit http://www.researchandmarkets.com/research/658681/uk_ipsip_trunking

Contacts

Research and Markets
Laura Wood, Senior Manager
press@researchandmarkets.com
U.S. Fax: 646-607-1907
Fax (outside U.S.): +353-1-481-1716

The Secrets of SIP Trunking : A Seminar At IAUG

The Secrets of SIP Trunking http://snapvoip.blogspot.com/
This Seminar aims to show people how SIP trunking is displacing traditional technologies on its way to becoming the leading service for Voice provisioning. The focus is in how SIP trunks works along with ensuring they are Secure, Reliable and able to deliver well into the future and offer new features as they become available. The Seminar also fully intends to shake things up and tell people what to watch out for when entering the SIP trunk market and why things will not always go as planned. With a whole list of Gotchas to run through, delegates will learn that SIP trunking is most certainly the future but the road to this end will not be a smooth one.By meeting the Seminar objectives of Informing, guiding and making people think; delegates will walk away with an understanding of SIP Trunking along with a lot of food for thought on a technology that's new and exciting but can also be fraught with interoperability issues that can quite often makes grown men cry.

Session Number: 10024
Track: Professional Development
Room: Neopolitan 4
Session Type: General Sessions
Presenter: (The SIP School)
Time: May 25, 2011 (08:30 AM - 09:30 AM)

REGISTRATION SITE IS HERE 

Professional Roadmap: Managerial
Experience Level: Basic/Introductory Overview
Category: General Overview
Objective #1:: Generate an understanding that SIP Trunking is the Future and to prepare for it.
Objective #2:: Understand how to approach Service providers in order to understand what they do offer and should offer an enterprise. i.e. How not to get caught out with a 'poor' provider.
Objective #3:: Understand some of the technical challenges of getting SIP Trunking right 'before' you even talk to a provider.

Keep Your PBX in a Flash ( PIAF ) Running During An Internet Outage!

PBX in a Flash ( PIAF ) Can Run without Internet http://snapvoip.blogspot.com/
If you experienced PIAF server was not able to route any calls after losing your Internet connection, even between local extensions Ward Mundy just Tweeted how we could avoid this.
The solution is to run a Bind DNS Caching Server running on your PIAF box to keep it going with an Internet outage. If you decided to use this solution, and if you are using FDQNs on your phones, the primary DNS server address must internal or external IP address of your server, where your phones are attached.
Ward Mundy, on the PIAF forum, has provided a simple script to run on your server and a set of files to get everything setup properly like ROOT server information and forwarding DNS queries to Google DNS servers.. The script is not shell script so you will have to run each line by themselves. (You can convert it into a shell script, if you have many PIAF servers running.)
After logging into your server with root capability (root or a sudo account) run the following code at the end of the article. Once you run the code and satisfied that everything is fine, run a dig command (e.g. dig inbound1.vitelity.net) for each of the FQDNs of your SIP trunk providers! This will cache the DNS information on your server to be used when the need arises. You can find the info with FreePBX: Tools, Asterisk Info, SIP Info, SIP Registry. Happy calling;

yum -y install bind*

cd /var/named
service named stop
wget http://pbxinaflash.net/source/bind/bind.tgz
tar zxvf bind.tgz
rm bind.tgz

echo nameserver 127.0.0.1 > /etc/resolv.conf

sed -i 's|$nameserver|127.0.0.1|' /sbin/dhclient-script

service named start
service network restart

dig pbxinaflash.com
dig pbxinaflash.com

# look at the output for the second dig command issued above
# 2d dig command should show a very short DNS lookup time, e.g.
#;; Query time: 1 msec
#;; SERVER: 127.0.0.1#53(127.0.0.1)
# IF ALL IS WELL...

#be sure to set named for auto start on bootup

chkconfig  named  on --level 2345
chkconfig --list named

# named should be configured to start with RunLevels 2,3,4,5

SIP Network Operators Conference ( SIPNOC ) Preliminary List Of Registered Attendees And Participants Announced.

SIPNOC 2011http://snapvoip.blogspot.com/
SIPNOC seem to be going places with ever expanding attention from CEOs, CTOs, network architects and engineers from carriers on five continents as well as government, research organizations and industry bigwigs. The first of its kind conference will be a good source of information for everyone involved in SIP Network operations or any communications operations involving SIP for that matter.
SIPNOC earlier announced that the Keynote Speech will presented by Dr. Douglas Sicker, FCC.

"When we initiated SIPNOC, we knew we were embarking on a new type of event for the industry and that there would be interest, and the response we have received has justified our enthusiasm, and then some, We expect this event to be far more than just an educational conference, but a meeting of the brightest industry minds from across the entire spectrum of the global networking operator community. Our vision for SIPNOC is to make it a place for network operators of all shapes and sizes to come together to share their perspectives and ideas on how to make IP communications better, and hopefully provide tangible results that further accelerate the development and propagation of SIP and SIP services in the years ahead." " said SIP Forum Managing Director, Marc Robins.

Follow the links below for registration and further information.

Press Release;

NORTH ANDOVER, MA (March 18, 2011) - The SIP Forum announced today a preliminary list of registered attendees and participants in the first SIP Network Operators Conference (SIPNOC), to be held April 25-27 at the Hyatt Dulles Hotel in Herndon, VA. The two-day conference, which is designed to be a first-of-its-kind international conference aimed a SIP network operations personnel, is attracting CEOs, CTOs, network architects and engineers from carriers on five continents, including AT&T, Comcast, Cbeyond, Time Warner Cable, Swisscom, COX Communications, Verizon, Uni-tel, A1Telekom Austria, Telecommunikasi Indonesia, XO Communications, and Sprint Nextel.

In addition to carrier participants, SIPNOC has also attracted a myriad of SIP community stakeholders from vendors, governments and research organizations such as Acme Packet (which has also signed on as a SIPNOC Diamond Sponsor), the FCC, Avaya, Cisco Systems, Microsoft Corp., IBM Research, CableLabs, Dialogic, Broadsoft, Ingate Systems, MetaSwitch, Polycom, Commetrex, Sonus, Sagemcom Canada, NGN Data Services Corp. and the Illinois Institute of Technology.

Registration for SIPNOC remains opens by visiting http://www.regonline.com/sipnoc_2011.

"When we initiated SIPNOC, we knew we were embarking on a new type of event for the industry and that there would be interest, and the response we have received has justified our enthusiasm, and then some," said SIP Forum Managing Director, Marc Robins. "We expect this event to be far more than just an educational conference, but a meeting of the brightest industry minds from across the entire spectrum of the global networking operator community. Our vision for SIPNOC is to make it a place for network operators of all shapes and sizes to come together to share their perspectives and ideas on how to make IP communications better, and hopefully provide tangible results that further accelerate the development and propagation of SIP and SIP services in the years ahead."

The SIP Forum has also developed a world-class lineup of speakers at SIPNOC, to be announced in detail at a later date.

SIPNOC will focus on issues critical to the reliable and successful deployment and operation of SIP-based services in carrier networks. The agenda will feature special presentations, panel discussions and workshops covering key topics by network operators related to SIP-based services and infrastructure, including testing, application development, SIP trunking, FoIP, call routing and peering, troubleshooting and monitoring, emergency services and more.

While the international carrier and service provider community is the lynchpin of the event, industry stakeholders involved in major SIP initiatives such equipment vendors, government representatives, large enterprises and research organizations are encouraged to attend.

The SIP Forum has gained an international reputation for developing important, educational events surrounding SIP. The organization's SIPit series of interoperability testing events (www.sipit.net) regularly provides a test bed for SIP-based applications and equipment that has been heralded as critical for the development of new products and services in the industry. The SIP Forum also has a number of committees and task groups made up of well-known industry experts examining a myriad of SIP-related topics, including the use of SIP in smart grid installations, FoIP, video, and user-agent configuration.

SIPNOC Registration

* To register for SIPNOC, please visit http://www.regonline.com/sipnoc_2011.

SIPNOC General Show Info

* For more information about SIPNOC, please visit www.sipnoc.org or send an email to sipnocinfo@sipforum.org.

SIPNOC Sponsorship Info

* For information about corporate sponsorship opportunities at SIPNOC, please contact Marc Robins, SIP Forum President and Managing Director, at +1-718-548-7245 or marc.robins "at" sipforum.org.

Grandstream’s Broad Portfolio of IP Voice and Video Solutions Certified With Elastix

Grandstream, Elastix, http://snapvoip.blogspot.com/
Grandstream announced today that it's VoIP IP Telephony portfolio has been certified with Elastix of Elastix Project, an IP-PBX Distribution for Unified Communication Server Solutions.

Press Release;

Grandstream’s Broad Portfolio of IP Voice and Video Solutions Certified With Elastix
Interoperability Gives Resellers, Integrators and SMBs High Quality, Turn-Key Solution

For Implementing Innovative and Cost-Effective Unified Communications

IT Expo, Miami, FL, Booth 817 (February 2, 2011) – Grandstream Networks, the leading manufacturer of IP voice/video telephony and video surveillance solutions, today announced that its award-winning SIP based voice and video CPE endpoints are fully certified and interoperable with Elastix, the largest open source software solutions for unified communications in Latin America. The coupling of Grandstream’s high quality and innovative endpoint solutions with Elastix’s unified communications software for IP PBX’s results in exceptional turn-key voice, visual communication and security surveillance solutions for resellers, integrators and small- to medium-sized businesses (SMBs) worldwide.
Interoperability assures implementers of telecom solutions are utilizing the most reliable and feature-rich endpoint equipments for business networks. It also provides a level of confidence that the CPE solutions being installed can be automatically provisioned through the Elastix software. A host of Grandstream’s advanced CPE solutions were rigorously tested and approved including the new GXP2120 and the GXP21xx and GXP14xx series enterprise HD SIP phone, GXP280 single-line phone, GXW IP analog gateways, GXV3611 and other GXV35xx/36xx series video surveillance camera, and the widely popular GXV3140 and GXV3175 IP multimedia phones.

“We are having much success to date fulfilling customer telephony needs utilizing a wide variety of Grandstream voice and video endpoints with Elastix,” said Edgar Landivar, CEO of PaloSanto Solutions, the company behind the Elastix Distro. “Grandstream is one of the first hardware manufacturers to support our certification program. We look forward to working with Grandstream and evolving Elastix’s telephony solutions to include new communication alternatives that make organizations more productive and efficient.”

“We are excited about our partnership with PaloSanto Solutions and the popular Elastix platform. Our relationship will benefit the many channel providers of Elastix unified communication solutions and business customers around the globe seeking ways to easily incorporate time and money saving technology, “ said David Li, CEO of Grandstream. “We will continue to deliver innovative SIP-based endpoint devices that fit the needs of all SMB and enterprise customers.” 

Grandstream IP CPE solutions will be visible in Grandstream Booth# 817 and Elastix Booth #1009 at ITExpo, February 2-4, 2011 in Miami, FL.

About Elastix
Elastix is appliance software that integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface. It also adds its own set of utilities and allows for the creation of third party modules to make it the best software package available for open source telephony. These characteristics added to the strong reporting capabilities make it the best choice for implementing an Asterisk-based PBX.

About PaloSanto Solutions
PaloSanto Solutions is an Ecuadorian company dedicated to development and support of open source solutions. Since 2006 is the company behind the Elastix Project, an IP-PBX Distribution for Unified Communication Server Solutions. PaloSanto began its operations in 1999 and now distributes its services through Ecuador and an extensive reseller network in America and other countries around the world. Visit www.elastix.org for more information.

ABOUT GRANDSTREAM NETWORKS
Founded in 2002, Grandstream Networks is the leading manufacturer of IP voice/video telephony and video surveillance solutions. Grandstream serves the small-to-medium business and consumer markets with innovative products that lower communication costs, increase security protection, and enhance productivity. Our open standard SIP-based products offer broad interoperability in the industry, unrivaled features, flexibility, and price-performance competitiveness. Visit www.grandstream.com for more information.
# # #
CONTACT
Lorena Leonard
Grandstream Networks
marketing@grandstream.com
617-566-9300

Samsung BCS, Broadvox Announce Interoperability Between Samsung OfficeServ™ 7000 Series And Broadvox GO! SIP Trunking Services

Samsung OfficeServ™ 7000 Series And Broadvox GO! SIP Trunking Certified To Work Together http://snapvoip.blogspot.com/
Samsung Business Communication Systems (BCS), and Broadvox, announced  the interoperability certification of their communications solutions. The certification confirms the seamless performance of Samsung OfficeServ™ 7000 Series telecommunications platforms with Broadvox GO! SIP Trunking services.
Samsung OfficeServ™ 7000 Series platforms with SIP-enabled Samsung VoIP phones, provides convergence solutions for SMBs leading to operational flexibility while increasing mobility. The communications unification becomes even more affordable with cost-saving Broadvox GO! SIP Trunking services, which is known to let businesses  reduce the phone bills costs. It is believed that the savings with SIP-enabled systems could range from 25–60% reduction of the phone bill.

“Through our strategic alliance with Broadvox, we are continually looking to foster acceptance for, and build awareness of, the benefits of SIP Trunking services for SMBs,This certification provides our dealers another tool for confidently demonstrating the seamless application of Broadvox SIP Trunking services with Samsung OfficeServ™ systems.” said Doug Wonson, Vice President and General Manager, Samsung BCS. “

The certification process included testing under rigorous conditions encompassing key elements of the systems to ensure seamless “real world” system operation. The core Session Initiation Protocol (SIP) curriculum testing comprises over sixty separate elements involving key aspects of call processing, both inbound and outbound.

 “Interoperability certification reaffirms the compatibility and combined strength of the Samsung OfficeServ™ and Broadvox GO! SIP Trunking solutions for SMB and other applications,” voiced Andre Temnorod, Chairman and Chief Executive Officer, Broadvox.

For further information;
Samsung BCS

Broadvox

SIPit 26 Test Results Released.

SIPit 26 Test Results http://snapvoip.blogspot.com/
After a very extensive week long interoperability testing event, SIPit 26 event held in Kista, Sweden, : The SIP forum has released the results. SIPit 26 drew twenty eight companies from fifteen countries to test forty two of distinct implementations. SIP Security and IPv6 internet working  were the highlights of this highly anticipated event.

"SIPit events bring together a diverse array of stakeholders from the global telecom community - SIP application developers, IP communications equipment manufacturers, and service providers - and provides them with an opportunity to glean valuable technical data about new SIP specifications and implementations important to their business, in a neutral setting, The event and the resulting data also give standards bodies such as the IETF a good progress report on the growth of SIP and the specific SIP implementations that are becoming more widely adopted in the service provider, enterprise, and consumer markets." voiced Marc Robins, SIP Forum President and Managing Director.

The test results are available at  (https://www.sipit.net/SIPit26_Summary)
 The next SIPit event, SIPit 27, will take place in coming fall.
Following is the complete press release;
NORTH ANDOVER, MA (July 7, 2010) - The SIP Forum has announced the results of its latest SIP Interoperability Testing event, SIPit 26, the world's premier interoperability testing event for Session Initiation Protocol (SIP) for SIP application developers, service providers, and IP communications equipment manufacturers, held May 17-21, 2010 in Kista, Sweden. The five-day testing event demonstrated continued growth and evolution in SIP-based technology as SIP-based consumer and enterprise applications become more broadly deployed across fixed and mobile IP network topologies.

The event, which was hosted by Edvina and TANDBERG and sponsored by Intertex, Ingate, and .se, with the IPv6 Forum serving as an association sponsor, provided 28 attending companies with valuable "real-world" deployment data gathered from the 42 SIP implementations conducted during the week as participants tested SIP interworking in a variety of simulated, live IP network conditions and settings. Of particular note was an increase over past events in the number of Secure Real-time Transport Protocol (SRTP) and TLS implementations for secure SIP interworking, and the continued and steady growth of IPv6 implementations.
"As SIP communications become more widely prevalent, securing SIP at both the application and media layers through SRTP and TLS is critical," said Robert Sparks, chair of the SIP Forum's Test Event Working Group and organizer of SIPit 26. "Other interesting developments were the growth of IPv6 and a large number of participants testing network edge-specific scenarios with ICE and TURN implementations."
A full summary report of SIPit 26 results is available at the SIP Forum website or by clicking here (https://www.sipit.net/SIPit26_Summary).
"SIPit events bring together a diverse array of stakeholders from the global telecom community - SIP application developers, IP communications equipment manufacturers, and service providers - and provides them with an opportunity to glean valuable technical data about new SIP specifications and implementations important to their business, in a neutral setting," said Marc Robins, SIP Forum President and Managing Director. "The event and the resulting data also give standards bodies such as the IETF a good progress report on the growth of SIP and the specific SIP implementations that are becoming more widely adopted in the service provider, enterprise, and consumer markets."
SIPit is organized by the SIP Forum's Test Event Working Group (TEWG) and serves as a "plugfest" for participating companies to perform SIP interoperability testing with other participants in a live network environment. Conducted twice a year, with events rotating in the United States, Europe, and Asia, the SIP Forum has hosted 26 plugfest events around the globe. In the coming months, the SIP Forum will release details about its next event - SIPit 27 - which will take place this fall. For information about past SIPits, please visit www.sipit.net.
About the SIP Forum The SIP Forum is an IP communications industry association that engages in numerous activities that promote and advance SIP-based technology, such as the development of industry recommendations, the SIPit interoperability and testing events, special interoperability workshops, educational seminars, special interest groups, and general promotion of SIP in the industry. One of the Forum's notable technical activities is the development of the SIPconnect Technical Recommendation - a standards-based recommendation that provides detailed guidelines for direct IP peering and interoperability between IP PBXs and VoIP service provider networks. The SIPconnect Compliant Certification Program is an associated program through which eligible companies can gain SIPconnect validation and the right to license the use of the SIP Forum's 'SIPconnect Compliant' certification mark - the official brand of the leading standard for SIP Trunking products and services. Other important Forum initiatives include work in Fax-over-IP interoperability, User Agent Configuration, and the opportunities for SIP in Smart Grid deployments. For more information, please visit: www.sipforum.org.

About Edvina

Edvina is a company with many years' experience in TCP/IP networking and IP Communications solutions. Edvina has been active in software development, software distribution, consulting, training and conferences since 1987. Since its inception, the keyword "open" has been Edvina's mantra - open networking, open standards and open source.

Edvina has been involved with an array of projects, including the installation, configuration and custom development and documentation of Asterisk - the Open Source PBX; VoIP training - including SIP Express Router and Asterisk; XMPP/Jabber installations and development; VoIP architectures for service providers; large-scale SIP PBX services for Universities; scalability, failover and load balancing for SIP infrastructures; and security audits of VoIP infrastructures for carriers and enterprises. For more information, please visit http://edvina.net.

About TANDBERG

TANDBERG is the leading global provider of telepresence, high-definition video conferencing and mobile video products and services with dual headquarters in New York and Norway. TANDBERG designs, develops and markets systems and software for video, voice and data. The company provides sales, support and value-added services in more than 90 countries worldwide. TANDBERG is publicly traded on the Oslo Stock Exchange under the ticker TAA.OL. Please visit www.tandberg.com for more information. TANDBERG is a trademark or registered trademark in the U.S. and other countries.

About Intertex Data AB

Intertex Data AB was founded in 1982 by current CEO Karl Erik Stahl. The company develops and markets communication and security products. Currently, the focus is on next generation broadband access products following the SIP standard and Live person-to-person IP Communication. The products include feature rich ADSL modems and firewalls prepared for SIP based IP Telephony, Presence, Instant Messaging and Video Conferencing, both over the Internet and over IMS networks. Intertex has won several International awards for creative product design as well as business excellence. For further information, see www.intertexdata.com.

About Ingate® Systems

Ingate® Systems develops firewall technology to enable global SIP-based person-to-person communication such as IP telephony, presence, Instant Messaging, video conferencing, IP-PBX connections and Unified Communications. With a history based in security, Ingate offers enterprises, service providers and carriers elegant solutions that make trusted SIP communications possible beyond the LAN. Among the company's award-winning products are the Ingate Firewall® and Enterprise Session Border Controllers (E-SBCs) including the Ingate SIParator® and the Intertex IX78 E-SBC for operator volume SIP trunking deployments. These are widely used to connect service providers' SIP trunks to enterprise IP-PBXs and Unified Communication solutions and to allow remote users outside the LAN. Ingate products are used by retail companies, financial institutions, industrial firms, government agencies and small-to-large enterprises throughout Europe, Asia and North America. Ingate Systems AB is headquartered in Stockholm. Sweden. Its U.S. headquarters is located in Hollis, New Hampshire. For more information on Ingate Systems, visit www.ingate.com.

About .se

.SE (The Internet Infrastructure Foundation) is an independent utility that acts to promote positive development of the Internet in Sweden. .SE is responsible for the Internet's Swedish top-level domain, .se, encompassing domain name registration and administration, as well as the technical operation of the national domain name register. Profits from domain name registrations are used to support projects that contribute to Internet development in Sweden through both its own operations and the financing of independent projects. For more information, visit http://www.iis.se.

About the IPv6 Forum

The IPv6 Forum is a world-wide consortium of leading Internet service vendors, National Research & Education Networks (NRENs) and international ISPs, with a clear mission to promote IPv6 by improving market and user awareness, creating a quality and secure Next Generation Internet and allowing world-wide equitable access to knowledge and technology. The key focus of the IPv6 Forum today is to provide technical guidance for the deployment of interoperability through its IPv6 Ready & Enabled Logo Programs. For more information, visit www.ipv6forum.com.

Media Contact John Gates Elevate Communications john@elevatecom.com (617) 861-3651

RFC 5727 : Change Process for the Session Initiation Protocol (SIP) and the Real-time Applications and Infrastructure Area

RFC 5727 Abstract http://snapvoip.blogspot.com/

Internet Engineering Task Force (IETF)                       J. Peterson
Request for Comments: 5727                                 NeuStar, Inc.
BCP: 67                                                      C. Jennings
Obsoletes: 3427                                            Cisco Systems
Updates: 3265, 3969                                            R. Sparks
Category: Best Current Practice                                  Tekelec
ISSN: 2070-1721                                               March 2010

        Change Process for the Session Initiation Protocol (SIP) and the Real-time Applications and Infrastructure Area

Abstract

   This memo documents a process intended to organize the future    development of the Session Initiation Protocol (SIP) and related work in the Real-time Applications and Infrastructure (RAI) Area.  As the
   environments in which SIP is deployed grow more numerous and diverse, modifying or extending SIP in certain ways may threaten the  interoperability and security of the protocol; however, the IETF process must also cater to the realities of existing deployments and serve the needs of the implementers working with SIP.  This document therefore defines the functions of two long-lived working groups in the RAI Area that are, respectively, responsible for the maintenance of the core SIP specifications and the development of new efforts to extend and apply work in this space.  This document obsoletes RFC 3427.

Status of This Memo

   This memo documents an Internet Best Current Practice.

   This document is a product of the Internet Engineering Task Force  (IETF).  It represents the consensus of the IETF community.  It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG).  Further information on BCPs is available in Section 2 of RFC 5741.

   Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at  http://www.rfc-editor.org/info/rfc5727.

You can find Free SIP Providers from the left column or from one of our links, like  Free SIP Accounts.

OpenSIPS will Be At SIPit 26, To Test Interoperability.

SIPit 26 http://snapvoip.blogspot.com/
The SIPit, or Session Initiation Protocol Interoperability Test, is a week long event held to assist people and companies involved in SIP implementations to ensure they work well with each other.
OpenSIPS project will be there on May 17-21 to perform tests on IPv6, presence integration and presence aggregation and B2BUA functionality of OpenSIPS.
SIPit 26 will be held in Kista - the telecom valley outside Stockholm, Sweden on May 17-21, and will be hosted by Edvina and TANDBERG and sponsored by Ingate, Intertex and .se.
I will not be attending SIPit this year.

Register at SIPit at EDVINA

The SIP Router Project, User Conference

The SIP Router Project snapvoip.blogspot.com

 VoIP Users Conference March 19, 2010, 17:00GMT

ZipDX using g722 or g711 sip:200901@login.zipdx.com
ZipDX PSTN: +1 567 252 2286 iNum: +883 5100 123 94882 

The SRP (SIP ROUTER PROJECT) a progressive project born out of Kamailio (former OpenSER) and SER (SIP Express Router) is holding a VoIP user conference, an audio conference which you could attend via SIP, PSTN, skype and other options.
The conference will focus on:

• the achievements of SIP Router Project so far
• the role of Kamailio (OpenSER) release within the SIP Router project
• new features in Kamailio (OpenSER) 3.0
• what it takes to upgrade from older Kamailio (OpenSER) versions
• new features in development version and future plans for SIP Router Project

The team also released Kamailio v3.0.1 recently.
 
http://www.voipusersconference.org/about/future-topics/

http://www.voipusersconference.org/how-to-participate/

Skype-OUTages

Skype-OUTages snapvoip.blogspot.com
Even though I have not been affected (I use skype from iPhone mainly) Skype is reporting outages. Starting from 29th (sign in problems and skype out), 30th had SIP channel subscription extensions problems and 31st seem to have brought in services problems. Growing Pains?
May be when my emotional shape changing  phone comes out, it will be able to report about skype heartbeat.
You can follow skype on twitter to findout about problems, status and solutions.

Skype for SIP Beta, Connect SIP Infrastructure To The Skype Community

(Skype has Approved Shoretel Gear to be compatible with Skype for SIP, with this Shoretel becomes the first IP PBX vendor to be approved by Skype.)

Skype for SIP.
Skype has announced Skype for SIP. a service to connect SIP comminications devices to Skype netwoek allowing to call any landline or mobile phones with low Skype rates. It also allows millions of Skype users to call you directly, for free.
One could also put Skype for SIP to work in many a ways like;

• With click-to-call buttons customers can contact you directly from your site or email.
• Customers can call you on a local online number.
• Your business will have a presence within the community of over 440 million registered Skype users.
• Free Skype calls to the office when travelling.
• Business as usual for home workers – the office is one click away.
• Your business will be globally accessible and visible to the Skype community and potential partners.
• Improve long distance business relationships with free inbound calling. 

 All for rates from just 1.7€¢ / 2.1US¢ per minute (A connection fee applies.) to more than 35 countries worldwide.
For technically minded,  Skype for SIP is a standards-based RFC3261 SIP interface.

• Signaling: SIP (rfc3261)
• Channels: Maximum of 300 channels (concurrent calls)
• Authentication: IP authentication or SIP DIGEST
• Transport: UDP, Support for Far end NAT traversal
• CODECS:G.711 µLaw/ALaw (20ms/40ms), G.729 (Annex A and B)
• DTMF: RFC2833 and SIP INFO
• Call Support: Call on Hold, Music on Hold, Call transfer, Early media
• Number Format: e.164 (country code and national number)

and all you need is a SIP-enabled PBX which supports either of the G.729/G.711 codecs.

You can Apply for Skype FOr SIP here.

Digium, Asterisk, SIP, Telematrix and Loquendo On A Winning Path!

Multilingual Asterisk.
I was reading an article by the VarGuy and taking into account the latest moves by Digium and it's alliances, a several light bulbs went on my head.
Digium, the Asterisk company has made alliances with TeleMatrix, (SIP and DECT SIP telephone solutions) Loquendo (speech recognition and speech synthesis) and it seems some wise people at Digium is doing the right thing for the Asterisk and basically for all of us.
The VarGuy focuses on hotel industry and what the Digium, Asterisk, Telematrix and Loquendo combination could do to wake up the industry. Telematrix is not a stranger to anyone who is involved in hospitality industry and people who stays in hotels like Best Western, Hilton, Hyatt, and Marriott, (the Varguy has the whole list, I just stated what is familiar to me). So it is up to Asterisk community to get together and approach these oppertunities with right solutions.

I travel extensively and eventhough I can get away with most international situations (I mange to converse in about six languages, all bad just like my English :) there have been many a time that I wished for assistance in the verbal communications department.
The Loquendo deal will let us customize and adopt Asterisk into many languages like Chinese, French, German, Italian, Portuguese, Spanish, Turkish and Russian. Even though it will not provide me the wisdom to speak to my Mongolian Hotel Manager in his language, might make it easier for me to dial into the hotels phone system and make my booking or reservation.
Varguy says he is going to some work at the Astricon, and I am looking forward to read his thoughts.
The Varguy, (Digium Asterisk: A New Wakeup Call for Hotels?)

Tags: Digium, Asterisk, Telematrix, Loquendo, speech recognition, speech synthesis, SIP, DECT SIP telephone solutions

GUAVA For Android, GoogleVoice Unauthorized Android Voice Application.

Update:
Sipdroid Project has left a comment as well as another reader regarding the GUAVA. I have had no time to investigate the issue further but I will. This is not the first time Micheal Robertson made me say "HUH", I did not like what he did with Lindows, but again it was his idea, his product and his company. Above link displays a letter from I will keep a tab on this story and followup
The creator of the SIPdroid, I-P-Tel GMBH also provides Free SIP Accounts. Need to Add them to my The Free SIP Account Provider List
GUAVA For Android
So the Gizmo people have released the GUAVA, GoogleVoice Unauthorized Android Voice Application for Android and now you can make calls for free over the wireless connection using Gizmo5 and Google Voice (Grandcentral). It seem to work even witout a SIM card in the phone.
In order to use GUAVA, you need:
- Gizmo5 software
- A Google Voice account
- An Android phone
- GUAVA for android software
Once you have got that combination together, using the wireless connection from the Android phone, you can do many things, according to Guava;
- GUAVA lets you make free calls to any US number using your Google Voice and Gizmo5 account.
- GUAVA lets you receive calls to your Google Voice number from wherever you have a net connection.
- GUAVA lets you make free calls to Skype users
- GUAVA lets you make free calls to Googletalk users
- GUAVA lets you send free SMS messages to US phones.
- GUAVA lets you read incoming messages (via the browser)
- GUAVA works fine without a SIM card in fact that's how it was tested.
- You can get GUAVA software from www.gizmo5.com/guava
So check out how a GUAVA tastes without having to go to Hawaii!

Tags: android, gizmo5, GUAVA For Android, NoSIMCard, skype, googletalk, Free SMS, Google Voice, SIPDROID, The Free SIP Account Provider List

VoIP Phone Supports H.323,MGCP,SIP,IAX2 And Many Codecs

Multi Protocol VoIP Phone, YWH500
A Company in China, Zhengzhou Yuneng Communication Co., Ltd, has released a VoIP Phone that fits most of the VoIP networks that exists today.
As well as supporting H.323 MGCP SIP IAX2, the phone supports G.711, G.723.1, G729 codec among others. The phones lists following features as well;
Multi-Language support (Chinese,English,Japanese,German,French,Spanish,Italian)
Tone generation and Local DTMF generation and detection according with ITU-T
call hold;Unattended/ attended call transfer
call waiting and per call-waiting blocking
call forward (Always, busy, no answer, power off)
Large 2*16 LCD with blue back light
12 function keys for operating and setting phone besides standard keys 0-9,#
Hotline
80 entries each for missed calls, answered calls and dialed calls
100 entries for speed dial number
voice promote,configurable jitter buffer size,configurable audio frame
adjustable user password and super password
http, telnet, keypad and dedicate manage tool
URL based FTP,HTTP and TFTP auto provision
E.164 dial plan and customized dial rules

You can read more info at VoIP-Info.ORG article. Pricing was not indicated.

Tags: E.164 dial plan, H.323, MGCP, SIP, IAX2, VoIP Phone, G.711, G.723.1, G729

SIP Softphone Application for the iPhone and iPod Touch, Media5-Fone.

Media5-Fone
Media5-fone from Media5, provides iPhone and iPod users with ability to connect with any SIP IP-PBX, be it an Enterprise telephony service or Service Provider Network, like the ones we have listed in The Free SIP Account Provider List
Media5-Fone lists following features;

Pure SIP standard softphone	Voice Mail Integration	Loudspeaker
VoIP over Wi-Fi	Native Contacts List	Hold
Easy Configuration	Call History	Mute

According to the Media5 press release, "the Media5-Fone enables offices as well as the mobile worker to use their iPhone as an IP-PBX extension, allowing them to access the same phone services and features as if they were in the office. Remote workers can be in contact with any other office or employee easily and efficiently, promoting collaboration across all locations.
VoIP service providers who offer calling plan can also benefit from the same SIP connectivity extension for their customers who own an iPhone."

"This new release of the Media5-Fone extends our mobile portfolio to iPhone users on the go, and offers them the key features needed to integrate an easy-to-use SIP IP-PBX extension within the iPhone,” explained Pascal Doré, Mobility Product Line Manager. "In addition to this Lite version, our engineers are already at work to bring the next fully featured Enterprise version of the Media5-Fone that will embed strong Voice security encryption among the key features. Worth mentioning that the Media5-Fone can be co-branded and custom-tailored for any specifics system needs."

The Media5-Fone application for the iPhone is available in the Apple iTunes.

Media5 Press Release could be found here.

Tags: Media5-Fone, Free SIP, SIP, iPhone, iPOD, VoIP over Wi-Fi, SIP connectivity, IP-PBX, Free SIP Account Provider List

SIP Security, A New Book From Wiley

SIP and SIP Security
The big guns at Fraunhofer Institute Fokus,who brought us the SIP Express Router (SER), has written a very good book on SIP Security.
Authors, Dr. Dorgham Sisalem, Dr. John Floroiu, Jiri Kuthan, Ulrich Abend, and Prof. Henning Schulzrinne are pioneers in SIP technology and have vested interest and knowledge of the subject SIP and SIP Security. If you need further assuarence, you can read a forward by security guru Phil Zimmerman on the site which is linked below.
SIP Accounts are used by many now for day to day communications and all major IP Telephony and VoIP carriers are deploying or already have deployed SIP Back Bones. I learned my SIP from SER, when most people were wondering what VoIP was.
I cant comment much on the book itself as I have not read it yet but I will assure you, it is something you need to have near by, if you are involved in Communications.
The authors have a website to provide information on the book.

You can read an excerpt of the book here.

Tags: SIP, SIP Security, SER