Jim Dalton's Blog

Interview with Rich Terani

2010-08-18T16:45:00.003-04:00

I was recently interviewed by Rich Terani, the CEO of TMC. The text of that interview is below.

[RT] What is the most significant trend in communications today? Why?

[JD] The growth in apps for mobile phones is the most significant trend. Mobile apps on the Internet are providing more value than just plain old voice service. Users are finally realizing the benefits of the data network, not just a wireless voice networks. As a result, it will be harder for carriers to resist the demand from customers for simple IP bandwidth instead of services that are bundled with the network.

[RT] What is the one product or service the market is most in need of?

[JD] A security mechanism that was decoupled from the network. For example, e-mail is totally non-secure and VoIP suffers the same problem. The market needs a secure way to authenticate the parties with whom they communicate. Relying on fax as the only secure way to transmit documents is ridiculous. The technology to solve the problem (public key cryptography) has been around for years, but no market mechanism has been able to drive wide spread adoption.

[RT] How is your company leveraging the growth of social media to enhance your own business?

[JD] Our products are technical and serve a small niche. We have not yet identified how social media can benefit our business.

[RT] Nearly every phone manufacturer is now incorporating support for wideband codecs. Will we finally see widespread HD voice deployments in 2011?

[JD] No, wideband codes will have to be widely deployed for 3-5 years before they are be used. Adoption always lags availability by several years.

[RT] What are your thoughts on the viability of mobile video chat or conferencing?

[JD] Mobile video chat and conferencing are inevitable.

[RT] Which wireless operating system (Android, iOS4, Microsoft, etc.) will see the greatest success over the next three years? Why?

[JD] Android – because it is the most open OS.

[RT] Some have suggested wireless networking will soon replace wired networks in the enterprise. Do you agree? Why or why not?

[JD] There will always be a mix of wired and wireless networks. I cannot foresee wireless completely replacing wired networks.

[RT] What impact has the growth of cloud-based services had on your business?

[JD]No impact.

[RT] If you had the opportunity to decide the Net neutrality debate, how would you rule?

[JD] If you want the Internet to grow and be vibrant, then it must be completely competitive and everyone must be able to make money from traffic that traverses their network. Networks must offer equal access to all other networks and must be able to charge those networks for the traffic they generate. The critical challenge is to make certain that pricing is applied equally to all participants with no subsidies or price discrimination.

[RT] You are exhibiting at ITEXPO West 2010. What is the most exciting thing attendees will see at your booth?

[JD] We will be releasing OSPrey-32 at the ITEXPO West 2010. OSPrey-32 is the 32-bit version of our commercial routing and CDR collection server. OSPrey-32 is a free software package and it perfect for enterprises that need a centralized routing and accounting platform to manage an enterprise VoIP network with multiple branch offices. OSPrey-32 is a really powerful package for enterprise VoIP managers, it is easy to use - and its free.

FreeSWITCH supports OSP

2010-08-17T13:57:00.006-04:00

TransNexus has contributed a new module, mod_osp, to FreeSWITCH. The new module enables FreeSWITCH to use the Open Settlement Protocol (OSP) to query an external route server and to report XML based Call Detail Records (CDRs). The new module uses the OSP Toolkit available from SourceForge.

One of the advantages of OSP route queries over SIP redirect messages or ENUM queries is that OSP was designed as a route query protocol. OSP is not limited to single protocol, such as SIP, or just to route discovery. OSP is designed to support all the information elements that might be needed for a fully intelligent query/response. For example, a standard OSP response can include rich details about all possible destinations such as which protocol to use (SIP, H.323. skype, SMS, MMS) calling and called number translations, all the number portability elements (LRN, SPID, alt-SPID, MCC, MNC), bandwidth requirements and even pricing information.

Likewise, XML based CDRs defined by the OSP standard define a large set of data elements and since the CDRs are XML based they can be easily extended to meet any need.

TransNexus has long been involved with open source VoIP projects such as SER, OpenSIPS, Asterisk and now FreeSWITCH. We see a lot of growth and innovation occurring in the FreeSWITCH community and we are looking forward to becoming more involved with the growing number of creative entrepreneurs using FreeSWITCH.

Acme Packet Interconnect 2010

2010-05-11T14:50:00.001-04:00

TransNexus will be one of a selected group of Acme Packet Pinnacle Partners participating at Interconnect 2010 in San Diego CA on May 16-19. A product demonstration planned by TransNexus will be the new SDReporter software for Acme Packet SBCs. The SDReporter is a easy to use, low cost solution for quickly collecting Call Detail Records from Acme Packet SBCs and generating near real time reports with call statistics by source, destination, time of day and dial code. TransNexus will also be demonstrating NexOSS, its complete least cost routing, number portability, traffic reporting, profit analysis and wholesale billing solution for Acme Packet SBCs.

Please contact alcina.defig@transnexus.com to schedule a meeting with TransNexus at Interconnect 2010

ITW 2010

2010-04-25T12:57:00.001-04:00

International Telecoms Week (ITW) 2010
May 24-26, 2010 Washington, DC

TransNexus will be exhibiting at the ITW 2010 show at the Marriott Wardman Park Hotel in Washington DC. TransNexus will be exhibiting NexOSS, the Least Cost Routing, Number Portability, Traffic Reporting, Profit Analysis and Wholesale Billing solution for VoIP carriers worldwide. TransNexus will also be demonstrating SDReporter a simple and low cost CDR collection and reporting platform for Acme Packet Session Border Controllers (SBCs). In addition, TransNexus will also being demonstrating its international number portability feature based on the GSMA's PathFinder service.

Please contact alcina.defig@transnexus.com to schedule a meeting with TransNexus at ITW 2010

Metaswitch Forum 2010

2010-04-25T08:45:00.005-04:00

I am looking forward to attending the Metaswitch Forum 2010 in Nashville, TN on May 19th. At TransNexus, we are very pleased to be a MetaSwitch partner. TransNexus has been a Metaswitch OSS/Network Management Partner since 2007. The NexOSS Least Cost Routing, Number Portability, Traffic Reporting, Profit Analysis and Wholesale Billing solution is an elegant solution for managing interconnection between the Metaswitch and other carriers. TransNexus will be providing demonstrations of its NexOSS platform at the Metaswitch Forum from 9:30 am to 6:00 pm on Wednesday May 19th.

Please contact alcina.defig@transnexus.com to schedule a meeting with TransNexus at the Metaswitch Forum 2010

Least Cost Routing Challenges

2010-04-07T13:20:00.003-04:00

Not too long ago, optimizing least cost routing was a relatively straight-forward exercise that could be managed part time by an industrious billing or switch technician. This is no longer the case and this article will describes the major developments that have increased the scope and complexity of least cost routing for the North American telephone network.

Routing Table Size
Area Codes were first established in 1947 with a total of seventy-eight area codes nationwide. Over the next 43 years, only thirty-six area codes were added. In the 1990s, the growth in wireless services drove the addition of 109 new area codes so new blocks of 10,000 numbers could be issued to carriers. These number blocks refer to the first six digits of a telephone number, or NPA-NXX where NPA is the area code and NXX is the switch code.

Faced with a dwindling supply of telephone numbers for new carriers to issue to their subscribers, the FCC in 2000 ordered that NPA-NXX number blocks be partitioned into 1000 blocks (NPA-NXX-X). Partitioning of NPA-NXX blocks, also known as number pooling, dramatically increased the supply of new numbers for expanding carriers which is good. But seven digit routing based on 1000 block partitions is more complicated than routing on six digit NPA-NXX blocks. In the Local Exchange Routing Guide (LERG) there are 166,431 six digit NPA-NXX dial codes. In contrast, there are 644,327 dial codes when routing based on seven digit one thousand blocks.

In telephone routing tables, a dial code matched with a destination (IP address or trunk group) is called a translation. Most least cost routing tables have an average of three destinations per dial code. This means that a basic optimized least cost routing tables for domestic US routing can have more than one and half million translations. Some TransNexus customers have 50 million translations in their routing table.

Jurisdictional Routing
Jurisdictional routing is another name for local, intra-state and inter-state routing. Call routing decisions are based on both the calling and called number. How a call gets rated depends on where the calling party is located. Paradoxically intra-state calls cost more than inter-state calls. For a service provider to optimize their least cost routing, they must have two routing tables, one for intra-state calls and a second for inter-state call.

Jurisdictional routing is determined by the ANI (Automatic Number Identification), or telephone number, of the calling party. With VoIP calls, it is common for the ANI to be an invalid value, such as a random number that is not a telephone number or the SIP URI of the calling party. Calls with an invalid ANI are rated at the higher intra-state rate if they are completed. Most major carriers will block calls that do not have a valid ANI. However, some service providers will accept calls with an invalid ANI so this creates an optimization for a third least cost routing table. One for inter-state, one for intra-state and a third for calls with invalid ANIs.

Local routing is another form of jurisdictional call routing which is determined by the NPA-NXX-X of the calling party and the called party. Interconnection of local calls is usually charged at very low rates, much less than intra-state, and offers another least cost routing optimization. A challenge with local call routing for most VoIP providers is the size of the table which defines the source and destination NPA-NXX-X combinations for each local calling area. Unlike Local Exchange Carrier switches that serve limited geographic areas, a VoIP service provider’s softswitch can serve customers anywhere in the US. A table that defines all the local calling areas in the US would have over a billion NPA-NXX-X combinations.

Deciphering Rates by LATA, OCN and Tier
Least cost routing is all about analyzing dial codes and the rates carriers charge to complete calls to those dial codes. Unfortunately, most carriers to not quote rates for US termination by dial code. Instead they quotes rate in terms of OCN, LATA and Tier. So the first step in least cost routing analysis is to normalize carrier rates from OCN, LATA and Tier to E.164 dial codes. OCN stands for Operating Company Number and is a four digit number in the LERG. The LERG lists all the dial codes (NPA-NXX-X) for each of the 3,936 OCNs. LATA stands for Local Access and Transport Area and is a regulatory relic of the AT&T breakup in 1984. The LERG can be used to determine all dial codes for each OCN within each of the 223 different LATAs. Tiers are the last abstraction used by carriers to quote rates. Carriers often have six to eight rate tiers per LATA. Tiers are not standard and every carrier’s tier structure is different. Manually normalizing carrier rates using the LERG is too big a task for even the most industrious technician.

Number Portability Correction
In 1996 the FCC ordered that telephone companies must let subscribers switch service providers and keep their telephone number. When you switch phone companies and keep your telephone number, your number is ported to the new service provider and added to the number portability database. Today, every call in North America starts with a query to the Number Portability Administration Center (NPAC) database to determine if the telephone number has been ported to a different service provider. If the number has been ported, then the NPAC returns the Location Routing Number (LRN) for the ported number. The call is then routed using the LRN, not the dialed telephone number.

Today there are 197 million ported telephone numbers in North America. TransNexus customers report that 40% of their calls are to ported telephone numbers. This development has a dramatic impact on least cost routing. If least cost routing is based on the dialed telephone number then 40% of all calls will not be routed to the lowest cost provider! TransNexus customers have found that number portability correction can decrease their total termination costs by over 15%.

President Clinton to speak at ITW

2010-04-07T09:38:00.004-04:00

The Annual GTM conference in Washington, D.C. every May was a must attend event for international telecom carriers for years. But as the show became more successful, the managers of the Intelsat who ran the show became increasingly arrogant. The GTM show had become a very lucrative franchise and its organizers, like all monopolists, gleefully gouged its sponsors for everything they could get. Three years ago, the ITW show was hastily convened, just blocks down the street from the GTM show, as a competing conference for international telecom carriers. The faithful GTM attendees moved down the street in mass to the upstart ITW conference and the GTM show was gone for good. The rapid collapse of the GTM conference is a stunning business case study of franchise destruction caused by arrogance toward customers.

Since that initial start, the managers of the ITW conference have run their conference like any well run business. They provide good service at a reasonable price. The result has been rapid growth and a major success story for ITW (International Telecoms Week). This year ITW is expanding its exposure and having President Clinton as a keynote speaker. TransNexus will be exhibiting at the ITW conference which will be held May 24-26 in Washington, D.C. Please stop by the TransNexus booth to say hello.

For more information about the ITW conference go to: www.internationaltelecomsweek.com

OSP Assigned Port 5045

2010-03-24T09:18:00.003-04:00

The ETSI Open Settlement Protocol, or OSP, has been assigned TCP port 5045 by IANA. IANA, the Internet Assigned Numbers Authority, is the body responsible for coordinating some of the key elements that keep the Internet running smoothly. Whilst the Internet is renowned for being a worldwide network free from central coordination, there is a technical need for some key parts of the Internet to be globally coordinated – and this coordination role is undertaken by IANA.

OSP is a set of XML messages defined for authorization, routing, pricing and recording usage of IP session transactions. The most common use of OSP is for routing and billing VoIP calls. The OSP standard defines that OSP XML messages should be transmitted over HTTP or HTTPS. So OSP servers are really nothing more than specialized web servers. In the original OSP standard, the tcp port defined for OSP was port 80, the port assigned for HTTP. However, as VoIP communications have become more prevalent, the need has arisen for OSP to have its own port assignment - and that port is 5045.

TransNexus will transition its OSPrey OSP server to use port 5045 by default in the next OSPrey release due out in May. Port assignment will be configurable so users can continue to whatever they port they choose for OSP.

The LERG is Becoming Obsolete

2010-03-05T14:47:00.004-05:00

The LERG, or Local Exchange Routing Guide, has been the routing table for the North American telephone companies for decades. But two forces are driving the LERG toward irrelevance. First, growing numbers of subscribers are porting their landline telephone numbers to cable and wireless services providers. Second, technology and cost trends indicate that the North American number portability database will be widely available as a routing resource for enterprise and VoIP service providers.

Number portability has been an issue for telephone companies since it was mandated by the FCC in 1996. Now number portability is becoming a serious issue for enterprises and VoIP service providers as the percentage of calls to ported numbers is approaching parity with calls to un-ported number. Our customers tell us that approximately 40% of their VoIP calls to ported numbers.

Virtually all enterprises and most retail VoIP service providers route calls based on the dialed telephone number. This is a problem when the called party has ported their number to different carrier. Routing the call to the dialed number directs the call toward the carrier that owns the number, not the carrier that serves the subscriber. The call will be completed, but the enterprise or service provider originating the call will get a surprise when the termination cost is greater than expected. The reason is because most long distance providers charge for wholesale termination based on the final carrier, not the dialed telephone number.

In almost every case, call termination charges based on the dialed number are different from termination charges based on the carrier serving the ported number. The result is that long distance invoices never match what customers expect, and usually the bill is significantly larger than expected. This issue is amplified by wholesale rate plans that include a high termination fees for calls to terminating carriers not explicitly included in the rate deck. Calls to ported number may unexpectedly be routed to carriers not included on a rate deck and charged at rates as high as five cents per minute or more.

The experience of our customers indicate the rate differences between dialed and ported numbers can increase termination costs by at least 15% and the problem will only get worse as more people port their numbers. If you want to manage your costs, you have to route based on the LRN. The LRN, or Location Routing Number, looks like a telephone number, but is a special routing number that links a dialed telephone number that has been ported to its new carrier.

The FCC mandated the creation of the Number Portability Administration Center, or NPAC, to manage the database of ported numbers in the U.S. In simple terms, the NPAC is a list of 300 million ported telephone numbers and their corresponding LRN. For every call, telephone carriers query the NPAC with the dialed telephone and receive the corresponding LRN needed for call routing.

Originally access to the NPAC was expensive and limited to only telephone companies. Now, however, subscription to NPAC is affordable for enterprises and VoIP service providers who need number portability correction. Companies such as NeuStar offer the complete NPAC available for download. Enterprises and VoIP service provides can now host the NPAC database locally in their network and correctly route calls based on the LRN just like phone companies.

Ten years ago, the specialized hardware needed to host the NPAC with a telephone switch cost one million dollars. Four years ago the cost of an ENUM network appliance to host the NPAC had fallen to one hundred thousand dollars. Now TransNexus is offering a SIP based Linux based software package that reduces the cost of hosting the NPAC by another order of magnitude.

We forecast that in five years, virtually all enterprise call centers and VoIP service providers will host the NPAC in their operation. Barry Augustinsky, CEO of SipGully, a growing SIP service provider says “SIP technology changes everything in telecom. We have a lower cost structure than our competitors because we are on the front of a dramatically falling cost curve. The TransNexus server is a great example, now anybody can afford to host their own number portability database.”

Start-up Camp

2010-01-22T08:15:00.005-05:00

Last night I attended the "Start-up Camp" at the IT-Expo in Miami. It was a showcase for four start-ups to make their pitch. The session was started with a rousing presentation by Jamie Siminoff - an accomplished start-up entrepreneur who gave an excellent presentation encouraging all innovators to follow their passion. The session was very well done. I am looking forward to attending future start-up camps.

All four companies had good ideas and gave good presentations. The presentation I found most interesting was given by a company called Close-Haul Communications. The founder is Robin Coxe, a small lady with a lot of energy and Ph.D in particle physics. The company's product is called GAPfiller - a low cost, open source based femto cell for the masses. The idea with GAPfiller is to place a GSM cell site anywhere there is a GAP in existing GSM coverage. The difference is that GAPfiller is directly connected to any SIP network. There is no MSC and so no handoff to other cell sites for full mobility. But GAPfiller enables is a very low cost access network that works with a billion or more existing GSM phones. The GAPfiller is based on Asterisk and OpenBTS.

Companies like Magic-Jack, which announced its new product at CES that can use any GSM phone as a wireless handest for its Magic-Jack service, and Close-Haul communications are the start of an exciting new trend. Companies that are directly attacking the last great walled gardens of telecom - the wireless carriers.

Telefonica Acquires Jajah

2009-12-23T15:09:00.003-05:00

Telefonica has acquired Jajah in an all cash transaction for 145 million euros. Jajah has been a TransNexus customer since 2008 and we are excited to see Jajah make this transition. Jajah deployed the TransNexus least cost routing, reporting and billing software to implement the class 4 functionality of its Acme Packet session border controllers. Jajah is a technically savvy company that has rapidly built a large VoIP operation based on software skills rather than investing a fortune with traditional telecom vendors. Jajah is very innovative, moves fast and we believe represents the next generation of telecom providers.

To view the press release go to: www.marketwire.com/press-release/Jajah-Inc-1095071.html

PathFinder Business Model Flaw

2009-11-11T09:05:00.010-05:00

"PathFinder is the GSMA’s multi-tiered, fully extensible number resolution system. It is designed to facilitate the efficient global interoperability of new and existing IP services using telephone numbers – all within a business framework consistent with operator needs. It provides a comprehensive suite of services designed to enable IP-to-IP interconnections between interconnect partners for voice, messaging, video, and other IP services."

TransNexus is a certified PathFinder partner. The TransNexus OSPrey server can launch a ENUM query to the PathFinder service as an alternative to least cost routing. We believe the efficiencies from IP to IP routing made possible by PathFinder will generate new profits for carriers and better services for their customers.

As a PathFinder partner, I have been promoting the service to carrier customers. Some of the feedback I have received is very clear. Carriers do not like the Called Party Pays business model of PathFinder. Here is a simplified description of the PathFinder business model. Service providers provision their telephone numbers and corresponding Service Provider Number (SPN) to the PathFinder database. When a service provider originates a call they send a query with the called number to PathFinder. PathFinder returns the SPN of the service provider that serves the called number. This type of service has become critical since number porting between service providers is widespread. All of this is good. Now here is the problem - the carrier serving the called party pays PathFinder for the query. This simple flaw in the PathFinder business model is a serious obstacle to its success.

There is some sound logic to the PathFinder business model and here are some reasons why it makes sense:
1) The called network may benefit from the PathFinder service because it may receive the call directly over an IP connection which could be less expensive than a TDM interconnect.
2) Also a direct IP interconnection will enable end to end IP services so the called party can benefit from the advantages of VoIP such as wideband codecs and better sound quality.
3) The carriers who terminate the most calls will be the biggest carriers since they serve the most telephone numbers. Therefore, the biggest carriers with the most financial resources will make the largest revenue contribution for funding PathFinder. This is an egalitarian concept that regulators would endorse.

Now here are the counter arguments:
1) The called party pays model means that the terminating carrier is forced to pay an expense that it has incurred on behalf of the calling network. Everyone fears an expense liability that is out of their control.
2) The called party network has no record of PathFinder dips that point to its network and no way to audit its PathFinder bill. Conversely, the Calling party network has complete control over its number of Pathfinder dips and perfect audit capability of its PathFinder expenses.
3) Why should large carriers, who will be the net terminators for PathFinder queries, subsidize smaller carriers for PathFinder expenses?

And here is the Partner's appeal:
Put logic and intellectual arguments aside when defining the PathFinder business model. Go with what works in the market. PathFinder, like so many network initiatives, has a chicken and egg problem. The idea makes total sense, but is worthless until there is a big database that offers value. Getting the world's largest carriers to participate and provision their numbers to PathFinder is goal number one. Any other focus is folly. If the big carriers want a calling party pays business model - the argument is over. PathFinder needs to implement a calling party pays business model. TransNexus is a big supporter of the PathFinder concept and looking forward to its success.

If you think I have this wrong, please post your thoughts here. This is all about getting it right, you won't hurt my feelings. This post is all about getting the economic incentives right so PathFinder and the promise of IP to IP communications become a reality for everyone.

i3 Forum vs IPIA

2009-11-10T07:09:00.008-05:00

In Andy Ory's opening speech at the Acme Interconnect 2009 conference he made the point that the emergence of new market segments requires disruptive change. But he added that it has to be just the right amount of disruption. Too little disruption and customers will just adjust and continue doing what they always do. Too much disruption and it becomes too difficult for customers to adopt a new opportunity even if it has major benefits.

This concept seemed to be illustrated in a panel discussion of the missions of the i3 Forum and the IPIA (IP Interworking Alliance). Carlos Da Silva from France Telecom represented the i3 Forum that has the goal to develop collaborative recommendations for an industry-wide transition of voice and related services to Internet Protocol (IP). John Boden of Vodafone spoke for the IPIA which has the mission “To provide an environment whereby stakeholders from mobile operators, fixed operators and carriers can agree to technical specifications and commercial templates for the IPX, which will facilitate the global interworking of IP services.” The IPIA is the users group for the GSMA IPX.

The mission of these two organizations sound very similar, but their plans and apparent success seem quite different. Early on I was a big fan of the GSMA IPX concept. The IPX is a big, bold idea and it seems to have a fundamental business model that makes sense. In contrast, my initial impression of the i3 Forum was that it just another group of PTT's trying to protect their incumbent business model. I still think the i3 Forum is just a stodgy group of incumbents trying to protect their business, but it sounds as if the organization, which has grown to 28 members, is making some progress on its goal of nudging the largest international SS7 networks toward VoIP. In comparison, there is no news from the IPIA. Its web site is closed to everyone except board members. It seems that the disruptive change offered by the GSMA IPX may be too much. Maybe the GSMA IPX be just another great idea which is ahead of its time.

One of the interesting points made by Da Silva in the panel discussion was about PTT bilateral agreements. Settlement based bilateral agreements are the bane of VoIP purists who believe in a bill and keep business model with settlement free peering. They claim it was the business model that drove the explosive growth of the Internet and can do the same for VoIP adoption. Da Silva explained the irony in this constant debate between incumbents and new entrants. Bilateral agreements are basically the bill and keep model. Each carrier manages their traffic flow to their bilateral partners so the net settlement is zero. Which is identical to the bill and keep model for networks that have equal traffic exchange. However, there is a funny money accounting twist that is important to incumbent carriers. Bilateral agreements with a zero net settlement enable each trading partner to book a significant amount of revenue. If incumbents changed to a bill and keep model they would take a big revenue loss. Of course the cash impact of the accounting change would be zero, but no incumbent telco executive wants his incentive pay plan to take a hit from a decline in revenue.

Acme Interconnect 2009

2009-11-09T08:22:00.004-05:00

I am at the Acme Interconnect 2009 conference in Miami, FL. The Acme folks do a great job hosting a customer. The conference sessions they have organized are excellent and the facilities and activities are first class. It is no surprise that Acme customers are here from around the world.

Yesterday the conference opened with your choice of golf or fishing. I took the fishing trip. It was a great outing with plenty of beer and good food on the high seas off of Miami beach. The swells were five to eight feet and it was a trip to remember. Seamus Hourihan made a tally of the catch and the 22 people on aboard caught an average of approximately two fish each. Not bad, and nobody got sea sick.

Comptel Update

2009-10-15T14:56:00.002-04:00

I was at the Comptel show in Orlando, FL earlier this week and everything appeared to be pretty much business as usual. The number of exhibitors were down about 10% from the last Comptel show, but overall the quantity and quality of business activity was good. The last year has not been a bad recession for the telecom sector.

The interesting new development apparent at this Comptel show is the new emphasis on managed services and the emergence of cloud computing. I suppose capital limitations are forcing some operators to avoid capital outlays and prompting others to repackage their services to meet the demand. I had been skeptical about cloud computing, thinking it was just another marketing buzzword for hosted services. However, Kelly Sparks, Chief Architect for Pac-West Telecomm gave me an insightful description of their cloud computing plans. It will be much more than just hosted servicess. The Pac-West Telecomm cloud computing offer will leverage their private network in ways that cannot be matched by cloud computing services offered over the Internet. Their plans make a lot of sense. They will not be competing with the likes of Amazon.com, instead they will focus on customers and telecom applications that need a dedicated network and service level assurance (SLA). In some respects Pac-West will become a distribution channel for their cloud application vendors. An interesting new carrier business model that is beyond simple network management.

Asterisk is 10

2009-10-15T00:02:00.005-04:00

Wednesday night at Astricon 2009 in Glendale, AZ, the Asterisk Community celebrated the 10th birthday of Asterisk. It was not your typical trade show party. It was a pretty quirky event with a lot techie guys drinking beer and playing video games, darts, pool, air hockey and foosball. One colleague noted that there were seven females present and that this was an increase from three females at the Astricon party in 2008. Hey, but what do you expect, this is the Asterisk community. It is a bright, eclectic mix of hackers, open source developers and a lot of independent entrepreneurs who install telephone systems for small and medium businesses.

The stereotypical open source developer is an irreverent young guy with an outrageous mohawk and yes of course those folks are an important part of the Asterisk community. But what I find interesting is the large number of older business owners who attend Astricon. These are experienced guys who have been installing data and telephone systems for 20-30 years. If you wonder who these people are, go to the yellow pages and look for telephone systems. You will see pages of small businesses that install telephone systems. This is a highly fragmented, grass roots distribution channel that adopted Asterisk early because it enables them to earn margins they could never make from reselling the traditional PBX solutions. According to sources at Digium, Asterisk based PBX systems serve more phone seats in North America than any other phone system. That is a pretty amazing accomplishment for a 10 year project.

While at Astricon, I gave a number on tutorial on Number Portability. Here is a download link for the Number Portability presentation.

Covad Deploys TransNexus

2009-10-08T13:20:00.006-04:00

I am happy to announce that we have been working closely with Covad Communications since the beginning of 2009 as they have expanded their VoIP services. Covad is a technically sophisticated carrier that rigorously explores every operating detail. They left no stone unturned as they tested our software. The Covad type of customer is our favorite type of customer. They know what they are doing and everything they do is planned and tested. They run a professional operation and there are no surprises and practically no technical support issues. More importantly, they are always looking for ways to innovate - a force that helps vendors like TransNexus lead the market with new features.

I am especially pleased with the testimonial from Jake Heinz, vice president and general manager of Covad. “Covad canvassed the vendor space for intelligent routing platforms, and based on feature/functionality, a fair price, and an excellent ROI, we chose TransNexus.” You can read more about this success story at www.TransNexus.com.

TransNexus Certified as a GSMA Pathfinder Partner

2009-10-07T18:13:00.003-04:00

On October 1st, the GSMA announced the launch of its PathFinder Number Portability Discovery Service. The GSMA is the global wireless organization consisting of nearly 800 wireless carriers in 219 countries. PathFinder was created by the GSMA to provide an interoperable, industry-wide solution that enables the routing of global IP service interconnect traffic.

A couple of years ago the GSMA presented its vision for an all IP world based on the IPX, a global IP network run by GSMA operators. Carriers connected to the IPX would be able to peer IP communucation applications such as VoIP and video end to end over IP networks. The global route discovery provided by PathFinder is a big step forward in implementing that vision.

At TransNexus, we are proud to be one of the first vendor partners certified for the PathFinder service. The TransNexus OSPrey route server is certified to query PathFinder for Number Portability discovery. The GSMA Pathfinder Announcement.

FISPA Annual Conference

2009-10-06T09:10:00.003-04:00

Last Friday, I had the pleasure of presenting to FISPA, the Federation of Internet Solution Providers of the Americas. FISPA had its annual meeting in Atlanta last week and I was invited to provide a tutorial on how Number Portability impacts VoIP service providers. I was unaware of FISPA until just last month. In general, when I hear the term ISP, the picture that comes to my mind is a struggling Internet Service Provider unable to compete with the giant carriers.

However, the folks I met at FISPA did not fit that stereotype. Primarily, the folks I met were owners of small service providers that provide all kind of data centric services. They had learned how to become more than just Internet Service Providers, they had in fact become solution providers for their small and medium size business customers. Being entrepreneurial and innovative, they had developed nice businesses serving the needs of their customers that could not be met by large competitors. It was a technically savvy group that also had the business sense for building a business through bootstrap financing. No VCs or corporate high flyers were in the crowd, these guys were all achievers, not marketing promoters. It was a nice change from the typical trade show conference that can be heavy on creating buzz, but lacking any real substance.

I estimate that about one half of the FISPA members provide VoIP services and I expect more will add VoIP to their suite of services. While FISPA is a small group, I expect this organization will grow as more innovative service providers learn how to serve vertical market niches that require focus and expertise that is beyond the scope of large competitors. You can learn more about FISPA at www.FISPA.org.

Astricon 2009 Interview

2009-10-01T10:28:00.007-04:00

Recently I was interviewed by Erik Linask, TMC's Editorial Director for Astricon 2009. Below is a transcript of that interview.

Erik Linask: This year marks the 10th birthday of Asterisk. What has driven its growth over a decade?

Jim Dalton: Pent-up demand for a reasonably priced PBX solution for small and medium businesses.

EL: What is your company’s involvement with Asterisk?

JD: TransNexus maintains the OSP module in Asterisk. OSP is the Open Settlement Protocol for secure peering among VoIP peers.

EL: Who is your target customer and what are their biggest pain points?

JD: Our target customers are any service provider who offers A to Z VoIP termination services. They use the TransNexus software for least cost routing, number portability, traffic reports, profitability analysis and billing.

EL: How does Asterisk help address your customers’ pain points?

JD: Actually, there is very little intersection between the TransNexus customer base and the universe of Asterisk users. Nevertheless, we like to be involved with Asterisk because we think it will become more than just a PBX. Asterisk and a couple of other open source telecom projects are the most significant leading indicators for the future of telecommunications. If you are not watching developments in the Asterisk community, you have your head in the sand.

EL: What are the key differentiators of your product over others on the market?

JD: Focus and Price.

For twelve years our focus has been on VoIP routing and accounting. From the beginning our solution architecture has been based on a standards based routing and CDR collection server. This is quite different from traditional least cost routing and billing solutions for telecom. Traditionally, these solutions have been customized to interoperate with each switch vendor’s proprietary routing table and CDR format. Customization for many different vendors is very expensive, but that is just the way it was in telecom. The TransNexus approach, however, is based on the OSP protocol which is an XML based standard for authorizing, routing and accounting for IP transactions. It is a simpler approach with much better scalability. For example, we have customers that manage over forty million routes with the TransNexus OSPrey routing and CDR collection server. By comparison, a softswitch or session border controller is limited to a routing table of just two million routes or less.

Price is the second differentiator. Since our standards based architecture is simpler and more efficient, our costs are lower which means we are profitable even though our solution is significantly lower in price than the competition. In addition, we are the only vendor that offers a free 90 day evaluation trial complete with technical support and remote training. We have had multiple customers save more money using TransNexus software during the free evaluation trial than they paid us at the end of 90 days. I do not know of any competitor that can profitably offer a solution with a payback period of zero months.

EL: You are speaking at AstriCon 2009 – describe your session and tell our readers why they should attend it.

JD: My session will be a tutorial on Number Portability. In the US, 40% of all calls are to telephone numbers that have been ported to a different service provider. If you are routing calls on the dialed telephone number, then 40% of your calls will be mis-routed! The call will still complete to the called party so end users will not know the call was mis-routed, but the cost of routing a call to the wrong carrier who then has to forward the call to the correct terminating provider is very expensive. So this tutorial will explain the basics of Number Portability technology and how Asterisk users and correct their call routing with Number Portability and most importantly save on termination fees.

EL: What else do you expect to see or hear at AstriCon that will be particularly interesting or innovative?

JD: Astricon attracts an eclectic group of innovators. You never know what you will learn at Astricon. That is what makes Astricon interesting.

EL: Where do you see the Asterisk market in five years?

JD: Asterisk will continue to be used in new ways that folks never imagined. But more importantly, Asterisk will have a greater share of the PBX market.

VON Peering Initiative

2009-09-28T16:55:00.005-04:00

My trip to VON last week was cut short last week by three feet of flood waters following through my home in Atlanta. Nevertheless I did get to spend some time at the show and these are a couple of my impressions.

First, Rick Martin and his colleagues put up a courageous effort to rebuild the VON show in a very tough economic climate. The conference agenda looked good and a couple of premier keynote speakers certainly made the conference content worth attending. Nevertheless, the attendance had to be less than they hoped for. For what its worth, I heard that the Channel Partners show, which followed immediately after VON, was a good success.

Second, a possible spark for re-igniting interest in VON may come from a grass roots VoIP peering effort being led by Dave Gilbert, CEO of SimpleSignal. As a VoIP operator, Dave organized a meeting at VON with a cadre of like minded VoIP service providers who want to build a "bill and keep" (settlement free) VoIP peering federation. This idea is certainly not new. A number of VoIP visionaries, starting with Jeff Pulver (the founder of VON), have created settlement free VoIP peering networks. At TransNexus, we were the Atlanta node for Jeff's second Free World Dial-up (FWD II) network in 1998 which used VocalTec software and Dialogic cards. FWD II was a lot of fun and was a fascinating testbed for the early days of VON, but it never took-off.

Other more commercially based peering federations such as the Voice Peering Fabric and XConnect have had success, but the dream of a rapidly growing, VoIP peering network which can challenge the Public Switched Telephone Network has yet to emerge. I do not think the VoIP peering pioneers made any mistakes other than being too early for the market. Perhaps the grassroots effort by Dave Gilbert and his group of VoIP service provider peers will have the good fortune of stumbling onto perfect market timing and starting a new peering federation that grows virally. If they do, it will be good for VON and good for everyone in the VoIP business.

Interview with Rich Tehrani

2009-09-17T12:24:00.004-04:00

Rich Tehrani is CEO of TMC which publishes Internet Telephony magazine.

RT: What has the economic crisis taught you, and how has it changed your customers?

JD: If you are not over-leveraged, then an economic downturn can actually create new opportunities as competitors and customers adjust to their businesses’ lower revenue and lower profit margins.

RT: How is this down economy affecting your decisions to reinvest in your company or market, if at all? Where will you invest?

JD: We are fortunate that in our niche of the VoIP market, demand is still growing. We are hiring engineering staff, but keeping all other expenses flat.

RT: What’s the strongest segment in the communications industry?

JD: Wireless.

RT: With the rise of smartphones and netbooks, many wireless technologies, such as WiFi appear to be poised for rapid growth. For example, we’re seeing more and more airlines add in-flight WiFi. In general, how widespread should WiFi be, in your view?

JD: I don’t know how widespread WiFi should be, but I hope it becomes universally available. WiFi is an inexpensive and reliable network access technology that anyone can provide. All other forms of wireless network access are controlled by spectrum license holders. WiFi offers hope for a competitive market of public network access.

RT: Which nation or region of the world will present the largest opportunity for your company in 2009/10?

JD: We serve customers around the world, but our best market is the United States.

RT: In what ways is President Barack Obama helping or hindering the technology markets? What more can he do?

JD: For this question, our perspective is limited to observations of FCC policy. In this regard, I think the FCC under the Obama administration will do more to encourage competition and innovation. For the last eight years, the FCC appeared to favor the policies of major telephone companies over smaller more innovative firms. I think this is changing, but it is still too early to be certain.

RT: What device or devices do you use, and what do you wish you used?

JD: I like to keep it simple. I use a laptop and it is all I need.

RT: I understand you are speaking during ITEXPO West, to be held Sept. 1 to 3 in Los Angeles. Describe your talk and tell us what companies or people should attend.

JD: I will be giving a tutorial on number portability and why it is important for all VoIP service providers. Telephony service providers have always relied on the Local Exchange Routing Guide for routing. VoIP service providers still use the LERG today to translate routing based on “Local Access & Transport Area” and “Operating Company Number” because the rate schedules they get from their providers are quoted in terms of LATA and OCN. However, we are now reaching a tipping point in number portability.

By some estimates, 25 percent of all numbers in the LERG have been ported to a different carrier. This means that traditional routing using the LERG without number portability correction causes up to 25 percent of all calls to be misrouted. Misrouted calls still get completed, so the end users do not realize a service issue. But the service provider gets stuck with a termination charge larger than necessary because the call was not routed to the lowest cost provider. TransNexus will be releasing a new version of its free OSPrey routing server which will include the ability to host locally, or query, the U.S. number portability database managed by NeuStar.

RT: Why should customers choose your company’s solutions? How do they justify the expense to management?

JD: We see two reasons why customers choose the TransNexus solution. First, we have been focused on developing software for VoIP routing and accounting since 1997. Few companies can match our experience and focus. TransNexus is not a start-up that will disappear when the funding runs out. We are here to stay. Also, we have a proven track record and are certified partners with leading VoIP vendors such as Acme Packet, Cisco andMetaSwitch. The second reason is price. We have always kept our expenses low so we can offer what I believe are the most competitive prices in the industry for sophisticated VoIP routing and reporting software.

Interconnect Business Models

2009-09-17T11:51:00.003-04:00

One of the most interesting white papers I have read on the economics of network interconnect models was written by Jay M. Atkinson and Christopher C. Barnekov who were working at the FCC. The white paper has a very odd title: "A Coasian Alternative to Pigovian Regulation of Network Interconnection". The title sounds comical for non-economists like me, but if you take the time to dig into this white paper it offers an insightful understanding of interconnect costs between networks. More importantly, it illustrates why interconnect costs between two networks are not equal. When a small and large network interconnect, the larger network tends to experience a disproportionate share of the interconnect cost. This explains why network operators fear a simple bill and keep peering model with smaller networks. You can download the white paper from http://web.si.umich.edu/tprc/papers/2004/348/CoasianAlternative040901b.pdf.

If you do not have the time to read the tedious details covered in this white paper, you can download a presentation I gave at PTC'06 entitled "Interconnect Business Models". It summarizes some of the main points addressed by Atkinson and Barnekov in their white paper.

VoIP Peering Discussion at VON

2009-09-17T10:14:00.005-04:00

I had an interesting discussion yesterday with Dave Gilbert (SimpleSignal), Dave Cullerot (Acme Packet) and Kevin Groth (XConnect) about the current state of VoIP peering and how it will evolve. VoIP peering is an old idea that is slowly catching on, but will a public VoIP peering network ever replace the Public Switched Telephone Network? There is plenty of speculation about this topic which the four of us will be discussing next Tuesday, September 22nd at the VON conference in Miami. We are participating on a panel titled"The Peering Puzzle: VoIP Interconnections and Peering in a Distributed-Network World".

One fundamental issue of contention is the definition of VoIP peering. Some folks define VoIP peering as a secure, dedicated IP connection between two VoIP carriers. This definition is how data peering is defined between to large Internet Service Providers. But other folks take a more expansive view and believe VoIP peering is a many to many, peer to peer connection among a large and open group of VoIP networks. This type of peering enables the full potential for two peers to realize all the application possibilities of peer to peer communications, such as the use of broadband codecs and video.

While most peering issues are discussed in terms of technical interoperability, I believe the most important issue is the economic model. The primary economic incentive of peering models are focused on cost avoidance for the source network. This approach in not sufficient. If you want a VoIP peering revolution, there needs to be an economic incentive for all VoIP networks to open their networks to any anonymous VoIP network. Peering models today lack a secure clearing and settlement mechanism that provides an efficient incentive for destination VoIP carriers to take traffic from an anonymous sources. When this problem is solved, a public VoIP network will replace the Public Switched Telephone Network. The white paper "Peer to Peer Settlement for VoIP Networks" provides a proposed solution.

Our panel discussion at VON is scheduled for 8:00 am next Tuesday. I hope you will join us.

Report on IT-Expo in Los Angeles

2009-09-04T16:17:00.006-04:00

I just returned from the Internet Telephony Expo in Los Angeles where I gave a tutorial on Number Portability. Overall the conference was good. I am always interested in trade show attendance and vibrance since it is a good business indicator for our market. Overall attendance at the keynotes seemed to be up from last year and the exhibit hall had about the same number of booths. More importantly, I did not detect any gloom. The continued growth of VoIP in a recession is a good sign for rapid growth of the VoIP business when the overall market rebounds.

Interestingly, my 8:30 am tutorial on Number Portability was well attended. By most measures, the topic of Number Portability registers about zero in terms of excitement. Couple this with an early morning time and I expected just a few of folks in the room reading the paper while they drank their coffee. Instead, the tutorial was well attended and the group was engaged, asking a lot of questions throughout. One gentlemen told me that a last minute call from his CEO, who was stuck in traffic, instructed him to attend the number portability tutorial and take notes. In the past, VoIP folks dismissed number portability as insignificant. But things have changed. We have found that up to 40% of our customer's VoIP traffic is to ported telephone numbers. This means that their least cost routing, without number portability correction, is wrong for 40% of their calls. The simple implementation of routing calls based on the ported number can increase a VoIP provider's net income by 4% to 8%. This is not a revenue increase, this is cash income straight to the bottom line. One TransNexus customer tells me that our analysis is much too conservative and that the savings is significantly greater. All driven by an easy to implement database dip. Everybody loves easy money, I guess that is why an early morning tutorial on a dry topic drew such a keen crowd. You can download the presentation from www.TransNexus.com.

Rate Plan Exasperation

2009-05-02T08:19:00.006-04:00

Telecom service provider rate plans come in a variety of formats, most of which are confusing. Telecom rates are quoted by NPA-NXX, LATA, LATA and OCN, LATA and Tier, State and OCN, State and LATA. Sometimes rates are quoted by geographic names, such as a single rate for USA. This practice sounds user friendly, but is ambiquous and can lead to rate disputes. For example, some carriers define USA as the lower 48 states and include Alaska and Hawaii as offshore. And what about Guam, US Virgin Islands, Puerto Rico and American Samoa? Are US territories part of the USA rate plan?

Then there are jurisdictional rates - that is different rates based on the calling and called number. Example are different rates for inter-state, intra-state and local calls. There can also be rates for local long distance. Local long distance, now that is a funny oxymoron, until you get hit with charges higher than expected.

All these different rate formats cause major headaches for enterprises and small service providers who must constantly struggle to understand their telecom bill and dispute over-charges from their carrier. In fact, a significant industry niche has developed to serve the need of managing the confusing array of telecom fees. Do a Google search on telecom expense management and you will find dozens of companies serving this need.

So all this effort struggling with confusing telecom rates would go away if carriers offered rates in a common format for explicit ranges of telephone numbers. Now I understand that there are a lot very good technical and business reasons for carriers to not do this. However, non-confusing rate plans are a common need for all telecom consumers. So to meet this need, John Todd, the VoIP innovator who created ITAD, is leading a VoIP community effort to define a telecom rate plan that is easy to understand and to load for routing and billing. To review the proposed rate plan format, and offer your edits, go to http://www.loligo.com/asterisk/misc/rates/.

IT Expo - Miami

2009-02-18T11:07:00.002-05:00

Last week I was at the Telephony Expo in Miami, Florida. Given the overall direction of the general economy, my expectations for the conference were low. I expected low attendance and a generally negative outlook from most attendees. What I found, however, was different. Traffic in the exhibition hall was good and just about everyone I spoke to was genuinely positive about their business prospects for 2009. I remember 2002 when VoIP companies were going bankrupt left and right. At that time trade shows were just a gossip hall about who would go broke next. There was no sign of that in Miami last week. Everyone I spoke to was looking for ways to grow their business. From what we see, it looks like the VoIP business will be good in 2009 as customers look for VoIP technology as a way to save costs and offer new services. Click here to see my interview at IT-Expo.

VoIP Outlook for 2009

2009-02-18T10:22:00.005-05:00

In early January I was interviewed by Rich Tehrani, TMC President and Editor-in-Chief. TMC has published that interview which is posted below.

[RT] Recently I had the opportunity to speak with Jim Dalton, CEO of TransNexus. Our conversation went something like this:

[RT] Are you generally optimistic, Pessimistic or realistic?

[JD] I am optimistic about VoIP and pessimistic about the overall economy. VoIP as a technology is losing its excitement as it matures, but in reality we are still very early in the process of replacing the PSTN with a global VoIP infrastructure. We think VoIP has at least 10 years of healthy growth ahead. As for the overall economy, we believe 2009 will be worse than the expectations we read in the press or the stock prices we see in the market. However, we think a very bad economy can be a positive for VoIP. VoIP technology is much more cost effective than traditional TDM telco technology. The cost advantages and flexibility of VoIP are more compelling in a bad economy.

[RT] How many frequent flyer miles have you amassed?

[JD] Same amount as last year.

[RT] How surprised are you at the global financial situation?

[JD] Our business has been good and the business of our customers appears good. So I was very surprised by the financial crash last October.

[RT] How is your company changing the way it does business as a result?

[JD] So far our strategy and plans for the last three years have worked and will continue unchanged in 2009. This is not because we are smarter than other folks, we are just fortunate to be in the VoIP business that is not suffering the problems of the overall economy.

[RT] How have customers reacted to the slowing global markets?

[JD] We see some caution and some customer’s reports longer time to collect receivables, but overall no significant changes.

[RT] Do you see this time as an opportunity or a rough spot to get through quickly?

[JD] We think this is a long term opportunity. We have already seen competing vendors who offer solutions for PTSN networks hit by financial troubles. We think this downturn is a good opportunity for vendors focused on VoIP solutions that are cost effective.

[RT] What will companies need to do to survive this downturn?

[JD] Manage their business so they can lower their prices and still generate positive cash flow.

[RT] How do your company’s’ products help customers in a slow market?

[JD] The simplest and most powerful value proposition we offer is Least Cost Routing. Service providers can increase their gross profit margins by 20% or more with least cost routing. It is a simple concept, but not trivial to implement. We make the implementation easy. We offer a free 90 day trial period complete with technical support and remote training. Some of our customers save more money from least cost routing during the free 90 trial than it costs to buy our software. This means that our software has a zero payback period for the customer. This is an easy business case for CFOs to understand.

[RT] What do you feel is the strongest segment of the communications space? Technology?

[JD] The RBOC and the Cable Companies enjoy a market duopoly for services to the residential market. A duopoly is every business operator's dream, limited competition but unregulated pricing. The RBOCs and Cable Companies do a great job telling the FCC that they face daunting challenges, but in reality they enjoy a fantastic business model. RBOCs and Cable Companies are the strongest segment in the communications space by far.

Having said that, however, I also think that open source VoIP will become much more important segment in the communications space. I do not foresee open source as a strong segment financially, but very powerful in moving the direction of the communications space. The success and impact ofAsterisk is well known, but we also see OpenSIPS and FreeSWITCH continuing to increase their market penetration.

[RT] Which would you rather be president of and why? Google, Yahoo, Microsoft, Cisco, the United States.

[JD] Google's management is shrewd, but more importantly they are very effective at what they do. They are great operators.

[RT] What does President Bush need to accomplish before he leaves office?

[JD] Making a gracious and cooperative transition to the new administration has been one of President Bush's best accomplishments.

[RT] What does an Obama administration need to do to help communications and technology become more pervasive?

[JD] Enable more competition for broadband services. The duopoly market structure of the RBOCs and the Cable Companies does not encourage innovation.

[RT] Will this slowdown present an opportunity to reinvest in your company/market? If so, where will you invest?

[JD] We see the slowdown as an opportunity for vendors of VoIP solutions. We plan to increase our investment in software development in 2009.

[RT] Which country will present the largest opportunity for your company in 2009/10?

[JD] We have customers in five continents, but we are a US company and the US is our best opportunity in 2009/10.

[RT] If Nokia, RIM, Google and Apple devices are stranded together on an island, who survives and why?

[JD] Apple. No one can beat Steve Jobs for usability and elegance. Apple is the obvious winner of any user device competition.

[RT] I understand you are exhibiting at ITEXPO which takes place Feb 2-4 2009 in Miami. What will you be showing there?

[JD] We are releasing a new software server named OSPrey. It is a VoIP Routing and CDR Collection server. It is not a sexy product, but OSPrey is very effective, scalable and reliable for routing VoIP calls and collecting CDRs. Traditionally, our target market has been the service provider market. But this new product targets enterprises that operate a VoIP network with multiple branch offices. It provides a central application for authenticating and routing inter-office and off-net VoIP calls. It also collects CDRs so enterprises can allocate VoIP charges back to specific branch offices. Our target market is enterprises that use Asterisk and the software is free - not open source, but cost free. We will be exhibiting in theDigium-Asterisk World. We are excited about being part of the Digium-Asterisk World because we think Digium has the right cost structure and value proposition to benefit from the tough economy in 2009.

[RT] What sorts of companies/people should come to your exhibit?

[JD] Enterprise Telecom Managers that manage multi-office PBX networks. The free OSPrey server is designed to work with Asterisk or OpenSER. But we also have a commercial version of the software that is optimized for Cisco VoIP networks (H.323 or SIP).

[RT] Why should customers choose your company’s solutions – and how do they justify the expense to management?

[JD] The cost savings from simplifying network operation and least cost routing makes a very powerful return on investment business case. As I said earlier, we have had customers save more than enough money using our software during the free trial period to pay for the product before the trial ended.

[RT] I am a purchasing decision-maker, why do I need to speak with you before I buy?

[JD] We are the price leader in our niche. Even if you do not purchase the TransNexus solution, the possibility of buying the TransNexus solution will enable you to negotiate a better deal with your vendor. Our mission is to help folks lower the cost of their VoIP network. So even if you do not buy our solution, but we help you save some money by getting a better price from your vendor, then we feel some satisfaction.

[RT] What is your favorite part of your job?

[JD] Getting positive feedback from our customers.

[RT] What do you look forward to in the future and why?

[JD] Continued growth in the VoIP market.

Rich Tehrani is President and Group Editor-in-Chief of TMC

Incentives for Peer to Peer to VoIP

2009-02-18T09:09:00.004-05:00

At the IT Expo in Miami I had the opportunity to speak with Dan York about a presentation I gave on Next Generation Network Peer to Peer Settlement Billing. The critical element missing from the vision of universal peer to peer VoIP communications is an economic incentive for carriers to open up their "walled gardens." Peer to peer settlement through competitive clearinghouses provides the economic incentive for all VoIP networks to openly peer. Peer to peer transaction settlement is a tried and true business model being used by multiple industries. Click here to view the interview with Dan York of Emerging Tech Talk.

Asterisk V1.4 Performance

2008-10-24T14:44:00.002-04:00

We recently ran a performance test of Asterisk configured as a SIP Back to Back User Agent (B2BUA). The test platform hosting Asterisk was a $1000 Dell PowerEdge 840 with a Quad Core Xeon X3220, 2x4M cache, 2.40 GHz, 1066 MHz FSB and 4 GB RAM. Redhat V5 was the operating system. The test was configured to simulate a wholesale VoIP operation with three minute call durations and an average of two call retries for every completed call. This was an "out of the box"Asterisk configuration with default settings and no optimizations.

We found that Asterisk on the test server could handle approximately 1000 simultaneous calls with no codec transalation. This works out to be about a $1 per port investment for a B2BUA platform.

When calls were transcoded from G.711 to G.729, the call capacity fell to 320 simultaneous calls. With the added cost of of the G.729 codec royalty and the lower call capacity, the cost increases to approximately $13.50 per port. You can download the test results and all the test plan details from: http://www.transnexus.com/White%20Papers/Performance_Test_of_Asterisk_v1-4.htm

Astricon 2008 - The Momentum Accelerates

2008-09-26T09:27:00.004-04:00

I was at Astricon in Phoenix Arizona this week. It was a very good conference with approximately 700 attendees. There was certainly no indication that the market share growth of Asterisk is going to slow any time soon. It was clearly evident at this show that Asterisk is still early in its adoption lifecycle, but the product and community are beginning to mature. This is a good thing.

There will always be nostalgia for the good old days of Asterisk and the first Astricon. The first Astriscon was much larger than expected and had all the excitement of a new movement that is catching fire. Those who were there will remember the eager cheering crowd when Mark Spencer announced the release of Asterisk 1.0 and a free IAXy to whomever could post the news on Slashdot.org first. Everyone there felt special because they had the luck or foresight to be on the ground floor of something that was going to be big. And they were right, Asterisk is going to big.

This Astricon was also exciting, but in a different way. If you are an open source purist, you probably were disappointed that the zany, geekiness has been diluted by the growing presence of boring folks. Boring folks who either use Asterisk in their business or are part of the Asterisk ecosystem. I have been an Asterisk fan since 2003, but as a boring person I find the new maturity of Asterisk more exciting than ever. Asterisk's feature set, scalability and reliability continue to improve. It is now competing successfully with the blue chip PBX vendors such as Cisco and Nortel for the small business market.

At Atsricon, I presented our latest performance test results for Asterisk as a B2BUA. We found that Asterisk running on a Dell PowerEdge 840 with a single quad core 2.4 GHz CPU, running Redhat v5.1 can handle 1000 simultaneous calls of G.711 traffic and 320 simultaneous calls with transcoding from G.711 to G.729. The economics of this solution work out to be $1 per port with no transcoding and $13.13 per port for transcoding from G.711 to G.729. We are working on the final document which we will publish to the Asterisk mailing list.

OpenSER and RTPproxy Performance Test

2008-09-20T19:03:00.002-04:00

We have published results from a performance test of OpenSER V1.3 and RTPproxy V1.0 running on the same host server. The purpose of the test was to understand the relationship between server CPU performance and the maximum number of simultaneous calls. The massive scalability of SIP signaling with OpenSER is well known. For this test the critical component being tested was RTPproxy and the proxying of media.

A quick summary of the test results show that RTPproxy running on a single core of a 2.33 GHz Intel Xeon 5140 CPU can manage up to 750 simultaneous calls. If both cores had been used on both Xeon CPUs, we expect that RTPproxy could have managed 3000 simultaneous calls.

For a summary of the performance test, and to download the complete test plan, go to www.transnexus.com/White Papers/OpenSER_RTPproxy_Benchmark_Test.pdf

We are finalizing a new performance test of Asterisk V1.4 used as a B2BUA. We will present our Asterisk performance test results next week at Astricon.

Internet Telephony Expo Update

2008-09-17T21:53:00.004-04:00

I have spent the last two days at the Internet Telephony Expo in Los Angeles. It appears that the Internet Telephony Expo has benefited from the demise of the Voice on the Net (VON) show. In a down economy, the number of attendees looks to be up slightly from last year and the number of vendor exhibits appears to have increased.

Yesterday I gave a presentation on Open Source and Open Standards (download presentation). What impressed me most about the audience was how little they knew about VoIP and telecom. I see this as a very positive development as more people from enterprises are being drawn to learn about VoIP. VoIP is now spreading beyond the developers and early adopters. The audience is now people who are looking to solve real world problems, not futurists.

The most popular sessions at the conference have been the sessions on SIP Trunking. The audience at these sessions appeared keen to get the practical insight they need to make decisions on how to replace PRI circuits with SIP trunking services. While I have not heard any buzz about the latest and greatest new innovations, it is nice to see VoIP technology maturing with real and wide spread deployments.

TransNexus Joins Intel Certified Solutions Program

2008-08-20T12:12:00.007-04:00

TransNexus has joined the Intel® Certified Solutions Program and will certify its NexOSS software for Intel processors. This is a good program with real technical benefits, not just a marketing gimmick. Given the $25,000 price to join the program, we did our homework to make certain the program offered more than just a rubber stamp certification logo. Admission to this program provides TransNexus with software tools that will assist us in our efforts to boost the performance of our software and reduce security threats. As part of the process, Intel will certify the performance of TransNexus software based on hard data and provide feedback on how our software can be made more secure.

According to Scott Harrison, director of Intel's Partner Programs Organization, “The Intel Certified Solutions Program moves in lockstep with technology development at Intel. Through certification, solution vendors can be more confident that their applications take advantage of the Intel technologies available to their customers, both now and in the future.” We are glad to hear this. We have spent a lot of time over the last year optimizing TransNexus software for multi-core processors. We are looking forward to the Intel certification to verify the competitive advantage we offer our customers.

TransNexus will support Kamailio and OpenSIPS

2008-08-06T14:35:00.002-04:00

There has been a lot of confusion and angst in the OpenSER community this week. It all started on July 28th when the OpenSER project renamed itself to Kamailio. According to the announcement, "Kamailio is a Hawaiian word. Kama'ilio means talk, to converse." The OpenSER name had to be changed for trademark reasons. The feedback on the mailing list about the new name was generally negative since Kamailio is a complicated name to pronounce and to remember. OpenSER version 1.3 will remain OpenSER. But starting with release V1.4, the project is official named Kamailio (www.kamailio.org).

Then on August 4th, Bogdan-Andrei Iancu, an early developer of SER and a founder of the OpenSER fork from SER, announced that he was forking a new open source project from OpenSER. The new project is called OpenSIPS - for Open SIP Server (www.opensips.org). OpenSIPS is a better name than Kamailio and the fork annoucement caused a flurry of e-mail messages on the Kamailio mailing list. Few folks were happy about the surprise.

TransNexus is not an OpenSER user, but our customers are. I think the general feeling among the user community is concern about the viability of the two projects going forward. The OpenSER fork from OpenSER was met with similar concern. In the end, however, OpenSER hastened the adoption of open source VoIP software. We are optimistic that Kamailio will continue to be successful and that OpenSIPS will grow into a thriving project that serves a different niche from Kamailio. The Telekom world is huge and there is plenty of market space for both projects.

TransNexus plans to continue its support for the original SER project, the OpenSER turned Kamailio project and the new OpenSIPS project because we believe this is what our customers want.

Voice Peering Forum Update

2008-06-24T19:42:00.003-04:00

I am at the closing plenary of the Voice Peering Forum in SanFrancisco. It has been a very interesting two days with a lot of informative and thought provoking conversations and presentations. The Voice Peering Fabric made two very notable announcements today. First the VPF is opening up its ENUM directory to the public Internet. Previously, the VPF ENUM registry has been limited to users of the VPF private ethernet network. Now common route discovery between networks on the VPF and public Internet is possible. More importantly, however, the VPF ENUM registry is no longer just a simple open database. The VPF ENUM registry users can now define the groups of users that have access to their ENUM entries - similar to the features offer by other ENUM solutions such as NetNumber.

The second announcement from the VPF was the launch of the VPF Automated Trading Platform (ATP) which enables a VPF user to define and control a hosted least cost routing and settlement solution. Users will deposit funds with the ATP which will automatically execute clearing and settlement of funds between peering partners.

In many respects, the VPF initiative is similar to Arbinet's VoIP peering service, however, their is one big difference. The VPF ENUM registry and ATP is completely free for VPF users. Automated clearing and settlement of VoIP peering transactions seems like a major infrastructure service that will have a material expense. Nevertheless the folks at the VPF feel confident that the float of prepaid deposits plus the addition of new VPF business will pay for the services

The Voice Peering Forum

2008-06-19T10:40:00.003-04:00

Yesterday I mentioned the Voice Peering Fabric (VPF), run by Stealth Communications, as an innovation to watch in NGN interconnection. Twice a year the VPF hosts its Voice Peering Forum. The Voice Peering Forum is a gathering of VPF customers and leaders of the VoIP industry for discussions about the future of peering applications, such as VoIP, over IP networks. I attended the last Voice Peering Forum in December 2007 in New York City and heard a lot interesting ideas and experiences about VoIP peering and new applications. Next Monday and Tuesday, June 23rd & 24th, the Voice Peering Forum will be meeting in San Francisco. On Tuesday, at 11:30am to 12:15pm, I will be participating in a panel discussing "To Couple or Decouple Routing Intelligence from SBC?". I expect it will be an interesting two days. I hope to see you there.

Ideas for VoIP Interconnect Evolution

2008-06-18T08:57:00.007-04:00

Like other folks involved with VoIP, we ponder how the new interconnect model for future IP services will evolve. I think we all share the same vision - open, global, peer to peer communication between any two applications on the Internet. This is the model we experience with the World Wide Web (WWW) and it works. Any browser can have a session with any web server. The technical challenge of HTTP inter-working is easy and the interconnect model is simple. The WWW business model is a broadcast model. The web site sponsors pay to have their content advertised on the Internet. It is a very simple economic model that benefits both the source (web browser) and destination (web site) of each session.

The interconnect challenge for Next Generation Networks (NGN) is much more complex. Like the WWW model, there are two different challenges: finding a technical and economic solution that encourages networks to interconnect. First, the technical inter-working challenges are much more complex. For example, VoIP, with its many features and codecs is much for complex that a typical web session. But, like all technical issues, the inter-working challenges are being overcome. The more challenging problem is finding the economic model that provides an incentive for for every network to interconnect.

The economic model for interconnecting networks for VoIP and other value added IP sessions such as video, gaming, and content distribution is much less simple than the WWW model. Many folks in the industry have advocated for a "Bill and Keep" model. In this model, the interconnecting networks bill their end user subscribers, keep all the revenue and interconnect with one another for free. There is no sharing of revenue between the networks of the calling and called parties. This simple model does have its merits, but the practical reality is that no network operators have indicated any interest in this model.

The economic model problem is simple, different applications might require more or less network infrastructure services and therefore have different costs, which may be small, but are still material for the network operators. In addition, the direction of the value exchange for session between the source and destination is unknown to the network. In some scenarios the calling party (source) is willing to pay for the entire session, in other scenarios the called party (destination) is willing to pay for the entire session. The major challenge is to find an economic model, coupled with a technical solution, that enables the efficient exchange of economic value that tracks the value of interconnected sessions. When this solution is found, NGN interconnection will flourish.

But back to the point we ponder - how will the winning model for NGN interconnection evolve? We see four possible paths.
1) Dominant Player Defines the Solution
In many industries a dominant player evolves that defines the industry wide standard. AT&T certainly did this for the US telephone network. But, today the telecom world is more competitive and fragmented than ever. It seems unlikely that a single company, or even a consortium of companies can dictate the industry standard.

2) Industry Cooperation
This is possible and we see these beginning of these efforts from CableLabs, GSMA, i3 Forum and others. But the process to define technical standards and standard practices is long and bureaucratic. In addition, since this is a global challenge, there will be competition between different interest groups which are huge, such as wireless versus wireline carriers, or the US versus the EU. This migration path is possible, but long.

3) Regulation
Mandating NGN interconnect standards is a scenario where regulators might add some value. But unfortunately regulators usually do not understand the issues, are overly influenced by the dominant players and lack innovation. It is doubtful that regulators will solve the problem.

4) Grass Roots
The last possible evolution path toward an NGN standard could from the users themselves. For example, this kind of solution could come from an open source project. Another scenario is that a group of small companies or new entrants to the industry develop a solution that works and grows to critical mass.

One possibility for the grass roots success scenario could be the Voice Peering Fabric (VPF) run by Stealth Communications. It is private ethernet network that enables VoIP peering among networks. The VPF is not a carrier and provides a neutral solution for network interconnection. The VPF provides an ENUM server to support the Bill and Keep economic model for interconnection. The VPF ENUM implementation has been successful, but the narrow appeal of the Bill and Keep model has limited its growth. We understand that the VPF will soon be providing infrastructure services that enable efficient interconnect billing settlements between peering VoIP networks. If they get the economic model right, VoIP peering on the VPF will flourish. If this happens, could the amount of network traffic using the VPF NGN interconnect model grow to critical mass and become a de facto standard before being countered by a solution from the industry giants? We will see.

The VPF is just one example of a possible grass roots solution for NGN interconnection. I am certain there are other innovators and entrepreneurs working to solve this global problem. If you know of any, I would like to hear from you.

CableLabs and VoIP Peering

2008-06-17T09:20:00.002-04:00

Well a lot of blogging from me last week about activities in the field of Next Generation Network (NGN) Interconnection. A more commonly used term for NGN Interconnection has been VoIP peering. However, the term VoIP peering may be is losing is usefulness since it limits the concept to VoIP. Anyway, one organization that has been surprisingly silent in the recent dialog about NGN Interconnection has been CableLabs.

In October 2006 CableLabs released a Request for Proposal (RFP) for "CableLabs® Voice over IP (VoIP) Peering Registry Services". The RFP was focused on delivering a solution for VoIP to VoIP calling between Cable MSOs (Multi-System Operators). The CableLabs RFP generated a lot of excitement when it was first released, but no progress has been reported by CableLabs. Rumors in the industry indicate a pilot trial is in place based on ENUM technology from Net Number deployed by NeuStar. More importantly, however, is the reality that Time Warner (TWC) - a critical player in any successful VoIP peering initiative, partnered with Sprint to handle all of TWCs offnet voice traffic.

So the lack of news about VoIP peering from CableLabs may be telling - there is not much to tell. Perhaps the Cable MSOs are too small a market niche to drive a standard NGN interconnection strategy, or may the initiative was just too early. Another possibility could be the limitation that the CableLabs approach had no mechanism for inter-carrier settlements. Open peering sounds great until you look more closely at the need to manage the profitability of your network. At that point, peering without any economic controls seems like a very risky concept.

IP Internetworking Alliance

2008-06-13T09:46:00.004-04:00

There is no shortage of new organizations focusing IP Interconnection standards. The latest organization to be formed is the IP Internetworking Alliance or IPIA. IPIA has been established by the GSMA to be the not for profit, governing body of the GSMA IPX. The IPIA mission statement is:

" To provide an environment whereby stakeholders from mobile operators, fixed operators and carriers can agree to technical specifications and commercial templates for the IPX, which will facilitate the global interworking of IP services."

The inaugural IPIA Board members are:
- AT&T
- Belagacom
- BT
- FT Orange
- GSMA
- iBasis (KPN)
- Softbank
- Telecom Italia
- Telefonica
- Telenor
- TeliaSonera
- Vodafone

The guiding principals for the IPIA are:

•1. Open to participation by all relevant stakeholders, i.e. those using and those providing IPX services;

•2. Transparent and fair decision making processes;

•3. Freedom for innovation and self determination in the development and implementation of IPX services;

•4. Support for national, regional and international requirements, including compliance with relevant regulation;

•5. The IPX Forum will not create standards, but will only define implementation guidelines for the use and implementation of existing standards.

The IPIA plans to begin accepting members in Q3 2008.

ATIS Inter-Carrier VoIP Workshop

2008-06-12T09:34:00.002-04:00

The migration to an all IP world takes another small step forward. This week ATIS is holding a workshop in Washington D.C. to discuss Inter-Carrier VoIP Call Routing (ICVR). If you do not know, ATIS stands for Alliance for Telecommunications Industry Solutions. ATIS is accredited by the American National Standards Institute (ANSI) and creates telecom industry standards for everything from provisioning to routing to billing.

The objective of the ICVR workshop will be to determine if there is consensus that it is desirable to reach a single inter-carrier VoIP routing solution. If a single routing solution is desired, then the workshop will begin development of a roadmap to achieve a single routing solution. This workshop is based on the work that has been done by the IVCR Focus Group which began in 2006. In January 2008 ATIS published the ATIS Inter-Carrier VoIP Call Routing Assessment and Work Plan. The Work Plan is a good document that describes alternative models for inter-carrier VoIP routing. The Work Plan identifies that inter-carrier billing and settlement are requirements, but no work has been done to define the requirements.

We are happy to see that IP interconnection is getting some serious focus by ATIS which is driven by the legacy telecom operators in the US. This will be good for VoIP vendors and good for consumers if the derived standards encourage an open competitive market.

The GSMA IPX

2008-06-11T08:23:00.003-04:00

Today's blog is a continuation of the discussion about Next Generation Network (NGN) interconnect. While there is a lot of discussion about the promise of replacing the Public Switch Telephone Network (PSTN) with an all IP network, few efforts seem to be getting past the discussion phase. The GSM Association's IPX effort, however, appears to have some real momentum. The GSM Association (GSMA) is the global association of wireless carriers who use the GSM wireless technology developed by ETSI. The majority of wireless users worldwide are on GSM networks. Since the number of wireless users surpass the number landline users, the GSMA is well positioned to define a NGN architecture which could replace the PSTN.

The GSMA' s proposed architecture for NGN interconnection is its IP eXchange or IPX. It is a grandiose plan that appears to address every possible carrier requirement. It appears so grandiose in fact that some may believe it will never happen. However, I think betting against the GSMA IPX would be a mistake. The concept was first publicized in 2006 and in February 2008 the first trials of the IPX were announced. One unique aspect of the IPX plan is that it addresses the requirement of intercarrier settlements upfront. If the IPX enables cascading settlement transactions among carriers, then the economic incentives will be in place to encourage open interconnect among carriers.

Today most carriers continue to interconnect via PSTN SS7 networks because it offers a proven economic model for interconnect compensation. However, carriers will follow the money. If the IPX has a better economic model for interconnect, then the IPX is positioned to replace the PSTN as the global voice network.

The i3 Forum

2008-06-10T08:57:00.004-04:00

As I mentioned in my blog yesterday, Next Generation Network (NGN) interconnection is a popular topic. On June 2nd, the opening plenary at the International Telecoms Week (ITW) conference in Washington D.C. was an announcement made by the International Interconnect Forum for IP services, also known as the i3 Forum. The i3 Forum was founded in Q3 2007 by the following eight international carriers:

- ATT Wholesale
- DT ICSS
- France Telecom Orange
- SingTel
- Telecom Italia Sparkle
- Telefonica
- Telekomunikacja Polska
- TeliaSonera

These eight carriers claim to represent 40% of all international traffic and serve more than 1 billion customers in 80 countries.

The mission and focus of the i3 Forum will be to provide specifications, based on existing standards, which can be used by the international wholesale carriers for migration of voice services to IP technology and IP-based multimedia services. The i3 Forum is not a standards body and will make use of existing standards from the ITU, ETSI-TISPAN, IETF and 3GPP. The group plans to liaise with other industry bodies with similar/competing interests such as GSMA and IPIA.

This group of former PTTs point out that domestic interconnection standards for TDM/IP or IP/IP are outside the scope of their mission. It seems to me that NGN interconnect, like the Internet, would not be constrained by geographic boundaries. But, since this is a group of PTTs that must feel the threat of wireless operators, I assume that it is best for them to keep local telecom interconnection a non-standard practice that creates an obstacle for new competing local carriers.

ETSI NGN Interconnect Workshop

2008-06-09T12:36:00.004-04:00

VoIP has been deployed for years, but VoIP is still just emerging as a viable alternative to replace the PSTN completely. The challenges of technical interconnection between networks are being solved as the standards mature and as VoIP equipment vendors improve their platforms. However, the challenge of solving the business problem of interconnect between carriers is just beginning. To replace the PSTN, the VoIP model must provide an economic incentive for ubiquitous interconnection among all VoIP networks - this requires intercarrier settlement.

Finding a solution for Next Generation Network (NGN) interconnect and settlement is becoming a relevant and popular topic. This week ETSI TISPAN is hosting a Workshop on Interconnection. The scope of the workshop will cover all aspects of the creation of a global standard for the IP interconnection of NGN services. It will aim to address interconnection requirements elaborated by GSMA and other organizations and how they could be taken to standardization in 3GPP and TISPAN, toward the timely creation of standards for the interconnection of next generation of IP services.

At TransNexus, we believe that the ETSI Open Settlement Protocol (OSP) has already defined a technical solution for the cascading settlement requirements defined by the GSMA. To learn more about how OSP can be used for NGN interconnect and settlement see: www.transnexus.com/White%20Papers/NGN_Settlement_using_OSP.pdf.

ITW brings the end of GTM

2008-06-03T16:00:00.002-04:00

I have spent the last two days at the International Telecoms Week (ITW) conference in Washington, DC. It has been a productive and well spent two days.

For wholesale carriers worldwide, this time of year has always been reserved for an annual trip to the GTM conference hosted by IntelSat in Washington DC. The GTM show had been very popular and growing for years. However, overtime, arrogance and over priced services began to permeate the IntelSat event. I do not think it was the fault of IntelSat, but rather the fault of the firm they hired to run the show for them.

Anyway, this year Capacity Magazine launched the competing ITW conference at the same time as GTM and located just one mile away on Connecticut Ave. in Washington. The success of the ITW show and the demise of the long running GTM conference is dramatic. The content of panel discussions I saw were meaningless, but it really does not matter. What the ITW participants want is a convenient venue for meeting partners and customers provided at a reasonable price. ITW provided that service. It is amazing how rapidly the market reacts when it is given a better alternative.

Like VON, I think the GTM conference is gone forever.

Atlanta Asterisk Users Group

2008-04-28T17:29:00.003-04:00

Last Saturday, April 26th, I had the pleasure of speaking to the Atlanta Asterisk Users Group (AAUG) at their annual Asterisk Installation Fest and Conference. The conference was held at the Georgia Tech Klaus Advanced Computing Building which is becoming a landmark for innovative computing activity in Atlanta. I was impressed with the vitality of the AAUG. They prepared an excellent program and had a strong showing of Asterisk users spending a beautiful spring day inside discussing various Asterisk topics.

Presenters included representatives from Pika and Sangoma. Mike Hunter, a Georgia Tech research scientist, briefed the audience on VoIP security work that is being done at the Georgia Tech Information Security Center. I followed Mike with a presentation on secure VoIP peering with Asterisk.

I recommend that any new or experienced Asterisk users in North Georgia to check the AAUG web site to learn more about this growing Asterisk community.

VoIP Peering Architectures & Strategies

2008-04-25T05:38:00.003-04:00

The link below is a video of the "VoIP Peering Architectures and Strategies" presentation I gave at the VoIP Peering Forum in December 2007.

VoIP Peering Architectures & Strategies - Video

VoIP Peering Architectures & Strategies - Powerpoint

Spring VON Presentations

2008-04-25T05:22:00.004-04:00

I have received several requests for copies of the presentations I made at Spring VON in San Jose. Here they are:

Secure Peering with Asterisk

OpenSER Summit - Commercial Business Models

Importance of Open Source Software

VoIP OSS and Billing

TransNexus & MetaSwitch

2008-04-25T05:08:00.004-04:00

I just returned from the MetaSwitch Customer Forum V in New Orleans. The MetaSwitch folks ran a great a conference for for their customers. A lot of good information and a good time. MetaSwitch has done a really great job of penetrating the small telco market. They really understand the phone business and know what telephone people want from a switch. More importantly, however, the MetaSwitch is a VoIP platform from the ground-up, not a TDM switch with a SIP interface

I have been too busy this year to announce on this blog that TransNexus is MetaSwitch partner. The TransNexus platform provides a really neat and clean least cost routing solution for the MetaSwitch. We are very happy to be associated with MetaSwitch - they have great operation.

TransNexus success story with MetaSwitch

Visions for Telecom Regulation

2008-01-15T15:47:00.000-05:00

The opening plenary at PTC'08 offered an interesting perspective from regulators about what they expect for the future. The plenary included regulatory officials from Germany, Japan, Singapore and an ITU representative.

To summarize, they all believe that the forces of network effects and economies of scale will drive more consolidation and that telecom regulation will be more important than ever in the future. What a vision - what else would you expect from regulators? In general, I heard the concern that network services are a commodity and there is declining incentive for carriers to build more infrastructure. Therefore, they argue, some regulatory policy must help shape an industry model that encourages infrastructure investment. Everyone agrees there has been limited network infrastructure investment in the last few years. But, this is healthy result of the bandwidth glut hangover from the telecom boom in the late 1990's, not a problem that needs to be fixed by regulators.

Fortunately, the plenary panel also included Paul Twomey, the CEO of ICANN, who was persausive in his counter arguments that the model of the future should be based on the Internet model. When the network is free from regulatory constriants, or the overbearing market power of a few carriers, innovation flourishes. He rightly noted that telecom operators narrowly think of convergence as the transport of multiple media over the same network. Internet entreprenuers, in contrast, see convergence beyond the network and all about taking money from other industries (i.e. advertising, retailing, travel).

Below are some of the interesting points from the PTC'08 opening plenary:
- There are 1.72 billion fixed telecom lines, 2.68 billion wireless subscribers and only about 250 million broadband connections.

- Singapore has a goal of 100% broadband penetration by 2015. To accomplish this public policy goal, the telecom industry will be divided into three parts:
1. Netco which provides dark fiber and is open to all network operators. The government will provide over $500 million USD to fund Netco.
2. Opcos which are competitive operators using Netco facilities.
3. Service Providers who sell Opco services.

In Japan:
- 36% of all broadband customers are served by fiber to the home.
- NTT has only 38% of the DSL market
- IP traffic grew by 40% in 2006
- 1% of the users consume 60% of the bandwidth.

My conclusions from the PTC'08 plenary are:
1) It was a excellent panel discussion with different perspectives and data from very qualified folks.
2) Bandwidth demand is outrunning supply and soon market forces will encouraging new infrastructure investment if the regulators resist the call from incumbent carriers to create regulatory barriers to entry.
3) The IP industry needs better cost accounting and billing mechanisms. IP infrastructure will be built if a return can be earned from the investment. Unfortunately, most IP network usage is based on a simple access charge for all you can use. The result is that a few heavy users are subsidized by the majority of low users. A vast amount of network value (return on investment) is left unrealized. If there is a role of regulators in the future, it should be focused on driving standards and practices that make the market for buying and selling of network interconnects and utilization more efficient.

Voice Peering Forum - New York

2007-12-05T20:33:00.000-05:00

I just wrapped up a full day at the Voice Peering Forum in New York City sponsored by The VPF (Voice Peering Fabric). There was a strong representation of service providers (AT&T, Verizon, XO, Sprint) and in a couple of sessions there was a lot of insightful discussion regarding the current and future state of VoIP peering. There is no doubt that VoIP peering is growing strongly, but it is still a small part of the total telecom interconnection market.

I think the good news for VoIP peering is that folks are starting to focus on business models for peering rather than the enabling technology. The focus on VoIP peering for the last three years has been on how to get it working for isolated interconnect cases. The initial VoIP peering concepts promoted a simple economic model of "bill and keep" (each VoIP peers bills their subscribers and keeps all the revenue without sharing with other networks involved with the call). The bill and keep model is attractive because it is simple, but it provides no economic incentive for interconnection or VoIP peering.

Now the discussion is beginning to focus on how to enable a common business model that gives all network operators an economic incentive to offer open interconnection with other VoIP networks. The general consensus is that the challenge is large and the following critical infrastructure components need to be defined for VoIP peering: route discovery, interoperability, accounting and settlement. The key question raised at the conference is when will the industry realize that VoIP interconnection is preferred over SS7 interconnection - when is the tipping point? Nobody knows the answer to this question, but for the first time I saw representatives from big incumbent carriers mulling over this problem and looking for a new market opportunity.

New Market VoIP Peering

2007-11-23T08:47:00.000-05:00

Next week, Wednesday through Friday, November 28-30 I will be at Pulver's New Market Peering conference in Orlando, FL. The conference is focused on many different issues related to VoIP peering, such as business models, technology choices, quality of services and security.

I willl speaking on a panel discussing Billing and OSS issues related to VoIP peering. I will be joined by Chris Celiberti and Etienne Pipers from Belgacom. Major issues I will be addressing are settlement billing, reconcilation and and the growing impact of local number portability.

I have a limited number of free passes for the conference, so contact me if you would like to attend. For more information about the conference go to http://www.newmarketpeering.com/2007/florida/web/

Asterisk Performance Test Results

2007-11-16T15:46:00.000-05:00

Our performance test results of Asterisk V1.4.11 configured as a SIP Back to Back User Agent (B2BUA) are now available.

For a summary of the results go to: http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm

To see all the details about the test plan, configuration files and test results go to: http://www.transnexus.com/White%20Papers/Asterisk_Performance_as_a_SIP_B2BUA.pdf

Report on Fall VON

2007-11-05T15:15:00.000-05:00

We found Fall VON in Boston to be busier than past VONs. There are more VoIP Service Providers, with more traffic, more problems to be solved and more revenue than ever. The prospects for years of growth ahead for the VoIP industry looks good.

Customer traffic where the TransNexus exhibit was located in the Digium Asterisk World was high. A lot of folks seem to enjoy complaining about the success of Digium and the company's shortcomings. But the simple truth is Asterisk and Digium continue to gain momentum as more service providers build their businesses on Asterisk and use Digium hardware. Attendance at the session on Directories, ENUM and DUNDi was good as interest in VoIP peering continues to grow. My presentation is available at http://www.transnexus.com/News%20and%20Events/2007/2007-10-30_Directories,_ENUM_and_DUNDI.ppt.

The OpenSER BoF was held on the final afternoon of the VON conference (the day after Halloween), and was very well attended with 50 OpenSER proponents. James Body, Research Director of TruPhone, gave a particularly interesting presentation about the success TruPhone has had with providing VoIP services to mobile phone users. Mr. Body gave a very strong testimonial for OpenSER which is the heart of the TruPhone SIP network. My presentation from the BoF is available at: http://www.transnexus.com/News%20and%20Events/2007/2007-11-01_OpenSER_Summit-TransNexus.ppt.

Another interesting VON attendee was Dean Elwood, the creator of www.voipuser.org. If you are not familiar with VoIP User, you should visit the site. It is an interesting site for VoIP developers and offers free UK local-rate telephone number (DID) to its members among other community benefits. Mr Elwood, an attorney, recently joined TruPhone but continues his after hours work with VoIP User. His next project under development is an open, secure VoIP peering network designed to hasten the vision of open peering among VoIP networks.

OpenSER BoF at Fall VON in Boston

2007-10-25T05:15:00.000-04:00

Next Thursday, November 1st, an OpenSER BoF (Birds of the Feather) meeting will be held to discuss ideas and issues regarding OpenSER. As part of the BoF, I will be joining the folks below to tell how OpenSER is being used in production VoIP networks.

Alan Crosswell, Senior Director - Network Infrastructure, Columbia University
Daniel-Constantin Mierla, Co-Founder OpenSER
James Body, Director - Networks, Truphone
Jim Dalton, CEO, TransNexus
Norman Brandinger, President, GOES.com
Xavier Casajuana, CEO, VozTelecom

The BoF meeting will be held at the VON conference in the Boston Convention Center on Thursday from 12:45 -2:00 pm. The meeting and admission to VON on Thursday is free to anyone who registers. Go to www.fallvon.com/register and use the priority code OPENSER to get a free one day Conference Pass for Thursday, November 1st.

VoIP Peering Panel at VON in Boston

2007-10-24T08:26:00.000-04:00

Next Tuesday (Oct 30th) at the VON conference in Boston, I will be joining Ed Guy from Truphone and Eli Katz from XConnect on a panel to discuss ”Directories, ENUM and DUNDI”. Both of these gentlemen are VoIP Peering experts. Ed was the CTO for Free World Dialup and Eli is the founder and CEO of XConnect, a rapiding growing VoIP peering provider.

Our discussion will focus on different VoIP peering models, enabling technologies, trends and visions for VoIP peering in the future. Directories and ENUM technologies follow the classic model of a central database that provides route discovery for all users. It is a simple model based on sound economic principles - a central database offers economies of scale and positive externalities that benefit everyone. Nevertheless, there is a lot of market resistance to the centralized model - everyone is suspicious of the power and control ceded to a central routing authority.

DUNDi is a completely different model with decentralized route discovery. However, DUNDi offers the same positive externalities offered by a centralized database. By positive externalities, I mean that the benefit of using DUNDi grows as more endpoints join the DUNDi network. DUNDi, created by Mark Spencer, is a clever way to remove the objections to a central routing authority while keeping the benefits.

Unfortunately, neither ENUM nor DUNDi are rapidly gaining acceptance in the market place. The reason is because these technologies only address route discovery. A VoIP peering model requires more than just route discovery, VoIP peering also requires access control and economic settlement. Today these VoIP Peering functions are fulfilled by Session Border Controllers that are architectually identical to a class 4 switch in a PSTN inter-exchange network. This technique does not offer the benefits of ubiquitous direct peering between service providers. So part of our discussion will cover what is required to fulfill the VoIP vision for direct peering between any two VoIP networks globally.

I hope you will join us at Tuesday, October 30th, from 3:30pm – 4:30pm, in Room 209 at the VON conference in the Boston Convention Center.

VoIP Peering at ISPCON

2007-10-15T08:15:00.000-04:00

The Internet industry will be meeting this week at ISPCON in San Jose. ISPCON has always been a show for ISPs and NetHeads and more recently it has expanded its focus to include non-data applications like voice. VoIP is a natural business extension for ISPs and I will be at ISPCON tomorrow giving a tutorial on the business case for how ISPCONs can leverage their current business to provide VoIP service.

The ISPCON session is:

Track: TECHNOLOGY 10:15 AM - 11:15 AM Tuesday Oct. 16, 2007
E2: SIP Peering: A VoIP Business Model that Works for ISPs Serving Enterprise Customers.
This session will describe how the SIP peering business model is a natural service extension for ISPs providing bandwidth to Enterprise customers. The presentation will cover the following points:
- Who is the target market
- Selling proposition
- Business model and economics
- Incremental infrastructure required
- Example network design
- Understanding least-cost routing of telecom traffic
This presentation will provide ISPs the information they need to determine if they want to expand their business to provide basic VoIP services.

ISNs, an innovative approach to VoIP Peering

2007-10-05T09:32:00.000-04:00

What's an ISN?

An ISN is an ITAD Server Number.

What the heck is an ITAD?

An ITAD is an Internet Telephony Administrative Domain.

If you are in the business of developing future VoIP products you need to visit http://freenum.org/ and include the ITAD concept into your long term strategic thinking. It could be just another geek idea that never flys, but it could also be another one of those unusual open source type concepts that take-off virally when the market enablers are in place.

The concept is simple - create an open numbering system for VoIP that is available to everyone, not just telephone companies.

A billion plus people in the world make telephone calls using telephone numbers provided exclusively by telephone carriers. ITADs are different. They are numbers that anyone can get from IANA (Internet Assigned Number Authority). An ISN is a subscriber number followed by a * followed by the ITAD for the subscriber.

For example, the TransNexus ITAD is 484. You can think of 484 as the domain name for TransNexus VoIP calls - just like transnexus.com is the domain name for e-mail messages sent to TransNexus. The ISN for phone extension 101 at TransNexus is 101*484.

A key feature of ISNs is that they are easy to dial on a plain old 12 digit telephone key pad. This is not the case with a SIP uri which could require a full alphanumeric keyboard.

At this point ITADs are still just an interesting idea being nurtured in a live trial by John Todd, a clever VoIP innovator. Nevertheless, it is concept that forward thinking product managers and strategic planners need to keep in mind as they look for new opportunities to out-flank their competition. Visit freenum.org for more information and to join the trial.

TransNexus at Comptel in Dallas

2007-10-05T08:46:00.000-04:00

TransNexus will be exhibiting at the Comptel Plus Fall 2007 Convention and Expo next week in Dallas. The exhibit floor is open on Sunday - Tuesday October 7th-9th. If you plan to attend the Comptel show, please stop by booth 904 to visit TransNexus.

When you visit TransNexus, be sure to sign-up for a free Least Cost Routing Analysis of your telecom traffic. TransNexus will analyze your vendor rate plans and Call Detail Records to calculate your telecom vendor fees with profit optimizing Least Cost Routing.

The TransNexus analysis will give you the hard data you need to maximize the value of your vendor contracts:
- Do you need to fine tune your Least Cost Routing Algorithm?
- Will adding more telecom vendors save you money?
- Should you add Least Cost Routing for intra-state calls?
- Should you add Least Cost Routing for Expanded Local Calling Areas?
- Can you generate new revenues from wholesale VoIP peering with other service providers?

I hope to see you at the Comptel show.

The VoIP Peering Puzzle

2007-09-19T08:31:00.000-04:00

TransNexus has been featured in this week's edition of "The VoIP Peering Puzzle" - a comprehensive set of articles examining VoIP Session Border Controllers. "The VoIP Peering Puzzle" is a series of articles is written by Mark Miller, President of DigiNet Corporation. Miller, who is a Professional Engineer, is a good writer on technical topics and his articles provide a quick, easy to read understanding of technical issues.

To the see the article on how TransNexus fits into a VoIP peering network architecture, go to http://www.voipplanet.com/backgrounders/article.php/3700166

Asterisk Performance Testing

2007-09-18T08:48:00.000-04:00

Over the last couple of months we have been stress testing Asterisk V1.4 to understand its performance. We started using V1.4.4 and advanced to V1.4.11 as new releases were made available. We have been stress testing Asterisk configured as a SIP Back to Back User Agent (B2BUA) installed on a server with two, dual core, Xeon 5140 CPUs running at 2.33 GHz.

The performance results look very good. When there is no transcoding, Asterisk can handle approximately 1500 simultaneous calls. When using a $3000 server to host Asterisk, this works out to be $2 per port which is a very low investment. When calls are transcoded from the G.711 codec to G.729, the maximum number of simultaneous calls is approximately 400. The economics are quite different when transcoding is required since there is a $10 per port patent royalty for using the G.711 to G.729 transcoding algorithm. The per port cost for using Asterisk with G.711 to G.729 transcoding is $17.50 per port. (Note, the $10 per port patent royalty is the amount charged if you purchase a transcoding license from Digium. OEM vendors could negotiate a lower royalty fee directly from the patent holders, but this is not practical for end users.)

More importantly, we have seen a big improvement is stability between V1.4.4 and V1.4.11. We have found that V1.4.11 has been very stable in prolonged, high volume stress tests. We expect that V1.4.11 will be the foundation of Asterisk Business Edition Rev C which should be released later this year. We see Asterisk Business Edition Rev C to be a quantum leap forward compared to earlier versions of Asterisk Business Edition.

We will publish all the details of our test plan, a summary of the test results and all the raw data on the TransNexus web site (www.TransNexus.com) in the coming weeks.

If you have any comments or experience sizing server hardware for Asterisk in carrier operations please post your information here.

Thank you

SPIDER Adoption is an important trend for VoIP Peering

2007-09-09T12:09:00.000-04:00

On August 6th 2007, NueStar (www.NeuStar.biz) issued a press release announcing that they have partnered with NetNumber (www.netnumber.com) "in response to customer demand for interoperability between NeuStar’s SIP-IX service and NetNumber’s SPIDER and TITAN technology platforms."

This announcement is significant because it is the first coordination between two companies on a common approach for VoIP peering. Up until now, the apparent strategy for most players in the VoIP peering market has been to build a dominant database of peering partners before anyone else and become the de facto VoIP peering registry monopoly. Fortunately, this has not happened for a variety of reasons such as competition, immature technology, lack of demand and, most importantly, business models that do not work.

A clear message from the market regarding VoIP peering is that customers do not want a monopoly VoIP peering registrar even if it brings simplicity and economies of scale. A market driven business model for VoIP peering will not be similar to the monopoly status given to telecom databases such as the Telcordia (www.telcordia.com) LERG (Local Exchange Routing Guide) or the NPAC (Number Portability Adminstration Center) run by NeuStar.

While service providers might loathe dependence on a centralized authority to coordinate global VoIP peering, at some level a root authority for route discovery seems impossible to avoid. This is why SPIDER and coordination between NeuStar and NetNumber is interesting. SPIDER stands for Service Provider Interconnect Data Exchange Resource and "is used by registry solution providers to enable the efficient exchange of interconnect address information between trusted communications service providers and VoIP communities" (quote from press release). Conceptually, SPIDER (spiderregistry.net) is a neutral, non-profit organization created by NetNumber and Arbinet (www.arbinet.com). I use the word conceptually, because up until now, SPIDER has appeared to be a marketing tool for Arbinet and NetNumber. The participation of NeuStar might be the first step in the evolution of SPIDER becoming a truely independent mechanism for enabling ubiquitous VoIP peering.

If you have used SPIDER or have an opinion about SPIDER I welcome your comments.

VoIP Peering Panel at Internet Telephony Conference

2007-08-22T08:56:00.000-04:00

I am looking forward to participating on a panel discussion about the State of VoIP Peering at the Internet Telephony Conference in Los Angeles on September 10th. At TransNexus, we believe the increasing success of bilateral VoIP Peering is the foundation of the industry model that will eventually replace the open interconnection model of the PSTN SS7 network.

Intercarrier VoIP Clearing and Settlement is a service TransNexus provided in 1998-2001, but the business evaporated when most of our ITSP customers went bankrupt in the the telecom meltdown of 2001. The concept of open multi-lateral interconnect among all VoIP networks, based on secure call clearing and settlement seems to be an inevitable business model, but it is still a long way off. The Peering panel at IT EXPO will be a good place to hear different opinions about the future of interconnection between VoIP carriers.

For more information about the State of VoIP Peering Panel see: http://www.tmcnet.com/voip/conference/west-07/w-07-conferences.aspx?t=SP#SP-05

Conversation with Rich Tehrani

2007-08-15T04:45:00.000-04:00

Rich Tehrani, Editor-in-Chief of TMCnet, recently asked me a few questions regarding the VoIP industry. I gave him my on the spot, candid answers and suffered the wrath of my diligent VP of Marketing for missing a promotion opportunity. You can see the questions and answers at:
http://internetcommunications.tmcnet.com/topics/enterprise/articles/9535-ip-communications-focus-transnexus.htm

NexSRS Peering Server gets press

2007-07-18T04:25:00.000-04:00

We have been quietly working hard this Summer preparing for new releases of NexOSS and NexSRS. Our focused effort, plus some vacation, explains my recent silience on this blog.

Nevertheless, we were very pleased to see the recent article covering our NexSRS V3.0 Peering Server and VoIP Service Control Point in "Service Provider Weekly" magazine. Go to www.spweekly.com/issue929.pdf and turn to page 7 for an informative article on the NexSRS server.

NexSRS V3.1 will be available in September and will offer important new features such as call burst control for inbound and outbound traffic, better error code reporting for faster trouble shooting and network analysis and perhaps the addition of a web services interface.

NexOSS 3.3, in beta testing since early June, will also be available in September and will include improved reporting, invoicing summaries, reconciliation reports to simplify auditing of bills from VoIP terminators and improved rate plan management.

Asterisk Developers Conference

2007-05-30T15:23:00.000-04:00

Last week the Asterisk Developers Conference, or AstriDevCon 2007, was held at the Georgia Institute of Technology Information Security Center in Atlanta, Georgia. Since a couple of TransNexus developers were attending, I visited the conference as a non-technical observer to see what was going on. I have to say that I was impressed with what I saw. Kevin Fleming has cultivated a productive relationship with Georgia Tech which has a long history of nurturing innovative new technologies and start-ups and Asterisk is a perfect fit with the Georgia Tech environment.

I was also impressed with the number and background of attendees. AstriDevCon 2007 was not a convention of beer drinking hackers (however, I think they would have been welcome) which some might suspect from an open source meeting. The attendees included developers from 31 different companies including heavy hitters IBM and Sun Microsystems. The topics discussed included improved security and new video features for Asterisk. There is no doubt that Asterisk will continue its robust growth and development. Asterisk version 1.6 promises to provide another great stride forward. If you are in the telecom business and not seriously considering the impact of Asterisk in your strategic marketing plans, then the assumptions in your long term plans are probably very wrong.

See you at GTM 2007 Next Week

2007-05-18T08:25:00.000-04:00

Intelsat's annual Global Telecommunications Meeting (GTM 2007) will occur next week in Washington, D.C. GTM is like the UN of telecommunications. Every major international carrier and perhaps hundreds of smaller international carriers meet to negotiate bilateral interconnect agreements. In the old days before telecom deregulation, GTM was a small event for PTTs from all parts of the world. With deregulation, the number of carriers attending has increased dramatically.

Unlike technology conferences which try to be cool or geeky, GTM is a telecom conference for old time PTT and Bellheads - every one is dressed in suits. While it may appear formal, GTM is a great conference hosted by Intelsat and the entertainment are always first class. Like the PTC conference in Honolulu, GTM is a great networking event

TransNexus has attended GTM for years to meet with our customers and friends. If you plan to attend GTM and would like to meet, please send me an e-mail at jim.dalton@transnexus.com.

OSP Peering Server V3.0 Released

2007-05-02T04:43:00.000-04:00

NexSRS OSP Peering Server V3.0 is now available for download from the TransNexus website.

New Features Include:
- Simpler route provisioning
- Support for trunk group routing and device partitioning based on called number prefix
- Better CDR reporting for stateless SIP proxies such as OpenSER and SER
- Improved integration with Cisco IP-IP VoIP gateways

For more information, please see http://www.transnexus.com/Products/NexSRS_VoIP_Peering_Server.htm

OpenSER vs SER Performance Test

2007-04-27T05:25:00.000-04:00

We often hear the question "Which is better OpenSER or SIP Express Router?". Well, now we know because we have completed a thorough and carefully controlled benchmark test of both SIP proxies. For an overview of our test, please see my blog from April 18th (OpenSER Performance Test). The complete test plan and results will be posted on www.transnexus.com soon.

The comparison results for 200 calls per second are:
.............................................CPU .........................Call
.........................................Utilization.............. Completion
OpenSER V1.1 .....................92.9% ......................97.17%
OpenSER V1.2 ....................83.3% .....................100.00%
SER 2.0 ..............................83.1% .....................100.00%

The comparison results for 220 calls per second are:
.............................................CPU .........................Call
........................................Utilization ...............Completion
OpenSER V1.1 .......................NA ...........................NA
OpenSER V1.2 ...................90.5% ......................99.79%
SER 2.0 .............................90.7% ......................93.43%

Conclusions:
- OpenSER V1.2 is significantly better than OpenSER V1.1 in terms of performance, call completion and post dial delay.

- OpenSER V1.2 and SER 2.o are identical in performance, but OpenSER V1.2 performed more reliably as it approached the limit of CPU utilization.

OpenSER V1.2 is 13.4% faster than V1.1

2007-04-19T05:19:00.000-04:00

This post is follow-up on my post yesterday about our performance testing of OpenSER V1.1, so please refer to that post for details on the test setup and test scenario.

We have completed our performance testing of OpenSER V1.2. The comparison results are:
V1.1 - 200 calls per second - 92.9% CPU Utilization - 97.17% call completion
V1.2 - 220 calls per second - 90.9% CPU Utilization - 99.79% call completion

OpenSER V1.2 performance is 13.4% better than V1.1 and most importantly its call completion rate under a heavy load is excellent.

These results indicate that OpenSER V1.2 running on server with two Xeon, dual core CPUs could conservatively handle a 600 million minutes per month of VoIP traffic.

OpenSER Performance Benchmark

2007-04-18T09:41:00.000-04:00

We continue to see a growing number of service providers using SIP Express Router (also known as SER) and OpenSER as infrastructure SIP proxies for their carrier operations. SER (www.iptel.org) and OpenSER (www.openser.org) are mature open source projects for a very stable, high performance SIP proxy. SER was the original project and OpenSER is fork of SER started a couple of years ago. Both projects are excellent and OpenSER has growing popularity because it more rapidly introduces new features.

We often hear the question, "What is the capacity of OpenSER?" There have been benchmark tests that show OpenSER and SER can manage thousands of SIP calls per second. But how about a benchmark test that simulates a real production environment with call failures, retries and call detail reporting? Since we have not seen such a test report, we decided to perform our own benchmark test. For our test we configured OpenSER V1.1 on a Dell Precision 490 server with two Intel Xeon 5140 2.33 GHz CPUs and 4 GB of RAM. To reduce the traffic load requirements, we disabled one CPU and one core on the remaining CPU. We ran the test on a single core of a dual core chip. The operating system was CentOS 4.4.

To simulate a production environment, we configured OpenSER to communicate with an OSP server for call routing and CDR collection. For each call there were five possible destinations returned in random order. Four of the five destinations were configured to simulate call setup failures (no TCP response, no SIP response, call rejected and no route). We found that 200 calls per second was the maximum the single core processer could handle. If we had used all four CPU cores we expect the results would have been 800 calls per second. To be conservative, we would recommend service providers to plan on maximum CPU utilization of about 60%. This would establish the OpenSER planning gauideline of 500 calls per second on a server with two, dual core Xeon CPUs.

If you assume 15% of a service provider's traffic occurs during the busy hour, a 50% Answer Seizure Ratio (ASR) and a 3 minute average call duration, then 500 calls per second equates to 540 million minutes of VoIP traffic per month! We think this is impressive for an open source SIP proxy running on a server with a retail price of $2,967. If you want a copy of the test plan and results, send an e-mail to me at jim.dalton@transnexus.com.

We will run the same benchmark test on OpenSER V1.2 and then on SER to get a clear comparison of the different releases. When we are done, we will publish the results on www.transnexus.com.

Secure VoIP Peering

2007-04-05T07:52:00.000-04:00

At the Spring VON show in San Jose, CA I had the opportunity to give a presentation on Secure VoIP Peering. The presentation is an overview tutorial on how standard Public Key Infrastructure (PKI) services and the OSP protocol are being used today to enable secure peer to peer VoIP calls.

For those who do not know, OSP is a protocol standard defined by ETSI, the European Telecommunications Standards Institute. ETSI is best known as the standards body that created the global mobile phone standard known as GSM. OSP is a protocol used by a softswitch or session border controller to query and external route server for call routing and then also report call detail records when the call is finished. OSP is a set of data rich XML meesages transported over HTTP so it is a natural fit with SIP and web based services. Further, OSP is designed for routing, authorization, pricing and usage reporting for any IP service, not just VoIP.

To view the Flash version of the Secure VoIP Peering presentation, go to http://www.transnexus.com/Flash/2007_Spring_VON_-_Secure_VoIP_Peering/player.html

VoIP Peering Panel at Spring VON

2007-04-04T11:00:00.000-04:00

Spring VON in San Jose was a very good show - the VoIP market is looking very positive. It has taken me two weeks to follow-up on actions items from customer prospects and get back to routine matters such as blogging.

One of the VON sessions I found most interesting was the panel discussion on IP Voice Peering. The panelists came from a good cross section of companies (iBasis, NeuStar, Ag Projects, XConnect and Arbinet) and each had there own unique perspective on peering. Some of the interesting points from the discussion were:

1. Public ENUM is dead. Many in the industry have pushed for the idea that all telephone numbers with SIP access should be included in a public ENUM directory to enable peer to peer calling. This is an appealing idea for VoIP users because it would eliminate telephone companies and enable global peer to peer VoIP calling free of telecom access charges. While the public ENUM concept makes sense, there are too many political and market sources determined to keep it from happening. Perhaps the panel agreed Public ENUM is dead because it is a threat to their business.

2. The SPIDER Registry. No one wants to share their numbers. While everyone in the VoIP world complains that telcos are "Walled Gardens", or closed networks, the reality is that VoIP service providers are just as proprietary with their networks and customers. VoIP service providers do not want to share the telehone numbers of their customers with any third party. So how to you get broader VoIP peering adoptance if VoIP service providers will not disclose their telephone numbers? One well thought out solution appears to be the non-profit SPIDER registery created by Arbinet. VoIP service providers can list their numbers in the SPIDER registry and still maintain control over who can access those numbers. The SPIDER registry acts as a central provisioning mechanism for making VoIP numbers available to selected peering partners. The idea is good, but trust is still an issue. As expected, all the panelists were SPIDER sceptics since it has been created by Arbinet.

3. VoIP peering depends on the "call context". This an interesting new term that means call routing, or peering, depends on who is calling who and when. Ah-ha, so VoIP peering is not just as simple as connecting to VoIP users. VoIP peering depends on a variety of issues such as competition between peers, accounting and settlement, and technical interoperability. The panel agreed that one of the great drawbacks of ENUM is that it cannot convey "call context". ENUM only conveys the called number. To determine peering partners for a call, the calling and called numbers, source IP address, VoIP protocol, service type and many other factors must be known. As the VoIP peering market develops, we expect implementations of the OSP supported VoIP peering will grow since this well established ETSI protocol already addresses all of these "call context" requirements.

The Missoula Plan - Good Grief

2007-03-02T06:10:00.000-05:00

We have just wrapped up three days exhibiting at the Comptel show in Las Vegas. I am happy to report that business activity at this show was better than ever. One of the more informative sessions at the Comptel Convention was a debate on Intercarrier Compensation and the Missoula Plan. The Missoula Plan is a proposal to the FCC by Rural ILECs for intercarrier compensation. The plan also has some support from large carriers - notably AT&T. The Plan is too long and complex to describe here so I will get straight to the important points in the intercarrier compensation (IC) debate:

- The public policy driving IC is the fact that everyone benefits when even the most remote parts of rural America have telecom access.
- IC is an arcane federal tax used to transfer revenue from urban telecos to rural telcos.
- Folks who argue that the Missoula Plan will provide rural ILECs with the funds they need to build broadband to the farm house are dreaming. Innovative new network services for rural America will be best delivered by entreprenuerial companies, not the aging management of rural telcos with obsolete technical skills.
- The Missoula Plan is too complicated to work and adds big ineffciencies to the telecom business. It needs to be killed.

The Alternative Plan
- Market driven IC settlement rates (not IC rates dreamed up by a lawyer) are an economic necessity for the efficient allocation of network costs.
- Eliminate the requirement that all telco networks must accept all traffic routed to them. Yes, this is telecom heresy. But the world has changed in the last 100 years. Market forces will motivate carriers to carry any and all telephone calls. This is the Internet model and it works. Would you choose an ISP that could only reach 95% of the web sites on earth? Of course not, the same applies for telephone service.
-Proponents of the Missoula Plan argue that the plan will lower IC to rural ILECs from 7 cents per minute (in some cases) to just 1.75 cents per minute. This is a mistake. If the real cost is 7 cents per minute, the rural ILEC should be charging calling parties 7 cents per minute to call the remote hunting lodge.
- Phantom traffic (unbillable intercarrier traffic) is the real economic inefficiency that needs to be addressed. This is the one point of the Missoula Plan that most carriers agree with.

The Missoula Plan is a bad plan and will fail. The entire IC regulatory regime is being undermined by market forces. The FCC needs to keep its focus on creating new inter-modal competition for telecom. Some rural ILECs are going to go bust, but market forces will drive new capital, firms and technology to fill the void - if the FCC lets them.

Cisco Uses OpenSER

2007-02-23T07:56:00.000-05:00

Yesterday, Daniel-Constantin Mierla reported on the OpenSER mailing list that OpenSER is being used in the Cisco® Service Node for the Linksys® One family of products. Greg Fausak of Cisco confirmed this report and posted the following:

> I guess the cat's out of the bag! We've been using OpenSER for quite some time. I'm very happy with it's performance and flexibility. New features are introduced quite quickly without creating a big unstable mess. The performance is very good, we are able to route and account for calls predictably.
>
> One of my favorite things to do around here is to watch the traffic labs stress the call routing implementation. We are able to sustain 60 call per second and burst to 200.

Wow, that is pretty strong endorsement for OpenSER! We were first introduced to SER (SIP Express Router) two years ago by our carrier customers. As I wrote last week about Asterisk, open source software is having a major impact on telecom carriers. If you are in the telecom business and you do not include open source software as part of your implementation strategy, then you are competing with a serious cost disadvantage. While SER and OpenSER do not have the brand reognition Asterisk has, they are both widely deployed with carriers. The number of serious carriers using SER and OpenSER is remarkable, but few will admit to it in public.

It is nice to see Cisco be transparent about their use of open source. A full description of the Cisco Linksys Service Node is available at http://www.cisco.com/en/US/products/ps7194/products_data_sheet0900aecd805c3cc1.html

TransNexus in Asterisk Pavilion at Spring VON

2007-02-16T08:10:00.000-05:00

For me Asterisk is one of the most interesting and exciting developments in the world of VoIP. It has been fascinating to watch Mark Spencer grow Asterisk from a little known open source project into a market force that is changing everything in telecom. Sure, the naysayers always tell you Asterisk won't scale, its not stable, the SIP stack is lousy, and on and on ..... They reality, however, is that Asterisk is getting better and better all the time. OK, Asterisk is not carrier grade now, but it will be - probably sooner that anyone expects. If you are in the telecom business and think Asterisk is a gimic that will not impact your business, you are either a dinosaur or a strategic planner working for an ILEC.

We first started working with Asterisk three years ago after hearing about Asterisk from customers. Since that time, I have been amazed by the number of Asterisk deployments. Just about every customer we have is using Asterisk or evaluating Asterisk. Only Cisco routers and Intel chips have greater penetration in the telecom marketplace. At Fall VON in 2006, a long time friend who is a product manager with a major VoIP equipment provider told me that they ship 30,000 SIP phones or ATAs to Asterisk users each month. 30,000 is a big number from a firm that is not an Asterisk partner and has a PBX solution that competes directly with Asterisk!

From my oberservations, Mark Spencer is a pretty shrewd businessman. Now that he has reorganized Digium and added outside capital, the market impact Asterisk is having will accelerate. We first exhibited with Digium in the Asterisk Pavilion at Spring VON 2005 and we are excitied about being part of the Asterisk Pavilion again this year at Spring VON 2007 in San Jose.

OSP Toolkit now on SourceForge

2007-02-13T05:24:00.000-05:00

I am pleased to post that TransNexus has completed the transition of moving the OSP Toolkit project from SIPfoundry to SourceForge (http://sourceforge.net/projects/osp-toolkit). We appreciate our relationship with SIPfoundry and value the good work they do promoting SIP. However, we decided to move the OSP Toolkit project to SourceForge since OSP serves an audience broader than SIP users. OSP has been deployed in global wholesale H323 networks since 2000 and the number of SIP networks using OSP exceeded H323 networks in 2006. But OSP is also useful with other protocols such as IAX, jabber and any peer to peer IP communication application that requires secure authorization and accounting. So we decided to move the OSP Toolkit to SourceForge where we also host the RAMS OSP test server and the GKTMP-OSP interface.