VoIP Media Session

The rest of this article describes the VoIPMediaSession class which is designed for use with SIP/VoIP applications. For WebRTC see the RTCPeerConnection article.

The VoIPMediaSession class acts as a bridge between the RTPSession class and a separate media library, such as SIPSorceryMedia.Windows or SIPSorceryMedia.FFmpeg.

The code snippet below shows how to instantiate an audio only VoIPMediaSession.

using SIPSorcery.Media;

var audioEndPoint = new SIPSorceryMedia.Windows.WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(audioEndPoint.ToMediaEndPoints());

The code snippet below shows how to instantiate an audio and video VoIPMediaSession.

using SIPSorcery.Media;
using SIPSorceryMedia.Abstractions.V1;

var audioEndPoint = new SIPSorceryMedia.Windows.WindowsAudioEndPoint(new AudioEncoder());
var videoEndPoint = new SIPSorceryMedia.Windows.WindowsVideoEndPoint();

MediaEndPoints mediaEndPoints = new MediaEndPoints
{
    AudioSink = audioEndPoint,
    AudioSource = audioEndPoint,
    VideoSink = videoEndPoint,
    VideoSource = videoEndPoint,
};

var voipMediaSession = new VoIPMediaSession(mediaEndPoints);

A different media library implementation can be used by swapping it in for WindowsAudioEndPoint and/or WindowsVideoEndPoint.

The code snippet below shows how it's possible to combine different media sinks and sources. In this case the audio and video sources are generated from an mp4 file and the SIPSorceryMedia.FFmpeg library while the audio and video samples from the remote party are handled with the SIPSorceryMedia.Windows library.

using System.Collections.Generic;
using SIPSorcery.Media;
using SIPSorceryMedia.Abstractions.V1;

var windowsAudioEndPoint = new SIPSorceryMedia.Windows.WindowsAudioEndPoint(new AudioEncoder(), -1, -1, true);
windowsAudioEndPoint.RestrictCodecs(new List<AudioCodecsEnum> { AudioCodecsEnum.PCMU });
var windowsVideoEndPoint = new SIPSorceryMedia.Windows.WindowsVideoEndPoint();

var mediaFileSource = new SIPSorceryMedia.FFmpeg.FFmpegFileSource("max_intro.mp4", true, new AudioEncoder());
mediaFileSource.Initialise();
mediaFileSource.RestrictCodecs(new List<VideoCodecsEnum> { VideoCodecsEnum.VP8 });
mediaFileSource.RestrictCodecs(new List<AudioCodecsEnum> { AudioCodecsEnum.PCMU });

MediaEndPoints mediaEndPoints = new MediaEndPoints
{
    AudioSink = windowsAudioEndPoint,
    AudioSource = mediaFileSource,
    VideoSink = windowsVideoEndPoint,
    VideoSource = mediaFileSource,
};

var voipMediaSession = new VoIPMediaSession(mediaEndPoints);
voipMediaSession.AcceptRtpFromAny = true;

Media Control during Calls

Once a SIP call has been established, typically with the SIPUserAgent, certain events, such as hanging up or placing on hold, need to adjust the way audio and/or video streams are handled.

The SIPUserAgent uses the IMediaSession interface to control the VoIPMediaSession. The reason for an additional interface, instead of relying on, IAudioSink etc, is to combine the signalling and media functions. For example placing a call on hold requires a SIP re-INVITE request to be sent as well as adjusting the media streams.

Applications do not hav to deal with signalling or media adjustments for standard SIP scenarios - such as on/off hold, transfers and hangup - they are able to be handled automatically by the SIPUserAgent.

There are cases where an application may want to adjust the media on a call other than the standard scenarios. For example playing a prompt for an Interactive Voice Response (IVR) system. For those cases the audio or video source can be called directly. The Play Sounds demo application is a good example.

It creates a VoIPMediaSession in the standard manner.

var windowsAudio = new WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(windowsAudio.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;

Once the call is established it pauses the microphone capture and switches the audio source to play pre-recorded prompts and signals from an internal generator.

await windowsAudio.PauseAudio();

Console.WriteLine("Sending welcome message from 8KHz sample.");
await voipMediaSession.AudioExtrasSource.SendAudioFromStream(new FileStream(WELCOME_8K, FileMode.Open), AudioSamplingRatesEnum.Rate8KHz);

await Task.Delay(200);

Console.WriteLine("Sending sine wave.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.SineWave);

await Task.Delay(2000);

Console.WriteLine("Sending white noise signal.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.WhiteNoise);
await Task.Delay(2000);

Console.WriteLine("Sending pink noise signal.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.PinkNoise);
await Task.Delay(2000);

Console.WriteLine("Sending silence.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.Silence);

await Task.Delay(2000);

Console.WriteLine("Sending goodbye message from 16KHz sample.");
await voipMediaSession.AudioExtrasSource.SendAudioFromStream(new FileStream(GOODBYE_16K, FileMode.Open), AudioSamplingRatesEnum.Rate16KHz);

voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.None);

await Task.Delay(200);

// Switch to the external microphone input source.
await windowsAudio.ResumeAudio();

The full example for the above code snippet can be found in the Play Sounds demo.