In late 2007, Global Crossing' Technology Office (GCTO) took a hard look at the company's internal VoIP network used to interconnect its home and field offices.  

Toll arbitrage via "on-net" dialing served as the main business justification to Global Crossing's internal VOIP network build out. But while "on-net" dialing helped the company recoup the cost of implementation within two years, there were still hurdles in the design that could not be overcome without Session Initiation Protocol (SIP) Trunking Services. By moving not only intra-office call flows "on-net", but centralizing the PSTN interconnectivity in a regional model, control over both call routing and costs was achieved. This paper from Global Crossing illustrates how, through the transition to SIP Trunking, the foundation was laid for removing complexity in both routing design, and at the same time helping the company to cut costs and enhance their productivity. 

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VoiceCon March 22-24, Orlando
Gaylord Palms Resort and Convention Center, Orlando, Florida | Panel, "SIP Trunking: Who is Providing What?"
 

Join Chris Smith from Global Crossing at VoiceCon on Monday, March 22 9AM EST during a Deep Dive on SIP Trunking.  The panel, located in room "Sun A" will have two panel sessions with Lisa Pierce, President, Strategic Networks Group, as the lead speaker.

Also speaking on the panel, "SIP Trunking - Who is providing what?" with Lisa Pierce, President, Strategic Networks Group:

Rupesh Chokshi, General Manager, Product Management, AT&T

Dan Jacobson, Senior Portfolio Manager, Converged Voice Services, Sprint Nextel

Thomas Dalrymple, Director, Global Voice Services Product Management, Verizon

Christopher Smith, Senior Manager VoIP Services, Global Crossing

People often talk about the evolution of the internet with regards to social media, news, content or multimedia, but the actual process of progression and growth is far more technical. It involves the inner workings of the web, which include protocols, addresses, bandwidth, security and capacity, and the latest development in this space is the introduction of IPv6. IPv6, which stands for Internet Protocol version 6, is here and should be adopted as soon as possible. Google already runs a number of its applications on IPv6 and earlier this week added YouTube to the new protocol due to the additional security benefits it provides for its users and capability to stream a wider choice of content.

The challenge is educating customers on the benefits IPv6 offers and motivating them into upgrading from IPv4. The benefits are clear, IPv6 is four times larger in address space which means more capacity and faster connections. It also has security in-built enabling an enhanced, more secure connection and network.

Global Crossing has been promoting IPv6 for almost a decade: since 2001 it has been adopted by 20 partners and 40 customers. But despite this, many are still using the IPv4, which means they are still vulnerable from a security angle.

One of the issues concerning late adoption levels of IPv6 is there are very limited drivers to move away from four even though the capacity is running out, many people have private addresses and are therefore not filling up the public networks. Further to this, many wireless handsets don’t require an IP address and this is also having an impact on the uptake of IPv6. Another major challenge is to ensure the migration of the infrastructure across the spectrum. Read more and comment...

All too often, SIP (Session Initiation Protocol) is associated only with VoIP.  It’s understandable, as the largest SIP deployments to date were done to move voice products (long distance, call centers, IVR, etc) from a TDM-based circuit-switch, to a packet-based IP architecture.  But voice is representative of a single example of what defines a “Session” in SIP.  SIP handles multiple forms of communications.  From quasi-realtime Instant Messaging to live video communications, all leveraging packet-based communications. 

The point above is very important to understand because SIP, as well as SIP Trunking provides the key foundational element to Unified Communications/Unified Collaboration (UC). In addition to cost savings and improved ROI as initial attractions, SIP Trunking’s ability to handle multiple communication strategies permits hybrid deployment models of solutions built for business communications, across both premised-based and hosted designs.  Read more and comment...

I stumbled across a series of question regarding SIP Trunking that deserve further thought and discussion. (http://www.200ok.info/) The first of these questions is:

•Why static VoIP peering? (Why not ENUM?) (http://en.wikipedia.org/wiki/Telephone_number_mapping - for a definition of ENUM before we get started)

 The answer is multifaceted. A colleague I know from my work within the IETF, Otmar Lendl, has expressed part of the reasoning in his Internet Draft (draft-lendl-speermint-background).  Main points of his argument are these: Read more and comment...

The 8 Most Important Publications to the Internet

Only because I can, and I have a particular affinity to the number eight right now, ;) I decided to share what I believe to be the eight most important publications to the Internet.  Have a read and enjoy the fun!

Adam Uzelac

1)     A Protocol for Packet Network Interconnection

Vint Cerf, Bob Kahn  IEEE Transactions on Communication Technology, 1974 Online copy (PDF)

2)     Internet Protocol

RFC 791, Information Sciences Institute, Marina Del Rey, California, September 1981 Online copy (HTML)

3)     Transmission Control Protocol Read more and comment...

MS says jump, you ought to ask “How high?”

Like it or not folks, Redmond, Washington will has become the focal point for Enterprise Telephony, Unified Communications, Cloud Telephony and/or anything else you want to repackage VoIP as. Microsoft is a significant player in the world of VoIP, today, tomorrow and then some. MS is lining up partners left and right to support SIP Trunking to OCS. OCS R2 (released Feb 2009) was the first iteration of SIP Trunking for OCS. (though it was possible in a limited way pre-R2, but that’s fodder for another post) So MS comes out with SIP Trunking support, and they made a decision that immediately forced their partner’s technological roadmap. If you wanted to be a MS-certified SIP Trunking provider, support for SIP over TCP was mandatory. The folks in Redmond chose not to use SIP over UDP for numerous technical reasons, which don’t pertain to this post. (but also fodder for another day)  Here’s the significance of the decision – the lion’s share of SIP was (and still is) SIP over UDP. SIP Service Providers, vendors, partners, etc – all had to enable SIP over TCP to support MS, near exclusive. MS ignored where the market was(is), and instead put the stake in the ground. Due to their emerging dominance in the Enterprise communication space, partners either had to adopt the shift, or be forced to the outside of this movement looking in on all the “fun”. Read more and comment...

Important yet subtle differences in SIP Trunking: Line-side versus Trunk-side

Whenever I get into a discussion around SIP Trunking, it’s important for me to establish the type of SIP Trunking that is being discussed.  The most important question for me is determining if the context is “line-side” SIP Trunking, or “trunk-side” SIP Trunking.  There are subtle, yet very important differences to consider when designing the SIP Trunking solution. 

The difference between “line-side” and “trunk-side” lies in the administrative domain that the Telephony Server (the network element that is supplying “dial-tone”) resides.  In a “line-side” configuration, the telephony engine supporting the SIP-based endpoint (typically a IP-phone) is “IP-Centrex”, “Hosted Telephony” (I like this link because she's holding a CD and CDs have everything to do with VoIP! ;) ), or “Cloud Telephony” (just google this term as it's just silly to try to find one definition).  The registration of the IP phone to the telephony server traverses the SIP Trunk.  Read more and comment...

T.38 Fax – been there, done that, 4 years running….

First the article here (http://tr.im/p5sv)  - about VoIP testing solutions.  And in this article this..

Devfoundry’s President Athir Nuaimi, says, “There’s been a renewed interest in fax-over-IP testing. The SIP Forum (News - Alert) has put together a task group to try and get that fast-tracked. T.38 was formulated over a decade ago, but it really never went very far. Many people put a lot of blood, sweat and tears to try to get it to work, and have had limited success in some small domains. But the biggest issue is that the large carriers really haven’t supported it until the last year or two. Global Crossing now supports it, and I believe most of the other major backbone VoIP providers either have T.38 support or it’s in the trial phase. It’s taken a long time, but the stuff is actually now available to customers if they want it, but there’s still some compatibility issues, enough so that a number of what are basically equipment vendors as well as some service providers, have put together this ‘task force’.”

Now for a little tidbit of information. -  we (Global Crossing) have been supporting T.38 for 4 years (since July of 2005). 

Allow me to speak for most VoIP Engineers, we would like to see faxing just go away all together. One would think in the age of PDF, email, digital signatures (and good old snail mail), faxing would just go away all together.  Like the high school football star that didn’t amount to anything and is clinging to his past accolades with all of his diminishing might, FAX (as with Mr. QB) just needs to give it up already!!!

Adam “voiploser” Uzelac Read more and comment...

SIP Virtualization

I believe that most folks within the “computer” field have heard of Virtualization.  I would like to introduce what may be a new term to the industry; SIP Virtualization.  The concept we have conceived here at Global Crossing borrows from many of the same constructs from the popular definition.  For reference, a brief history of Virtualization is in order.

Virtualization was conceived back in days of the Mainframe (1960s).  The mainframes of the day were expensive and underutilized.  To increase the value of the investment, a logical (read as, not physical) partitioning of the computing resource was developed.  This partitioning created “Virtual Machines” within the greater single physical form factor. In order to achieve this, an abstraction of the physical elements and the software-based operating system was introduced.  In this manner, Operating Systems were able to leverage and share common hardware.  It was no longer necessary to dedicate a hard-drive or a network interface to a specific instance of software.  This notion has since enjoyed a rebirth using x86 computing architectures.

How ambiguous is the term ‘virtualization’ or ‘virtual’ within the computing industry?  Here’s some that am personally aware of and use often…
-    Virtual Reality
-    Virtual Memory
-    Virtual Private Network
-    Desktop Virtualization
-    Virtual IP Address
-    Virtual Router Redundancy Protocol
-    Virtual Router

For sake of this discussion, I will cherry-pick a rather simple definition from whatis.com.  

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The Unstated Value of SIP Trunking Providers

Speaking as one that has an intimate technical knowledge of SIP Trunking in my professional life, I feel it’s my public duty to increase public discourse around SIP Trunking.  First, let’s start by establishing the context of the discussion with a definition of SIP Trunking.  A google search (June 2009)  on a ‘SIP Trunking’ returns ~500k references. 

From Wikipedia (replace ‘connection’ with Trunk)
 A SIP (Session Initiation Protocol) connection is a service offered by many ITSP (Internet Telephony Service Providers) that connects a company's PBX to the existing telephone system infrastructure(PSTN) via Internet using the SIP VoIP standard. 

HD VoIP – Ridding Us of the Music-on-hold Tease

On the heels of the hyper marketing of HD Television, makers of VoIP endpoints supporting HD Voice have been making a huge push as of late.  The difference between standard voice and HD voice is quantifiably noticeable – HD voice is better sounding without a doubt.  Traditional voice samples voice from our vocal cords between 300Hz and 3.4kHz.  HD voice, on the other hand, samples in the range of 50Hz to 7kHz.  This means that the very low and high ranges of voice are being captures in a world of HD Voice, and lost in land of POTS phones.  We are talking about “FM versus AM on our radio dial” kind of differences here.  Read more and comment...