All posts in P2P VoIP

WebRTC is live. Flash, take cover!

By Erik Lagerway | June 1, 2011 | api, communications, google, ietf, IP Communications News, iphone, microsoft, P2P VoIP, RTCWEB, sip, SIP phone software, Skype, softphone, voip, xmpp | 1 Comment

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.

Written by Erik Lagerway - Visit Website

So it begins. Skype for Asterisk falls.

By Erik Lagerway | May 24, 2011 | Hosted VoIP for SMB, IP Videoconferencing, mobile sip, P2P VoIP, pbx, phone services, phone system, sip trunk, Skype, small office phone systems, softphone, voip, VoIP for the Masses | 5 Comments
skype-for-asterisk-no-more

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.

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IETF 80 – RTCWEB BOF

By Erik Lagerway | March 28, 2011 | ietf, P2P VoIP, RTCWEB, sip, SIP Standards | Comments Off

The RTCWEB Bof meeting starts in less than 5 minutes. To some, the most anticipated meeting of IETF 80.

Proposed charter (truncated)…

The WG will perform the following work:
1.     Define the communication model in detail, including how session management is to occur within the model.
2.     Define a security model that describes the security goals and how the communication model can achieve these goals.
3.     Define how NAT and Firewall traversal is to occur.
4.     Define which RTP functions and extensions that shall be supported in the client and their usage for real-time media, including media adaptation to ensure congestion safe usage.
5.     Define what functionalities in the solution, such as media codecs, security algorithms, etc., that can be extended and how the extensibility mechanisms works.
6.     Define a set of media formats that must or should be supported by a client to improve interoperability.
7.     Define how non RTP datagram and byte stream data communication between the clients can be done securely and in a congestion safe way.
8.     Provide W3C input for the APIs that comes from the communication model and the selected components and protocols that are part of the solution.

Milestones:
Aug 2011 Architecture and Security and Threat Model sent to W3C
Aug 2011 Use cases and Scenarios document sent to W3C
Sept 2011 Architecture and Security and Threat Model to IESG as Informational
Sept 2011 Use cases and Scenarios for RTCWeb document sent to IESG as Informational
Dec 2011 RTCWeb and Media format specification(s) to IESG as PS
Dec 2011 Information elements and events APIs Input to W3C
Apr 2012 API to Protocol mapping document submitted to the IESG as Informational (if needed)

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Canada Gets Skype for iPhone, Not.

By Erik Lagerway | March 31, 2009 | apple, communications, IP Communications News, iphone, iphone, mobile sip, P2P VoIP, pbx, Skype, softphone, voip, VoIP for the Masses, wireless | 6 Comments
Skype Banned

Tom does some handy investigative work and finds out that Skype has been banned from use in Canada due to a legal issue around what seems to be a codec related patent.

Excerpt:
I then asked if other countries were affected or if it was just Canada and was informed it was just Canada. When asked whose patent it was or what category it involved (i.e. mobile VoIP), the representative told me, “I can’t go into many more details other than it’s codec related.”

That really bites. I was hoping to do some testing via Skype for iPhone on the new Skype for SIP on Response Point.

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Skype for SIP, it’s about time!

By Erik Lagerway | March 23, 2009 | aastra, communications, d-link, Erik Lagerway, IM & Presence with SIMPLE and XCAP, Media Engines, microsoft, mobile sip, Open Source, P2P SIP, P2P VoIP, phone services, Recording, response point, response point, SIP phone software, SIP Standards, Skype, Small Medium Business in Canada, small office phone systems, syspine, voip, VoIP for the Masses, Xten - Latest and Greatest | 4 Comments

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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.

Written by Erik Lagerway - Visit Website

P2P VoIP is Hot and Getting Hotter

By Erik Lagerway | September 25, 2005 | P2P VoIP | Comments Off

With Avaya’s purchase of Nimcat and eBay’s purchase of Skype I think it’s obvious that P2P VoIP is one hot topic with investors and press lately.

Something that surprises me is the lack of Video in these offerings. Neither Skype nor Nimcat offer video. Yes, Skype has said it is working on Video but it would seem as though the technology needs work. At the recent VON show Niklas Zennstrom was supposed to make his keynote via Skype’s new video component but it failed and hence no keynote.

I think P2P VoIP has merit but in order for it to really take off it must be standards-based and offer all the features.

If 2004 was the year of VoIP maybe 2005 is the year of P2P VoIP.

Written by Erik Lagerway - Visit Website
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