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.
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?
Skype for SIP, it's about time!
Back in 2004 I wrote a post relating to the VON Canada Panel I sat on with Niklas Zennstrom. It was an interesting debate on open standards (SIP in this case) and closed networks, specifically Skype. I was quite vocal about how silly I thought Skype was not to include SIP, a few of you picked up on that 😉
It looks like something good came of the eBay purchase as we now see a Skype pushing towards open standards, good stuff!
On a similar note, I heard a rumour that it’s likely Jason Fischl the current CTO at Counterpath (Xten) will be going over to work with Jonathan Christensen (General Manager – Media Platform) at Skype. Jason was an early advocate of SIP in the IETF and works with some of the best minds on the subject: Cullen Jennings, Robert Sparks, Alan Duric come to mind.
This could get interesting.
I will do some testing with SkypeforSIP & Response Point and post the results along with my comments on what this new offer from Skype might mean for Response Point.
64th IETF P2P SIP Adhoc BOF – Part 3 – Video
This is part 3 of the footage that was shot at the last IETF P2P SIP Adhoc BOF in Vancouver, BC.
http://sipthat.com/movies/p2p-sip-3256K_Stream.mov
If you are interested in getting your hands on the AVI files (for DVD production for instance) I would be happy to send them to you, just send me an email: erik AT sipthat.com.
For meeting presentations and notes: http://www.p2psip.org
P2P SIP at the 64th IETF – Part 2 – Video
This is part 2 of the footage that was shot at the last IETF P2P SIP Adhoc BOF in Vancouver, BC. There is one more reel to follow, I will post that early next week. In an effort to conserve quality and reduce file size I have made the display only 180×240.
http://sipthat.com/movies/p2p-sip-2256K_Stream.mp4
If you are interested in getting your hands on the AVI files (for DVD production for instance) I would be happy to send them to you, just send me an email: erik AT sipthat.com.
For meeting presentations and notes: http://www.p2psip.org
P2P SIP at the 64th IETF – Video Footage
This footage was shot at the last IETF P2P SIP Adhoc BOF. This is the first reel, there are 2 to follow, early next week. In an effort to conserve quality and reduce file size I have made the display only 180×240.
http://sipthat.com/movies/p2p-sip-1256K_Stream.mp4
The video is a little blurry at points due to my wide angle lens, which ended up being a bad idea. Parts 2 and 3 are much better.
If you are interested in getting your hands on the AVI files (for DVD production for instance) I would be happy to send them to you, just send me an email: erik AT sipthat.com
For meeting presentations and notes: http://www.p2psip.org
P2P SIP Draft Updated
We are seeing more progress in P2P SIP, an update to the IETF draft has been recently submitted by David Bryan and Cullen Jennings. The changes are mostly surrounding syntax and handling methods as opposed to incorporating new components. I am excited about the progress being made here and I am hopeful that we will see some integrated NAT traversal and auto-provisioning with P2P SIP in mind. This work could easily have a profound effect on VoIP, Video and IM as we know it. Any operator running a SIP network today cold benefit greatly from this technology.
Good Stuff!