Open orhanyk opened 4 months ago
Using the Sip Sorcery library, I create a voice call on the sip server and answer this call. After a while, the call closes automatically. What could be the reason for this? Do I need any extra configuration?
`var rtpSession = CreateRtpSession(userAgent, null); var callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
if (callResult) { await rtpSession.Start(); //_calls.TryAdd(ua.Dialogue.CallId, ua); }`
`private static VoIPMediaSession CreateRtpSession(SIPUserAgent ua, string dst) { List codecs = new List { AudioCodecsEnum.PCMU, AudioCodecsEnum.PCMA };
var audioSource = AudioSourcesEnum.SineWave; if (string.IsNullOrEmpty(dst) || !Enum.TryParse(dst, out audioSource)) { audioSource = AudioSourcesEnum.Music; } _log.Info($"RTP audio session source set to {audioSource}."); AudioExtrasSource audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions { AudioSource = audioSource }); audioExtrasSource.RestrictFormats(formats => codecs.Contains(formats.Codec)); var rtpAudioSession = new VoIPMediaSession(new MediaEndPoints { AudioSource = audioExtrasSource }); rtpAudioSession.AcceptRtpFromAny = true; // Wire up the event handler for RTP packets received from the remote party. rtpAudioSession.OnRtpPacketReceived += (ep, type, rtp) => OnRtpPacketReceived(ua, type, rtp); rtpAudioSession.OnTimeout += (mediaType) => { if (ua?.Dialogue != null) { _log.Info($"RTP timeout on call with {ua.Dialogue.RemoteTarget}, hanging up."); } else { _log.Info($"RTP timeout on incomplete call, closing RTP session."); } ua.Hangup(); }; return rtpAudioSession;
}`
Using the Sip Sorcery library, I create a voice call on the sip server and answer this call. After a while, the call closes automatically. What could be the reason for this? Do I need any extra configuration?
`var rtpSession = CreateRtpSession(userAgent, null); var callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
if (callResult) { await rtpSession.Start(); //_calls.TryAdd(ua.Dialogue.CallId, ua); }`
`private static VoIPMediaSession CreateRtpSession(SIPUserAgent ua, string dst) { List codecs = new List { AudioCodecsEnum.PCMU, AudioCodecsEnum.PCMA };
}`