使用Ozeki库在C#中实现SIP呼叫触发状态异常问题排查

问题描述 #

使用Ozeki库在C#中实现SIP呼叫，对接Asterisk电话系统，已成功完成SIP账号注册（输出Registration succeeded - Online!），但呼叫状态进入Setup后直接变为Error，程序输出如下：

Registration succeeded - Online!
Call state: Setup.
Call state: Error.

错误排查 #

1. 端口配置错误：代码中domainPort设置为5038，这是Asterisk的AMI（管理接口）端口，而非SIP服务默认端口（默认是5060），这是导致呼叫失败的核心原因。
2. 错误信息未捕获：当前代码未处理CallState.Error状态的具体异常信息，无法定位具体错误原因。
3. 拨号号码格式问题：拨号号码00994122000020可能不符合Asterisk的拨号规则，比如需要添加出局前缀（如9）或号码格式不被路由规则匹配。

代码修改方案 #

核心修改点 #

1. 修正SIP端口为Asterisk默认的5060
2. 添加Error状态的异常信息捕获
3. 可选：调整拨号号码格式适配Asterisk路由规则

修改后的完整代码 #

class Program
{
    static ISoftPhone softphone;
    static IPhoneLine phoneLine;
    static IPhoneCall call;
    static Microphone microphone;
    static Speaker speaker;
    static MediaConnector connector;
    static PhoneCallAudioSender mediaSender;
    static PhoneCallAudioReceiver mediaReceiver;

    public static void Main(string[] args)
    {
        // 创建软电话对象，指定RTP端口范围
        softphone = SoftPhoneFactory.CreateSoftPhone(5000, 10000);

        // SIP账号注册信息
        var registrationRequired = true;
        var userName = "111";
        var displayName = "111";
        var authenticationId = "111";
        var registerPassword = "pin111";
        var domainHost = "localhost";
        var domainPort = 5060; // 修正为Asterisk SIP默认端口

        var account = new SIPAccount(registrationRequired, displayName, userName, authenticationId, registerPassword, domainHost, domainPort);

        RegisterAccount(account);

        microphone = Microphone.GetDefaultDevice();
        speaker = Speaker.GetDefaultDevice();
        mediaSender = new PhoneCallAudioSender();
        mediaReceiver = new PhoneCallAudioReceiver();
        connector = new MediaConnector();

        Console.WriteLine("\nDone");
        Console.ReadLine();
    }

    static void RegisterAccount(SIPAccount account)
    {
        try
        {
            phoneLine = softphone.CreatePhoneLine(account);
            phoneLine.RegistrationStateChanged += line_RegStateChanged;
            softphone.RegisterPhoneLine(phoneLine);
        }
        catch (Exception ex)
        {
            Console.WriteLine("SIP注册错误: " + ex.Message);
        }
    }

    static void line_RegStateChanged(object sender, RegistrationStateChangedArgs e)
    {
        if (e.State == RegState.NotRegistered || e.State == RegState.Error)
            Console.WriteLine("注册失败!");

        if (e.State == RegState.RegistrationSucceeded)
        {
            Console.WriteLine("注册成功 - 在线!");
            CreateCall();
        }
    }

    private static void CreateCall()
    {
        // 根据Asterisk拨号规则调整号码，比如添加出局前缀9
        var numberToDial = "900994122000020";
        call = softphone.CreateCallObject(phoneLine, numberToDial);
        call.CallStateChanged += call_CallStateChanged;
        call.Start();
    }

    private static void SetupDevices()
    {
        connector.Connect(microphone, mediaSender);
        connector.Connect(mediaReceiver, speaker);

        mediaSender.AttachToCall(call);
        mediaReceiver.AttachToCall(call);

        microphone.Start();
        speaker.Start();
    }

    static void call_CallStateChanged(object sender, CallStateChangedArgs e)
    {
        Console.WriteLine("呼叫状态: {0}.", e.State);

        // 捕获Error状态的具体异常信息
        if (e.State == CallState.Error)
        {
            Console.WriteLine("呼叫错误详情: " + e.Error.Message);
        }

        if (e.State == CallState.Answered)
            SetupDevices();
    }
}

额外验证步骤 #

• 确认Asterisk中已正确配置SIP账号111，允许呼出权限
• 检查Asterisk的拨号规则（extensions.conf）是否包含目标号码的路由
• 确保防火墙允许5060（SIP）及5000-10000（RTP）端口的UDP流量

适用于Asterisk的SIP呼叫库推荐 #

• SIP.NET：轻量级开源库，原生支持SIP协议，与Asterisk兼容性出色，适合基础到中级VoIP开发
• PJSIP.NET：基于成熟的PJSIP库封装，功能全面，支持复杂呼叫场景（如视频呼叫、加密通话）
• Twilio Voice SDK：适合云呼叫集成场景，无需维护本地Asterisk，快速实现呼叫功能

内容的提问来源于stack exchange，提问作者Farid Huseynli