Editors note:

There was movement at the station, for the word had passed around,
That the colt from old Regret had got away,
And had joined the wild bush horses - he was worth a thousand pound,
So all the cracks had gathered to the fray...

Just a little poetry by AB (Banjo) Patterson to lighten up your day today. There was plenty of movement at the SIP Center station in September. Two new Principal Sponsors gathered to the fray. A herd of new resources for developers have been added to the site as well as some whitepapers and all kinds of other bits and pieces. Take a trip to our news section to see last months frenzied press coverage with plenty of fantastic new product announcements. More than twenty new SIP products have been added to the products directory yet again this month. 'How high can we go' I ask?

Many thanks for taking the time to read our humble newsletter and I hope there is something helpful here for you.

Best wishes,

Kathleen Misson
Editor

Contents:

• Thinkengine Networks and GL Communications join The SIP Center!
• Stacks of Stacks and other stuff for SIPpie Developers
• Ubiquity Software Re-launches Developer Program
• Delivering IP-Based Voice Services with SIP
• SIPit news from The SIP Forum
• Answering the Top 10 Questions on Residential Broadband Voice
• Some forthcoming events for your diaries
• SIP Webinars

Regular Features:

• New SIP products and services
• Events diary
• Unsubscribe and copyright info
  

Thinkengine Networks and GL Communications join The SIP Center!

It is exciting to see The SIP Center growing in numbers and members and our greatest ambassadors are our sponsors - They have fully embraced SIP technology and they contribute invaluable content and expertise to The SIP Center. Their contributions help us to promote SIP and improve The SIP Center resource for everyone.

And so, we are delighted to welcome Thinkengine Networks and GL Communications to the team as Principal Sponsors. I'm sure it will be worth just a few moments of your time to check out these fantastic SIP products:

Thinkengine's flagship product, the VSR1000, combines the features of a Media Gateway, Media Server, Conferencing Engine, Call Routing Engine, and Voice Recognition Server into a cost effective, NEBS compliant 1U platform. The VSR1000 simultaneously terminates traffic from both PSTN and VoIP networks "out of the box". Service providers can utilize the VSR1000 to save money in today’s PSTN network and position themselves for migration to VoIP without further equipment or card purchase. Standard SIP signaling is utilized throughout ensuring a future-proof solution.

www.thinkengine.com

GL Communications Inc. is a leading provider of test and measurement tools for TDM, VoIP, and Wireless networks. In the VoIP arena, GL offers their Packet Series™ products: PacketGen™, PacketScan™, RTP Toolbox™, and H.323 Call Emulator / Analyzer. In addition, GL supports full Voice Quality Testing over all types of networks using the industry standard PESQ, PAMS, and PSQM algorithms.

PacketGen™ is a SIP bulk call generator that can emulate thousands of SIP calls with real-time and file based traffic over established calls, such as Voice, Digits, Tones, Noise, Live Speech (all codecs), and Fax.

PacketScan™ is a realtime protocol analyzer that can analyze thousands of SIP sessions in detail. Call logging, call records, call trace, and statistics are just a few of the capabilities.

www.gl.com

Stacks of Stacks and other stuff for SIPpie Developers

There is an incredible number of resources out there for developers to play around with and create amazing new SIP products and services. Our Developers resources section has been updated with the latest and greatest under the following headings:

• SIP Parsers and Stacks
• SIP and DNS
• Testing Tools
• Robustness testing
• SIP Client Software
• Codecs
• SIP servers, proxy servers and gateways

A few of the resources that were sent to us this month are:

Free SIP Phone from Center for Network Research (CNR), Prince of Songkla University, Thailand.

EarthLink SIPshare, a simple, SIP-based proof-of-concept content sharing application, demonstrates the viability of SIP as a protocol over which peer-to-peer (P2P) applications other than the well-known voice and video cases may be implemented.

LIVE.COM Streaming Media
This code forms a set of C++ libraries for multimedia streaming, using open standard protocols (RTP/RTCP, RTSP, SIP). These libraries - which can be compiled for Unix (including Linux and Mac OS X), Windows, and QNX (and other POSIX-compliant systems) - can be used to build streaming applications.

Far too many to list all of them for you here in this newsletter! Here's the link - Enjoy!

Please let us know! If you have a link or resource that we could share with others on this page.

Delivering IP-Based Voice Services with SIP

By By Tom Baumgartner, Director of Product Marketing, NMS Communications.

A recent survey of 300 business technology executives1 shows 29% use VoIP, 18% are testing it, and 34% have plans to deploy VoIP - so there is no doubt that VoIP is making its mark. Key to the successful implementation of any voice over IP solution is a call control protocol that is widely accepted, easy-to-use, and provides all the user features demanded of today's telecommunications services. While the H.323 ITU-T standard for the transmission of real-time audio, video, and data information over packet-switching networks currently commands a large installed base, mainly because it existed when VoIP first emerged in the mid 1990s, the Session Initiation Protocol (SIP) from the Internet Engineering Task Force (IETF) is now in contention for leadership.

In recent years there has been a lot of innovative work done to define clarifications, bug fixes, tweaks, and extensions to SIP. Notable examples include the IETF clarifications of procedures for transferring calls, and new requests for comment (RFCs) and drafts to support the administration and attendant features common to PBXs and call centers. Recognized problems for SIP, namely operation through firewalls and Network Address Translation (NAT), have been address via RFCs and drafts. In addition a body of literature is now available to describe how to use SIP, the Session Description Protocol (SDP), and other IETF standards to provide every imaginable voice service over an IP network.

The downside of all this progress for developers is the continued fluidity of the SIP standards. Successful products must sometimes implement drafts in anticipation of the final standards. But the eventual dominance of SIP as the VoIP call control method of choice seems inevitable. The SIP advantages - scalability and interoperability - are derived from the architecture of the Internet that has intelligence located at the edge of the network (no central control) and devices communicating using standard protocols.

Read the full article >>>

SIPit news from The SIP Forum

The 15th SIPit produced some outstanding results towards interoperability issues, fixing "Bugs" and expanding SIP. 70 Companies and 200 Attendees braved a typhoon to gather for the first ever SIPit in Asia! Nearly a quarter of the attendees were first-time participants in SIPits, which demonstrated that holding the event in Asia enabled many companies from that region to join the SIP interoperability effort.

The challenges addressed by the sessions involved pushing SIP into new territory, ironing out interoperability issues with the use of multiple media audio, video, instant messaging within SIP sessions. The majority of SIP end points (telephones, soft phones) today still function mostly just as phones.

SIPit Events are organized by The SIP Forum. Read the full press release here.

Some forthcoming events for your diaries

Internet Telephony Conference and Expo and SIP Developers Workshop are underway! Don't forget VON on the 18-21st - We'll be posting the top stories from these events here on The SIP Center. Here are a couple of new events that may be of interest to you:

VoIP Services Forum 2004
Is VoIP Really Becoming Mainstream?
3-5 November 2004 - Central Amsterdam, The Netherlands

"It is estimated that by 2008, VoIP traffic in the EMEA could reach 57 billion minutes. 1,771 billion minutes of voice traffic from the region will also have a VoIP component. These projections make observers believe that VoIP is the "new pot of gold""

VoIP Services forum will bring together industry leaders and innovators to shed light on the strategic direction(s) of the "pot of gold" that is VoIP.

Push to Talk 2 World Summit
29 November - 2 December, Four Points Sheraton Rome

Over 200 delegates from across the globe attended IIR’s first Push to Talk World Summit and shared opinions, problems and questions with their colleagues from many different markets around the world. If you weren’t there the first time don’t miss out again! For operators preparing to launch push to talk services this summit cannot be missed. Hear the latest developments on SIP based push to talk solutions and on IMS standardisation. Additionally, learn how to attract new users, increase mobile usage of existing customers and deliver a high quality PTT user experience. Attend the conference and learn how to Maximize PTT ARPU opportunities.

Winter 2005 SIP Summit
January 16-18, 2005, Honolulu, Hawaii.

The pulver.com SIP Events are the place to go if you need to understand the issues effecting the international SIP industry today and would benefit from having a heads up on the future directions of the space. At our SIP Summit events we speak about the present future rather than the present past. Since 2001, our SIP events have directly contributed to the growth of the SIP industry and has been the place where ideas were exchanged, companies got started and where some companies received funding.

For information on how you can sponsor, exhibit at or attend the Winter 2005 SIP Summit, please contact Christopher Erb at (631) 961-8987 or via email at cerb@pulver.com.

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