public void MediaOrderMatchesRemoteOfferUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
// By default offers made by us always put audio first. Create a remote SDP offer
// with the video first.
string remoteSdp =
@"v=0
o=- 1986548327 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=video 60638 RTP/AVP 100
a=rtpmap:100 VP8/90000
a=sendrecv
m=audio 60640 RTP/AVP 0 111
a=rtpmap:0 PCMU/8000
a=rtpmap:111 OPUS/48000/2
a=sendrecv";
// Create a local session and add the video track first.
RTPSession rtpSession = new RTPSession(false, false, false);
MediaStreamTrack localAudioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU),
new SDPAudioVideoMediaFormat(SDPMediaTypesEnum.audio, 110, "OPUS/48000/2")
});
rtpSession.addTrack(localAudioTrack);
MediaStreamTrack localVideoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPMediaTypesEnum.video, 96, "VP8", 90000)
});
rtpSession.addTrack(localVideoTrack);
var offer = SDP.ParseSDPDescription(remoteSdp);
logger.LogDebug($"Remote offer: {offer}");
var result = rtpSession.SetRemoteDescription(SIP.App.SdpType.offer, offer);
logger.LogDebug($"Set remote description on local session result {result}.");
Assert.Equal(SetDescriptionResultEnum.OK, result);
var answer = rtpSession.CreateAnswer(null);
logger.LogDebug($"Local answer: {answer}");
Assert.Equal(111, rtpSession.AudioLocalTrack.Capabilities.Single(x => x.Name() == "OPUS").ID);
Assert.Equal(100, rtpSession.VideoLocalTrack.Capabilities.Single(x => x.Name() == "VP8").ID);
//Assert.True(SDPAudioVideoMediaFormat.AreMatch(offer.Media.Single(x => x.Media == SDPMediaTypesEnum.audio)., answer.Media.First().Media));
//Assert.Equal(offer.Media.Last().Media, answer.Media.Last().Media);
rtpSession.Close("normal");
}
private static RTPSession CreateRtpSession(List <SDPAudioVideoMediaFormat> audioFormats, List <SDPAudioVideoMediaFormat> videoFormats)
{
var rtpSession = new RTPSession(false, false, false, IPAddress.Loopback);
bool hasAudio = false;
bool hasVideo = false;
if (audioFormats != null && audioFormats.Count > 0)
{
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, audioFormats, MediaStreamStatusEnum.SendRecv);
rtpSession.addTrack(audioTrack);
hasAudio = true;
}
if (videoFormats != null && videoFormats.Count > 0)
{
MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, videoFormats, MediaStreamStatusEnum.SendRecv);
rtpSession.addTrack(videoTrack);
hasVideo = true;
}
var sdpOffer = rtpSession.CreateOffer(null);
// Because the SDP being written to the file is the input to ffplay the connection ports need to be changed
// to the ones ffplay will be listening on.
if (hasAudio)
{
sdpOffer.Media.Single(x => x.Media == SDPMediaTypesEnum.audio).Port = FFPLAY_DEFAULT_AUDIO_PORT;
}
if (hasVideo)
{
sdpOffer.Media.Single(x => x.Media == SDPMediaTypesEnum.video).Port = FFPLAY_DEFAULT_VIDEO_PORT;
}
Console.WriteLine(sdpOffer);
using (StreamWriter sw = new StreamWriter(FFPLAY_DEFAULT_SDP_PATH))
{
sw.Write(sdpOffer);
}
string ffplayCommand = String.Format(FFPLAY_DEFAULT_COMMAND, FFPLAY_DEFAULT_SDP_PATH);
Console.WriteLine($"Start ffplay using the command below:");
Console.WriteLine(ffplayCommand);
Console.WriteLine($"To request the remote peer to send a video key frame press 'k'");
rtpSession.Start();
rtpSession.SetDestination(SDPMediaTypesEnum.audio, new IPEndPoint(IPAddress.Loopback, FFPLAY_DEFAULT_AUDIO_PORT), new IPEndPoint(IPAddress.Loopback, FFPLAY_DEFAULT_AUDIO_PORT + 1));
rtpSession.SetDestination(SDPMediaTypesEnum.video, new IPEndPoint(IPAddress.Loopback, FFPLAY_DEFAULT_VIDEO_PORT), new IPEndPoint(IPAddress.Loopback, FFPLAY_DEFAULT_VIDEO_PORT + 1));
return(rtpSession);
}
public void MediaOrderMatchesRemoteOfferUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
// By default offers made by us always put audio first. Create a remote SDP offer
// with the video first.
string remoteSdp =
@"v=0
o=- 1986548327 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=video 60638 RTP/AVP 100
a=rtpmap:100 VP8/90000
a=sendrecv
m=audio 60640 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv";
// Create a local session and add the video track first.
RTPSession localSession = new RTPSession(false, false, false);
MediaStreamTrack localAudioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
localSession.addTrack(localAudioTrack);
MediaStreamTrack localVideoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
localSession.addTrack(localVideoTrack);
var offer = SDP.ParseSDPDescription(remoteSdp);
logger.LogDebug($"Remote offer: {offer}");
var result = localSession.SetRemoteDescription(SIP.App.SdpType.offer, offer);
logger.LogDebug($"Set remote description on local session result {result}.");
Assert.Equal(SetDescriptionResultEnum.OK, result);
var answer = localSession.CreateAnswer(null);
logger.LogDebug($"Local answer: {answer}");
Assert.Equal(offer.Media.First().Media, answer.Media.First().Media);
Assert.Equal(offer.Media.Last().Media, answer.Media.Last().Media);
localSession.Close("normal");
}
private static async Task StartFfmpegListener(string sdpPath, CancellationToken cancel)
{
while (!File.Exists(FFMPEG_SDP_FILE) && !cancel.IsCancellationRequested)
{
await Task.Delay(500);
}
if (!cancel.IsCancellationRequested)
{
var sdp = SDP.ParseSDPDescription(File.ReadAllText(FFMPEG_SDP_FILE));
// The SDP is only expected to contain a single video media announcement.
var videoAnn = sdp.Media.Single(x => x.Media == SDPMediaTypesEnum.video);
_ffmpegVideoFormat = videoAnn.MediaFormats.Values.First();
_ffmpegListener = new RTPSession(false, false, false, IPAddress.Loopback, FFMPEG_DEFAULT_RTP_PORT);
_ffmpegListener.AcceptRtpFromAny = true;
MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPAudioVideoMediaFormat> {
_ffmpegVideoFormat
}, MediaStreamStatusEnum.RecvOnly);
_ffmpegListener.addTrack(videoTrack);
_ffmpegListener.SetRemoteDescription(SIP.App.SdpType.answer, sdp);
// Set a dummy destination end point or the RTP session will end up sending RTCP reports
// to itself.
var dummyIPEndPoint = new IPEndPoint(IPAddress.Loopback, 0);
_ffmpegListener.SetDestination(SDPMediaTypesEnum.video, dummyIPEndPoint, dummyIPEndPoint);
await _ffmpegListener.Start();
}
}
public void NoLocalTracksTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
// Create two RTP sessions. First one acts as the local session to generate the offer.
// Second one acts as the remote session to generate the answer.
// A local session is created but NO media tracks are added to it.
RTPSession localSession = new RTPSession(false, false, false);
// Create a remote session WITH an audio track.
RTPSession remoteSession = new RTPSession(false, false, false);
// The track for the track for the remote session is still local relative to the session it's being added to.
MediaStreamTrack remoteAudioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
remoteSession.addTrack(remoteAudioTrack);
var offer = remoteSession.CreateOffer(IPAddress.Loopback);
// Give the offer to the local session that is missing any media tracks.
var result = localSession.SetRemoteDescription(SIP.App.SdpType.offer, offer);
logger.LogDebug($"Set remote description on local session result {result}.");
Assert.Equal(SetDescriptionResultEnum.NoMatchingMediaType, result);
localSession.Close("normal");
remoteSession.Close("normal");
}
public async void SortChecklistUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
RTPSession rtpSession = new RTPSession(true, true, true);
MediaStreamTrack dummyTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(dummyTrack);
var iceSession = new IceSession(rtpSession.GetRtpChannel(SDPMediaTypesEnum.audio), RTCIceComponent.rtp, null);
iceSession.StartGathering();
Assert.NotNull(iceSession);
Assert.NotEmpty(iceSession.Candidates);
foreach (var hostCandidate in iceSession.Candidates)
{
logger.LogDebug(hostCandidate.ToString());
}
var remoteCandidate = RTCIceCandidate.Parse("candidate:408132416 1 udp 2113937151 192.168.11.50 51268 typ host generation 0 ufrag CI7o network-cost 999");
await iceSession.AddRemoteCandidate(remoteCandidate);
var remoteCandidate2 = RTCIceCandidate.Parse("candidate:408132417 1 udp 2113937150 192.168.11.51 51268 typ host generation 0 ufrag CI7o network-cost 999");
await iceSession.AddRemoteCandidate(remoteCandidate2);
foreach (var entry in iceSession._checklist)
{
logger.LogDebug($"checklist entry priority {entry.Priority}.");
}
}
public void NoMatchingMediaTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
RTPSession localSession = new RTPSession(false, false, false);
MediaStreamTrack localAudioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
localSession.addTrack(localAudioTrack);
RTPSession remoteSession = new RTPSession(false, false, false);
// The track for the track for the remote session is still local relative to the session it's being added to.
MediaStreamTrack remoteVideoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPMediaTypesEnum.video, 96, "VP8", 90000)
});
remoteSession.addTrack(remoteVideoTrack);
var result = localSession.SetRemoteDescription(SIP.App.SdpType.offer, remoteSession.CreateOffer(IPAddress.Loopback));
logger.LogDebug($"Set remote description on local session result {result}.");
Assert.Equal(SetDescriptionResultEnum.NoMatchingMediaType, result);
localSession.Close("normal");
remoteSession.Close("normal");
}
public void GetHostCandidatesForRTPBindUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
var localAddress = NetServices.InternetDefaultAddress;
RTPSession rtpSession = new RTPSession(true, true, true, localAddress);
// Add a track to the session in order to initialise the RTPChannel.
MediaStreamTrack dummyTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(dummyTrack);
RTPChannel rtpChannel = rtpSession.GetRtpChannel(SDPMediaTypesEnum.audio);
logger.LogDebug($"RTP channel RTP socket local end point {rtpChannel.RTPLocalEndPoint}.");
var iceSession = new IceSession(rtpChannel, RTCIceComponent.rtp, null);
iceSession.StartGathering();
Assert.NotNull(iceSession);
Assert.NotEmpty(iceSession.Candidates);
Assert.True(localAddress.Equals(IPAddress.Parse(iceSession.Candidates.Single().address)));
foreach (var hostCandidate in iceSession.Candidates)
{
logger.LogDebug(hostCandidate.ToString());
}
}
public void CheckSelectedAudioForamtAttributeUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
string remoteSdp =
@"v=0
o=- 1986548327 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 60640 RTP/AVP 0 111 8
a=rtpmap:0 PCMU/8000
a=rtpmap:111 OPUS/48000/2";
// Create a local session with an audio track.
RTPSession rtpSession = new RTPSession(false, false, false);
MediaStreamTrack localAudioTrack = new MediaStreamTrack(SDPWellKnownMediaFormatsEnum.PCMA, SDPWellKnownMediaFormatsEnum.G723);
rtpSession.addTrack(localAudioTrack);
var offer = SDP.ParseSDPDescription(remoteSdp);
logger.LogDebug($"Remote offer: {offer}");
var result = rtpSession.SetRemoteDescription(SIP.App.SdpType.offer, offer);
logger.LogDebug($"Set remote description on local session result {result}.");
Assert.Equal(SetDescriptionResultEnum.OK, result);
Assert.Equal(8, rtpSession.AudioLocalTrack.Capabilities.Single(x => x.Name() == "PCMA").ID);
Assert.Equal("PCMA", rtpSession.GetSendingFormat(SDPMediaTypesEnum.audio).Name());
rtpSession.Close("normal");
}
public void CheckDuplicateBindPortFailsUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
// Create two RTP sessions. First one acts as the local session to generate the offer.
// Second one acts as the remote session to generate the answer.
RTPSession localSession = new RTPSession(false, false, false, IPAddress.Loopback);
MediaStreamTrack localAudioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
localSession.addTrack(localAudioTrack);
var rtpEndPoint = localSession.GetRtpChannel(SDPMediaTypesEnum.audio).RTPLocalEndPoint;
logger.LogDebug($"RTP session local end point {rtpEndPoint}.");
// Now attempt to create a second RTP session on the same port as the previous one.
RTPSession duplicateSession = new RTPSession(false, false, false, IPAddress.Loopback, rtpEndPoint.Port);
MediaStreamTrack duplicateTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
Assert.Throws <ApplicationException>(() => duplicateSession.addTrack(duplicateTrack));
localSession.Close(null);
}
public void AudioVideoOfferNoLocalAudioUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
// Create two RTP sessions. First one acts as the local session to generate the offer.
// Second one acts as the remote session to generate the answer.
// A local session is created but only has a video track added to it.
RTPSession localSession = new RTPSession(false, false, false);
MediaStreamTrack localVideoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPMediaTypesEnum.video, 96, "VP8", 90000)
});
localSession.addTrack(localVideoTrack);
// Create a remote session with both audio and video tracks.
RTPSession remoteSession = new RTPSession(false, false, false);
// The track for the track for the remote session is still local relative to the session it's being added to.
MediaStreamTrack remoteAudioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
remoteSession.addTrack(remoteAudioTrack);
MediaStreamTrack remoteVideoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPMediaTypesEnum.video, 96, "VP8", 90000)
});
remoteSession.addTrack(remoteVideoTrack);
var offer = remoteSession.CreateOffer(IPAddress.Loopback);
// Give the offer to the local session that is missing a video tracks.
var result = localSession.SetRemoteDescription(SIP.App.SdpType.offer, offer);
logger.LogDebug($"Set remote description on local session result {result}.");
Assert.Equal(SetDescriptionResultEnum.OK, result);
var answer = localSession.CreateAnswer(null);
Assert.Equal(MediaStreamStatusEnum.Inactive, answer.Media.Where(x => x.Media == SDPMediaTypesEnum.audio).Single().MediaStreamStatus);
Assert.Equal(MediaStreamStatusEnum.SendRecv, answer.Media.Where(x => x.Media == SDPMediaTypesEnum.video).Single().MediaStreamStatus);
localSession.Close("normal");
remoteSession.Close("normal");
}
private static RTPSession CreateRtpSession(List <SDPMediaFormat> videoFormats)
{
var rtpSession = new RTPSession(false, false, false, IPAddress.Loopback);
MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, videoFormats, MediaStreamStatusEnum.SendRecv);
rtpSession.addTrack(videoTrack);
rtpSession.SetDestination(SDPMediaTypesEnum.video, new IPEndPoint(IPAddress.Loopback, FFPLAY_DEFAULT_VIDEO_PORT), new IPEndPoint(IPAddress.Loopback, FFPLAY_DEFAULT_VIDEO_PORT + 1));
return(rtpSession);
}
public void AudioOnlyOfferAnswerTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
// Create two RTP sessions. First one acts as the local session to generate the offer.
// Second one acts as the remote session to generate the answer.
RTPSession localSession = new RTPSession(false, false, false);
MediaStreamTrack localAudioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
localSession.addTrack(localAudioTrack);
// Generate the offer to send to the remote party.
var offer = localSession.CreateOffer(IPAddress.Loopback);
logger.LogDebug("Local offer: " + offer.ToString());
RTPSession remoteSession = new RTPSession(false, false, false);
// The track for the track for the remote session is still local relative to the session it's being added to.
MediaStreamTrack remoteAudioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
remoteSession.addTrack(remoteAudioTrack);
var result = remoteSession.SetRemoteDescription(SIP.App.SdpType.offer, offer);
logger.LogDebug($"Set remote description on remote session result {result}.");
Assert.Equal(SetDescriptionResultEnum.OK, result);
// Get the answer from the remote session.
var answer = remoteSession.CreateAnswer(IPAddress.Loopback);
logger.LogDebug("Remote answer: " + offer.ToString());
// Provide the answer back to the local session.
result = localSession.SetRemoteDescription(SIP.App.SdpType.answer, answer);
logger.LogDebug($"Set remote description on local session result {result}.");
Assert.Equal(SetDescriptionResultEnum.OK, result);
localSession.Close("normal");
remoteSession.Close("normal");
}
public void CreateInstanceUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
RTPSession rtpSession = new RTPSession(true, true, true);
// Add a track to the session in order to initialise the RTPChannel.
MediaStreamTrack dummyTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, new List<SDPMediaFormat> { new SDPMediaFormat(SDPMediaFormatsEnum.PCMU) });
rtpSession.addTrack(dummyTrack);
var iceSession = new IceSession(rtpSession.GetRtpChannel(SDPMediaTypesEnum.audio), RTCIceComponent.rtp);
Assert.NotNull(iceSession);
}
public async Task HandleInvalidSdpPortOnAnswerUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport();
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = @"INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK57441c4980b94e1686a06ae080be2935;rport
To: <sip:[email protected]>
From: <sip:0.0.0.0:0>;tag=MYILIYPHQD
Call-ID: ddf0e5a9687b4745925438da9000445d
CSeq: 1 INVITE
Max-Forwards: 70
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE
Content-Length: 0
v=0
o=- 1838015445 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 79762 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv";
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Any, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
var uas = userAgent.AcceptCall(inviteReq);
RTPSession rtpSession = new RTPSession(false, false, false);
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(audioTrack);
var result = await userAgent.Answer(uas, rtpSession);
Assert.False(result);
rtpSession.Close("normal");
}
public void ModifiedWellKnownFormatIDUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
string remoteSdp =
@"v=0
o=- 1986548327 0 IN IP4 127.0.0.1
c=IN IP4 127.0.0.1
m=audio 60640 RTP/AVP 8 12
a=rtpmap:8 OPUS/48000/2
a=rtpmap:12 PCMA/8000";
// Create a local session with an audio track.
RTPSession rtpSession = new RTPSession(false, false, false);
MediaStreamTrack localAudioTrack = new MediaStreamTrack(
SDPWellKnownMediaFormatsEnum.PCMU,
SDPWellKnownMediaFormatsEnum.PCMA,
SDPWellKnownMediaFormatsEnum.G722);
rtpSession.addTrack(localAudioTrack);
var offer = SDP.ParseSDPDescription(remoteSdp);
logger.LogDebug($"Remote offer: {offer}");
var result = rtpSession.SetRemoteDescription(SIP.App.SdpType.offer, offer);
logger.LogDebug($"Set remote description on local session result {result}.");
Assert.Equal(SetDescriptionResultEnum.OK, result);
Assert.Equal(12, rtpSession.AudioLocalTrack.Capabilities.Single(x => x.Name() == "PCMA").ID);
Assert.Equal("PCMA", rtpSession.GetSendingFormat(SDPMediaTypesEnum.audio).Name());
var answer = rtpSession.CreateAnswer(null);
logger.LogDebug($"Local answer: {answer}");
Assert.Equal(12, answer.Media.Single().MediaFormats.Single().Key);
Assert.Equal("PCMA", answer.Media.Single().MediaFormats.Single().Value.Name());
rtpSession.Close("normal");
}
public void SetRemoteSDPNoMediaStreamAttributeUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
string remoteSdp =
@"v=0
o=- 1986548327 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 60640 RTP/AVP 0 111
a=rtpmap:0 PCMU/8000
a=rtpmap:111 OPUS/48000/2";
// Create a local session with an audio track.
RTPSession rtpSession = new RTPSession(false, false, false);
MediaStreamTrack localAudioTrack = new MediaStreamTrack(new AudioFormat(SDPWellKnownMediaFormatsEnum.PCMU));
rtpSession.addTrack(localAudioTrack);
var offer = SDP.ParseSDPDescription(remoteSdp);
logger.LogDebug($"Remote offer: {offer}");
var result = rtpSession.SetRemoteDescription(SIP.App.SdpType.offer, offer);
logger.LogDebug($"Set remote description on local session result {result}.");
Assert.Equal(SetDescriptionResultEnum.OK, result);
var answer = rtpSession.CreateAnswer(null);
logger.LogDebug($"Local answer: {answer}");
Assert.Equal(0, rtpSession.AudioLocalTrack.Capabilities.Single(x => x.Name() == "PCMU").ID);
Assert.Equal(MediaStreamStatusEnum.SendRecv, rtpSession.AudioLocalTrack.StreamStatus);
Assert.NotNull(rtpSession.AudioRemoteTrack);
Assert.Equal(MediaStreamStatusEnum.SendRecv, rtpSession.AudioRemoteTrack.StreamStatus);
rtpSession.Close("normal");
}
public void NoRemoteMediaTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
RTPSession localSession = new RTPSession(false, false, false);
MediaStreamTrack localAudioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
localSession.addTrack(localAudioTrack);
var remoteOffer = new SDP();
var result = localSession.SetRemoteDescription(SIP.App.SdpType.offer, remoteOffer);
logger.LogDebug($"Set remote description on local session result {result}.");
Assert.Equal(SetDescriptionResultEnum.NoRemoteMedia, result);
localSession.Close("normal");
}
public void InvalidPortInRemoteOfferTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
RTPSession localSession = new RTPSession(false, false, false);
MediaStreamTrack localAudioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
localSession.addTrack(localAudioTrack);
var remoteOffer = new SDP();
remoteOffer.SessionId = Crypto.GetRandomInt(5).ToString();
remoteOffer.Connection = new SDPConnectionInformation(IPAddress.Loopback);
SDPMediaAnnouncement audioAnnouncement = new SDPMediaAnnouncement(
SDPMediaTypesEnum.audio,
66000,
new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
audioAnnouncement.Transport = RTPSession.RTP_MEDIA_PROFILE;
remoteOffer.Media.Add(audioAnnouncement);
var result = localSession.SetRemoteDescription(SIP.App.SdpType.offer, remoteOffer);
logger.LogDebug($"Set remote description on local session result {result}.");
Assert.Null(localSession.AudioDestinationEndPoint);
localSession.Close("normal");
}
public void CheckCreateOfferWithIPv6BindAddressAnswerTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
// Create two RTP sessions. First one acts as the local session to generate the offer.
// Second one acts as the remote session to generate the answer.
RTPSession localSession = new RTPSession(false, false, false, IPAddress.IPv6Loopback);
MediaStreamTrack localAudioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
localSession.addTrack(localAudioTrack);
// Generate the offer to send to the remote party.
var offer = localSession.CreateOffer(null);
logger.LogDebug("Local offer: " + offer.ToString());
Assert.True(IPAddress.IPv6Loopback.Equals(IPAddress.Parse(offer.Connection.ConnectionAddress)));
localSession.Close("normal");
}
static void Main()
{
Console.WriteLine("SIPSorcery SIP to WebRTC example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
Log = AddConsoleLogger();
//EnableTraceLogs(sipTransport);
// Start web socket.
Console.WriteLine("Starting web socket server...");
var webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT);
webSocketServer.AddWebSocketService <WebRTCWebSocketPeer>("/", (peer) => peer.CreatePeerConnection = CreatePeerConnection);
webSocketServer.Start();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
// Create a SIP user agent to receive a call from a remote SIP client.
// Wire up event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null, true);
// We're only answering SIP calls, not placing them.
userAgent.OnCallHungup += (dialog) =>
{
Log.LogInformation($"Call hungup by remote party.");
exitCts.Cancel();
};
userAgent.ServerCallCancelled += (uas) => Log.LogInformation("Incoming call cancelled by caller.");
userAgent.OnIncomingCall += async(ua, req) =>
{
Log.LogInformation($"Incoming call request from {req.RemoteSIPEndPoint}: {req.StatusLine}.");
var incomingCall = userAgent.AcceptCall(req);
var rtpSession = new RTPSession(false, false, false);
rtpSession.AcceptRtpFromAny = true;
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false,
new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(audioTrack);
await userAgent.Answer(incomingCall, rtpSession);
rtpSession.OnRtpPacketReceived += ForwardMediaToPeerConnection;
Log.LogInformation($"Answered incoming call from {req.Header.From.FriendlyDescription()} at {req.RemoteSIPEndPoint}.");
_rtpSession = rtpSession;
};
Console.WriteLine($"Waiting for browser web socket connection to {webSocketServer.Address}:{webSocketServer.Port}...");
var contactURI = new SIPURI(SIPSchemesEnum.sip, sipTransport.GetSIPChannels().First().ListeningSIPEndPoint);
Console.WriteLine($"Waiting for incoming SIP call to {contactURI}.");
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
_rtpSession?.Close("app exit");
if (userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
public async Task HandleInvalidSdpPortOnPlaceCallUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
// This transport will act as the call receiver. It allows the test to provide a
// tailored response to an incoming call.
SIPTransport calleeTransport = new SIPTransport();
// This transport will be used by the SIPUserAgent being tested to place the call.
SIPTransport callerTransport = new SIPTransport();
RTPSession rtpSession = new RTPSession(false, false, false);
try
{
calleeTransport.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0));
calleeTransport.SIPTransportRequestReceived += async(lep, rep, req) =>
{
if (req.Method != SIPMethodsEnum.INVITE)
{
SIPResponse notAllowedResponse = SIPResponse.GetResponse(req, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await calleeTransport.SendResponseAsync(notAllowedResponse);
}
else
{
UASInviteTransaction uasTransaction = new UASInviteTransaction(calleeTransport, req, null);
var uas = new SIPServerUserAgent(calleeTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
var answerSdp = @"
v=0
o=- 1838015445 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 79762 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv";
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp, null, SIPDialogueTransferModesEnum.NotAllowed);
}
};
SIPUserAgent userAgent = new SIPUserAgent(callerTransport, null);
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(audioTrack);
SIPURI dstUri = new SIPURI(SIPSchemesEnum.sip, calleeTransport.GetSIPChannels().First().ListeningSIPEndPoint);
var result = await userAgent.Call(dstUri.ToString(), null, null, rtpSession);
Assert.False(result);
}
finally
{
rtpSession?.Close("normal");
callerTransport?.Shutdown();
calleeTransport?.Shutdown();
}
}
//private delegate void MediaSampleReadyDelegate(SDPMediaTypesEnum mediaType, uint duration, byte[] sample);
//private static event MediaSampleReadyDelegate OnMediaFromSIPSampleReady;
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery SIP to WebRTC example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
Log = AddConsoleLogger();
// Start web socket.
Console.WriteLine("Starting web socket server...");
_webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT, true);
_webSocketServer.SslConfiguration.ServerCertificate = new X509Certificate2(WEBSOCKET_CERTIFICATE_PATH);
_webSocketServer.SslConfiguration.CheckCertificateRevocation = false;
//_webSocketServer.Log.Level = WebSocketSharp.LogLevel.Debug;
_webSocketServer.AddWebSocketService <SDPExchange>("/", (sdpExchanger) =>
{
sdpExchanger.WebSocketOpened += SendSDPOffer;
sdpExchanger.SDPAnswerReceived += SDPAnswerReceived;
});
_webSocketServer.Start();
Console.WriteLine($"Waiting for browser web socket connection to {_webSocketServer.Address}:{_webSocketServer.Port}...");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//EnableTraceLogs(sipTransport);
RTPSession rtpSession = null;
// Create a SIP user agent to receive a call from a remote SIP client.
// Wire up event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null);
// We're only answering SIP calls, not placing them.
userAgent.OnCallHungup += (dialog) =>
{
Log.LogInformation($"Call hungup by remote party.");
exitCts.Cancel();
};
userAgent.ServerCallCancelled += (uas) => Log.LogInformation("Incoming call cancelled by caller.");
sipTransport.SIPTransportRequestReceived += async(localEndPoint, remoteEndPoint, sipRequest) =>
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (userAgent?.IsCallActive == true)
{
Log.LogWarning($"Busy response returned for incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
// If we are already on a call return a busy response.
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
SIPResponse busyResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BusyHere, null);
uasTransaction.SendFinalResponse(busyResponse);
}
else
{
Log.LogInformation($"Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = userAgent.AcceptCall(sipRequest);
rtpSession = new RTPSession(false, false, false);
rtpSession.AcceptRtpFromAny = true;
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(audioTrack);
await userAgent.Answer(incomingCall, rtpSession);
rtpSession.OnRtpPacketReceived += (ep, mediaType, rtpPacket) => ForwardMedia(mediaType, rtpPacket);
Log.LogInformation($"Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}
}
else
{
Log.LogDebug($"SIP {sipRequest.Method} request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowedResponse);
}
};
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
rtpSession?.Close("app exit");
if (userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}