Build FreeSWITCH from source on Ubuntu 12.04


FreeSWITCH has a fairly detailed Wiki page on download and installation. This post cuts to the chaff.

Execute the following commands from terminal in the order specified. You should be fine doing it in your home folder.

sudo apt-get install git-core build-essential autoconf automake libtool libncurses5 libncurses5-dev make libjpeg-dev pkg-config unixodbc unixodbc-dev zlib1g-dev libcurl4-openssl-dev libexpat1-dev libssl-dev libtiff4-dev libx11-dev unixodbc-dev zlib1g-dev libzrtpcpp-dev libasound2-dev libogg-dev libvorbis-dev libperl-dev libgdbm-dev libdb-dev python-dev uuid-dev bison autoconf g++ libncurses-dev speex libspeexdsp-dev libedit-dev libpcre3-dev

git clone https://freeswitch.org/stash/scm/fs/freeswitch.git

cd freeswitch

./bootstrap.sh

./configure

make

sudo make install cd-sounds-install cd-moh-install

Configuration files are located under /usr/local/freeswitch/conf, if you want to edit any.

Execute FreeSWITCH as superuser thus

sudo /usr/local/freeswitch/bin/freeswitch -nc

Remove -nc option to run in console mode.

To stop FreeSWITCH

sudo /usr/local/freeswitch/bin/freeswitch -stop

If you get “libspandsp.a: No such file or directoryerror when executing make after a make clean, execute the following commands and resume make.

cd libs/spandsp
make clean
make
cd ../..

To ensure a clean build, use git clean -f -x instead of make clean.

Kudos: Henrique Borges, Vitória Vasconcelos

VoIP calls from the browser using WebRTC and FreeSWITCH


There was a time when making phone calls from the browser would have meant installing a native extension. Thanks to WebRTC, we can now make phone calls from the browser. This post is my recent experiment with doing exactly that, using readily available open source components.

Let’s start by installing FreeSWITCH (FS). I am assuming a Windows based setup but Linux or Mac should also work. Once you have FS installed (I’m on 1.5.8b+git~20131213T181356Z~87751f9eaf~64bit) and sanity-tested, you’ll need to enable websocket support. This can be done by editing the configuration file <FS folder>\conf\sip_profiles\internal.xml so the the following line is uncommented

    <param name="ws-binding"  value=":5066"/>

You can also use secure websockets, I’ll leave that setup for a future exercise. At this point, restart the FreeSWITCH service.

The next step is to find a suitable browser-based SIP client. Luckily, there is exactly such a client provided by jsSIP, and you don’t even have to install it. Fire up your browser (I’m using the latest version of Chrome) and access that URL. Assuming that your IP address is 10.211.55.3, this is the information you can provide

Name: Your Name
SIP URI: sip:1000@10.211.55.3
SIP Password: 1234
WS URI: ws://10.211.55.3:5066

Hit ENTER and you’ll be taken to the dialer. Dial 9195 to make a call, FS will relay your voice back to you, after a five second delay. You’ll need to allow the browser to use your microphone.

sipML5 is another nice alternative to jsSIP. You’ll need to edit the WebSocket Server URL in expert mode.

Happy RTCing!

FFmpeg on Windows


A quick post to document how audio and video can be captured on Windows using FFmpeg into different container formats like MKV, MP4 and WebM.

List DirectShow devices

The following command lists the names of audio and video devices currently installed

ffmpeg -list_devices true -f dshow -i dummy

Capture from webcam and microphone

With container format mkv the default video codec is H.264 and audio codec is Vorbis. With the mp4 container format (change output.mkv to output.mp4) the video codec is H.264 and audio code is MPEG AAC. With the webm container format the video codec is the Google/On2 VP8 and audio codec is Vorbis.

ffmpeg -f dshow -i video="video device name":audio="audio device name" -r 25 -s 320x240 output.mkv

I added the -s (video size) and -r (video frame rate) options because with mkv and mp4 I was getting lots of dropped frames.

References
ffmpeg directshow

Video streaming using jpeg encoding


Here’s an example of a GStreamer pipeline that produces a less CPU intensive and low latency video stream using jpeg encoding. Audio in vorbis is muxed, along with the video, into a matroska stream. I have tested this on Ubuntu 11.04.

gst-launch v4l2src decimate=3 ! video/x-raw-yuv,width=320,height=240 ! jpegenc ! queue2 ! m. alsasrc device=hw:2,0 ! audioconvert ! vorbisenc ! queue2 ! m. matroskamux name=m streamable=true ! tcpclientsink host=localhost port=9002

A server can stream it with a content type of video/x-matroska. Most browsers will not play it directly, but external plugins can be used.

Adjusting attributes of v4l2src and vp8enc elements for video conferencing


Video conferencing is real time in nature. The default encoding parameters of vp8enc element of GStreamer are not always appropriate. Let us start with the following pipeline

gst-launch v4l2src ! video/x-raw-rgb,width=320,height=240 ! ffmpegcolorspace ! vp8enc ! vp8dec ! ffmpegcolorspace ! ximagesink sync=false

The CPU usage, on a PandaBoard with Ubuntu 11.04, is close to 100% (since there are 2 cores, that translates to 50%).

Now, modify the pipeline as follows

gst-launch v4l2src decimate=3 ! video/x-raw-rgb,width=320,height=240 ! ffmpegcolorspace ! vp8enc speed=2 max-latency=2 quality=5.0 max-keyframe-distance=3 threads=5 ! vp8dec ! ffmpegcolorspace ! ximagesink sync=false

Note the decimate attribute of the v4l2src element, and the attributes speed, max-latency, max-keyframe-distance, threads and quality of the vp8enc element. With these changes the CPU usage drops to 40% and the video playback is more real time.

HP sells Visual Collaboration business; Google announces WebRTC


In two unrelated events, HP has sold its Visual Collaboration business to Polycom, and Google has announced the WebRTC initiative. Polycom has also announced an industry wide initiative called The Open Visual Communications Consortium, whereas the approach that Google is taking is to support synergistic initiatives from the W3C, IETF and the WHATWG. Google also released IP they acquired from Global IP Solutions under an open project.

The Architecture of Open Source Applications


A new book regarding the architecture of several prominent open source projects has been released. They have the whole book online at http://www.aosabook.org/. It can be purchased from Lulu.com and Amazon.com. Considering that they’ll be donating the royalties from the proceeds, buying from Lulu.com makes the most sense, see the breakdown of the proceeds of sale at their site for more details.

Here’s a summary of some chapters I read, for their relevance to my current line of work.

Chapter 1 – Asterisk

VoIP is hard, and Asterisk does it well. I have used OpenSIPS (then OpenSer) to implement an OMA PoC client and a prototype server. That work was scrapped but it taught me enough to know that VoIP projects take significant effort. The Asterisk architecture has evolved over ten years and is quite elegant. Even so, they are working hard on making Asterisk scale. A chapter worth reading.

Chapter 10 – Jitsi

Jitsi is an open source communicator written in Java. They use a plugins based architecture based on OSGi. I have briefly been involved with OSGi as a TCK developer, so I am not surprised by the fact that its adoption is now widespread. A pity though that, in my opinion, Java has lost its momentum. New smartphones don’t run Java byte code.

It is hard to use Java in the problem domain that Jitsi serves. Accessing media devices from Java using native code is a pain, so is having to render media with good performance. Using OSGi can help a bit, native code can be isolated into decoupled services. Read the chapter for some excellent design tips on protocol, device, media codec and stream handling. Even if your application is not written in Java, this separation of concerns is by itself very useful.

Chapter 20 – Telepathy

Telepathy is a service oriented approach to providing communications capabilities to all Linux apps. It uses the D-Bus to link together different components (and clients). Using Telepathy along with Farsight2, which is a media streaming framework based on GStreamer, one can build VoIP applications for Linux. Empathy, the default messaging program on Ubuntu, is one prominent application that uses this infrastructure.

Conclusion

Each chapter is a treatise on the architecture of a particular open source project, but it is disheartening to observe the lack of reuse between them. Also, there are some significant open source applications missing:

  • Apache HTTPD
  • The Linux Kernel
  • WebKit engine or the Chrome browser