Just a guy in Vermont trying to connect all the dots...
Author's posts
Feb 18
FIR #691 – 2/18/13 – For Immediate Release
Feb 15
Ending A Time Of Quiet And Returning To Action
As you’ve perhaps noticed, it’s been basically two weeks since we last posted anything to the Deploy360 site and I thought I’d just take a moment to explain the quiet time. On the afternoon of February 1, 2013, we had a short outage of the site and while it was only offline for a brief period, the experience highlighted a couple of weaknesses we had in our setup both in terms of technology and process. We took some time to analyze the issues and to make some changes, and I’m pleased to report that the site is now running on a new server in a different architecture and we’ve made a number of other changes as well behind the scenes.
As we were working through this analysis and change, we deliberately wanted to not attract large numbers of visitors to the site and so we put a pause on publishing new content. While we still have a few more changes we may be making in the weeks ahead, we’re ready to get back in action and so you’ll start seeing new posts and resources coming out from us starting today!
Feb 11
FIR #690 – 2/11/13 – For Immediate Release
Feb 06
WebRTC Passes Huge Milestone In Rewiring The Web – Video Calls Between Chrome and Firefox
They also published the video I've embedded below. On the surface, the video doesn't appear terribly exciting: two guys having a basic conversation over video. But consider this:
- The video conversation was initiated from within web browsers.
- There were NO plugins used... no Flash, Java or anything else.
- The entire conversation was securely encrypted.
- The call used "wideband audio" (also called "HD audio") to provide a much richer experience that far exceeds any kind of conversation you can have on traditional telecom and mobile networks.
- The call did not have to involve any external telecom networks or services and could have been initiated directly from one browser to the other. (I don't know exactly how they set up this call.)
And perhaps most importantly:
Any web developer can now create this kind of real-time communication using a few lines of JavaScript and other web programming languages.
As I'm said before, WebRTC will fundamentally disrupt telecommunications and add a real-time communications layer to the Internet that is based on open standards and is interoperable between systems. Creating applications that use voice, video and chat is being removed from the realm of "telecom developers" and made truly accessible to the zillions of "web developers" out there.
Congrats to the Google and Mozilla teams... this is a huge step forward for WebRTC!
You can see the video below... and if you are a developer interested in playing with WebRTC further, both the Google and Mozilla blog posts offer pointers to source code. The team over at Voxeo Labs also released a new version of their Phono SDK yesterday with WebRTC support that may be helpful as well.
UPDATE #1: The discussion threads on Hacker News related to the Google and Chrome blog posts make for quite interesting reading and provide many additional links for exploration:
- Hacker News thread about Google blog post (long thread about relation to Skype)
- Hacker News thread about Mozilla blog post
UPDATE #2: Over at Forbes, Anthony Wing Kosner weighed in with a similar piece and proved he can write far more poetic headlines than mine: Google And Mozilla Strike The Golden Spike On The Tracks Of The Real Time Web
UPDATE #3: And over on No Jitter, Tsahi Levant-Levi gets the "wet blanket" award for dampening enthusiasm with his post: WebRTC Browser Interoperability: Heroic. Important. And...Expected
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Feb 04
Oracle Buys Acme Packet For $2 Billion To Gain SIP Session Border Controllers (SBCs) And More
As Andy Abramson writes, the fascinating aspect of this acquisition is this:
This is an interesting grab by one of the tech world's true giants because it sqaurly puts Oracle into a game where they begin to compete with the giants of telecom, many of whom run Oracle software to drive things including SBC's, media gateways and firewall technology that's sold.
This acquisition does put Oracle VERY firmly into the telecom sector at a carrier / large enterprise level, as Acme Packet's products are widely used within that tier of companies. As the news release notes:
"The company's solutions are deployed by more than 1,900 service providers and enterprises globally, including 89 of world's top 100 communications companies."
Acme Packet has also long been recognized as a leader by analyst firms such as Gartner. People from Acme Packet, in particular Hadriel Kaplan, have also been extremely involved with industry efforts such as the SIP Forum and standards activity in the IETF.
As far as integration, Oracle already has a wide array of "communications" products, including several unified communications (UC) products that could potentially interact with Acme Packet products extremely well. Beyond all of that, though, this acquisition will have Oracle being a strong player in providing telecom infrastructure as we continue to collectively move to basing all our communications on top of IP.
Congratulations to my friends at Acme Packet and Oracle... and I wish them the best as they proceed down the path to completing this acquisition.
More information here:
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Feb 04
FIR #689 – 2/4/13 – For Immediate Release
Jan 28
Next SIPit Test Event Feb 18-22 – Deadline of Feb 4 For Registration
Are you a vendor of SIP-based products and services? Do you have software or hardware (or cloud-based products) that use SIP? If so, are you planning to attend the next SIPit test event planned for February 18-22, 2013, in Raleigh, North Carolina?
The SIPit events are an outstanding place to test your SIP implementations. Where else will you have so many other vendors also testing their equipment? It's a great place to go, test... and iterate your code even while you are there so that you can test again.
The registration deadline is Feb 4, 2013 for SIPit 30, so you need to act soon if you want to attend.
Olle Johansson posted a great set of slides about why you should go to SIPit:
And reaching back to 2009, here's a video interview I did with Robert Sparks about the SIPit test events:
If you are a vendor of SIP products or services, I would strongly encourage you to consider attending the next SIPit. It's a great way to make sure your SIP works as best as it can.
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Jan 28
FIR #688 – 1/28/13 – For Immediate Release
Jan 25
New Internet-Draft: Balanced IPv6 Security for Residential CPE
What should the appropriate IPv6 security policy be for residential customers? How can they get the benefits of IPv6 while still ensuring that their home networks are secure? These are the questions pursued in a new Internet-Draft available today:
http://tools.ietf.org/html/draft-v6ops-vyncke-balanced-ipv6-security
The abstract and introduction explain quite well how this applies to “customer premise equipment (CPE)”:
Internet access in residential IPv4 deployments generally consist of a single IPv4 address provided by the service provider for each home. Residential CPE then translates the single address into multiple private IPv4 addresses allowing more than one device in the home, but at the cost of losing end-to-end reachability. IPv6 allows all devices to have a unique, global, IP address, restoring end-to-end reachability directly between any device. Such reachability is very powerful for ubiquitous global connectivity, and is often heralded as one of the significant advantages to IPv6 over IPv4. Despite this, concern about exposure to inbound packets from the IPv6 Internet (which would otherwise be dropped by the address translation function if they had been sent from the IPv4 Internet) remain. This document describes firewall functionality for an IPv6 CPE which departs from the “simple security” model described in [RFC6092] . The intention is to provide an example of a security model which allows most traffic, including incoming unsolicited packets and connections, to traverse the CPE unless the CPE identifies the traffic as potentially harmful based on a set of rules. This model has been deployed successfully in Switzerland by Swisscom without any known security incident.
This document is applicable to off-the-shelves CPE as well to managed
Service Provider CPE.
The authors welcome comments to the draft and their email addresses can be found at the end of the document. It’s definitely a worthwhile contribution to the IPv6 security discussion and could provide useful guidance to operators seeking to understand how they should configure customer equipment to allow IPv6 yet still remain secure.
Jan 25
Last Day To Submit Speaking Proposals for SIPNOC2013
The SIP Network Operators Conference (SIPNOC) is an outstanding event happening in Herndon, Virginia, USA, from April 22-25. It brings together network operators working with SIP / VoIP networks for several days of talks, networking (of the human kind) and education. I've gone the past two years, speaking about IPv6, and they are truly excellent conferences. Not too big, not too small... and with an extremely high quality of people both attending and speaking.
If you think you'd like to present, TODAY, January 25, 2013, is the end of the call for presentations for SIPNOC 2013. They are seeking presentations on topics such as (see the CFP for more detail):
- Peering
- SIP Trunking
- Congestion Control
- Applications/content Development
- Interoperability
- Call Routing
- Security
- Monitoring/Troubleshoooting and Operational Issues
- Testing Considerations and Tools
- Availability/Disaster-Recovery
- WebRTC and SIP
- SIP-Network Operations Center Best Practices
- Standardization Issues and Progress
- FoIP/T.38 Deployment
- User-Agent Configuration
- IPv6 Deployment Challenges
- Emergency Services
- Scaling and Capacity Issues
- HD-Voice Deployment Challenges
- Video Interop Issues
They are seeking individual talks, panel sessions, research sessions and BOFs.
Even if you just have an idea for a session, I'd encourage you to submit a proposal so that the SIPNOC 2013 Program Committee will know of your interest and can reach out to you for more details. More info about the process can be found on the CFP page.
If you aren't interested in speaking, but are now intrigued by SIPNOC and would like to be learning from all the excellent sessions, you can go to the SIPNOC 2013 main page and find out information about how to register and attend.
If you work at or for a telecom/network operator who is involved with SIP and VoIP, I highly recommend SIPNOC as a conference you should attend - you'll learn a huge amount and make great connections.
P.S. I have no affiliation with SIPNOC other than being a speaker there in the past. SIPNOC is a production of the SIP Forum, a great group of people focused on advancing the deployment and interoperability of communications products and services based on SIP.
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