Onnowpurbo: New page: This is the most important part as it’s where you’ll decide which features to support/enable. The Telepresence system use Doubango VoIP Framework (requires SVN r989 or later) and it...

2014-02-18T20:54:56Z

New page: This is the most important part as it’s where you’ll decide which features to support/enable. The Telepresence system use Doubango VoIP Framework (requires SVN r989 or later) and it...

New page

This is the most important part as it’s where you’ll decide which features to support/enable. The Telepresence system use Doubango VoIP Framework (requires SVN r989 or later) and it’s highly recommended to rebuild it if you already have an old version installed because of some new and required features.

This section explains how to build the product on CentOS64 but could be easily adapted for any Linux, Windows (MinGW) or OS X (MacPorts).

For any issue, please ask on the telepresence developer group.

Preparing the system
Building thirdparties libraries
Required libraries
Building libsrtp
Building OpenSSL
Building libogg, libvorbis and libtheora
Building libspeex and libspeexdsp
Building YASM
Building libvpx
Building opencore-amr
Building libopus
Building libgsm
Building g729
Building iLBC
Building x264
Building libfreetype
Building libfaac
Building FFmpeg
Building OpenAL Soft
Building openOffice
Notes
Known Issues
Building Doubango
Building the Telepresence system
Installing the configuration and fonts files
Testing

Preparing the system

sudo yum update
sudo yum install make libtool autoconf subversion git wget cmake gcc gcc-c++ pkgconfig

Building thirdparties libraries
Required libraries

Always: libspeexdsp, FFmpeg and Doubango
For WebRTC clients: libspeexdsp, libsrtp, libvpx, OpenSSL 1.0.1+(Firefox only), FFmpeg and Doubango

... all other libraries are optional.
Building libsrtp

libsrtp is optional unless you want to use WebRTC SIP clients. It’s highly recommended. The WebRTC Telepresence demo client requires a system with SRTP enabled.

git clone https://github.com/cisco/libsrtp/
cd libsrtp
CFLAGS="-fPIC" ./configure --enable-pic && make && make install

You should not use any libsrtp package because the latest dev version is required and building the source by yourself is highly recommended.
Building OpenSSL

OpenSSL is required if you want to use TLS, WSS (Secure WebSocket) or DTLS-SRTP (also requires libsrtp). OpenSSL version 1.0.1 is required if you want support for DTLS-SRTP which is mandatory for WebRTC implementation from Mozilla (Firefox Nightly or Aurora).

This section is only required if you don’t have OpenSSL installed on your system or using version prior to 1.0.1 and want to enable DTLS-SRTP. A quick way to have OpenSSL may be installing openssl-devel package but this version will most likely be outdated (prior to 1.0.1). Anyways, you can check the version like this:

openssl version

To build OpenSSL:

wget http://www.openssl.org/source/openssl-1.0.1c.tar.gz
tar -xvzf openssl-1.0.1c.tar.gz
cd openssl-1.0.1c
./config shared --prefix=/usr/local --openssldir=/usr/local/openssl && make && make install

Some known issues when you have more than one openssl libraries installed is discussed at https://groups.google.com/forum/#!topic/opentelepresence/JctxtEyW-dg (see comment 4).
Building libogg, libvorbis and libtheora

These libraries are optional unless you want to use .webm or .mkv containers. You can install the devel packages (recommended):

sudo yum install libogg-devel libvorbis-devel libtheora-devel

Or build the source code by yourself:

--This section intentionally left blank--

Building libspeex and libspeexdsp

libspeex (audio codec) is optional but libspeexdsp (audio resampler, jitter buffer…) is required. You can install the devel packages:

sudo yum install speex-devel

Or build the source by yourself:

wget http://downloads.xiph.org/releases/speex/speex-1.2beta3.tar.gz
tar -xvzf speex-1.2beta3.tar.gz
cd speex-1.2beta3
./configure --disable-oggtest --without-libogg && make && make install

Building YASM

YASM is only required if you want to enable and build VPX (VP8 video codec) or x264 (H.264 codec). It’s highly recommended.

wget http://www.tortall.net/projects/yasm/releases/yasm-1.2.0.tar.gz
tar -xvzf yasm-1.2.0.tar.gz
cd yasm-1.2.0
./configure && make && make install

Building libvpx

libvpx adds support for VP8 and is optional but highly recommended if you want support for video when using Google Chrome or Mozilla Firefox. libvpx is required if you want to use .webm container or our WebRTC SIP Telepresence client. You can install the devel packages:

sudo yum install libvpx-devel

Or build the source by yourself:

git clone http://git.chromium.org/webm/libvpx.git
cd libvpx
./configure --enable-realtime-only --enable-error-concealment --disable-examples --enable-vp8 --enable-pic --enable-shared --as=yasm
make && make install

Building opencore-amr

opencore-amr is optional. Adds support for AMR audio codec.

git clone git://opencore-amr.git.sourceforge.net/gitroot/opencore-amr/opencore-amr
cd opencore-amr && autoreconf --install && ./configure && make && make install

Building libopus

libopus is optional but highly recommended as it’s an MTI codec for WebRTC. Adds support for Opus audio codec.

wget http://downloads.xiph.org/releases/opus/opus-1.0.2.tar.gz
tar -xvzf opus-1.0.2.tar.gz
cd opus-1.0.2
./configure --with-pic --enable-float-approx && make && make install

Building libgsm

libgsm is optional. Adds support for GSM audio codec. You can install the devel packages (recommended):

sudo yum install gsm-devel

Or build the source by yourself:

wget http://www.quut.com/gsm/gsm-1.0.13.tar.gz
tar -xvzf gsm-1.0.13.tar.gz
cd gsm-1.0-pl13 && make && make install
#cp -rf ./inc/* /usr/local/include
#cp -rf ./lib/* /usr/local/lib

Building g729

G729 is optional. Adds support for G.729 audio codec.

svn co http://g729.googlecode.com/svn/trunk/ g729b
cd g729b
./autogen.sh && ./configure --enable-static --disable-shared && make && make install

Building iLBC

iLBC is optional. Adds support for iLBC audio codec.

svn co http://doubango.googlecode.com/svn/branches/2.0/doubango/thirdparties/scripts/ilbc
cd ilbc
wget http://www.ietf.org/rfc/rfc3951.txt
awk -f extract.awk rfc3951.txt
./autogen.sh && ./configure
make && make install

Building x264

x264 is optional but highly recommended and adds support for H.264 video codec (requires FFmpeg). x264 is required if you want to use .mp4 container.

wget ftp://ftp.videolan.org/pub/x264/snapshots/last_x264.tar.bz2
tar -xvjf last_x264.tar.bz2
# the output directory may be difference depending on the version and date
cd x264-snapshot-20121201-2245
./configure --enable-shared --enable-pic && make && make install

Building libfreetype

libfreetype is required and used for video overlays. You can install the devel packages (recommended):

sudo yum install freetype-devel

Or build the source by yourself:

wget http://download.savannah.gnu.org/releases/freetype/freetype-2.4.12.tar.bz2
tar -xvjf freetype-2.4.12.tar.bz2
cd freetype-2.4.12
./configure && make && make install

Building libfaac

libfaac is optional unless you want support for AAC audio codec or .mp4 container for recording.

wget http://downloads.sourceforge.net/faac/faac-1.28.tar.bz2
tar -xvjf faac-1.28.tar.bz2
cd faac-1.28 && ./configure && make && make install

Note: building the tests could fails but you can safely ignore it.
Building FFmpeg

FFmpeg is required even if you don’t want support for video.

# [1] checkout source code
git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg
cd ffmpeg

# [2] grap a release branch
git checkout n1.2

# [3] configure source
./configure \
--extra-cflags="-fPIC" \
--extra-ldflags="-lpthread" \
\
--enable-pic --enable-memalign-hack --enable-pthreads \
--enable-shared --disable-static \
--disable-network --enable-pthreads \
--disable-ffmpeg --disable-ffplay --disable-ffserver --disable-ffprobe \
\
--enable-gpl \
\
--disable-debug \
\
--enable-libfreetype \
\
--enable-libfaac \
\
--enable-nonfree
# [4] build and install
make && make install

Remove "--enable-nonfree" and "--enable-libfaac" if you don’t want support for AAC audio codec.
Add --enable-libx264 for force building with support for H.264.

Building OpenAL Soft

OpenAL Soft is optional. Adds support for Stereoscopic (spatial) 3D audio.

wget http://kcat.strangesoft.net/openal-releases/openal-soft-1.15.1.tar.bz2
tar -xvjf openal-soft-1.15.1.tar.bz2
cd openal-soft-1.15.1/build
cmake ..
make && make install

Building openOffice

OpenOffice (or LibreOffice) are optional and add support for presentation sharing. For information about this feature, check the technical details. Version 4.0 or later is required. Both the application and SDK are required.

This section explain how to install (building would take hours) OpenOffice. LibreOffice could also be used but not recommended (not fully tested).

IMPORTANT: These instructions are for Linux x86-64 and you must change the paths if you’re using a 32-bit system. Run uname -m to get your CPU type. All rpms could be found at http://www.openoffice.org/download/other.html.

## Application (x64) ##
wget http://sourceforge.net/projects/openofficeorg.mirror/files/4.0.0/binaries/en-US/Apache_OpenOffice_4.0.0_Linux_x86-64_install-rpm_en-US.tar.gz
mkdir -p OpenOfficeApplication && tar -zxvf Apache_OpenOffice_4.0.0_Linux_x86-64_install-rpm_en-US.tar.gz -C OpenOfficeApplication
rpm -Uvih OpenOfficeApplication/en-US/RPMS/*rpm

## SDK (x64) ##
wget http://sourceforge.net/projects/openofficeorg.mirror/files/4.0.0/binaries/SDK/Apache_OpenOffice-SDK_4.0.0_Linux_x86-64_install-rpm_en-US.tar.gz
mkdir -p OpenOfficeSDK && tar -zxvf Apache_OpenOffice-SDK_4.0.0_Linux_x86-64_install-rpm_en-US.tar.gz -C OpenOfficeSDK
rpm -Uvih OpenOfficeSDK/en-US/RPMS/*rpm

Both OpenOffice application and SDK should be installed into /opt/openoffice4. If not, you’ll need to edit the script used to prepare the SDK headers.

Prepare the SDK headers:

LD_LIBRARY_PATH=/opt/openoffice4/program:/opt/openoffice4/sdk/lib /opt/openoffice4/sdk/bin/cppumaker -BUCR -O /opt/openoffice4/sdk/includecpp /opt/openoffice4/program/types.rdb

Please note that the destination folder must be named includecpp.

Install java runtime (required):

yum install java-1.7.0-openjdk

Notes

Installing OpenOffice application will not add the binary (soffice) in your $PATH environment variable. The TelePresence system will try to start the program in the background using a relative path unless you have changed presentation-sharing-app configuration entry. You can change your $PATH environment variable to avoid editing presentation-sharing-app but this is not recommend if you’re testing different OpenOffice versions. It’s also highly recommended to append the folder containing the binary AFTER $PATH. Appending the folder before $PATH will force using shared libraries (e.g. libssl, libcurl…) installed with OpenOffice instead of yours.

Correct: export PATH=$PATH:/opt/openoffice4/program

NOT correct: export PATH=/opt/openoffice4/program:$PATH
Known Issues

"/usr/bin/ld: skipping incompatible /opt/openoffice4/sdk/lib/libuno_sal.so when searching for -luno_sal": CPU type mismatch (e.g. installed 64-bit libraries on 32-bit OS).
Building Doubango

Doubango VoIP framework 2.0 SVN r989 or later is required.

svn checkout http://doubango.googlecode.com/svn/branches/2.0/doubango doubango
cd doubango && ./autogen.sh && ./configure --with-speexdsp --with-ffmpeg
make && make install

Only few options are used to configure the source code and force enabling mandatory libraries. Any optional library is automatically detected. For example, use "--with-opus" to force using Opus audio codec or "--without-opus" to avoid automatic detection. You can also specify a path where to search for a library (e.g. "--with-opus=/usr/local"). Use "configure --help" for more information on supported options.
Building the Telepresence system

svn checkout https://telepresence.googlecode.com/svn/trunk/ telepresence
cd telepresence
./autogen.sh && ./configure
make && make install

If no prefix is defined then, the binaries will be installed into /usr/local/sbin.
Installing the configuration and fonts files

This is only required for first-time installations and will override any existing configuration file.

make samples

Testing

We highly recommend using our WebRTC SIP telepresence client to test the system.

For more information on how to test the system: click here
Comment by muthuvel...@gmail.com, Jul 30, 2013

tinywrap does not have autogen.sh and even telepresence. how to solve this

==Referensi==

* http://code.google.com/p/telepresence/
* http://code.google.com/p/telepresence/wiki/Support_BuildingSourceCode
* http://code.google.com/p/telepresence/wiki/Support_BuildingSourceCode#Installing_the_configuration_and_fonts_files

VoIP: Instalasi Telepresence - Revision history

Onnowpurbo: New page: This is the most important part as it’s where you’ll decide which features to support/enable. The Telepresence system use Doubango VoIP Framework (requires SVN r989 or later) and it...