Tech Talk- IT & Code

JUN 2014

Posted by Will at 05:22 PM EDT

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Trusted Payments with SIP: Technical Overview

Last weekend, I ventured to Spain to attend TADHack 2014. Overall, it was a great event; developers from all around the world gathered in Madrid as well as the many satellite locations across the globe to share creative new ideas spanning topics like distributed GSM networks for rural villages, SMS APIs to summon garbage pickup, and, of course, WebRTC. One of the themes included Bitcoin and payment systems, so I asked myself what happens when you combine instant communications in the browser with distributed, electronic crypto-currencies. What if you could tie payments directly into a session?

Many applications connect you with a premium endpoint, at a cost. Bitcoin could be used to eliminate the need for trust. You wouldn’t have to worry about paying for an hour, just to have the person hang up after two minutes. At the same time, the tutor wouldn’t have fear that you will talk with them for hours and never send payment.

The result was my hack: Trusted Payments With SIP.

APR 2014

Posted by Kevin Bartley at 12:55 PM EDT

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Announcing The OnSIP Network: We've Slain Those Signaling Dragons for WebRTC Developers

You may have noticed some changes on our website— We've added a whole new section For Developers. That's because OnSIP is happy to announce The OnSIP Network, a Platform as a Service that enables developers to add the vital signaling layer to their WebRTC applications in a matter of minutes.

The OnSIP Network locates and negotiates communications between WebRTC application peers, solving a similar problem that the Public Switched Telephone Network once did for telephones.

JAN 2014

Posted by Will at 03:23 PM EST

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WebRTC Developers Ecstatic: SIP over WebSocket Approved!

Update 2014-01-29: RFC7118 has now been officially published. The full specification can be found on the IETF Tools site.

WebRTC developers rejoice! Yesterday, the IETF assigned an RFC number to the SIP WebSocket specification, a drafted document to standardize transporting SIP signaling using HTML5's realtime WebSocket API. The new document, The WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP), will be published as RFC7118.

This is big news for WebRTC developers. WebRTC is a free, open source project that enables browsers with Real-Time Communications. When building an application with real time communications, however, one of the first things developers realize is that while WebRTC provides a high quality media stack right in your browser, it does not mandate a certain signaling protocol. This leaves application developers to research many complex protocols or to write their own homegrown signaling.

OCT 2013

Posted by Kevin Bartley at 10:58 AM EDT

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OnSIP Launches Free Browser-Based Video Calling With GetOnSIP

As we've blogged about many times, WebRTC, an open project that enables web browsers with Real-Time Communications (RTC) capabilities, has the potential to change the way we communicate. This potential is recognized in the growing number of WebRTC events and media buzz, but WebRTC application development is only beginning to catch fire across industries. Today, OnSIP is happy to announce a significant milestone in product development with WebRTC: The relaunch of

AUG 2013

Posted by Nicole at 04:36 PM EDT

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OnSIP CTO John Riordan Presents on "Geographically Distributed SIP Applications with OnSIP" at ClueCon 2013

We had another great year at ClueCon in Chicago! It was my third time, earning me a ClueCon "Hat Trick" pin. By comparison, Will Mitchell, OnSIP Engineer, picked up his first year "Rookie" pin. I asked the ClueCon Rookie about his first experience at the conference.

"ClueCon was a lot of fun," said Will. "While the days were chock full of dev talks, I think my favorite part was meeting and hanging out with all of the people I only knew before from mailing lists and GitHub comments. It's a really great crowd."

John Riordan, CTO and ClueCon Hat Trick, also enjoyed his time at ClueCon. "Once again ClueCon proved to be an excellent festival of ideas and characters," he described. "A particular highlight for me was once again the evening VoIPSec BoF with Phil Zimmermann, Travis Cross and Alan Johnston."

JUL 2013

Posted by Eric Phipps at 03:47 PM EDT

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Skype Connect Makes Modest Progress as Viable SIP Service

OnSIP gets a lot of questions about Skype as a business solution. Customers are interested in using Skype either as their phone service or concurrently with OnSIP to try and leverage Skype's installed user base to make their services less expensive. Skype Connect is a business class SIP based service for Skype. After documented testing, we found that Skype Connect is almost pure SIP, although we were still unable to make outbound SIP calls. Because of these findings, we can only recommend Skype for communications in which immediacy is more important than quality.

JUN 2013

Posted by Kevin Bartley at 11:28 AM EDT

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Under the Hood: What makes OnSIP Hosted PBX platform superior?

When Hurricane Sandy ravaged New York, Internet and phone connections were disrupted across the city, but OnSIP experienced no downtime. What makes OnSIP’s hosted PBX platform so reliable? How do our redundant servers, routers, Tier 1 Internet connections, colocation centers, and upstream carriers interact to offer a platform that is uniquely superior and seamlessly scalable?

For starters, OnSIP is not comprised of customer PBXs deployed in a datacenter. OnSIP consists of a geographically distributed cloud of SIP proxies that present themselves as a single SIP server ( OnSIP phones connect to the geographically closest SIP proxy in the cloud. This unique infrastructure allows for optimal call routing, no service outages if a single box fails, and massive scalability. In the case of Hurricane Sandy, we routed customers from our NY data center to our LA data center in preparation for the storm. As result, OnSIP was not affected by the storm. In 2012, OnSIP had 99.9% uptime.

FEB 2013

Posted by Leo Zheng at 01:05 PM EST

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Why hasn't VoIP made more of a dent in traditional telephony?

During my first few weeks working at OnSIP, I spent a significant portion of my time learning about VoIP, and more specifically, about SIP, the de facto signaling protocol most widely used for setting up voice communication sessions over the Internet. I tried out other SIP applications such as video, presence, and IM, and found it convenient that I could access all of these applications using a single point of contact, my SIP address. With SIP, I discovered that both the media type and what a person uses as his access point don’t matter (SIP phones or anything that can run SIP software will allow the user to get access to his applications).

It clicked with me. I could make a high definition call, conduct a video chat, send an IM, etc. using an address that looks like my email address. Best of all, I could do all of this for free using the Internet I already pay for.

I distinctly remember thinking to myself that it was only a matter of time before the excitement around this technology started having serious impacts on traditional telephony.

That was almost 4 years ago, and as far as I can tell, my prediction has yet to happen. What happened?

JAN 2013

Posted by Samantha Avignone at 02:47 PM EST

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Recap: OnSIP Presents at NYC Telephony Hackers Meetup

Earlier this week, CTO John Riordan and Software Engineer Will Mitchell attended the NYC Telephony Hackers Meetup to demonstrate and discuss OnSIP’s three APIs.

Written By Will Mitchell

Monday night John Riordan and I went to the Rain Agency, located in NYC, to attend the Telephony Hackers Meetup. This was the third of the monthly meetups and my first time attending. Organizer Doug Crescenzi did a great job with the event, attracting about 15 people to discuss telephony APIs and share in some pizza and beer.

JAN 2013

Posted by Will at 04:00 PM EST

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WebRTC, SIP, and HTML5: A Brief Introduction

WebRTC. It has certainly generated a lot of interest in the web community. Last month, you may have even caught us saying we believe the browser to be the ultimate destination of SIP communications. And with another Java security flaw being discovered (and patched) this month, the idea of a purely browser-based option is very appealing. So what is this great new technology? It's actually a couple of different HTML5 specifications, each with its own role. Let's take a look.

Note: For the sake of brevity, I have left off the use browser-specific prefixes. Be sure to check resources such as Can I Use... when implementing your web app.