Howto setup Asterisk for recording a podcast over the Internet

April 6, 2008

Some friends and I are planing a to make a podcast and as we IT guys we needed something to support our distributed recordings over the Internet. One of my friend lives about 200km away but also with the others it would be not that easy to get all into one room at the same time. When we looked around for various ways to do distributed recordings we found mostly Skype howto’s and we knew only a few podcasts which did the recording with Asterisk (in a not that good audio quality I think). But we didn’t find what we really wanted and so I started to look into that topic. At first I wrote our requirements down:

the recording should be done centrally on a server automatically without user interaction (so it is not forgotten and to minimize the lag and sound quality problems)
the recorded files should be easy available as OGG files to all podcast members after the recording, as the post production is maybe not done always by the same people
various client operation systems and VoIP clients should be supported, therefore an open standard protocol should be used
possibility to record each participant separately to allow changes in the volume or exclusively applying audio filters
it should be possible to invite guest for interviews via the same system without requiring more than a VoIP client which support the choosen protocol. (no registration somewhere)
optional it should be possible to connect the system to the POTS (plain old telephone service) for interviews with people which cannot use a VoIP client.
And as a requirement by my knowledge and existing infrastructure, the server should run on one of my Linux servers I’ve running in a big data center, within a OpenVZ virtual environment.

After some research I decided to go with Asterisk. This howto describes what I’ve done to reach the above goals. After the completion of this howto you should have following:

A SIP server where you participants can connect to and talk with each other.
As soon as they go into a special virtual conference room everything they say will be recorded
After they leave the conference room a background process will reencode the recorded WAV file to OGG and make it available via web.
Every participant gets its own OGG file with the starting timestamp in the filename, you need to use the recording at the correct place in the post production.
The inclusion of a SIP provider with connection to the POTS network is not described in this howto as there are may others describing it.

Some points of this howto are specific to my OpenVZ setup and the chosen distribution but most are generic and should work for any setup. Anyway here is the software I used.

OpenVZ for virtualization
Ubuntu 8.04 Hardy (x86_64) as distribution for the virtual environment
Asterisk and Zaptel for VoIP part (Ubuntu packages)
sox for translating the WAV files to OGG (Ubuntu package)
lighttpd as small and fast web server for the OGG files (Ubuntu package)
Twinkle as SIP client under Linux. (apt-get install twinkle)

Part 1: Hardware node setup

You can ignore that part if you don’t use OpenVZ and your kernel/distribution comes with ztdummy modules. The hardware node in my case runs on a Centos4 (x86_64). This is important as Asterisk needs the ztdummy kernel module, which comes with zaptel, for the meetme Asterisk module which is used for the conference rooms. As it is not possible to load kernel modules within a VE (virtual environment) (that’s a security feature!) I needed it on my hardware node. As the kernel of the hardware is a OpenVZ patched kernel and also Centos 4 does not come with a ztdummy module anyway, I needed to compile it.

I used the same version of zaptel as Ubuntu 8.04 does and it is also very important that you use a 64bit VE if you hardware node is 64bit, otherwise the device cannot be accessed correctly.

The install is quite straightforward

# tar xzf zaptel-1.4.8.tar.gz # cd zaptel-1.4.8 # ./install_prereq test

Install the required packages and then continue with:

# ./configure # make # make install

but no â€œmake configâ€, as we don’t need init scripts or that stuff. Now load the kernel module with modprobe ztdummy (and make sure that this is done after very boot, before the VEs start). Make sure the device is working with:

# ztcfg -v

The output should be something like:

Zaptel Version: 1.4.8 Echo Canceller: MG2 Configuration ====================== 0 channels to configure.

At last we need the VE be able to access the ztdummy device, so we need to tell openVZ this.

# for x in `ls /dev/zap`; do /usr/sbin/vzctl set XXX --devnodes zap/${x}:rw --save; done

Replace the XXX with the ID of your podcast VE. Now we’re done with the hardware node and we can take a look at the user space stuff.

Part 2: Virtual Environment setup

At first we install the packages we need with following command:

# apt-get install asterisk-h323 asterisk-doc speex vpb-utils sox libsox-fmt-all lighttpd zaptel

Now we configure zaptel and check if it works:

# genzaptelconf # ztcfg -v # ztcfg -d

No error should be given. If a device is not found check if they got created by vzctl. After that make the devices in /dev/zap read and writable for the asterisk user:

# chown root:asterisk /dev/zap/* # chmod 660 /dev/zap/*

Now we can work on the Asterisk configuration. We set following values in /etc/default/asterisk:

RUNASTERISK=yes AST_REALTIME=no

The real time stuff does not work in a VE and gives audio problems. Now we need to do some configuration for NAT users in /etc/asterisk/sip.conf:

externip = you're_external_IP ; this is needed as asterisk has problem with the venet0 stuff otherwise localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation localnet=169.254.0.0/255.255.0.0 ;Zero conf local network

nat=yes qualify=yes canreinvite=no

After this global setup we configure for each of our podcasters one section, as shown here:

[firstPodcaster] ; this is also the user name type=friend context=sip secret=the_password_of_this_user callerid="Your_Name" <1> ; it is recommended to use no spaces here, as we use this as part of the filename. You need the â€œ, â€œ and < ,> exactly as show here host=dynamic dtmfmode=info disallow=all allow=alaw callingpres=allowed_passed_screen

We only support alaw so every client uses G.711a and we don’t need to translate. I believe in the US you need to use G.711u and therefore ulaw. Now we need a conference room for which we add following line to /etc/asterisk/meetme.conf:

conf => 10 conf => 20

Now we need to tie that together with the dial plan in /etc/asterisk/extensions.conf:

[globals] MONITOR_EXEC=/usr/local/bin/wavIn2ogg.sh

And add following section:

[sip]

; so we can talk directly with another exten => 31,1,Dial(SIP/firstPodcaster,20,tr) exten => 32,1,Dial(SIP/secondPodcaster,20,tr) exten => 33,1,Dial(SIP/thirdPodcaster,20,tr)

; this conference room is not recorded, for preparations exten => 10,1,Answer exten => 10,2,Wait(1) exten => 10,3,Meetme(10,s)

; this conference room records automatically exten => 20,1,Answer exten => 20,2,Wait(1) exten => 20,3,Set(CALLFILENAME=podcast_X_${CALLERID(name)}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) exten => 20,4,Monitor(wav,${CALLFILENAME},m) exten => 20,5,Meetme(20,s)

; from the demo code â€“ useful to look at the quality of your connection ; Create an extension, 60, for evaluating echo latency. exten => 60,1,Playback(demo-echotest) ; Let them know what's going on exten => 60,n,Echo ; Do the echo test exten => 60,n,Playback(demo-echodone) ; Let them know it's over exten => 60,n,Goto(s,60) ; Start over

Now you can restart Asterisk and connect with you SIP client. Call 60 to check if the audio stream works in both directions (try it without firewall on the Asterisk server if don’t hear anything). After that go into the conference room 10 with 1 or 2 friends and test it. If that all works you can work on the recording stuff. Create following script /usr/local/bin/wavIn2ogg.sh (don’t forget chmod 755):

#!/bin/bash # wavIn2ogg.sh - creates ogg of the input mono stream # used for recording each participant in a meeting room separately # Written by Robert Penz

SOX=/usr/bin/sox NICE=/usr/bin/nice

InFile="$1" OutFile="$2" OggFile=`echo $3|sed s/.wav//`.ogg FinalDir=/var/www/production

#test if input files exist test ! -r $InFile && exit test ! -r $OutFile && exit

$NICE -n 19 $SOX -t wav "$InFile" -t vorbis "$OggFile"

#remove input files if successfull test -r "$OggFile" && rm "$InFile" "$OutFile"

# at last set the permissions and move it in an atomic way chmod 644 "$OggFile" mv "$OggFile" "$FinalDir/."

Create the directory /var/www/production and set the correct permissions:

# mkdir /var/www/production # chown asterisk:www-data /var/www/production

Now go the the conference room 20 and say something and disconnect. If it worked you should see with your browser under http:///production/ the recorded OGG file(s). If there are none, take a look at /var/spool/asterisk/monitor/ if there are 2 WAV files. If so call the wavIn2ogg.sh script by hand and look for any errors.

So thats the end of the story â€“ you’ve now a system for recording podcasts over the internet in a cool way! Any comments, ideas or questions? Post them here.

In HowTo, Linux | 6 Comments

6 Comments »

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[…] Check out this article: Howto setup Asterisk for recording a podcast over the Internet. […]

Pingback by Richard Spindlerâ€™s Weblog » Blog Archive » Lazy Podcast Setup — April 12, 2008 #
I can’t seem to get the script working. I get an “/usr/bin/sox stio: Can’t open input file `’: No such file or directory
chmod: cannot access `.ogg’: No such file or directory
mv: cannot stat `.ogg’: No such file or directory” error when i try to run the script

Comment by andy — May 12, 2008 #
Andy, if you don’t configure the wavIn2ogg.sh script in extensions.conf, can you find the wav files after making a conference call? If not, than there is an error in you asterisk config (maybe the files are created at an other location). If there are files call the script by hand, to check if it works.

Comment by admin — May 13, 2008 #
Yes i have the 2 wav files but and i run the script from the CLI and it comes up with that error still

Comment by andy — May 14, 2008 #
With which parameters? You need 3 parameters, which are used by asterisk. Look at /var/log/asterisk/messages

there should be something like this (if the script is called by asterisk):

NOTICE[13748] res_monitor.c: monitor executing /usr/local/bin/wavIn2ogg.sh parameter1..3

try to call the script with the same parameters.

Comment by admin — May 14, 2008 #
[…] written a Howto for recording a podcast with asterisk. This setup is working for our podcast (in German) for some time now, but we wanted to boost our […]

Pingback by Local audio recording to boost the audio quality for asterisk recording | Robert Penz Blog — November 23, 2008 #

Howto setup Asterisk for recording a podcast over the Internet

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