Microsoft Teams, Direct Routing and SBCs (pt.1)
My journey creating a scalable SBC as a Service for Microsoft Teams Direct Routing is over at Snapsonic.com.
Hookflash, Google and Microsoft lead on ORTC / WebRTC 1.1 Public Draft
The first ORTC Public Draft Specification has been published, authored by Hookflash, Microsoft, and Google. (http://ortc.org/wp-content/uploads/2014/08/ortc.html ) This specification extends WebRTC 1.0 with new functionality to create a WebRTC 1.1 API with exceptional flexibility and no loss of compatibility.
Like WebRTC, ORTC (Object Real-time Communication) enables plugin-free real-time communications for mobile, web and cloud, but is specifically tailored to provide the direct control needed to enable advanced multimedia and conferencing features.
“We heard developers say that they wanted more direct control over the technologies available in WebRTC. At the same time, we didn’t want existing developers to have to start over with a new API. ORTC is our proposal for how we can accomplish both of these things – a new set of APIs for direct control, that builds off the existing WebRTC 1.0 API set. As an evolution of the existing API, we consider this WebRTC 1.1” comments Justin Uberti, Google Tech Lead, WebRTC. “We’re grateful to Hookflash for their work to get ORTC off the ground. They have been instrumental in making this cross-industry collaboration happen, and we look forward to continuing our work with them.”
This newly published public draft has come a long way since the W3C ORTC Community Group was formed in mid-2013. As it has progressed from an initial set of ideas to a fleshed-out draft complete enough for implementations, several companies have gotten closely involved, with Microsoft and Google now joining Hookflash as authors of the emerging specification.
“We have been working hard to get the ORTC API to the point where it can be implemented. This would not have been possible without the initial and continuing work of Hookflash”, commented Bernard Aboba, Principal Architect, Skype, “We also are excited by the ORTC API’s support for advanced video features such as SVC (Scalable Video Coding) and simulcast. The Javascript Object API approach has made these advanced video technologies more accessible, which has been difficult in the past.”
The W3C ORTC Community Group now numbers more than 60 participants.
“We believe the contributions to WebRTC 1.1 / ORTC will allow web communications technology to become ubiquitous and transcend nearly all communications technologies that came before it” says Hookflash Co-founder, Erik Lagerway, “We are honored to be working with some of the brightest minds at Google, Microsoft, and the other contributing members in the ORTC CG to mature WebRTC into a universal go-to toolkit enabling communications across the globe.”
For more information on ORTC, see:
W3C ORTC Community Group
ORTC.org – History and FAQs
WebRTC.is – ORTC & WebRTC news
Hookflash enables real-time social, mobile, and web communications for integration of voice, video, messaging with federated identity into world leading software, enterprise, applications, networks, mobile and computing devices. Hookflash and Open Peer are trademarks of Hookflash Inc.
Developers can register at (http://fly.hookflash.me) to start using the Hookflash RTC service and toolkits today. For more information on Hookflash RTC toolkits and White Labeling please visit Hookflash http://hookflash.com.
Come and work at one of the coolest companies in the space! We’re now hiring for these development positions: iOS, Android, Node.js & C++ send us your resume: jobs@hookflash.com.
Hookflash – Trent Johnsen
855-466-5352 Ext: 1
New CU-RTC-Web Prototype – Roaming
The crew at Microsoft is forging ahead with their “CU-RTC-Web” specification as a counter proposal to the new WebRTC / RTCWEB proposed standard in the W3C and IETF. My colleague Robin Raymond and I certainly align with Microsoft on some issues, more specifically around SDP but it would have been good if this work took place inside the IETF.
I really can’t see Microsoft changing their tune anytime soon, which means that Enterprise web application developers will likely need to support both WebRTC and CU-RTC-Web if they are to be a plugin-less solution enabling RTC across all browsers. Not ideal.
From the Sidelines, My Introduction into RTCWEB
I’ve been following the RTCWEB standardization for a while now from an architecture and technology standpoint. For the most part, I’ve been quiet and I’ve assumed a rather neutral stance in regards to the RTCWEB process when it comes to Open Peer, but my opinion has changed and I can no longer maintain a neutral standpoint.
There are many companies taking stances who all need to have a say in what happens because they want to make sure their technology does not get left out in the cold when RTCWEB comes into reality, as most people think this technology will be huge with consumers and businesses. The big guys with SIP, XMPP and Skype have various established offerings and they are married to existing technology that is difficult to change. They need to make sure that RTCWEB closely follows, or at the very least, does not hinder their own technology from functioning otherwise they will get left behind. The process of adapting existing systems to a new standard is understandably costly.
Thus, I have to ask, who am I with Open Peer to come along and push back against these tides as a new protocol when I have much more flexibility in our implementation than existing deployed systems? Further, the Open Peer protocol particularities didn’t even exist until recently and it has been under revision as Hookflash tested the implementation. We’ve just recently published our specification and source code and we’ve just undergone a significant update based on internal and external feedback from our initial implementations.
To be honest, architecting, designing and implementing a brand new protocol with such an ambitious scale for a small company has kept me extremely engaged and busy. I could listen to what’s happening from a 1,000-foot high perspective, but unfortunately that has also been a factor in my personal ability to participate. I don’t think it’s a great secret for those already involved that it takes immense devotion of time resources to follow the details, let alone participate in these long drawn procedures in ratifying a specification complex as RTCWEB that spans two organizations, namely the IETF and W3C groups. This is unfortunate that such time commitments are so huge as I think having those on the front lines much more actively involved would be healthy, but I digress.
In reality though, Hookflash is in a unique position with Open Peer. I am working on this protocol with a clean slate and a future thinking sense. I do not have the old technology shackles and I didn’t have to design with legacy deployed services in mind which would no doubt confound my decision making process. Likewise, I’ve had the experience of these legacy systems to help avoid their pitfalls (specifically as the original author of the X-Lite/X-Pro SIP softphone client for CounterPath years back with SIP).
For those unaware, Open Peer is an open peer-to-peer signaling protocol that has an initial implementation in C++ and Hookflash is in the process of writing a pure JavaScript version. The idea is to allow secure peer-to-peer signaling communication straight from browser-to-browser and capability to talk to native mobile device applications as well.
The Open Peer implementation goes beyond basic call flow signaling and even beyond peer-to-peer signaling and incorporates identity and federation concepts with strong privacy and security considerations in mind.
Having just completed the next iteration of the protocol that is going through internal testing, I plan to spend much more time actively examining the details of RTCWEB standards. Even though I’m later to the table representing a newer company with newer technology, I hope the input will be welcome to the discussion. I do understand that decisions may be too immovable to change peoples’ minds and there is an active amount of established legacy systems, but hopefully coming from a unique perspective will help bring fresh blood and deeper insight. Forgive me if I argue points already lost, but I will always explain my reasoning for wanting to push certain aspects even if I am ignored in the end.
Ultimately, Open Peer will leverage RTCWEB and the implementation will adapt accordingly to the mutually agreed standards. I’m still going to give my opinion for whatever it is worth and I hope to prove it worthy, unique and valuable.
There are many bright people involved in the process and many companies with unique corporate political angles and agendas. My perspective and motivation will be straight up front. I want RTCWEB to succeed as soon as possible, but with an equal emphasis on ensuring the technology is sound from the future perspective as well, obviously in relation to plans with Open Peer utilizing RTCWEB.
What's up with Skype for Mac?
Can’t wait for Skype for iPad, it looks awesome, but here is an example of a one-on-one video call on Skype with my Macbook AIR…
Anyone else seeing Skype hog resources like this?
WebRTC is live. Flash, take cover!
Update 2: To the hundreds/thousands of repetitive spam tweets / twits, “Will WebRTC replace / kill Skype”, the answer is NO!! It will not. WebRTC is using broken Jingle in the browser, it does not support chat and can only make and receive calls., there is no buddy / contact list to speak of etc etc. NO it will not replace Skype. Stop with the spam tweets already, please!
Update: It seems to me that until all the browsers are on board, native clients will be required to make this go. Which is not outside the realm of possibility, considering Google has open sourced the GIPS audio and video engine along with WebRTC.
Something to remember, WebRTC is not RTCWEB! It may sound silly but it’s true. WebRTC is a Google-centric project using Google code etc. RTCWEB is essentially an IETF effort, a working group driving towards open real-time communications on the web. They are not the same, which can be rather confusing.
— Original Post —
Google has been busy it would seem, last night WebRTC appeared to the public for the first time. This has some pretty serious implications for Flash, which was the de-facto technology one had to use to get real-time communications in a browser, that has now been circumvented, at least to a certain degree.
The sessions are not run by a signaling protocol per se, not Jingle, no XMPP, not SIP not anything we have seen before. All the session management looks to be coming from libjingle. Which, to me means Jingle is in the browser.
A few early comments:
1. Where does Google stand on websockets? Google have said they will block it if an exploit emerges.
2. Chrome, Opera & Firefox are the supported browsers. Where does Safari and IE land? My guess is that Microsoft will not be in any hurry to implement this considering their recent Skype acquisition.
3. Web-cam captures from HTM5 has not been ratified, although this is likely not as serious as the former points.
Skype's email to me re: Skype for Asterisk
I just received this email from Skype’s PR firm…
Hi Erik,
Here is Skype’s official comment regarding Skype for Asterisk. You can attribute this to Jennifer Caukin, 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.”
Thank you,
Cassie
Call me crazy but if I have to pay to integrate Skype into my phone system, where I already have a phone service that I am happy with, why would I do that? Maybe I just want to be able to make/receive Skype calls on my SIP-enabled desk phone? If it doesn’t hit the PSTN why do I have to pay? Seems like an odd approach for a company that has a long history of working around POTS, much to the delight of their users.
Integration with SIP is great, don’t get me wrong, but it would be nice if Skype talked SIP and was ‘still’ free. Seems like a massive oversight on behalf of Skype or am I missing something?
Open and secure alternative to Skype
Imagine a new secure P2P (Skype like) offer that also supported SIP in the client. You could use the client software on it’s own (just like Skype) or attach it to just about any VoIP service or phone system for free.
Does it make sense for consumers?
Does it make sense for business users?
Is there room in the market?
Would you use it?
Martyn Davies chimes in…
I would use it, but as a telecom industry insider, I know that I’m not the average business user or consumer. As to whether there is room in the market, I think that depends a lot on what Microsoft do with Skype now that they own it. From a business point-of-view, their efforts are focused around OCS/Lync (and software licenses), so Skype there is not adding to their central proposition. Skype has a lot of users, but produces very little revenue, since the majority just use the free services. As a Skype competitor you would have the same problems getting to the cash.
Skype was really the first company to take VoIP and make it completely trivial to install and use. To do that, they had to take some liberties and deviate from standards (like SIP), so that they could add the magic that made it work from behind firewalls, add security and self-configuration, and integrate video so seamlessly. Like Facebook, once it is clearly the biggest of its kind of services, it becomes the community that everyone must join. I can’t see that another Skype-alike has a way in, unless Microsoft significantly change the rules now.
What do you think?
So it begins. Skype for Asterisk falls.
It looks like the first victim in the Microsoft acquisition of Skype is Digium and the open source PBX – Asterisk. The following is an email sent to existing Skype for Asterisk users…
Skype for Asterisk will not be available for sale or activation after July 26, 2011.
Skype for Asterisk was developed by Digium in cooperation with Skype. It includes proprietary software from Skype that allows Asterisk to join the Skype network as a native client. Skype has decided not to renew the agreement that permits us to package this proprietary software. Therefore Skype for Asterisk sales and activations will cease on July 26, 2011.
This change should not affect any existing users of Skype for Asterisk. Representatives of Skype have assured us that they will continue to support and maintain the Skype for Asterisk software for a period of two years thereafter, as specified in the agreement with Digium. We expect that users of Skype for Asterisk will be able to continue using their Asterisk systems on the Skype network until at least July 26, 2013. Skype may extend this at their discretion.
Skype for Asterisk remains for sale and activation until July 26, 2011. Please complete any purchases and activations before that date.
Thank you for your business.
Digium Product Management
One has to wonder what will become of Skype Connect, Skype’s answer to SIP Trunking. Will Microsoft shut off the Skype Connect vendors (Cisco, Avaya, Grandstream, etc.) as well?
Original forum post here.
VoIP over 3G now officially allowed on iPhone & iPad, confirmed.
UPDATE: It’s looking good folks!
No more Jail breaking iPhones for VoiP over 3G
In the agreement…
3.3. 23 Because some mobile network operators may prohibit or restrict the use of Voice over Internet Protocol (VoIP) functionality over their network, such as the use of VoIP telephony over a cellular network, and may also impose additional fees, or other charges in connection with VoIP, You agree to inform end-users, prior to purchase, to check the terms of agreement with their operator, for example, by providing such notice in the marketing text that You provide accompanying Your Application on the App Store.
9. Third Party Terms of Agreement: You must state in the EULA that the end-user must comply with applicable third party terms of agreement when using Your Application, e.g., if You have a VoIP application, then the end-user must not be in violation of their wireless data service agreement when using Your Application.
Now that we know VoIP over the cellular data network is allowed, and ATT has said they will support it, and ATT has a cheap unlimited data plan (Listen up Rogers, Telus, Bell!), the iPad and iPhone has just become something I think we should be excited about.
Previous Post:
Apparently the new iPhone dev agreement has officially been modified allowing for VoIP over the cellular data networks. Trying to confirm that myself.
If this is the case, the iPad and iPhone just got a whole lot more interesting.
WebRTC.is 