ortclib / ortclib-cpp

ORTC (Object RTC) C++ Library wrapper for WebRTC

INSTRUCTIONS

From your terminal, please clone the "ortclib-sdk" git repository (before cloning, please read IMPORTANT NOTES section below): git clone --recursive https://github.com/ortclib/ortclib-sdk.git

This repository will yield the ORTC Lib SDK and dependency libraries.

Directory structure:

• ortc-lib-sdk\bin contains scripts for preparing an environment for Windows
• ortc-lib-sdk\common contains samples applications that can use Ortc and WebRtc libraries. Currently it is only the case for PeerCC sample
• ortc-lib-sdk\ortc contains code of Ortc and dependency libraries
• ortc-lib-sdk\webrtc contains code of WebRtc and dependency libraries
• ortc-lib-sdk\docs contains documentation

IMPORTANT NOTES:

1. Ensure that Git is properly configured to handle line endings. Git setting for core.autocrlf needs to be set to true; you can check current setting for core.autocrlf using command: 'git config -l', or 'git config core.autocrlf', and you can change setting to true using 'git config --global core.autocrlf true'. Usefull links: https://help.github.com/articles/dealing-with-line-endings/ https://git-scm.com/book/en/v2/Customizing-Git-Git-Configuration#__code_core_autocrlf_code (section core.autocrlf)
2. Visual Studio 2015 is required
3. Latest supported Win 10 SDK is 10.0.14393.x

How to Build:

WINDOWS BUILD :

1. Run prepare script, from Windows Command Prompt (not from a terminal emulator):

cd ortclib-sdk
bin\prepare.bat

This script will prepare environment but it won't build webrtc projects. In this case webrtc project will be built from Visual Studio once you try to build Org.Ortc or Org.WebRtc.

2. From VS2015, load ortc\windows\solutions\Ortc.sln for Ortc or webrtc\windows\solutions\WebRtc.sln for WebRtc development.

3. Now you can build winrt libraries for Ortc or WebRtc and deploy sample apps ChatterBox (WebRtc only) and PeerCC.

====

ortclib-cpp's Issues

Build instructions and scripts appear to be out of date

Hi! Looks like an interesting project, but I think your README is out of date for build instructions. Additionally scripts seem to be looking for boost, although it seems as if boost isn't really required anymore (at least I don't see it in the code at all). I'm going to poke around a bit and try to give it a build (linux first) and make a PR if I succeed at it.

create RTC sender / receiver objects

Enable security in WebRTC PeerConnection_client on Windows

Security is currently disabled due to lack of OpenSSL port for Windows inside webRTC solution. OpenSSL should be added and linked, proper defines should be adjusted in order to build openssl related code

Change nominated candidate pair to selected candidate pair

RTP parameters translation between received and new created forwarding packets

$feature #25

Implement step() pattern for MediaManager, MediaStream and MediaStreamTrack classes

$feature #19

Create DTLS skeleton object for wiring

Related:
#12

Audio - GetUserMedia interface, MediaStream, MediaStreamTrack classes

Create iOS client to SFU server conference scenario with 3 participants

$feature #36

Sync MediaStream/MediaStreamTrack to latest ORTC class pattern changes

$feature #19

Server side forwarding channels on OSX - proof of concept

https://github.com/openpeer/ortc-lib/tree/feature-20140924-sfu-audio-conference

factory add if (this) {} to members

To protect against non-virtual factor members.

RTCP packets filtering and forwarding to the source end point

$feature #25

Instructions

Changes for instructions:

git clone --recursive https://github.com/openpeer/ortc-lib.git

1) Build boost, from your terminal:

pushd ortc-lib/libs/boost/
./boost.sh
popd


2) Build curl, from your terminal:

pushd ortc-lib/libs/curl/
./build_ios.sh
popd

WebRTC VoiceEngine API changes for SFU support

$feature #25

android build error

common.copy ../build/include/boost-1_53/boost/tr1/tr1/locale
common.copy ../build/include/boost-1_53/boost/tr1/tr1/fstream
common.copy ../build/include/boost-1_53/boost/tr1/tr1/memory
common.copy ../build/include/boost-1_53/boost/tr1/tr1/valarray
common.copy ../build/include/boost-1_53/boost/tr1/tr1/utility
common.copy ../build/include/boost-1_53/boost/tr1/tr1/cmath
common.copy ../build/include/boost-1_53/boost/tr1/tr1/unordered_set
...updated 10232 targets...
Done!
~/ortc/ortc-lib/libs/boost/projects
----------- Installing boost 1.53.0 for ANDROID platform -----------------
~/ortc/ortc-lib/projects/android
~/ortc/ortc-lib/projects/android ~/ortc/ortc-lib/projects/android
./buildall_android.sh: line 54: cd: ./../../libs/openssl/projects/android: No such file or directory
./buildall_android.sh: line 55: ./build-libssl_android.sh: No such file or directory
~/ortc/ortc-lib/projects/android
~/ortc/ortc-lib/projects/android ~/ortc/ortc-lib/projects/android
./buildall_android.sh: line 61: ./build_curl_android.sh: No such file or directory
~/ortc/ortc-lib/projects/android
~/ortc/ortc-lib/projects/android ~/ortc/ortc-lib/projects/android
Host machine : x86_64
----------------- NDK Path is : /home/lee/test/android-ndk-r8e ----------------
----------------- Exporting the android-ndk path ----------------
------ Building cryptopp 5.6.2 for ANDROID platform ------
~/ortc/ortc-lib/libs/cryptopp/projects/android-patches ~/ortc/ortc-lib/libs/cryptopp/projects/android-patches
Archive: cryptopp.zip
creating: TestData/
inflating: TestData/3desval.dat

inflating: TestData/3wayval.dat

inflating: TestData/camellia.dat

inflating: TestData/cast128v.dat

inflating: TestData/cast256v.dat

inflating: TestData/descert.dat

inflating: TestData/dh1024.dat

inflating: TestData/dh2048.dat

inflating: TestData/dlie1024.dat

i downloaded at https://github.com/openpeer/openssl

build_curl_android.sh: No such file or directory

GetUserMedia / MediaStream refactoring

Issue with libs?

I see that the project as switched to gclient. But I'm facing issues sync'ing the project.

C:\workspace\ortc-lib\libs>gclient sync
Oh hai! You are using subversion. Chrome infra is eager to get rid of
svn support so please switch to git.
Tracking bug: http://crbug.com/475320
If you are a project owner, you may request git migration assistance at:
https://code.google.com/p/chromium/issues/entry?template=Infra-Git

________ running 'svn checkout file:///./webrtc C:\workspace\ortc-lib\libs\webrtc --non-interactive --ignore-externals' in 'C:\workspace\ortc-lib\libs'
svn: Unable to open an ra_local session to URL
svn: Unable to open repository 'file:///webrtc'

webrtc (ERROR)

[0:00:00] Started.

Error: Command 'svn checkout file:///./webrtc C:\workspace\ortc-lib\libs\webrtc --non-interactive --ignore-externals' returned non-zero exit status 1 in C:\workspace\ortc-lib\libs

Make a console SFU test application for OSX

$feature #36

Fix ortclibTest_ios application build for LLVM/libc++ set of tools

https://github.com/openpeer/ortc-lib/tree/feature-20141109-push-messaging-database

Settings interface

Make an option for audio channel not to use output mixer and device

$feature #25

This type of channel is intended to be used on a SFU server side, as an incoming channel.

Handle partial messages in data channels

I've had a glance at your code and it seems you're not handling the MSG_EOR flag which means you will treat incoming partial SCTP messages as whole messages, thus violating the message-oriented principle of SCTP (and data channels). This is a widespread issue in many implementations we're currently trying to clean up.

If you need help, you can look at the relevant section in RAWRTC's code or at Mozilla's implementation. I'm the author of RAWRTC and I've also worked on Mozilla's data channel code, so feel free to ping me if you have any questions.

Tracked (and resolved) for Firefox in bug 979417. For Chromium, this is tracked in issue 7774.

Make support for SFU audio channel establishment

$feature #25

SFU audio channel will not make use of microphone, transmit mixer, codecs and output mixer. This should be one way (outgoing only) channel.

Initialize OpenSSL to be able to use DTLS

SCTP remote port need start() not constructor

To allow for remote port to be signalled instead of in channel data.

Video - GetUserMedia interface, MediaStream, MediaStreamTrack classes

Support for multiple remote tracks

Multiple channel support in WebRTC

Windows build

Hi guys,

Is there a chance that you publish the latest changes in WinRT branches so I can do Windows build?
So far there are absent project files like
"......\libs\webrtc\third_party\boringssl\boringssl_win.vcxproj"

Regards, Maxim.

RTCConnection

Wire up ICE and demux capabilities into RTCConnection object

Turn off audio recording and playout components when SFU channel is used

$feature #25

RTCSocket implementation

Prepare fingerprints for DTLS

Implement asynchronous callback for getUserMedia

$feature #15

Media testing

Integration of ORTC components, extending features of getUserMedia interface.

OpenSSL for iOS 7

You can find and download on link bellow, along with the sample app.

https://github.com/x2on/OpenSSL-for-iPhone

Link OpenSSL to iOS test app

Create test environment for DTLS-SRTP

Expose any default settings for ortc-lib

$feature #22

Issue during building the project in Mac OSX for android

Hi,

When i tried to build the android with the command

cd ./projects/android
./buildall_android.sh

I am getting the below mentioned issue. and libraries are not getting generated. Please check whether this is configuration issue or os.

/Development/ortc/ortc-lib/projects/android-test /Development/ortc/ortc-lib/projects/android-test
Android NDK: ERROR:/Development/ortc/ortc-lib/projects/android-test/jni/Android.mk:jpeg_turbo: LOCAL_SRC_FILES points to a missing file **
Android NDK: Check that /Development/ortc/ortc-lib/projects/android-test/jni/../../../libs/build/android/webrtc/libjpeg_turbo.a exists or that its path is correct
/Development/Android/android-ndk-r8e/build/core/prebuilt-library.mk:45: ** Android NDK: Aborting . Stop.
/Development/ortc/ortc-lib/projects/android-test
-------- Installing ortc-lib -----
/Development/ortc/ortc-lib/projects/android
_BUILD COMLETED FOR ALL ORTC LIBS_

Make SFU audio conferencing model using ORTC classes

https://github.com/openpeer/ortc-lib/tree/feature-20140924-sfu-audio-conference

Settings need to be wired to openpeer::services layer

$feature #22

Handle forwarding audio channel closing process

$feature #25

DTLS using OpenSSL example

Please check DTLS examples of blocking and non-blocking implementation from the link bellow. It looks a bit different from webrtc implementation, but much more simpler.

It is located in comment 2 as attachment.

http://openssl.6102.n7.nabble.com/DTLS-examples-td2143.html

Wire RTC sender / receiver objects to ICE DTLS session

Add turn token credential support

getUserMedia/MediaStream - support for 1 channel

Make use of getUserMedia interface and MediaStream/MediaStreamTrack objects to establish a 1 channel local call using WebRTC media engine.