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Friday May 1st #vuc539  +FreeSWITCH Labor of Love

With expêrimental Android Meerkat stream: http://meerkatapp.co/voipusers/sch-a3ddb324-5e9f-4b46-b730-627397e97125 starting 20 minutes before
 
Broadcasting live on YouTube from the FreeSWITCH conference bridge.


#vuc539  
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Jason Baird

SIP & PBX  - 
 
Hi Everyone! I've got a quick question about SIP trunking. I've mostly been using IAX2 trunks for different things. They are easy to setup and seem to find their way around a firewall with no port forwarding on my part. If I wanted to SIP trunk two machines together over the internet, am I going to have to monkey with the firewall or will it work like the IAX2 trunks I've been using. Just curious. Thanks!
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Jason Baird's profile photoCorrado Mella's profile photo
7 comments
 
Here's where I stop. Never used IPKall, so can't help you there, sorry.

Guidelines are, if call can't be established it's a signalling problem (SIP, and it's a minefield), if no audio (in and/or out) it's firewall / ports or codec(s).

VoIP is a complex, multidisciplinary beast.

It demands you know everything about networks, telephony, electronics, psychoacoustics, change management and legislation, all up to date and on the tip of your fingers.

All to make two people talk between them, sitting in two adjoining cubicles 😒
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See what the industry has for you to do Push Notifications for your #WebRTC Web App http://bit.ly/1GGKjVX
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We recently published a research paper about our "Last N" approach to videoconferencing. It describes how it all works and shows our evaluation based on the implementation in Jitsi Meet and Jitsi Videobridge: http://goo.gl/piwXjy
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Randy Resnick (randulo)
moderator

Security & Fraud  - 
 
"The experts seized the household server and a personal desktop computer, both running Linux. Noticeably, they left the other two Windows machines."
Some of the volunteers who keep the Tor network running are raided for their services. "Richard" was one of them.
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Jeff Fincher's profile photo
 
Time to start up more exit nodes.
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Randy Resnick (randulo)
moderator

VUC Sessions  - 
 
This Friday
 
Friday May 1st #vuc539  is +FreeSWITCH compatible!
The FreeSWITCH team has been working nonstop on code and infrastructure improvements and are ready to share it. The new BETA is being released this week with video codec transcoding and video MCU functionality in mod_conference. The whole team will be broadcasting directly from FreeSWITCH to YouTube live and demonstrating some of the latest functionality.
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Randy Resnick (randulo)
moderator

Water Cooler  - 
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Here's the first draft of the service that I described few days ago.
This is a simple public service for testing VoIP call quality. It provides PSTN numbers which are terminated on FreeSWITCH servers. Anyone can call these numbers, and the server plays a standard speech test audio file and records the incoming audio. Then people can download this recorded audio ...
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Stanislav Sinyagin's profile photo
 
I will add a warning that the caller id is revealed to the public
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Randy Resnick (randulo)
moderator

Water Cooler  - 
 
In case you were wondering. We've had a couple manufacturers of these on way back when...
 
Excellent clear video educating SBC's
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Michael Graves's profile photoVlad Markov's profile photo
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+Michael Graves SBC is not my primary topic, I am using it because it is part of our infrastructure, so I can't present Cisco product on SBC subject with Power Point and the rest of marketing diagrams, simply because it is not my BU. I can only describe SBC solution as our recommended deployment infrastructure. As per life audience - cue cards ;-) to push more information.
As for small business, I don't think SBC is necessary, because it is intended for multi-locations, regardless Data Centers or on prem solutions. Most modern PBX and end points capable to handle STUN/TURN servers, which is removing necessity for port traversing
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Randy Resnick (randulo)
moderator

WebRTC  - 
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Alex Eleftheriadis's profile photo
 
Congrats!
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About this community

PLEASE READ: This Community is people, not pages or companies. Only individuals with real names are accepted. Members are very familiar with the technologies and need no intro level articles. Links to your commercial sites will be muted. Spam = ban. We know how VoIP & security works. Be aware of what is appropriate. All about SIP, IAX, pbx, WebRTC, Audio and Video Conferencing, XMPP, SMS, Mobile and fixed Telephony and anything else that can be sent over IP. Weekly VoIP Users Meetings Fridays at 12 Noon Eastern US Time. Call sip:vuc@vuc.me IRC: #vuc on Freenode.net Originally, VoIP Users Conference since 2007
On your screens
 
Looks like a great project for enabling web devs to use WebRTC. I can't wait to try it.
 
Kurento WebRTC media server

Browser to browser video streaming and great tutorials in Java, Node.js

"Kurento is a WebRTC media server and a set of client APIs making simple the development of advanced video applications.

Tutorial 1 - Hello world
Tutorial 2 - WebRTC magic mirror
Tutorial 3 - WebRTC one-to-many broadcast
Tutorial 4 - WebRTC one-to-one video call
Tutorial 5 - WebRTC one-to-one video call with recording and filtering

Kurento is distributed as Open Source Software basing LGPL v2.1 license. Visit Kurento github repo to get it."

#nodejs   #webrtc   #webdev  
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Paul Collins

SIP & PBX  - 
 
Integrating and recording ISDN calls together with VOIP calls.

Has anyone got experience of installing an ISDN line card into a VOIP server such as Asterisk?  Can anyone recommend an ISDN interface card e.g. the Sangamoa ISDN BRI Voice Boards?

Also what are people's experiences of VOIP over ADSL either hosted VOIP or running their own server? I have several ADSL lines one dedicated to VOIP and I only ever use one channel of VOIP at a time but I still suffer massive quality problems with not being able to hear the remote party. Thanks.
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Paul Collins's profile photoStanislav Sinyagin's profile photo
16 comments
 
Paul, see my contact details in my g+ profile. Let's define the scope of work and I'll make an offer.
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Ever thought how to wakeup your #WebRTC Web Application? http://buff.ly/1AbwSFX
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Stanislav Sinyagin

Water Cooler  - 
 
here's a test scenario that I used for non-voice software testing, but it can also be easily used for VoIP tests.
I needed to test some master-slave software in a situation that the master communicated to the slave over NAT (master's IP address was replaced with the firewall's external address), and then NAT w...
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From now on it is a lot easier to get started using OpenWebRTC in your applications:
We know that many have been itching to get started writing applications since the OpenWebRTC project was published just over 6 months ago. We also know that many have been struggling to build the many dependencies that go into making OpenWebRTC. So... We are happy to announce our first (pre-)release - OpenWebRTC 0.3.0! The main goal of this release is to publish binaries of the current state of OpenWebRTC so that you can download and...
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Is +Google​ starting to make the network providers just data providers? Maybe until such a time that they (Google) can cover the entire planet with their Wi-Fi? Also, do you think that +Google Voice has reached its potential by Google implementing this "phone number in the cloud" that can ring every device on every platform?
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Andy Abramson's profile photoRandy Resnick (randulo)'s profile photo
8 comments
 
Needs to degrade to satellite!
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+JamKazam CEO David Wilson joins #vuc538  to chat about this effort. There have been many attempts at building a platform for musicians to play live together over the Internet, but this one actually works. There's a lot of interesting things about this, lots of features to check out so if you're into music, please join us! 
This Hangout On Air is hosted by Voipusers TV Hangouts. The live video broadcast will begin soon.
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#vuc 538 - Jamkazam Live Jam over the Net
Fri, April 24, 12:00 PM
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Jason Baird

SIP & PBX  - 
 
Hi Guys! I've got a question and you've been so helpful in the past. Here's my situation.

I need some help regarding IAX2 trunks. In the class I'm putting together, each person will be playing with their own version of FreePBX. I've been successful on having multiple PBXs on the same local subnet. People can trunk to each other's PBX and get their phones to register. 

What I want to do is have them establish a trunk over the internet to another PBX I have waiting behind a second router. To do this with a single PBX from behind one routing device to another across the internet is trivial. IAX2 does everything without having to have anything forwarded through the firewall. It's beautiful. To do this with 2 or more different PBXs on the same subnet behind one routing device to another one PBX across the web is giving me a headache. 

I understand that IAX2 operates on port 4569, so I tried to specify in the peer details of one of the PBXs to run off of port 4570 (port=4570). I did the same to the far end. No dice.

I'm trying to do all this within FreePBX without breaking into and editing various Asterisk files. The reason is because the students are mostly physical security types who are taking the class as a VoIP primer.

If anyone has any suggestions, I would be greatly appreciative. Thanks!
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Corrado Mella's profile photoRandy Resnick (randulo)'s profile photo
5 comments
 
Exactly, +Corrado Mella. That's also why I reject the majority of requests to join the community. Either they are companies that want to spam their routes, or people who have never posted and have no visible interest in what we do here. The 800+ members know what they're talking about, or want to learn more. Both of these are why we are here!
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Randy Resnick (randulo)
moderator

WebRTC  - 
 
+Chad Hart had some thought about the +Jitsi news
This morning Atlassian announced it was acquiring BlueJimp. Chad examines why and what this means for WebRTC.
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Alex Eleftheriadis's profile photoRandy Resnick (randulo)'s profile photo
2 comments
 
Thanks for the references, +Alex Eleftheriadis
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IETF targeting NETVC to rule them all video codecs  http://bit.ly/1D9cFR0
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