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Archive for the ‘VoIP’ Category

Upcoming WebRTC World Conference

Thursday, October 24th, 2013

Last month, WebRTC was included in Firefox for Android. It has been available for a while in Chrome, Mozilla, and Opera browsers. Justin Uberti of Google claims that this adds up to a billion devices running WebRTC.

To get up to speed on the challenges and opportunities of WebRTC, and the future of real-time voice/video communications, Wirevolution’s Charlie Gold will be attending the WebRTC World Conference next month and writing about it here. The conference runs Nov 19-21 at the Convention Center in Santa Clara, CA.

At the conference we hope to make sense of the wide range of applications integrating WebRTC, and to relate them to integration opportunities for service providers. The applications range from standard video enabled apps such as webcasting and security, to enterprise applications such as CRM and contact centers, to emerging opportunities such as virtual reality and gaming. Service providers can combine webRTC with IMS and RCS, and can use it to manage network capabilities and end users’ quality of experience.

The conference is organized around four tracks: developer workshops, B2C, enterprise, and service providers.

  • The developer workshop topics include concepts and structures of WebRTC, implementation and build options, standardization efforts, signaling options, applications to the internet of things (IoT), codec evolution, monitoring and alarms, firewall traversal with STUN/TURN/ICE, and large scale simultaneous delivery in applications such as webcasting, gaming and virtual reality (VR) and security.
  • Business and consumer applications sessions cover successful deployment strategies of use cases like collaboration and conferencing, call centers and the Internet of Things (IoT). Other sessions on this track cover security, device requirements and regulatory issues.
  • Service provider workshops include IMS value in a world of WebRTC, how to use WebRTC, deployment strategies, how to extend existing services and offer new services using WebRTC, using WebRTC to acquire new users, and understanding the network impact of WebRTC.
  • The enterprise track has additional sessions on integrating WebRTC into your contact center and websites (public, supplier, internal). These sessions cover details like mapping out your integration strategy between WebRTC and SIP, using a media server vs. direct media interoperation; and how to deploy a WebRTC portal.

Keynotes will be from Ericsson, Alcatel-Lucent, Mozilla, Genband, Mavenir, Radisys, CafeX and presumably others.

To round it out, there will be a plethora of special workshops, realtime demos, panels and round tables.

With the momentum of WebRTC growing in leaps and bounds, we are looking forward to attending and sharing more on webRTC next month.

Clearing the Cloud for Reliable, Crystal-Clear VoIP Services

Tuesday, June 25th, 2013

The compelling advantage of VoIP is that it is far cheaper than circuit switched technology. But VoIP calls often sound horrible. It doesn’t have to be this way. Although VoIP is intrinsically prone to jitter, delay and packet loss, good system design can mitigate all these impairments. The simplest solution is over-provisioning bandwidth.

The lowest bandwidth leg of a VoIP call, where the danger of delayed or lost packets is the greatest, is usually the ‘last mile’ WAN connection from the ISP to the customer premises. This is also where bandwidth is most expensive.

On this last leg, you tend to get what you pay for. Cheap connections are unreliable. Since businesses live or die with their phone service, they are motivated to pay top dollar for a Service Level Agreement specifying “five nines” reliability. But there’s more than one way to skin a cat. Modern network architectures achieve high levels of reliability through redundant low-cost, less reliable systems. For example, to achieve 99.999% aggregate reliability, you could combine two independent systems (two ISPs) each with 99.7% reliability, three each with 97.8% reliability, or four each with 94% reliability. In other words, if your goal is 5 minutes or less of system down-time per year, with two ISPs you could tolerate 4 minutes of down-time per ISP per day. With 3 ISPs, you could tolerate 30 minutes of down-time per ISP per day.

Here’s a guest post from Dr. Cahit Jay Akin of Mushroom Networks, describing how to do this:

Clearing the Cloud for Reliable, Crystal-Clear VoIP Services

More companies are interested in cloud-based VoIP services, but concerns about performance hold them back. Now there are technologies that can help.

There’s no question that hosted, cloud-based Voice over IP (VoIP) and IP-PBX technologies are gaining traction, largely because they reduce costs for equipment, lines, manpower, and maintenance. But there are stumbling blocks – namely around reliability, quality and weak or non-existent failover capabilities – that are keeping businesses from fully committing.

Fortunately, there are new and emerging technologies that can optimize performance without the need for costly upgrades to premium Internet services. These technologies also protect VoIP services from jitter, latency caused by slow network links, and other common unpredictable behaviors of IP networks that impact VoIP performance. For example, Broadband Bonding, a technique that bonds various Internet lines into a single connection, boosts connectivity speeds and improves management of the latency within an IP tunnel. Using such multiple links, advanced algorithms can closely monitor WAN links and make intelligent decisions about each packet of traffic to ensure nothing is ever late or lost during communication.

VoIP Gains Market Share

The global VoIP services market, including residential and business VoIP services, totaled $63 billion in 2012, up 9% from 2011, according to market research firm Infonetics. Infonetics predicts that the combined business and residential VoIP services market will grow to $82.7 billion in 2017. While the residential segment makes up the majority of VoIP services revenue, the fastest-growing segment is hosted VoIP and Unified Communications (UC) services for businesses. Managed IP-PBX services, which focus on dedicated enterprise systems, remain the largest business VoIP services segment.

According to Harbor Ridge Capital LLC, which did an overview of trends and mergers & acquisitions activity of the VoIP market in early 2012, there are a number of reasons for VoIP’s growth. Among them: the reduction in capital investments and the flexibility hosted VoIP provides, enabling businesses to scale up or down their VoIP services as needed. Harbor Ridge also points out a number of challenges, among them the need to improve the quality of service and meet customer expectations for reliability and ease of use.

But VolP Isn’t Always Reliable

No business can really afford a dropped call or a garbled message left in voicemail. But these mishaps do occur when using pure hosted VoIP services, largely because they are reliant on the performance of the IP tunnel through which the communications must travel. IP tunnels are inevitably congested and routing is unpredictable, two factors that contribute to jitter, delay and lost packets, which degrade the quality of the call. Of course, if an IP link goes down, the call is dropped.

Hosted, cloud-based VoIP services offer little in the way of traffic prioritization, so data and voice fight it out for Internet bandwidth. And there’s little monitoring available. IP-PBX servers placed in data centers or at the company’s headquarters can help by providing some protection over pure hosted VoIP services. They offer multiple WAN interfaces that let businesses add additional, albeit costly, links to serve as backups if one fails. Businesses can also take advantage of the various functions that an IP-PBX system offers, such as unlimited extensions and voice mail boxes, caller ID customizing, conferencing, interactive voice response and more. But IP-PBXes are still reliant on the WAN performance and offer limited monitoring features. Thus, users and system administrators might not even know about an outage until they can’t make or receive calls. Some hosted VoIP services include a hosted IP-PBX, which typically include back-up and storage and failover functions, as well as limited monitoring.

Boosting Performance through Bonding and Armor

Mushroom Networks has developed several technologies designed to improve the performance, reliability and intelligence of a range of Internet connection applications, including VoIP services. The San Diego, Calif., company’s WAN virtualization solution leverages virtual leased lines (VLLs) and its patented Broadband Bonding, a technique that melds various numbers of Internet lines into a single connection. WAN virtualization is a software-based technology that uncouples operating systems and applications from the physical hardware, so infrastructure can be consolidated and application and communications resources can be pooled within virtualized environments. WAN virtualization adds intelligence and management so network managers can dynamically build a simpler, higher-performing IP pipe out of real WAN resources, including existing private WANs and various Internet WAN links like DSL, cable, fiber, wireless and others. The solution is delivered via the Truffle appliance, a packet level load balancing router with WAN aggregation and Internet failover technology.

Using patented Broadband Bonding techniques, Truffle bonds various numbers of Internet lines into a single connection to ensure voice applications are clear, consistent and redundant. This provides faster connectivity via the sum of all the line speeds as well as intelligent management of the latency within the tunnel. Broadband Bonding is a cost effective solution for even global firms that have hundreds of branch offices scattered around the world because it can be used with existing infrastructures, enabling disparate offices to have the same level of connectivity as the headquarters without the outlay of too much capital. The end result is a faster connection with multiple built-in redundancies that can automatically shield negative network events and outages from the applications such as VoIP. Broadband Bonding also combines the best attributes of the various connections, boosting speeds and reliability.

Mushroom Networks’ newest technology, Application Armor, shields VoIP services from the negative effects of IP jitter, latency, packet drops, link disconnects and other issues. This technology relies on a research field known as Network Calculus, that models and optimizes communication resources. Through decision algorithms, Application Armor monitors traffic and refines routing in the aggregated, bonded pipe by enforcing application-specific goals, whether it’s throughput or reduced latency.

VoIP at Broker Houlihan Lawrence – Big Savings and Performance

New York area broker Houlihan Lawrence – the nation’s 15th largest independent realtor – has cut its telecommunications bill by nearly 75 percent by deploying Mushroom Networks’ Truffle appliances in its branch offices. The agency began using Truffle shortly after Superstorm Sandy took out the company’s slow and costly MPLS communications network when it landed ashore near Atlantic City, New Jersey last year. After the initial deployment to support mission-critical data applications including customer relationship management and email, Houlihan Lawrence deployed a state-of-the-art VOIP system and runs voice communications through Mushroom Networks’ solution. The ability to diversify connections across multiple providers and multiple paths assures automated failover in the event a connection goes down, and the Application Armor protects each packet, whether it’s carrying voice or data, to ensure quality and performance are unfailing and crystal clear.

Hosted, cloud-based Voice over IP (VoIP) and IP-PBX technologies help companies like Houlihan Lawrence dramatically reduce costs for equipment, lines, manpower, and maintenance. But those savings are far from ideal if they come without reliability, quality and failover capabilities. New technologies, including Mushroom Networks’ Broadband Bonding and Application Armor, can optimize IP performance, boost connectivity speeds, improve monitoring and shield VoIP services from jitter, latency, packet loss, link loss and other unwanted behaviors that degrade performance.

Dr. Cahit Jay Akin is the co-founder and chief executive officer of Mushroom Networks, a privately held company based in San Diego, CA, providing broadband products and solutions for a range of Internet applications.

Enterprise SBC and UC Security Essentials Discussion Group

Thursday, October 4th, 2012

The conversation at the IT Expo ‘Enterprise SBC and UC Security Essentials‘ session got so lively that one audience member requested an on-line continuation when we ran out of time. So if you are interested in posing a question to the panelists, please do it at the new Google Group: https://groups.google.com/d/forum/uc-security

ITExpo: Enterprise SBC and UC Security Essentials

Friday, September 7th, 2012

If you are going to ITExpo West 2012 in Austin, make sure you attend my panel on this topic at 10:00am on Wednesday, October 3rd.

The panelists are Scott Beer of Ingate Systems, Jeff Dworkin of Sangoma, Eric Hernaez of NeSatpiens, Mykola Konrad of Sonus Networks, Jack Rynes of Avaya and John Nye of Genband.

The pitch for the panel is:

Supported by Session Border Controllers (SBCs) and Unified Communications (UC), enterprises can enable workers to essentially carry their desk phone extensions and features with them, wherever they are working on any given day – via VoIP clients and other UC applications on smartphones, tablets, and other mobile devices. With rich UC applications features such as call transfer, conference call, corporate directory listings, and presence, workers can collaborate and communicate in real-time, increasing productivity by maintaining an always one presence.

But wireless and Internet connected mobile devices present unique security challenges that differ dramatically from traditional communications and data security methods that rely on firewalls, user authentication, and encryption. Further, these mobile devices can expose sensitive network traffic, and proprietary or confidential data and communications, to multiple vulnerabilities.

Enterprises that have embraced SBCs, and other components of UC security, are proving they can securely protect and extend communications to external parties, unlocking new ways of collaborating with clients, partners, distributed employees and the supply chain. This session will consider the Enterprise SBC as a means of satisfying security and privacy requirements, with signaling and traffic encryption, media and signaling forking, network demarcation, and threat detection and mitigation, enabling enterprises to capture the cost benefits of VoIP and UC, while maintaining essential security postures and access to multi-mobile communications across the network, anytime, anywhere.

The Post PSTN Telco Cloud

Thursday, December 1st, 2011

I will be moderating a panel on this topic at ITExpo East 2012 in Miami at 3:00pm on Thursday, February 2nd.

The panelists are Brian Donaghy of Appcore, LLC, Jan Lindén of Google, Hugh Goldstein of Voxbone and Danielle Morrill of Twilio.

The pitch for the panel is:

The FCC has proposed a date of 2018 to sunset the Public Service Telephone Network (PSTN) and move the nation to an all IP network for voice services. This session will explore the emerging trends in the Telco Cloud with case studies. Learn how traditional telephone companies are adapting to compete, and new opportunities for service providers, including leveraging cloud computing and Infrastructure as a Service (IaaS) systems that are being deployed with scalable commodity hardware to deliver voice and video services including IVR, IVVR, conferencing plus Video on Demand and local CDNs.

In related news, a group of industry experts is collaborating on a plan for this transition. The draft can be found here. I volunteered as the editor for one of the chapters, so the current outline roughs out some of my opinions on this topic. This is a collaborative project, so please contact me if you can help to write it.

ITExpo: The Realities of Mobile Videoconferencing

Thursday, December 1st, 2011

I will be moderating a panel on this topic at ITExpo East 2012 in Miami at 1:00pm on Thursday, February 2nd.

The panelists will be Girish Khavasi of Dialogic, Trent Johnsen of Hookflash, Anatoli Levine of RADVISION and Al Balasco RadiSys. This is a heavy hitting collection of panelists. Come with your toughest questions – you will get useful, authoritative answers.

The pitch for the panel is:

As 4G mobile networks continue to be rolled out and new devices are adopted by end users, mobile video conferencing is becoming an increasingly important component in today’s Unified Communications ecosystem. The ability to deliver enterprise-grade video conferencing including high definition voice, video and data-sharing will be critical for those playing in this space. Mobile video solutions require vendors to consider a number of issues including interoperability with new and traditional communications platforms as well as mobile operating systems, user interfaces that maximize the experience, and the ability to interoperate with carrier networks. This session will explore the business-class mobile video platforms available in the market today as well as highlight some end-user experiences with these technologies.

ITExpo East 2011: NGC-02 “The Next Generation of Voice over WLAN”

Monday, January 24th, 2011

I will be moderating this panel at IT Expo in Miami on February 2nd at 10:00 am.

Voice over WLAN has been deployed in enterprise applications for years, but has yet to reach mainstream adoption (beyond vertical markets). With technologies like mobile UC, 802.11n, fixed-mobile convergence and VoIP for smartphones raising awareness/demand, there are a number of vendors poised to address market needs by introducing new and innovative devices. This session will look at what industries have already adopted VoWLAN and why – and what benefits they have achieved, as well as the technology trends that make VoWLAN possible.

The panelists are:

  • Russell Knister, Sr. Director, Business Development & Product Marketing, Motorola Solutions
  • Ben Guderian, VP Applications and Ecosystem, Polycom
  • Carlos Torales, Cisco Systems, Inc.

All three of these companies have a venerable history in enterprise Wi-Fi phones; the two original pioneers of enterprise Voice over Wireless LAN were Symbol and Spectralink, which Motorola and Polycom acquired respectively in 2006 and 2007. Cisco announced a Wi-Fi handset (the 7920) to complement their Cisco CallManager in 2003. But the category has obstinately remained a niche for almost a decade.

It has been clear from the outset that cell phones would get Wi-Fi, and it would be redundant to have dedicated Wi-Fi phones. And of course, now that has come to pass. The advent of the iPhone with Wi-Fi in 2007 subdued the objections of the wireless carriers to Wi-Fi and knocked the phone OEMs off the fence. By 2010 you couldn’t really call a phone without Wi-Fi a smartphone, and feature phones aren’t far behind.

So this session will be very interesting, answering questions about why enterprise voice over Wi-Fi has been so confined, and why that will no longer be the case.

Video calling from your cell phone

Wednesday, December 15th, 2010

Although phone numbers are an antiquated kind of thing, we are sufficiently beaten down by the machines that we think of it as natural to identify a person by a 10 digit number. Maybe the demise of the numeric phone keypad as big touch-screens take over will change matters on this front. But meanwhile, phone numbers are holding us back in important ways. Because phone numbers are bound to the PSTN, which doesn’t carry video calls, it is harder to make video calls than voice, because we don’t have people’s video addresses so handy.

This year, three new products attempted to address this issue in remarkably similar ways – clearly an idea whose time has come. The products are Apple’s FaceTime, Cisco’s IME and a startup product called Tango.

In all three of these products, you make a call to a regular phone number, which triggers a video session over the Internet. You only need the phone number – the Internet addressing is handled automatically. The two problems the automatic addressing has to handle are finding a candidate address, then verifying that it is the right one. Here’s how each of those three new products does the job:

1. FaceTime. When you first start FaceTime, it sends an SMS (text message) to an Apple server. The SMS contains sufficient information for the Apple server to reliably associate your phone number with the XMPP (push services) client running on your iPhone. With this authentication performed, anybody else who has your phone number in their address book on their iPhone or Mac can place a videophone call to you via FaceTime.

2. Cisco IME (Inter-Company Media Engine). The protocol used by IME to securely associate your phone number with your IP address is ViPR (Verification Involving PSTN Reachability), an open protocol specified in several IETF drafts co-authored by Jonathan Rosenberg who is now at Skype. ViPR can be embodied in a network box like IME, or in an endpoint like a phone of PC.
Here’s how it works: you make a phone call in the usual way. After you hang up, ViPR looks up the phone number you called to see if it is also ViPR-enabled. If it is, ViPR performs a secure mutual verification, by using proof-of-knowledge of the previous PSTN call as a shared secret. The next time you dial that phone number, ViPR makes the call through the Internet rather than through the phone network, so you can do wideband audio and video with no per-minute charge. A major difference between ViPR and FaceTime or Tango is that ViPR does not have a central registration server. The directory that ViPR looks up phone numbers in is stored in a distributed hash table (DHT). This is basically a distributed database with the contents stored across the network. Each ViPR participant contributes a little bit of storage to the network. The DHT itself defines an algorithm – called Chord – which describes how each node connects to other nodes, and how to look up information.

3. Tango, like FaceTime, has its own registration servers. The authentication on these works slightly differently. When you register with Tango, it looks in the address book on your iPhone for other registered Tango users, and displays them in your Tango address book. So if you already know somebody’s phone number, and that person is a registered Tango user, Tango lets you call them in video over the Internet.

QoS meters on Voxygen

Wednesday, October 27th, 2010

The term “QoS” is used ambiguously. The two main categories of definition are first, QoS Provisioning: “the capability of a network to provide better service to selected network traffic,” which means packet prioritization of one kind or another, and second more literally: “Quality of Service,” which is the degree of perfection of a user’s audio experience in the face of potential impairments to network performance. These impairments fall into four categories: availability, packet loss, packet delay and tampering. Since this sense is normally used in the context of trying to measure it, we could call it QoS Metrics as opposed to QoS Provisioning. I would put issues like choice of codec and echo into the larger category of Quality of Experience, which includes all the possible impairments to audio experience, not just those imposed by the network.

By “tampering” I mean any intentional changes to the media payload of a packet, and I am OK with the negative connotations of the term since I favor the “dumb pipes” view of the Internet. On phone calls the vast bulk of such tampering is transcoding: changing the media format from one codec to another. Transcoding always reduces the fidelity of the sound, even when transcoding to a “better” codec.

Networks vary greatly in the QoS they deliver. One of the major benefits of going with VoIP service provided by your ISP (Internet Service Provider) is that your ISP has complete control over QoS. But there is a growing number of ITSPs (Internet Telephony Service Providers) that contend that the open Internet provides adequate QoS for business-grade telephone service. Skype, for example.

But it’s nice to be sure. So I have added a “QoS Metrics” category in the list to the right of this post. You can use the tools there to check your connection. I particularly like the one from Voxygen, which frames the test results in terms of the number of simultaneous voice sessions that your WAN connection can comfortably handle. Here’s an example of a test of ten channels:

Screen shot of Voxygen VoIP performance metrics tool

ITExpo West — Building Better HD Video Conferencing & Collaboration Systems

Friday, October 1st, 2010

I will be moderating a session at ITExpo West on Tuesday 5th October at 9:30 am: “Building Better HD Video Conferencing & Collaboration Systems,” will be held in room 306A.

Here’s the session description:

Visual communications are becoming more and more commonplace. As networks improve to support video more effectively, the moment is right for broad market adoption of video conferencing and collaboration systems.

Delivering high quality video streams requires expertise in both networks and audio/video codec technology. Often, however, audio quality gets ignored, despite it being more important to efficient communication than the video component. Intelligibility is the key metric here, where wideband audio and voice quality enhancement algorithms can greatly improve the quality of experience.

This session will cover both audio and video aspects of today’s conferencing systems, and the various criteria that are used to evaluate them, including round-trip delay, lip-sync, smooth motion, bit-rate required, visual artifacts and network traversal – and of course pure audio quality. The emphasis will be on sharing best practices for building and deploying high-definition conferencing systems.

The panelists are:

  • James Awad, Marketing Product Manager, Octasic
  • Amir Zmora, VP Products and Marketing, RADVISION
  • Andy Singleton, Product Manager, MASERGY

These panelists cover the complete technology stack from chips (Octasic), to equipment (Radvison) to network services (Masergy), so please bring your questions about any technical aspect of video conferencing systems.