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VOIP & IRC - Virtual Attendance

Overview:  BTIP has Voice over Internet (VOIP) & IRC for realtime communication.

Connecting via IRC

We have a regular IRC channel #berkeleytip on freenode.net. Connecting to the IRC channel is the first thing you should do for every meeting. Here we help/advise/assist on anything, such as how to get your VOIP setup configured & working.  :)   Join the discussions, say "Hi" & introduce yorself, get assistance, have fun, and chat.  You may also join the channel 24/7/366 for discussion any time someone is there.

Connecting via VOIP

We use VOIP (Voice over Internet Protocol) technology to create a conference environment between people attending the Berkeley meetings in person, and everyone who connects from anywhere globally.  Get a VOIP headset, Read the VOIP instructions below, try the voice loopback test, & join the IRC channel for help in getting your VOIP connection working. :)


== IRC

Freenode channel #berkeleytip, irc.freenode.net


== VOIP - Contents:

0) Quick summary of connection configurations

There are several aspects to using VOIP with a computer:

1) Your client software,

2) Your client hardware,

3) Your network (& firewall if any)

4) The VOIP server system (hardware, sw & network). 

4a) BTIP's own VOIP Server

4b) Original Ekiga.net server

4c) Darkvoip.net server

5) Connecting via telephone from any regular or cell phone.


==========

=====  0) Quick summary of connection configurations

SFLPhone is our latest recommendation as the best VOIP software client to use on your computer.

http://www.sflphone.org/download.php

Ekiga or Zopier were the previous VOIP SW many people used on their computer.

Macintosh Client:

Windows Client:

Server info:

Test #:

BTIP conference server connection #'s  (Server name & number) : 

a) Primary: Berkeley, California -   ??Correct??  btip@169.229.15.141

b) Backup: Darkvoip, Texas?- sip:87654@darkvoip.net

c) Backup: Ekiga, France - sip:*987687654@ekiga.net

One can also connect via regular phone by dialing this USA number:

PSTN: +1.206.279.5528 ext 87654


You should also definitely Read this Great document about BTIP VOIP Configuration , on docs.google.com


===== 1) Client software

GNU(Linux), Ubuntu, BSD:

SFLPhone is our latest recommendation as the best VOIP software client to use on your computer.  There are two programs you need - the "core" program, & the client program (either for the GNOME or KDE desktop).  This is for computers running GNU(Linux).

http://www.sflphone.org/download.php

Previously, the most recommended sw was the Ekiga (Gnome Meeting) software, which is available for GNU(Linux), Macs & other personal computer OSs, or the Zopier client.

http://ekiga.org/index.php?rub=5

http://www.zoiper.com/

Macintosh client info:


We will be working to add Skype & regular phone dial in connection ability sometime this year.

There are two main communication methods for communicatig voice: SIP & IAX.


=====  2) Your client hardware

A VOIP headset is the best (clearest, less noise, less echo) device for connecting with the conference. There are two main kinds, depending on the connector: 1) USB - newer technology, 2) Audio mic & speaker mini-jack connector. These typically cost about $10 - $40. They can be purchased from many retail electronic stores (Radio Shack, Best Buy, office supply stores, etc) & online.

Headset recommendations:
http://www.buy.com/prod/kensington-bluetooth-stereo-headset-over-the-ear-kensington-bluetooth/q/loc/101/206684060.html

Looks like a fantastic price - 68% savings - off from $80.  includes mic.  Great for being unwired from your computer - can walk around & do things while still on BTIP voip.

===== 3) Your network (& firewall if any)

The SFLPhone & Zopier client software seem to be better at working through firewalls than the Ekiga client.  SFLPhone is the current recommended client.

===== 4) The VOIP server system (hardware, sw & network).

4a) New BTIP's own Asterisk Server

BTIP has a server located in Berkeley. We are continually working on improving the VOIP server software.  Currently we use the Asterisk VOIP system, but hope to move to the FreeSwitch system, due to its seemingly better design & features. Please send a message to the BTIP list if you have any questions.

sip:8000@169.229.15.141 ??

Account Information:

Connect using a SIP (e.g. Ekiga) or IAX2 client (e.g. Zoiper) using the following information. Currently tested working with: Ekiga 2.0.12, Twinkle, Xlite, Zoiper.

Server IP: 169.229.15.141

IAX2 user name: btip
IAX2 password: btip

SIP user name: btip
SIP password: btip


Client Settings for Asterisk Test Server:


(using an IAX2 account):

IP: as above
user name: as above
password: as above

Current Extensions On the Berkeley Asterisk Test Server:

Echo test: 600
Conference: 8000

Note: Once the profile information is entered, these extensions should be able to be dialed directly, e.g. simply dial 600 for the echo test. You won't need any prefixing, or '@' symbols in the dial string.

So, does that mean also that one can connect as:

sip:8000@169.229.15.141 ??


4b) Original Ekiga.net server

VOIP-Ekiga - Get your own VOIP client working
    Ekiga - do a loopback test with mic & headset

sip:5012345@ekiga.net   BTIP voip conf #
sip:500@ekiga.net           loopback test # for Ekiga.net
sip:501@ekiga.net           test only conf # use 5012345

== Ekiga.com Ekiga.net

Documentation http://www.ekiga.org/index.php?rub=3

Wiki http://wiki.ekiga.org/index.php/Main_Page

FAQ http://wiki.ekiga.org/index.php/FAQ

Services, Conference Server https://www.ekiga.net/index.php?page=services

Conference calling:
  http://wiki.ekiga.org/index.php/Fun_Numbers
    use conference numbers.


4c) Darkvoip.net server

[TODO: Add Darkvoip info. Conf #: ]

Is it sip:87654@darkvoip.net  ?

PSTN: +1.206.279.5528 ext 87654

===== 5) Connecting via telephone from any regular or cell phone.

PSTN: +1.206.279.5528 ext 87654  - DarkVOIP server



== Hopes for the Future


As BerkeleyTIP grows, we hope to make it a kind of web conference. To that end, here is the Wikipedia comparison of web conferencing software:
http://en.wikipedia.org/wiki/Comparison_of_web_conferencing_software

As you can see, only two projects are completely Free and Open Source:
https://www.webhuddle.com/ and
http://code.google.com/p/openmeetings/

If you are a developer, please see what your skills can contribute to these!


Other related VOIP technology

dukaUS Launches Free Conference Calling, Realizes It Won’t Make Money 

http://www.techcrunch.com/2008/11/11/dukaus-launches-free-conference-calling-realizes-it-wont-make-money/


http://freeconferencepro.com/


http://www.freeconferencecall.com/

http://gizmo5.com/pc/products/features/conference-calling/


Things to do - Suggestions
Make a way to see who is in the conference. Probably needs to be suggested upstream to ekiga.net. Thus far, it doesn't seem possible.

Improve hearing clarity using a pair of earbuds (or using a voip headset, such as the logitech 350).

--

Debian VoIP packages live on #debian-voip in OFTC . 
Tzafrir Cohen  http://tzafrir.org.il

http://www.oftc.net/oftc/