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.
sudo yum update
sudo yum install make libtool autoconf subversion git wget cmake gcc gcc-c++ pkgconfig nasm
- 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.
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
git checkout v1.5.0
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.
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A quick way to have OpenSSL may be installing openssl-devel package but this version will most likely be outdated (prior to 1.0.1).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).
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--
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
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
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
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
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
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
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
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
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
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
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.
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 --enable-encoder=libx264 --enable-decoder=h264 to force building with support for H.264.
OpenH264 is optional. Adds support for H.264 constrained baseline video codec.
git clone https://github.com/cisco/openh264.git
cd openh264
git checkout v1.1
make ENABLE64BIT=Yes # Use ENABLE64BIT=No for 32bit platforms
make install
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
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 forLinux 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
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
"/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).
"MSG: TELEPRESENCE FFmpegOverlay Not valid No such filter: 'drawtext'": libfreetype is missing. You'll have to rebuild FFmpeg with this library or disable overlays.
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.
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.
This is only required for first-time installations and will override any existing configuration file.
make samples
We highly recommend using our WebRTC SIP telepresence client to test the system.
For more information on how to test the system: click here