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
Archive for the ‘VoIP’ Category
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 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.
I will be moderating a panel on this topic at ITExpo East 2012 in Miami at 3:00pm on Thursday, February 2nd.
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.
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.
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.
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.
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:
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.
I discussed last September how AT&T was considering opening up the 3G data channel to third party voice applications like Skype. According to Rethink Wireless, Steve Jobs mentioned in passing at this week’s iPad extravaganza that it is now a done deal.
Rethink mentions iCall and Skype as beneficiaries. Another notable one is Fring. Google Voice is not yet in this category, since it uses the cellular voice channel rather than the data channel, so it is not strictly speaking VoIP; the same applies to Skype for the iPhone.
According to Boaz Zilberman, Chief Architect at Fring, the Fring iPhone client needed no changes to implement VoIP on the 3G data channel. It was simply a matter of reprogramming the Fring servers to not block it. Apple also required a change to Fring’s customer license agreements, requiring the customer to use this feature only if permitted by his service provider. AT&T now allows it, but non-US carriers may have different policies.
Boaz also mentioned some interesting points about VoIP on the 3G data channel compared with EDGE/GPRS and Wi-Fi. He said that Fring only uses the codecs built in to handsets to avoid the battery drain of software codecs. He said that his preferred codec is AMR-NB; he feels the bandwidth constraints and packet loss inherent in wireless communications negate the audio quality benefits of wideband codecs. 3G data calls often sound better than Wi-Fi calls – the increased latency (100 ms additional round-trip according to Boaz) is balanced by reduced packet loss. 20% of Fring’s calls run on GPRS/EDGE, where the latency is even greater than on 3G; total round trip latency on a GPRS VoIP call is 400-500ms according to Boaz.
As for handsets, Boaz says that Symbian phones are best suited for VoIP, the Nokia N97 being the current champion. Windows Mobile has poor audio path support in its APIs. The iPhone’s greatest advantage is its user interface, it’s disadvantages are lack of background execution and lack of camera APIs. Android is fragmented: each Android device requires different programming to implement VoIP.
This will make international calls much cheaper for people who are willing to put up with the latency issues of the data channel.