Category: SIP
-
/
Last Day To Submit Speaking Proposals for SIPNOC2013
Continue Reading: Last Day To Submit Speaking Proposals for SIPNOC2013Got a great idea for a talk to give to an excellent gathering of SIP/VoIP network operators? Have a new way of handling security? Have a case study you’d like to present for how you solved an operational issue?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…
-
/
The Fascinating Interest in Using Google Voice With SIP Addresses
Continue Reading: The Fascinating Interest in Using Google Voice With SIP AddressesWhy are so many people interested in using Google Voice with SIP? Is this a sign that people really want to use SIP-based services for VoIP? Is this all hobbyists or people looking to play around with Google Voice? Or is it people trying to solve real interconnection issues? What are people trying to do with Google Voice and SIP?All these questions came to my mind today when I dipped into Google Analytics and noticed that for the month to date in November 2012, my old (March 2011) post about Google Voice and SIP addresses continues to receive a large amount of traffic:
Slightly over 3,000 pageviews in the first 13 days of November – and if I go back a bit I see over 71,000 pageviews since January 1, 2012. In fact, it’s had about 232K pageviews since I wrote it over 1.5 years ago, and has accounted for almost 25% of all traffic to this site in that time.
And this particular article was just one in a series of articles I wound up writing about Google Voice and SIP as we all collectively tried to figure out what was going on.
Digging into the traffic sources…
-
/
Slides: How The Hidden Secret of TCP/IP Affects Real-time Communications
Continue Reading: Slides: How The Hidden Secret of TCP/IP Affects Real-time CommunicationsRecently at Voip2day + ElastixWorld in Madrid 2012, Olle E Johansson gave a great presentation outlining where we are at with telecom and VoIP in 2012 – and where we need to go! Olle is a long-time, passionate and tireless advocate for the open Internet, IPv6, SIP and standards and interoperability. I’ve known Olle for years via Asterisk-related issues, via the VUC calls and via work on SIP over IPv6.This presentation (slides available) really hits a number of key points about where we are at now:
The secret of TCP/IP and how it affects your PBX from Olle E Johansson
In particular I was struck by his slides 24-28 that strike the same theme I’ve been writing about across multiple blogs, namely the way we are reversing the “open Internet” trend and retreating back inside walled gardens of messaging:
He goes on to walk through what happened with SIP and how the protocol evolved – and evolved away from interoperability. His conclusion is that we as customers need to take back control, avoid vendor lock-in and demand interoperability.
He also points people over to his “SIP…
-
/
SIPNOC 2012 Photos Now Available On Flickr
Continue Reading: SIPNOC 2012 Photos Now Available On FlickrAt this year’s SIP Network Operators Conference (SIPNOC) on June 25-27, 2012 in Reston, VA, I was shooting photos of the various presenters as well as trying to take some shots that captured the general feel of the excellent event. As with shooting any event, I find the actual taking photos to be the insanely easy part… it is the curation of the photos that takes the longest amount of time. Over the past bit, though, I finally was able to reduce the 500+ photos I shot down to a meaningful set and I’ve now posted the SIPNOC 2012 photos up to Flickr:A special thanks to Spencer Dawkins who took some shots of me speaking.
I’ve licensed them all under a simple Creative Commons Attribution license so that they can be used by others. If you’re in the photos and want an original, you can download them from Flickr… and you’re also welcome to contact me if you have any issues downloading a file.
SIPNOC 2012 was a great event and kudos to the SIP Forum for making the event happen! I’m looking forward to next year’s event!
If you found this post interesting or useful, please consider…
-
/
The Big Question On Avaya’s Acquisition of Radvision – What About The SIP and H.323 Stacks?
Continue Reading: The Big Question On Avaya’s Acquisition of Radvision – What About The SIP and H.323 Stacks?With today’s big news in the VoIP / Unified Communications (UC) / telecom space of Avaya’s acquisition of Radvision, pretty much all of the coverage has predictably focused on the video angle. While that’s certainly important, I have a far bigger question:What about Radvision’s SIP and H.323 stacks?
More specifically –
will Avaya continue to support and promote the strong usage of Radvision stacks by other vendors?
Of all the coverage I’ve seen so far, only Tom Keating touched on this in his brief post:
They also developed a H.323 stack used in hundreds of VoIP and videoconferencing products before SIP became the dominant VoIP protocol of choice.
Beyond the popular H.323 stack, Radvision’s SIP stack has also been used in a good number of products out there – and Radvision also developed stacks for RTP, MGCP and many other VoIP protocols. Just follow the links off of Radvision’s developer page at:
http://www.radvision.com/Products/Developer/
to see the wide range of developer solutions they have developed over the years.
For those not familiar with this topic, a “stack” in developer-speak is basically a set of libraries that you can incorporate into your products to enable those products…
-
/
44% of SIP Implementations at SIPit 29 Supported IPv6!
Continue Reading: 44% of SIP Implementations at SIPit 29 Supported IPv6!Last week (Oct 24-27) was the 29th SIPit interoperability test event hosted by ETSI in Monaco. Organizer Robert Sparks has provided his usual outstanding summary of what occurred:
https://www.sipit.net/SIPit29_summary
The key point for me, given my new role, was right up at the top:
44% of the implementations present supported IPv6.
Now, of course ideally we’d like that to be 100%, but hey, it’s at least a good start!
There is also some narrative further down the report about “IPv6 Focused Tests” with some interesting info. One interesting note seems to be this:
Most UAs that supported dual-stack had a configuration to tell the application to ignore any returned AAAAs due to issues encountered in deployments where endpoints autoconfigured IPV6 that didn’t actually work.
In the web world this has been referred to as the “happy eyeballs” problem where a browser will try a DNS AAAA record to get to a site over IPv6 and then eventually will fail back to trying the A record to go over IPv4. The delay will cause the user to be very UNhappy. There are a couple of ways to address the issue with the usual one being to try both IPv6 and IPv4…
-
/
Video: How to Communicate at Burning Man using OpenBTS and Tropo
Continue Reading: Video: How to Communicate at Burning Man using OpenBTS and TropoHeading to Burning Man this coming week? Would you like to use your mobile phone to connect up with others on the playa in Black Rock City?If so, check out this video from Chris Pirillo about the work being done by a team of folks to supply local cell phone coverage… the vans with satellite and cell hookups are already enroute… it uses software from OpenBTS and Tropo.com to let burners leave each other voice messages, exchange SMS messages and more. Here’s the video:
And here are some blog posts that provide more information:
- The Long and Winding Road to Burning Man
- Tropo + OpenBTS + Burning man = Awesome
- Voice Board and Group SMS for Burning Man
- Papa Legba FAQ (about the deployment at Burning Man and what you need to do to participate)
I’m not personally going to be at Burning Man, but this does sound very cool!
If you found this post interesting or useful, please consider either:
- The Long and Winding Road to Burning Man
-
/
Slides from eComm 2011: How IPv6 Will Kill Telecom – And What We Need To Do About It
Continue Reading: Slides from eComm 2011: How IPv6 Will Kill Telecom – And What We Need To Do About ItLast week at the eComm conference in San Francisco I spoke about how IPv6 will impact telecommunications and what we as a community should be doing about it. The session was recorded on video and will be posted at some point in the next few months. In the meantime, I have posted my slides on SlideShare – they may not be terribly useful without my narrative… but you may find some of the links of interest:How IPv6 Will Kill Telecom – And What We Need To Do About It View more presentations from Dan York
P.S. If you are interested in me giving a presentation like this to an audience you know of, please feel free to contact me.
If you found this post interesting or useful, please consider either:
-
/
eComm Starts June 27 – The Event Showcasing Emerging Communications
Continue Reading: eComm Starts June 27 – The Event Showcasing Emerging CommunicationsWill you be out in San Francisco next week for eComm, the “Emerging Communications Conference & Awards“? If so, I’ll see you there, as I’m speaking on Wednesday. If not… are you interested in going? I still have a pass or two available as a speaker. (Drop me a note… quickly!)
I’ve always enjoyed eComm as it truly is a gathering of the thought leaders of the communications space… whether those people are working with VoIP or mobile or unified communications or whatever. It’s where the “alpha geeks” of comms go to hang out… to network… to listen and learn as they share with each other what they are working on truly out there on the bleeding edge of communications.
Some people have called eComm the “TED of Telecom” and in many ways that’s an apt comparison… quick, focused presentations to a high quality audience.
Check out the schedule and the long list of speakers… some truly great people will be there. I expect to be learning a good bit.
My own talk on Wednesday will be on the IPv6 theme I’ve been on lately. I titled it “How IPv6 Will Kill Telecom – And What We…
-
/
Skype Issues Official Statement About The End Of Skype For Asterisk
Continue Reading: Skype Issues Official Statement About The End Of Skype For AsteriskBefore writing my story yesterday about Skype killing off Skype For Asterisk, I had reached out to Skype’s PR agency to see if there was any statement from Skype. There wasn’t at the time, but today they sent over this statement from Jennifer Caukin, a spokeswoman for Skype:Skype made the decision to retire Skype for Asterisk several months ago, as we have prioritized our focus around implementing the IETF SIP standard in our Skype Connect solution. SIP enjoys the broadest support of any of the available signaling alternatives by business communications equipment vendors, including Digium. By supporting SIP in favor of alternatives, we maximize our resources and continue to reinforce our commitment to delivering Skype on key platforms where we can meet the broadest customer demand.
Being a huge advocate of open standards, I of course applaud Skype’s commitment to supporting SIP. However, as I noted two years ago in my detailed review of what was then “Skype For SIP” (and is now “Skype Connect”) the fundamental difference between Skype For Asterisk and Skype’s SIP offering is this:
Skype For Asterisk is/was two-way – you can make outbound calls TO Skype users.
You can’t do that with Skype Connect.…