VoIP


To use VoIP you need a VoIP client and an account with a VoIP connection service provider.

Skype - don't use!

I looked at Skype, who provide both a client and a connection service, but rejected them because:

sipgate

sipgate provide a connection service. They charge (cheaply) for calls to landline numbers, but everything else is free (including a landline number for incoming calls).

They sell hardware telephones that connect to your ADSL line. They also supply a free software client, but for Windows only.

However, since they use SIP, you should be able to use any VoIP client that supports SIP.

They appear to be a German company with a strong presence in the UK (good UK website and wide selection of UK telephone numbers to choose from).

Registration is easy - just follow the instructions on the site.

Ekiga - couldn't get to work!

Ekiga (formerly known as GnomeMeeting) is an open-source VoIP client, which is intended (as its old name implies) for use with Gnome.

It's available as a Slackware package from linuxpackages, but has several dependencies. I ended up downloading and installing the following from linuxpackages:
and also the following from slackware.com:
Obviously, there may be later versions of these packages available now.

When I tried running Ekiga, I got a window pop-up with the error "Gconf key error ..." and some suggested URLs that might help diagnose the problem (FAQ at http://www.ekiga.org/,  troubleshooting section at http://www.gnome.org/projects/gconf/, and mailing list archives at http://mail.gnome.org).

The first URL (Ekiga FAQ) suggests doing:
   chmod -R 4755 /etc/gconf/gconf.xml.defaults/
   killall -9 gconfd-2
but that didn't help. Couldn't find anything useful at the other URLs or from a Google search - so gave up!

Twinkle

Twinkle is another open-source VoIP client. This one has to be built from source, and also has several dependencies, some of which have to be built from source too.

Four dependencies are available as Slackware packages:
Having downloaded and installed those, on to the source packages. The following instructions use checkinstall to build Slackware packages from the source downloads. I've put all the Slackware packages that I built in this way in /usr/local/packages on sage, so it should be possible to simply install those packages instead of downloading and building the sources.

1. Download and install commoncpp2-1.5.0.tar.gz: 
tar xvf commoncpp2-1.5.0.tar.gz
cd commoncpp2-1.5.0
./configure
make
mkdir doc-pak
cp -a doc config.log AUTHORS COPYING COPYING.addendum ChangeLog INSTALL INSTALL.w32 NEWS README THANKS TODO doc-pak
vim description-pak
checkinstall

2. Download and install ccrtp-1.5.0.tar.gz:
tar xvf ccrtp-1.5.0.tar.gz
cd ccrtp-1.3.6
./configure
make
mkdir doc-pak
cp -a doc config.log AUTHORS COPYING COPYING.addendum ChangeLog INSTALL NEWS README THANKS TODO doc-pak
vim description-pak
checkinstall

3. Download and install libzrtpcpp-0.9.0.tar.gz:
tar xvf libzrtpcpp-0.9.0.tar.gz
cd libzrtpcpp-0.9.0
./configure
make
mkdir doc-pak
cp config.log AUTHORS COPYING README doc-pak
vim description-pak
checkinstall

4. Download and install ilbc-rfc3951.tar.gz:
tar xvf ilbc-rfc3951.tar.gz
cd ilbc-rfc3951
./configure
make
mkdir doc-pak
cp config.log AUTHORS COPYING ChangeLog INSTALL NEWS README doc-pak
vim description-pak
checkinstall
5. Download and install twinkle-0.9.tar.gz:
tar xvf twinkle-0.9.tar.gz
cd twinkle-0.9
cp /usr/lib/qt-3.3.4/lib/qt-mt.pc /usr/lib/pkgconfig    # should have happened as part of Qt installation???
./configure --without-kde
make
mkdir doc-pak
cp config.log AUTHORS COPYING ChangeLog INSTALL NEWS README THANKS doc-pak
vim description-pak
checkinstall

You should now be able to run twinkle (but you may want to look at the README and/or the start of the manual first).

When you first run twinkle, you have to create a user profile. You are given a choice of a wizard or a profile-editor to do this. I chose the wizard (which already knows about sipgate), but you can easily change things afterwards using Edit->User Profile.

My settings (mainly from "Connection Information" on "My Account" page at sipgate website) were:
Some other twinkle settings under Edit->System Settings that I changed:

Logging and recording calls

Twinkle can be configured (under Edit->User profile->Scripts) to call a script or program at various points during a call. See the manual for details.

In order to log calls, I have configured twinkle to call the following script for all trigger conditions (except call-release - see below):

#! /bin/bash

# This script is called by twinkle at various points during a call.
# It logs information about the call.

TWINKLE_DIR=~/twinkle/
SCRIPT_DIR=$TWINKLE_DIR/scripts
CALLS_DIR=$TWINKLE_DIR/calls
LOG_FILE=$CALLS_DIR/log

if [[ $TWINKLE_TRIGGER == 'in_call' || $TWINKLE_TRIGGER == 'out_call' ]]
then
echo >> $LOG_FILE
date +%c >> $LOG_FILE
echo "From: $SIP_FROM" >> $LOG_FILE
echo "To: $SIP_TO" >> $LOG_FILE
fi
echo " $(date +%H:%M:%S) $TWINKLE_TRIGGER" >> $LOG_FILE

Calls are recorded by configuring the audio devices (Edit->System settings->Audio) as follows:

Speaker: ALSA: Other device
Other device: tee:"plughw:Headset",/home/fog/twinkle/calls/spk.raw
Microphone: ALSA: Other device
Other device: tee:"plughw:Headset",/home/fog/twinkle/calls/mic.raw

and configuring twinkle to call the following script for the two call-release conditions:

#! /bin/bash

# This script is called by twinkle when a call ends.
# It saves the recording of the conversation.
   
TWINKLE_DIR=~/twinkle/
SCRIPT_DIR=$TWINKLE_DIR/scripts
CALLS_DIR=$TWINKLE_DIR/calls

$SCRIPT_DIR/log.sh

cd $CALLS_DIR
   
time=$(date +%Y%m%d-%H%M%S)
mv mic.raw mic-$time.raw
mv spk.raw spk-$time.raw

sox -swr 8000 mic-$time.raw -c 2 mic-$time.wav pan 1
sox -swr 8000 spk-$time.raw -c 2 spk-$time.wav pan -1
soxmix -v 1.0 mic-$time.wav -v 1.0 spk-$time.wav $time.wav
lame --quiet -V 2 $time.wav $time.mp3
rm -f mic-$time.raw spk-$time.raw mic-$time.wav spk-$time.wav $time.wav

This method of recording calls is based on suggestions from a thread in the twinkle form.

NAT and firewall

The SIP and RTP protocols used by VoIP have problems getting though the NAT (Network Address Translation) that allows us to have several computers sharing the same public IP address.

There are two methods of dealing with this:
The STUN method seemed to work for a single client without any change to firewalls on fennel or the laptop. But it didn't work reliably for two clients (on different machines).

So we have to use the first method, which requires a bit more setup. We need to configure:
Fennel configuration:
Service Type Ports
SIP1 (UDP) 5060 - 5061
RTP1 (UDP) 8000 - 8005
SIP2 (UDP) 5062 - 5063
RTP2 (UDP) 8006 - 8011

Service Name Action LAN Server IP address WAN Users Log
RTP2 ALLOW always 192.168.183.13 Any Never
SIP2 ALLOW always 192.168.183.13 Any Never
RTP1 ALLOW always 192.168.183.7 Any Never
SIP1 ALLOW always 192.168.183.7 Any Never

Laptop firewall configuration - use following line in /etc/rc.d/rc.firewall:

PERMIT="192.168.183.0/24 5060-5063/udp 8000-8011/udp"

Twinkle configuration:

DTMF

If you have to use the keypad when you are in a call (e.g. to select from a menu, enter a number, etc.) then your keypresses are transmitted using "DTMF". There seem to be three methods of doing this with SIP:
Twinkle supports the first two options.

It's not clear what sipgate supports, but their voicemail doesn't work with twinkle. Calls via sipgate to ordinary phones work with "in-band", but not with RFC2833-DTMF.

There's a long thread on this issue in the twinkle forum. It sounds as though the "INFO" method might be added to twinkle fairly soon.

For now, though, configure twinkle Edit->User Profile->RTP->DTMF->DTMF Transport to "Inband".

UPDATE 1 Oct 2006: There's a new version (0.9) of Twinkle that supports INFO DTMF. It seems to work initially with sipgate voicemail, but then stops. I've switched back to using Inband... (but that also seems to stop working part way through a call).

codecs

A codec is used to encode/decode speech in order to send it digitally.

sipgate supports the following codecs:
Twinkle supports:
They seem to end up agreeing to use G.711A  - so sipgate's preference ordering seems to take priority over Twinkle's.

There is useful information here and a comparison table here.

G.711 is the standard codec, used over landlines. It uses no compression, so there is no encoding/decoding delay and loss of quality due to compression. It is, however, expensive on bandwidth (64kbps).

iLBC and GSM are lower-bandwidth (around 13kbps) than G.711. GSM was designed for mobile phones. iLBC is specifically designed for the internet, and is supposedly more tolerant of lost packets - so it might be the best choice when the bandwidth/reliability of the connection is dubious (like over my wireless link!). G.711 should provide better quality when the bandwidth is available.