Switching to SIP? Four Facts to Check When Switching to SIP

SIP Trunking

SIP Communication in the business sector is getting traction as the businesses realize they can save a packet, sometimes up to 70% from traditional phone communications costs, by switching to SIP. So the SIP trunking is too gaining attraction to fill the technical needs of SIP communications.
SIP is traditionally a simple calling protocol yet there are some things to watch out when making the switch. Sangoma gives us Four most important and how to tackle them,

SIP Needs Security

Since SIP is in plain text, it doesn’t take an IT genius to decipher a SIP session (call). Without the proper security measures, anyone can run a packet capture tool (such as Wireshark) and extract audio from calls.

The good news is that this is an easy problem to solve with Secure Real-Time Transport protocol (SRTP) and a Session Border Controller (SBC), which is basically an application-layer firewall designed for SIP.

Make SIP (a) Priority

Between the media streaming and file sharing, chances are you have a lot of traffic on your network. If you implement SIP trunking on top of that without preparation, you may have delays in video buffering, slow email send speeds, or decreased audio quality on your VoIP calls.

To avoid this, make sure you utilize the standard QoS feature found in most business-grade routers and switches. The QoS feature will ensure your voice calls receive priority on the network, ensuring the available bandwidth is delivered straight to them first before less important network traffic.
Prior to SIP, Know Your Network

When switching to SIP, it’s important to have an accurate idea of how many concurrent calls your business makes. Review your call logs and understand the total number of calls your business makes, as well as how many concurrent calls are made at your busiest times.

SIP Bandwidth Needs 

The second thing to understand about your network is that SIP depends on bandwidth, and a lack of it can cause poor audio, dropped calls, and busy signals. Luckily, bandwidth is cheap and usually readily available!

Make sure you have enough to support the maximum number of concurrent calls your business requires. (To calculate the necessary bandwidth, simply multiply the maximum number of concurrent calls by Number of max concurrent calls x 100kb/sec = average bandwidth per call needed.) Simply add the bandwidth required to the amount you already use for business duties, and you should be good to go.

If You Fax Over SIP, Be Smart.

Faxing over IP can be very messy, particularly when it comes to the T.38 fax limitation. In very simple terms, a fax message cannot be compressed in the same manner as a voice packet. Packet loss with a phone call can result in poor audio quality, but you can often still understand what the person is saying. With faxing over IP, any packet loss can potentially cause the fax to fail completely.

That being said, if faxing is important to your business, then you should consider a reliable FoIP (fax over IP) solution such as FaxStation. With Sangoma’s FaxStation you can enjoy an analog fax experience without the added cost of an additional PSTN line.

You can find more information and solutions at Sangoma.

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.

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.

SIPNOC Keynote Address Will Be Given By Dr. Douglas Sicker, FCC Chief Technologist.

SIPNOC Keynote Address Will Be Given By Dr. Douglas Sicker, FCC http://snapvoip.blogspot.com/
Federal Communications Commission Chief Technologist Dr. Douglas C. Sicker will be the keynote speaker for the first annual SIP Network Operators Conference (SIPNOC), being held at the Hyatt Dulles Hotel in Herndon, VA, April 25 -27, 2011, SIP Forum announced in a press release.
Dr. Douglas C. Sicker who also hold a position as an associate professor in the Department of Computer Science at the University of Colorado at Boulder, is responsible for advising the FCC on communications technology issues, through the Office of Strategic Planning and Policy Analysis.

“I'm delighted to participate in this inaugural meeting of SIPNOC, In recognizing the importance of interconnected VoIP service as a growing part of the U.S. voice service market, I look forward to gaining a richer perspective on the future of SIP-enabled communications among service providers.” said Dr. Sicker on the announcement.

SIP Network Operators Conference (SIPNOC), aim to advance the knowledge of anyone interested in the operational and technical aspects of service provider networks. SIP Service Providers will be able to hone their knowledge in testing, application development, SIP trunking, FoIP, call routing and peering, troubleshooting and monitoring, emergency services and more assuring reliable and successful deployment and operation of SIP-based services in carrier networks.
The SIPNOC Conference Topics will include many related subjects spread over the course of two days that includes a mix of special presentations and panel discussions -- an agenda designed to address the most important topics for today’s network operator.
Sessions will focus on topics such as:

• Application development
• Testing
• SIP Trunking
• FoIP
• User-Agent Configuration
• Emergency Services
• Policy Servers
• Security
• Operational Issues
• Call Routing and Peering
• Troubleshooting and Monitoring
• SIP Interconnection (SIP-to-SS7 signaling gateways)
• HD-Voice/Video Deployment Challenges

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


Sip Forun Press Release;

NORTH ANDOVER, MA (January 25, 2011) - The SIP Forum announced today Federal Communications Commission Chief Technologist Dr. Douglas C. Sicker as the keynote speaker for the first annual SIP Network Operators Conference (SIPNOC), being held at the Hyatt Dulles Hotel in Herndon, VA, April 25 -27, 2011.
Aimed at an international audience focused on the operational and technical aspects of service provider networks, 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. The event website is located at www.sipnoc.org.
“I'm delighted to participate in this inaugural meeting of SIPNOC,” said Dr. Sicker. “In recognizing the importance of interconnected VoIP service as a growing part of the U.S. voice service market, I look forward to gaining a richer perspective on the future of SIP-enabled communications among service providers.”
As Chief Technologist for the FCC, Dr. Sicker is responsible for advising the agency on communications technology issues, a position that resides in the Office of Strategic Planning and Policy Analysis. Dr. Sicker is also an associate professor in the Department of Computer Science at the University of Colorado at Boulder with a joint appointment in the Interdisciplinary Telecommunications Program. He previously served on the Omnibus Broadband Initiative working on research and development issues. His other leadership roles include director of global architecture at Level 3 Communications, Inc, and chief of the network technology division at FCC.
Dr. Sicker has chaired and served on the program committees of numerous technical conferences, including IEEE, DySPAN, ISART and TPRC. He has also had research funded through the NSF, DARPA, the Internet Society and the Federal Aviation Administration.
“Dr. Sicker is a leading mind in both domestic and international communications, and we are thrilled to have him on board as the first keynote speaker of SIPNOC,” said Eric Burger, SIP Forum Chairman Emeritus and Chairman of the SIPNOC Program Committee.
Attendees at SIPNOC will include telecommunications providers, major backbone operators, interconnect and wholesale solution providers, ISPs, cable operators, wireless network operators as well as large enterprises deploying major SIP initiatives.
The SIP Forum has gained an international reputation for developing important, educational events surrounding SIP. The organization’s SIPit series of interoperability testing events 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.

For More Information, Submit Speaking Proposals, and to Register

For more information about SIPNOC, please visit www.sipnoc.org or send an email to sipnoc@sipforum.org.This e-mail address is being protected from spam bots, you need JavaScript enabled to view it
Proposed papers and presentations are currently being accepted by the SIPNOC Program Committee. To view the official call for papers, which includes instructions on submitting material and specific SIPNOC policies, please visit www.sipnoc.org or download the document here: http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,469/Itemid,261/
A special early-bird registration rate of $595 – $200 off the regular rate of $795 -- is now in effect until February 11, 2011! To register for SIPNOC, please visit http://www.regonline.com/sipnoc_2010.
For information about corporate sponsorship opportunities at SIPNOC, please contact Marc Robins, SIP Forum President and Managing Director, at 718-548-7245 or marc.robins “at” sipforum.org.

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, 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.
Media Contact
John Gates
Elevate Communications
(617) 861-3651
m: (617) 548-8972
john@elevatecom.comThis e-mail address is being protected from spam bots, you need JavaScript enabled to view it

World Business VoIP services forecast 2008 - 2015, It Is Trending Up.

VoIP services forecast 2008 - 2015 http://snapvoip.blogspot.com/
According to  ABI Research, VoIP markets across the globe seems to be healthy. The economic down turn may actually have helped the trend with business' looking to cut costs from every corner. The report, ABI Research’s new study, Business VoIP Services focuses on global VoIP Market, covering VoIP integrated access, SIP trunking, hosted IP-PBX/IP Centrex and managed IP-PBX services, is set to double over the next five years, is believed to exceed $20 billion by 2015,
“The trend of companies outsourcing their communications infrastructure has taken off in earnest, as enterprise decision-makers look for cost savings without having to compromise on features and functionality, Among VoIP access technologies, SIP trunking, with its promise of trunk consolidation, centralized management and administration, and application federation across enterprises, has seen considerable growth in terms of installed base and revenues in the last 12 – 18 months.”commented senior analyst Subha Rama.
“Overall, the growth momentum seen in 2009 was sustained in the first half of 2010 as companies continued to shy away from capital investments, opting, instead, for the OPEX model of hosted VoIP services.” .Rama adds, 
The forecasts suggest a 13% CAGR in terms of revenues and 19.6% CAGR in terms of lines between 2010 and 2015, despite growing competition from low-cost providers and price pressures in the market

Press Release

SIPVicious : Auditing And Protecting Yourself From SIP Registration Attacks.

SIPVicious http://snapvoip.blogspot.com/
SIPVisious is a tool developed to audit your SIP based VoIP IP Telephony system. Basically it is four tools that runs on any system capable of supporting python

• svmap - this is a sip scanner. Lists SIP devices found on an IP range
• svwar - identifies active extensions on a PBX
• svcrack - an online password cracker for SIP PBX
• svreport - manages sessions and exports reports to various formats
• svcrash - attempts to stop unauthorized svwar and svcrash scans 

While it is an excellent tool, there are many who misuse it as well, usually this is the tool used in SIP Registration Attacks. These are attempts to steal your SIP Services or basically minutes. Initially the tool bombarded IP PBXes with registration requests, it amounted to a DOS attack. So two rivaling small town SIP service providers could bombard each other everyday and forget about selling services.But mostly there are people who are looking to steal services than launching DOS attacks against a SIP Network. So the so called "friendly Scanner : SIPVisious" users are smart to tune the tool to such levels that you would not know they are attacking. There are some ways you can prevent or slow down these attempts of breaking in;

1. SIP Endpoint registration, always use SIP authentication! and encourage / force users to use proper passwords.
2. If you have a Session Border Controller (SBC), you can blacklist devices after they fail a few REGISTER attempts.
3. If you’re using non-registering SIP (such as SIP peering for SIP Trunking), limit access by SIP signaling IP addresses using firewall rules / or ACLs to block all connections except from the your peers.
4. Find methods to spot SIP devices sending abnormal traffic loads, and alarm your staff and/or block them for a certain period. 

Also, let your staff, at least the technical people to get their hands on SIPVisious and learn how it works. SIPVisious has instructions for running the attacks against a Trixbox VMware image.
SIPVisious via ecg

Gizmo5 SIP Trunks Trixboxed!

If you are a user of Gizmo Project, (Gizom5) and Trixbox CE, for your communications needs, you should be happier now. No more manual bungling to connect your Trixbox to Gizmo Project SIP Service. (We wrote earlier about how to connect Astrisk to Gizmo5)

The Gizmo5 team has released a module that takes care of all of your SIP needs regarding Trixbox and Gizmo Project. This module allows you to manage your account tasks like purchasing trunks, checking your account balance, top off minutes by purchasing. What is more is that it automatically setup your inbound and outbound routes.

You can get the module via Trixbox Package manager and will also be available via future ISO releases.

Following is the press release;

Gizmo5 SIP Trunks

World's Largest Open Source Telephony Community to Benefit From Gizmo5's Leading Standards-Based SIP Trunk Services

September 24, 2008 - Gizmo5, the leading standards-based consumer calling service, today announced Gizmo5 for trixbox CE, the world's largest free and open source telephony community. Gizmo5 is now providing SIP trunk services to trixbox CE, which will help SMBs save up to 98 percent on business calls. The new offering also includes pay-as-you-go and is fully integrated with the trixbox CE open source project. The company has also joined Fonality's FACE program (Fonality Authorized Certified Ecosystem) as a Gold partner to ensure its products are optimized and compatible with the trixboxCE platform.

"Gizmo5 for trixbox CE is delivering great value to the trixbox CE community by providing cost effective phone services that can easily purchased and managed directly from within the trixbox CE dashboard. Users just create an account, purchase credit, and click one of the outbound route setup options and they can be making calls within minutes," said Chris Vuillaume, vice president of channels and business development at Fonality. "Gizmo engineers even went a step further with their module by providing the simplest way possible to setup SIP trunk services and inbound and outbound routes."

New users of trixbox CE or those performing upgrades can now set up a full range of SIP trunking services and be making calls within minutes from the same easy-to-use interface they already use to set up office phone networks. Administrators can purchase inbound telephone numbers from more than 20 countries and quickly map them to their business. Economical pay-as-you-go calling credits can be leveraged by the entire business to avoid costly per line setup or monthly fees. In addition calling credits can be rolled over each month for a more affordable option than phone systems that require a per user fee.

"For over a hundred years the phone company has charged businesses vast premiums for domestic and international business calling without providing any increase in quality. Now, with trixbox CE and Gizmo5 businesses can pay for only for what they use, save up to 98% over the phone company and have superior call quality", said Jason Droege, President at Gizmo5.

The calling service for trixbox CE is pre-configured to use the Gizmo5 calling network and includes a new UI for easy administration. Also included is a Tech Check system that confirms basic setup of a trixbox CE system and notifies users when new Gizmo modules are available.

Snap VoIP

Tags: Gizmo Project, gizmo5, trixbox, SIP, SIP Trunking

Rich Tehrani Interviews Steven J. Johnson, President of Ingate Systems

The dynamic duo talks about shaping of the VoIP IP Telephony and Unified Communications for the coming year. They also discuss the value of SIP trunking that is driving the enterprise SIP adoption. I only wish they spoke more. Follow the link to find out all the above and who will get 12 gold medals in next Olympics.

SIP Trunking Driving Enterprise SIP Adoption

Tags: TMC, SIP trunking, VoIP IP Telephony, VoIP, SIP,