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Fixed RTP event negotiation to set common ID if remote party is not using 101. Resolves #1251. (#1280)
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Lines changed: 152 additions & 55 deletions

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‎examples/SIPExamples/ReceiveDtmf/Program.cs‎

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//-----------------------------------------------------------------------------
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// Filename: Program.cs
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//
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// Description: Sample program of how to receive DTMF tones using RTP events
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// as specified in RFC2833.
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//
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// Author(s):
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// Aaron Clauson ([email protected])
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//
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// History:
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// 12 Jan 2025 Aaron Clauson Created, Dublin, Ireland.
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//
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// License:
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// BSD 3-Clause "New" or "Revised" License, see included LICENSE.md file.
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//-----------------------------------------------------------------------------
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17+
using System;
18+
using System.Linq;
19+
using System.Net;
20+
using Microsoft.Extensions.Logging;
21+
using Microsoft.Extensions.Logging.Abstractions;
22+
using Serilog;
23+
using Serilog.Extensions.Logging;
24+
using SIPSorcery.Media;
25+
using SIPSorcery.Net;
26+
using SIPSorcery.SIP;
27+
using SIPSorcery.SIP.App;
28+
using TinyJson;
29+
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namespace demo
31+
{
32+
class Program
33+
{
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private static int SIP_LISTEN_PORT = 5060;
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36+
private static Microsoft.Extensions.Logging.ILogger Log = NullLogger.Instance;
37+
38+
private static SIPTransport _sipTransport;
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40+
static void Main()
41+
{
42+
Console.WriteLine("SIPSorcery Receive DTMF Demo");
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44+
Log = AddConsoleLogger();
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46+
_sipTransport = new SIPTransport();
47+
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
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49+
var userAgent = new SIPUserAgent(_sipTransport, null, true);
50+
userAgent.ServerCallCancelled += (uas, cancelReq) => Log.LogDebug("Incoming call cancelled by remote party.");
51+
userAgent.OnCallHungup += (dialog) => Log.LogDebug("Call hungup.");
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userAgent.OnIncomingCall += async (ua, req) =>
53+
{
54+
Log.LogDebug($"SDP Offer:\r\n{req.Body}");
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56+
VoIPMediaSession voipMediaSession = new VoIPMediaSession();
57+
voipMediaSession.AcceptRtpFromAny = true;
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59+
var uas = userAgent.AcceptCall(req);
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await userAgent.Answer(uas, voipMediaSession);
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var sdpAnswer = voipMediaSession.CreateAnswer(IPAddress.Loopback);
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Log.LogDebug($"SDP Answer:\r\n{sdpAnswer.ToString()}");
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};
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userAgent.OnDtmfTone += (tone, duration) => Log.LogInformation($"DTMF tone {tone} of duration {duration}ms received.");
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Console.WriteLine("press any key to exit...");
70+
Console.Read();
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// Clean up.
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_sipTransport.Shutdown();
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}
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/// <summary>
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/// Adds a console logger. Can be omitted if internal SIPSorcery debug and warning messages are not required.
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/// </summary>
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private static Microsoft.Extensions.Logging.ILogger AddConsoleLogger()
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{
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var serilogLogger = new LoggerConfiguration()
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.Enrich.FromLogContext()
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.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
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.WriteTo.Console()
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.CreateLogger();
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var factory = new SerilogLoggerFactory(serilogLogger);
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SIPSorcery.LogFactory.Set(factory);
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return factory.CreateLogger<Program>();
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}
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}
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}

‎examples/SIPExamples/ReceiveDtmf/ReceiveDtmf.csproj‎

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<Project Sdk="Microsoft.NET.Sdk">
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<PropertyGroup>
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<OutputType>Exe</OutputType>
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<TargetFramework>net8.0-windows10.0.17763.0</TargetFramework>
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</PropertyGroup>
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<ItemGroup>
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<PackageReference Include="Microsoft.Extensions.Logging" Version="9.0.0" />
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<PackageReference Include="NAudio" Version="2.2.1" />
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<PackageReference Include="Serilog.Extensions.Logging" Version="9.0.0" />
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<PackageReference Include="Serilog.Sinks.Console" Version="6.0.0" />
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<PackageReference Include="System.Net.Security" Version="4.3.2" />
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</ItemGroup>
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<ItemGroup>
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<ProjectReference Include="..\..\..\src\SIPSorcery.csproj" />
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</ItemGroup>
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</Project>

‎examples/SIPExamples/SIPExamples.sln‎

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@@ -43,6 +43,8 @@ Project("{9A19103F-16F7-4668-BE54-9A1E7A4F7556}") = "NotifierClient", "NotifierC
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EndProject
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Project("{9A19103F-16F7-4668-BE54-9A1E7A4F7556}") = "SIPSorcery", "..\..\src\SIPSorcery.csproj", "{DE20DB0F-AB95-47DD-B51F-A2A7EC12FA01}"
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EndProject
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Project("{FAE04EC0-301F-11D3-BF4B-00C04F79EFBC}") = "ReceiveDtmf", "ReceiveDtmf\ReceiveDtmf.csproj", "{428BFA9A-F082-4444-9E04-30AD60B5E794}"
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EndProject
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Global
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GlobalSection(SolutionConfigurationPlatforms) = preSolution
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Debug|Any CPU = Debug|Any CPU
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{DE20DB0F-AB95-47DD-B51F-A2A7EC12FA01}.Release|x64.Build.0 = Release|Any CPU
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{DE20DB0F-AB95-47DD-B51F-A2A7EC12FA01}.Release|x86.ActiveCfg = Release|Any CPU
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{DE20DB0F-AB95-47DD-B51F-A2A7EC12FA01}.Release|x86.Build.0 = Release|Any CPU
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{428BFA9A-F082-4444-9E04-30AD60B5E794}.Debug|Any CPU.ActiveCfg = Debug|Any CPU
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{428BFA9A-F082-4444-9E04-30AD60B5E794}.Debug|Any CPU.Build.0 = Debug|Any CPU
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{428BFA9A-F082-4444-9E04-30AD60B5E794}.Debug|x64.ActiveCfg = Debug|Any CPU
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{428BFA9A-F082-4444-9E04-30AD60B5E794}.Debug|x64.Build.0 = Debug|Any CPU
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{428BFA9A-F082-4444-9E04-30AD60B5E794}.Debug|x86.ActiveCfg = Debug|Any CPU
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{428BFA9A-F082-4444-9E04-30AD60B5E794}.Debug|x86.Build.0 = Debug|Any CPU
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{428BFA9A-F082-4444-9E04-30AD60B5E794}.Release|Any CPU.ActiveCfg = Release|Any CPU
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{428BFA9A-F082-4444-9E04-30AD60B5E794}.Release|Any CPU.Build.0 = Release|Any CPU
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{428BFA9A-F082-4444-9E04-30AD60B5E794}.Release|x64.ActiveCfg = Release|Any CPU
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{428BFA9A-F082-4444-9E04-30AD60B5E794}.Release|x64.Build.0 = Release|Any CPU
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{428BFA9A-F082-4444-9E04-30AD60B5E794}.Release|x86.ActiveCfg = Release|Any CPU
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{428BFA9A-F082-4444-9E04-30AD60B5E794}.Release|x86.Build.0 = Release|Any CPU
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EndGlobalSection
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GlobalSection(SolutionProperties) = preSolution
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HideSolutionNode = FALSE

‎examples/SIPExamples/SendDtmf/Program.cs‎

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@@ -1,7 +1,7 @@
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//-----------------------------------------------------------------------------
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// Filename: Program.cs
33
//
4-
// Description: An example of how top send DTMF tones in band (with specific RTP
4+
// Description: An example of how to send DTMF tones in band (with specific RTP
55
// packets) as specified in RFC2833.
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//
77
// Author(s):
@@ -20,7 +20,7 @@
2020
//
2121
// This example calls a destination (represented by DEFAULT_DESTINATION_SIP_URI)
2222
// and expects a SIP agent capable of receiving DTMF with RFC2833 to be listening.
23-
// What the receiving SIP does with the received DTMF is up to it. A good example
23+
// What the receiving SIP ageent does with the received DTMF is up to it. A good example
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// is to playback the presses via speech synthesis. The dialplan below is an
2525
// example of how to do that with Asterisk.
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//
@@ -43,16 +43,16 @@
4343
using Serilog;
4444
using Serilog.Extensions.Logging;
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using SIPSorcery.Media;
46+
using SIPSorcery.Net;
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using SIPSorcery.SIP;
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using SIPSorcery.SIP.App;
48-
using SIPSorceryMedia.Windows;
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5050
namespace SIPSorcery
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{
5252
class Program
5353
{
5454
//private static readonly string DEFAULT_DESTINATION_SIP_URI = "sip:*[email protected]"; // Custom Asterisk dialplan to speak back DTMF tones.
55-
private static readonly string DEFAULT_DESTINATION_SIP_URI = "sip:[email protected]:5080";
55+
private static readonly string DEFAULT_DESTINATION_SIP_URI = "sip:[email protected]:5060";
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5757
private static Microsoft.Extensions.Logging.ILogger Log = NullLogger.Instance;
5858

@@ -68,8 +68,7 @@ static async Task Main()
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6969
var sipTransport = new SIPTransport();
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var userAgent = new SIPUserAgent(sipTransport, null);
71-
var winAudioEP = new WindowsAudioEndPoint(new AudioEncoder());
72-
var voipMediaSession = new VoIPMediaSession(winAudioEP.ToMediaEndPoints());
71+
var voipMediaSession = new VoIPMediaSession();
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Console.WriteLine($"Calling {DEFAULT_DESTINATION_SIP_URI}.");
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await Task.Delay(1000);
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// Send the DTMF tones.
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Console.WriteLine("Sending DTMF tone 5.");
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await userAgent.SendDtmf(0x05);
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await Task.Delay(2000);
96+
Console.WriteLine("Sending DTMF tone 9.");
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await userAgent.SendDtmf(0x09);
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await Task.Delay(2000);
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Console.WriteLine("Sending DTMF tone 2.");
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await userAgent.SendDtmf(0x02);
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await Task.Delay(2000);
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116118
sipTransport.Shutdown();
117119
}
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119-
/// <summary>
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/// Enable detailed SIP log messages.
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/// </summary>
122-
private static void EnableTraceLogs(SIPTransport sipTransport)
123-
{
124-
sipTransport.SIPRequestInTraceEvent += (localEP, remoteEP, req) =>
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{
126-
Log.LogDebug($"Request received: {localEP}<-{remoteEP}");
127-
Log.LogDebug(req.ToString());
128-
};
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130-
sipTransport.SIPRequestOutTraceEvent += (localEP, remoteEP, req) =>
131-
{
132-
Log.LogDebug($"Request sent: {localEP}->{remoteEP}");
133-
Log.LogDebug(req.ToString());
134-
};
135-
136-
sipTransport.SIPResponseInTraceEvent += (localEP, remoteEP, resp) =>
137-
{
138-
Log.LogDebug($"Response received: {localEP}<-{remoteEP}");
139-
Log.LogDebug(resp.ToString());
140-
};
141-
142-
sipTransport.SIPResponseOutTraceEvent += (localEP, remoteEP, resp) =>
143-
{
144-
Log.LogDebug($"Response sent: {localEP}->{remoteEP}");
145-
Log.LogDebug(resp.ToString());
146-
};
147-
148-
sipTransport.SIPRequestRetransmitTraceEvent += (tx, req, count) =>
149-
{
150-
Log.LogDebug($"Request retransmit {count} for request {req.StatusLine}, initial transmit {DateTime.Now.Subtract(tx.InitialTransmit).TotalSeconds.ToString("0.###")}s ago.");
151-
};
152-
153-
sipTransport.SIPResponseRetransmitTraceEvent += (tx, resp, count) =>
154-
{
155-
Log.LogDebug($"Response retransmit {count} for response {resp.ShortDescription}, initial transmit {DateTime.Now.Subtract(tx.InitialTransmit).TotalSeconds.ToString("0.###")}s ago.");
156-
};
157-
}
158-
159121
/// <summary>
160122
/// Adds a console logger. Can be omitted if internal SIPSorcery debug and warning messages are not required.
161123
/// </summary>

‎examples/SIPExamples/SendDtmf/SendDtmf.csproj‎

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@@ -2,14 +2,13 @@
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<PropertyGroup>
44
<OutputType>Exe</OutputType>
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<TargetFramework>net8.0-windows10.0.17763.0</TargetFramework>
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<TargetFramework>net8</TargetFramework>
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</PropertyGroup>
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<ItemGroup>
99
<PackageReference Include="Microsoft.Extensions.Logging" Version="9.0.0" />
1010
<PackageReference Include="Serilog.Extensions.Logging" Version="9.0.0" />
1111
<PackageReference Include="Serilog.Sinks.Console" Version="6.0.0" />
12-
<PackageReference Include="SIPSorceryMedia.Windows" Version="8.0.7" />
1312
<PackageReference Include="System.Net.Security" Version="4.3.2" />
1413
</ItemGroup>
1514

‎src/app/Media/VoIPMediaSession.cs‎

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@@ -69,7 +69,7 @@ public VideoTestPatternSource TestPatternSource
6969
/// Default constructor which creates the simplest possible send only audio session. It does not
7070
/// wire up any devices or video processing.
7171
/// </summary>
72-
public VoIPMediaSession(Func<AudioFormat, bool> restrictFormats = null) : base(false, false, false)
72+
public VoIPMediaSession(Func<AudioFormat, bool> restrictFormats = null, bool noDtmfSupport = false) : base(false, false, false)
7373
{
7474
_audioExtrasSource = new AudioExtrasSource();
7575
_audioExtrasSource.OnAudioSourceEncodedSample += SendAudio;
@@ -81,6 +81,7 @@ public VoIPMediaSession(Func<AudioFormat, bool> restrictFormats = null) : base(f
8181
}
8282

8383
var audioTrack = new MediaStreamTrack(_audioExtrasSource.GetAudioSourceFormats());
84+
audioTrack.NoDtmfSupport = noDtmfSupport;
8485
base.addTrack(audioTrack);
8586
base.OnAudioFormatsNegotiated += AudioFormatsNegotiated;
8687

@@ -185,6 +186,11 @@ private void AudioFormatsNegotiated(List<AudioFormat> audoFormats)
185186
logger.LogDebug($"Setting audio source format to {audioFormat.FormatID}:{audioFormat.Codec} {audioFormat.ClockRate} (RTP clock rate {audioFormat.RtpClockRate}).");
186187
Media.AudioSource?.SetAudioSourceFormat(audioFormat);
187188
_audioExtrasSource.SetAudioSourceFormat(audioFormat);
189+
190+
if(AudioStream != null && AudioStream.LocalTrack.NoDtmfSupport == false)
191+
{
192+
logger.LogDebug($"Audio track negotiated DTMF payload ID {AudioStream.NegotiatedRtpEventPayloadID}.");
193+
}
188194
}
189195

190196
private void VideoFormatsNegotiated(List<VideoFormat> videoFormats)

‎src/net/RTP/RTPSession.cs‎

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@@ -175,7 +175,7 @@ public class RTPSession : IMediaSession, IDisposable
175175

176176
protected RtpSessionConfig rtpSessionConfig;
177177

178-
private Boolean m_acceptRtpFromAny = false;
178+
private bool m_acceptRtpFromAny = false;
179179
private string m_sdpSessionID = null; // Need to maintain the same SDP session ID for all offers and answers.
180180
private ulong m_sdpAnnouncementVersion = 0; // The SDP version needs to increase whenever the local SDP is modified (see https://tools.ietf.org/html/rfc6337#section-5.2.5).
181181
internal int m_rtpChannelsCount = 0; // Need to know the number of RTP Channels
@@ -1050,17 +1050,17 @@ public virtual SetDescriptionResultEnum SetRemoteDescription(SdpType sdpType, SD
10501050
else
10511051
{
10521052
// As proved by Azure implementation, we need to send based on capabilities of remote track. Azure return SDP with only one possible Codec (H264 107)
1053-
// but we receive frames based on our LocalRemoteTracks, so its possiblet o receive a frame with ID 122, for exemple, even when remote annoucement only have 107
1053+
// but we receive frames based on our LocalRemoteTracks, so it's possible to receive a frame with ID 122, for exemple, even when remote annoucement only have 107
10541054
// Thats why we changed line below to keep local track capabilities untouched as we can always do it during send/receive moment
10551055
capabilities = SDPAudioVideoMediaFormat.GetCompatibleFormats(currentMediaStream.RemoteTrack?.Capabilities, currentMediaStream.LocalTrack?.Capabilities);
1056-
//Keep same order of LocalTrack priority to prevent incorrect sending format
1056+
// Keep same order of LocalTrack priority to prevent incorrect sending format
10571057
SDPAudioVideoMediaFormat.SortMediaCapability(capabilities, currentMediaStream.LocalTrack?.Capabilities);
10581058

10591059
currentMediaStream.RemoteTrack.Capabilities = capabilities;
10601060

1061-
// Keep the local track's RTP event capability
1061+
// Adjust the local track's RTP event capability if the remote party has specified a different payload ID.
10621062
var currentLocalTrackCapabilities = currentMediaStream.LocalTrack.Capabilities;
1063-
SDPAudioVideoMediaFormat localRTPEventCapabilities;
1063+
SDPAudioVideoMediaFormat? localRTPEventCapabilities = null;
10641064
if (currentLocalTrackCapabilities.Any(x => x.Name().ToLower() == SDP.TELEPHONE_EVENT_ATTRIBUTE))
10651065
{
10661066
localRTPEventCapabilities = currentLocalTrackCapabilities.First(x => x.Name().ToLower() == SDP.TELEPHONE_EVENT_ATTRIBUTE);
@@ -1071,7 +1071,10 @@ public virtual SetDescriptionResultEnum SetRemoteDescription(SdpType sdpType, SD
10711071
}
10721072

10731073
currentMediaStream.LocalTrack.Capabilities = capabilities.Where(x => x.Name().ToLower() != SDP.TELEPHONE_EVENT_ATTRIBUTE).ToList();
1074-
currentMediaStream.LocalTrack.Capabilities.Add(localRTPEventCapabilities);
1074+
if (localRTPEventCapabilities != null)
1075+
{
1076+
currentMediaStream.LocalTrack.Capabilities.Add(localRTPEventCapabilities.Value);
1077+
}
10751078

10761079
if (currentMediaStream.MediaType == SDPMediaTypesEnum.audio)
10771080
{
@@ -1080,6 +1083,8 @@ public virtual SetDescriptionResultEnum SetRemoteDescription(SdpType sdpType, SD
10801083
if (!commonEventFormat.IsEmpty())
10811084
{
10821085
currentMediaStream.NegotiatedRtpEventPayloadID = commonEventFormat.ID;
1086+
currentMediaStream.LocalTrack.Capabilities.RemoveAll(x => x.Name().ToLower() == SDP.TELEPHONE_EVENT_ATTRIBUTE);
1087+
currentMediaStream.LocalTrack.Capabilities.Add(commonEventFormat);
10831088
}
10841089
}
10851090

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