All posts in voip

Job: VoIP Network / Systems Administrator (remote)

I am looking for a kick ass VoIP Network Admin (remote worker).

Must-haves:
2-4 years of VoIP and Video network admin and development.
2-4 years of Networking background (TCP/IP, NAT Traversal, VoIP protocol, SIP protocol)
5+ years of Linux server experience
2-4 years of experience managing VoIP monitor and test tools
Codec experience in VP8, SILK, G.722, H.264 and H.264 SVC etc.
SIP protocol expertise Day-to-Day.

Email me your resume: erik@sipthat.com

Written by Erik Lagerway - Visit Website

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.

Written by Erik Lagerway - Visit Website

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?

Written by Erik Lagerway - Visit Website

So it begins. Skype for Asterisk falls.

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.

Written by Erik Lagerway - Visit Website

With a wave of the hand and, hey presto, the revenue is gone!

Andy has a point here, http://andyabramson.blogs.com/voipwatch/2011/04/did-magicjack-lose-its-magic.html. It’s very reminiscent of the old RLEC + VoIP strategy that allowed RLECs to profit from a loophole in the system to drive revenues around inbound calls subsidized heavily by the larger ILECs. A fund called the USF (Universal Service Fund) was a cash cow for certain RLECs and their strategic partners, that became a real hornet’s nest.

Written by Erik Lagerway - Visit Website

SIP + RTCWEB marriage in question?

The RTC WEB sessions in Prague made it pretty clear, to me at least, that everyone knows that SIP is broken and we also know that it’s not going away anytime soon. That being said it will likely not be the only protocol to be used in conjunction with RTC WEB.

I think it’s a consensus, at least of the IETF participants of the RTC WEB BOF, that we should not be discussing signaling protocols within RTCWEB at this early stage in the process of creating a WG (working Group) in the IETF.

It’s likely the correct approach. If we pigeon hole the community into using one protocol over another we are not really doing the future of communications any great service. In the same breath I also think it is important that we do not lose sight of the fact that the business world today runs on SIP and will continue to do so for some time to come.

The one thing that stuck out is the obvious gap that exists between what we have today and what we need in order to make RTCWEB a huge success, although I do know of a few companies that can move rather quickly when presented with a challenge as well so maybe it’s not such a big deal.

Prague was great, on many levels. It will be very interesting to see the progress we make between now and Quebec.

Written by Erik Lagerway - Visit Website