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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

Skype’s email to me re: Skype for Asterisk

By Erik Lagerway | May 25, 2011 | asterisk, ietf, microsoft, P2P SIP, sip, SIP Standards, Skype, VoIP Standards | 1 Comment

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?

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Open and secure alternative to Skype

By Erik Lagerway | May 25, 2011 | asterisk, communications, ietf, IP Video, itsp, microsoft, mobile sip, mobile voip, Open Source, pbx, phone services, phone system, Secure VoIP, sip, SIP Standards, Skype, Small Medium Business in Canada, softphone, voip, VoIP for the Masses, Vonage | Comments Off

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?

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How much bandwidth do I need for Response Point? G.711 vs. G.729

By Erik Lagerway | April 27, 2009 | aastra, aastra, Canadian VoIP Startups, communications, d-link, microsoft, mobile sip, pbx, phone services, response point, response point, SIP phone software, SIP Standards, Small Medium Business in Canada, small office phone systems, syspine, voip | 1 Comment
dsp-chip

G.711 is the default audio CODEC for most Response Point phones and requires approximately 90Kbps bandwidth upstream (your voice going out) and 90Kbps bandwidth downstream (your caller’s voice coming in).

To calculate peak usage take the peak concurrent callers x 90Kbps. For example: 5 concurrent calls x 90Kbps = 450Kbps is the required bandwidth for each direction. Keep in mind, this does not account for VPN usage for remote users or voice mail to email etc.

As an example, if you have a 1Mbps ADSL connection from your service provider, you might have an upstream bandwidth of approximately 700 Kbps. A conservative approach is to estimate just over half of the upstream bandwidth is available, ISPs generally over-sell their bandwidth. In this case, you could safely support 4 simultaneous G.711 calls if you were not doing anything else (e.g. downloading email, listening to online radio, downloading large files, etc.) on that connection.

The SMB Digital Voice network also supports G.729, which uses approximately 20Kbps bandwidth upstream (your voice going out) and 20Kbps bandwidth downstream (your caller’s voice coming in) for each call. G.729 provides very good call quality while minimizing bandwidth usage. The only noticeable difference would likely arise during on-net calls (calling other users on the SMB Phone network). G.711 offers a higher quality on-net call because G.711 does not compress audio, but as soon as the the call is handed off to the PSTN the call quality between G.711 and G.729 is hardly noticeable.

G.729 offers some real benefits, the most obvious is the 400% decrease in bandwidth capacity requirements. G.729 also handles Jitter more efficiency during times where low bandwidth / high congestion would likely render a similar call using G.711 unintelligible.

You can force your phone to use G.729 on Response Point handsets but some are harder to configure than others. For example, on Aastra 675x phones the global SIP settings are grayed out out via Javascript on page load making it tough to set the codec.

As a general rule of thumb, we like to recommend an independent broadband connection that you can use for Response Point. You may want to acquire a router that has dual WAN link failover, VPN Server (for remote sites) and some QOS traffic shaping functionality.

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Response Point VPNs and Remote Workers

By Erik Lagerway | April 22, 2009 | aastra, aastra, canadian startups, Canadian VoIP Startups, communications, d-link, microsoft, pbx, phone services, response point, response point, SIP phone software, SIP Standards, Small Medium Business in Canada, small office phone systems, syspine, syspine, vancouver, Vancouver Startups, voip, VoIP Standards | Comments Off

I wrote an article over at the SMB Phone blog on Response Point VPNs and remote workers. If you are having some issues with VPNs and Response Point this might help.

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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
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