Student Application Template

To fill this in, copy the source text. Please don't rename the template.

This is a suggestion for the kind of information we'll find useful from students in their submissions. Remember -- you're going to be committing to several months' work. The more information and planning you can provide up-front, the more we (and Google!) will have to go on when we're ranking your application. Do not forget adding your submission at SummerOfCode2016/StudentApplications

* Read http://paste.debian.net/776169/ for better understanding.

ConversationManager.cxx

createMediaInterface(..)

CreatePortAllocator(..)

ConversationManager.cxx

setSpeakerVolume(..)

GetOutputVolume(..)

ConversationManager.cxx

setMicrophoneGain(..)

GetInputLevel()

ConversationManager.cxx

muteMicrophone(..)

GetInputLevel()

ConversationManager.cxx

setAudioAECMode(..)

bool aec

ConversationManager.cxx

enableAGC(..)

AdjustAgcLevel(..)

ConversationManager.cxx

setAudioNoiseReductionMode(..)

SetConferenceMode(..)

BridgeMixer.cxx

setMixWeightsForOutput(..)

--

BridgeMixer.cxx

setMixWeightsForInput(..)

--

MediaResourceParticipant.cxx

startTone(..)

PlayRingbackTone(..)

MediaResourceParticipant.cxx

stopAudio()

SetPlayout(..)

MediaResourceParticipant.cxx

playBuffer(..)

PlaySound(..) / SetPlayout(..)

MediaResourceParticipant.cxx

createPlayer(..)

--

RemoteParticipantDialogSet.cxx

createConnection(..)

createConnection(..)

RemoteParticipantDialogSet.cxx

deleteConnection(..)

delete port

RemoteParticipantDialogSet.cxx

getCapabilities(..)

GetCapabilities()

RemoteParticipantDialogSet.cxx

getConnectionPortOnBridge(..)

connected()

RemoteParticipant.cxx

setConnectionDestination(..)

WEBRTC_STUB(..)

RemoteParticipant.cxx

start/stopRtpSend(..)

SendRtp(..)/SendRtcp(..)

RemoteParticipant.cxx

start/stopRtpReceive(..)

OnPacketReceived(..)/OnRtcpReceived(..)

RemoteParticipant.cxx

isReceivingRtpAudio(..)

WEBRTC_FUNC(..)/WEBRTC_STUB(..)

RemoteParticipant.cxx

isSendingRtpAudio(..)

--

* Some NAT, RTP, RTCP, DTLS-SRTP, SDES-SRTP functions from reflow*

1 - Manages a set of STUN requests (send, delay, remove, clear)

2 - Creates a STUN server, which will listen on the given socket

3 - Handling different types of STUN/TURN requests

4 - SRTP support

5 - STUN, Turn configuration

6 - Class used for describing what media a ?PeerConnection can receive.

7 - Handle RTP and RTCP packets using ?MediaSinkInterface

8 - NAT traversal