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SIP:200901@login.zipdx.com (g722, g711) -
Skype:vuc.me and ld.vuc.me - IRC #vuc
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WebRTC promises to disrupt not only the telecommunications industry but the OTT establishment too, by making real-time audio and video ubiquitous and free on the web. Crocodile RTC Technical Director Peter Dunkley explains what they’re doing in this session.
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This coming Friday marks the third annual appearance of Doug Mohney, Editor-In-Chief of HDVoiceNews. Doug recently completed the latest edition of TMC’s annual HDVoice Report detailing the state of HDVoice deployment in the world.
There are myriad factors impacting the broad roll-out of HDVoice. Does anyone support HDVoice over LTE in mobile space? Does anyone actually enjoy HDVoice on T-Mobile? Will CableCo’s leverage their strength in residential phone service to deploy HDVoice in a big way? We’ll go to the man who makes it his business to know with these questions and more.
And of course, we will walk-the-walk as we talk-the-talk, conducting this call in HDVoice via ZipDX.
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Thomas Quintana, Telestax Director of Cloud Engineering, joins us to talk about TelScale RestComm, a Cloud Communications Platform to build voice, fax and text messaging applications, using mainstream web development skills. TelScale RestComm is a turnkey Telco 2.0 SaaS solution ready to deploy in your cloud based on the Twilio simple and effective APIs and 100% Open Source built on the next generation TelScale Cloud Communications platform.
TeleStax provides Open Source Communications software and services that facilitate the shift from legacy SS7 based IN networks to IP based LTE and IMS networks hosted on private (on-premise), hybrid or public clouds. TeleStax products include JSLEE, SS7 Resource Adaptors, SIP Servlets, Rich Multimedia Services, Presence Services/RCS, Diameter/AAA Services, XMPP Services, REST, WebRTC and others.
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What is DNSSEC (Domain Name System Security Extensions) and why is it important enough to send the likes of Mr. York around the globe? This and more Friday on VUC. The graphic below is from an analysis by DNSViz.
How do you know that you are in fact connecting to the correct VoIP system when you initiate a call or launch a client? What if someone redirected you to a different server? How would you know? Are there ways that we can improve the security of VoIP/UC systems? We rely on the Domain Name System (DNS) to resolve domain names to IP addresses and connect our computers to the correct destinations – but there are some weaknesses there that can be exploited. The “DNS Security Extensions” (DNSSEC) provide a mechanism to cryptographically ensure that the information you are getting out of DNS is the same information the operator of a domain put into DNS. A companion protocol, DANE, provides a mechanism to publish TLS/SSL certificates (or fingerprints of certificates) in DNS and add a stronger layer of trust to certificates. Both DNSSEC and DANE can potentially have a role to play with VoIP/UC.
In this VUC episode, Dan York will explain what DNSSEC and DANE are all about, how they work and will open a discussion about how they might be used within VoIP/UC to strengthen the security of communication. He’ll also talk about the DNSSEC support included in both the latest version of the Jitsi softphone and the Kamailio server. He’s looking for feedback and really just trying to raise questions around how all this work happening in the DNS area could benefit what we’re doing in the world of VoIP / UC / RTC / whatever. What could we do in VoIP if we had a global PKI we could use? Please come with your questions and ideas.
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As we have no guest this week, I thought we could do a few experiments. (Yeah, like the entire conference isn’t an experiment every week!)
I will post a URL to join the Hangout this week at around 11:30AM Eastern Time. Please join and let’s see if it’s true that Hangout can go to 15 people in Apps.
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