yate学习--关于yrtpchan模块

    转载请声明出处：http://blog.csdn.net/u012377333

    本文简介：yrtpchan是一个关于yate自定义的处理rtp协议的模块（关于rtp协议的知识，大家可以网上百度、谷歌）。本文主要是对于yrtpchan处理rtp.chan消息做了简要的分析和理解。

    对于yate的每个模块如何去分析和查看（因为没有main函数和其他的函数调用该模块的函数，所以一般常规的阅读代码的方式不是很好），首先是从继承Module类去查看。关于yrtpchan模块，就是从这里开始查看了：

class YRTPPlugin : public Module{public:    YRTPPlugin();    virtual ~YRTPPlugin();    virtual void initialize();    virtual bool received(Message& msg, int id);    virtual void statusParams(String& str);    virtual void statusDetail(String& str);    virtual void genUpdate(Message& msg);private:    bool reflectSetup(Message& msg, const char* id, RTPTransport& rtp, const char* rHost, const char* leg);    bool reflectStart(Message& msg, const char* id, RTPTransport& rtp, SocketAddr& rAddr);    void reflectDrop(YRTPReflector*& refl, Lock& mylock);    void reflectExecute(Message& msg);    void reflectAnswer(Message& msg, bool ignore);    void reflectHangup(Message& msg);    bool m_first;};

还定义了一个静态的全局变量

static YRTPPlugin splugin;

在加载模块的时候会调用

<pre name="code" class="cpp">void YRTPPlugin::initialize(){    Output("Initializing module YRTP");    Configuration cfg(Engine::configFile("yrtpchan"));    s_ipv6 = SocketAddr::supports(SocketAddr::IPv6) &&cfg.getBoolValue("general","ipv6_support",false);    s_minport = cfg.getIntValue("general","minport",MIN_PORT);    s_maxport = cfg.getIntValue("general","maxport",MAX_PORT);    s_bufsize = cfg.getIntValue("general","buffer",BUF_SIZE);    s_minJitter = cfg.getIntValue("general","minjitter",50);    s_maxJitter = cfg.getIntValue("general","maxjitter",Engine::clientMode() ? 120 : 0);    s_tos = cfg.getIntValue("general","tos",Socket::tosValues());    s_udpbuf = cfg.getIntValue("general","udpbuf",0);    s_localip = cfg.getValue("general","localip");    s_autoaddr = cfg.getBoolValue("general","autoaddr",true);    s_anyssrc = cfg.getBoolValue("general","anyssrc",true);    s_padding = cfg.getIntValue("general","padding",0);    s_rtcp = cfg.getBoolValue("general","rtcp",true);    s_interval = cfg.getIntValue("general","rtcp_interval",4500);    s_drill = cfg.getBoolValue("general","drillhole",Engine::clientMode());    s_monitor = cfg.getBoolValue("general","monitoring",false);    s_sleep = cfg.getIntValue("general","defsleep",5);    RTPGroup::setMinSleep(cfg.getIntValue("general","minsleep"));    s_priority = Thread::priority(cfg.getValue("general","thread"));    s_rtpWarnSeq = cfg.getBoolValue("general","rtp_warn_seq",true);    s_timeout = cfg.getIntValue("timeouts","timeout",3000);    s_udptlTimeout = cfg.getIntValue("timeouts","udptl_timeout",25000);    s_notifyMsg = cfg.getValue("timeouts","notifymsg");    s_warnFirst = cfg.getBoolValue("timeouts","warnfirst",true);    s_warnLater = cfg.getBoolValue("timeouts","warnlater",false);    setup();    if (m_first) {m_first = false;installRelay(Execute,50);installRelay(Ringing,50);installRelay(Progress,50);installRelay(Answered,50);installRelay(Private,"chan.hangup",50);Engine::install(new AttachHandler);Engine::install(new RtpHandler);Engine::install(new DTMFHandler);    }}

读取配置文件里面的参数，如果没有配置文件的话，使用的都是默认参数

    Configuration cfg(Engine::configFile("yrtpchan"));    s_ipv6 = SocketAddr::supports(SocketAddr::IPv6) &&cfg.getBoolValue("general","ipv6_support",false);    s_minport = cfg.getIntValue("general","minport",MIN_PORT);    s_maxport = cfg.getIntValue("general","maxport",MAX_PORT);    s_bufsize = cfg.getIntValue("general","buffer",BUF_SIZE);    s_minJitter = cfg.getIntValue("general","minjitter",50);    s_maxJitter = cfg.getIntValue("general","maxjitter",Engine::clientMode() ? 120 : 0);    s_tos = cfg.getIntValue("general","tos",Socket::tosValues());    s_udpbuf = cfg.getIntValue("general","udpbuf",0);    s_localip = cfg.getValue("general","localip");    s_autoaddr = cfg.getBoolValue("general","autoaddr",true);    s_anyssrc = cfg.getBoolValue("general","anyssrc",true);    s_padding = cfg.getIntValue("general","padding",0);    s_rtcp = cfg.getBoolValue("general","rtcp",true);    s_interval = cfg.getIntValue("general","rtcp_interval",4500);    s_drill = cfg.getBoolValue("general","drillhole",Engine::clientMode());    s_monitor = cfg.getBoolValue("general","monitoring",false);    s_sleep = cfg.getIntValue("general","defsleep",5);    RTPGroup::setMinSleep(cfg.getIntValue("general","minsleep"));    s_priority = Thread::priority(cfg.getValue("general","thread"));    s_rtpWarnSeq = cfg.getBoolValue("general","rtp_warn_seq",true);    s_timeout = cfg.getIntValue("timeouts","timeout",3000);    s_udptlTimeout = cfg.getIntValue("timeouts","udptl_timeout",25000);    s_notifyMsg = cfg.getValue("timeouts","notifymsg");    s_warnFirst = cfg.getBoolValue("timeouts","warnfirst",true);    s_warnLater = cfg.getBoolValue("timeouts","warnlater",false);

声明该模块可以处理那些消息

installRelay(Execute,50);installRelay(Ringing,50);installRelay(Progress,50);installRelay(Answered,50);installRelay(Private,"chan.hangup",50);Engine::install(new AttachHandler);Engine::install(new RtpHandler);Engine::install(new DTMFHandler);

这里我们重点查看处理RTP消息的类

class RtpHandler : public MessageHandler{public:    RtpHandler() : MessageHandler("chan.rtp",100,splugin.name()) { }    virtual bool received(Message &msg);};

每当有模块发布消息"chan.rtp"的时候，都会进入received函数

bool RtpHandler::received(Message &msg){    bool udptl = false;    const String& trans = msg[YSTRING("transport")];    if (trans && !trans.startsWith("RTP/")) {if (trans &= "udptl")    udptl = true;else    return false;    }    Debug(&splugin,DebugAll,"%s message received",(trans ? trans.c_str() : "No-transport"));    bool terminate = msg.getBoolValue(YSTRING("terminate"),false);    const String& dir = msg[YSTRING("direction")];    RTPSession::Direction direction = terminate ? RTPSession::FullStop : RTPSession::SendRecv;    bool d_recv = false;    bool d_send = false;    if (dir == YSTRING("bidir")) {d_recv = true;d_send = true;    }    else if (dir == YSTRING("receive")) {d_recv = true;direction = RTPSession::RecvOnly;    }    else if (dir == YSTRING("send")) {d_send = true;direction = RTPSession::SendOnly;    }    CallEndpoint* ch = YOBJECT(CallEndpoint,msg.userData());    DataEndpoint* de = YOBJECT(DataEndpoint,msg.userData());    const char* media = udptl ? "image" : "audio";    media = msg.getValue(YSTRING("media"),(de ? de->name().c_str() : media));    RefPointer<YRTPWrapper> w = YRTPWrapper::find(ch,media);    if (w)Debug(&splugin,DebugAll,"Wrapper %p found by CallEndpoint %p",(YRTPWrapper*)w,ch);    else {const String& rid = msg[YSTRING("rtpid")];w = YRTPWrapper::find(rid);if (w)    Debug(&splugin,DebugAll,"Wrapper %p found by ID '%s'",(YRTPWrapper*)w,rid.c_str());    }    if (w)w->deref();    if (terminate) {if (w) {    if (w->host())msg.setParam("localip",w->host());    if (w->port())msg.setParam("localport",String(w->port()));    w->terminate(msg);    msg.setParam("status","terminated");    return true;}return false;    }    if (!(ch || de || w)) {Debug(&splugin,DebugWarn,"Neither call channel nor RTP wrapper found!");return false;    }    const String& rip = msg[YSTRING("remoteip")];    const char* status = "updated";    if (!w) {// it would be pointless to create an unreferenced wrapperif (!(d_recv || d_send))    return false;String lip(msg.getValue(YSTRING("localip")));bool ipv6 = msg.getBoolValue(YSTRING("ipv6_support"),s_ipv6);if (lip.null())    YRTPWrapper::guessLocal(rip,lip,ipv6);if (lip.null()) {    Debug(&splugin,DebugWarn,"RTP request with no local address!");    return false;}status = "created";w = new YRTPWrapper(lip,ch,media,direction,msg,udptl,ipv6);w->setMaster(msg.getValue(YSTRING("id")));w->deref();    }    else if (w->valid())w->addDirection(direction);    elsereturn false;    if (d_recv) {if (ch && !ch->getSource(media)) {    DataSource* s = w->getSource();    ch->setSource(s,media);    s->deref();}else if (de && !de->getSource()) {    DataSource* s = w->getSource();    de->setSource(s);    s->deref();}    }    if (d_send) {if (ch && !ch->getConsumer(media)) {    DataConsumer* c = w->getConsumer();    ch->setConsumer(c,media);    c->deref();}else if (de && !de->getConsumer()) {    DataConsumer* c = w->getConsumer();    de->setConsumer(c);    c->deref();}    }    if (w->refcount() <= 1)return false;    w->setParams(rip,msg);    w->setFaxDivert(msg);    msg.setParam("localip",w->host());    msg.setParam("localport",String(w->port()));    msg.setParam("rtpid",w->id());    msg.setParam("status",status);    if (msg.getBoolValue(YSTRING("getsession"),!msg.userData()))msg.userData(w);    return true;}

这个里面的内容比较多，我感觉可以分成两个部分来查看：

一个是

RefPointer<YRTPWrapper> w = YRTPWrapper::find(ch,media);

<pre name="code" class="cpp">w = YRTPWrapper::find(rid);

另外一个

w = new YRTPWrapper(lip,ch,media,direction,msg,udptl,ipv6);

不知道大家有没有注意过，yate对于转发两个yate client的rtp包的时候，是创建了两个ip、port去对应两个客户端的接受和发送。这里举一个比较简单的例子：

S是服务器，A和C是客户端，假如是A呼叫C，从S上面抓包（rtp）的格式大致是这样的：

a.ip:port-------------------------------------------->s.ip2:port2

s.ip1:port1----------------------------------------->c.ip:port

s.ip2:port2----------------------------------------->a.ip:port

c.ip:port--------------------------------------------->s.ip1:port

思考：

也就是说服务器创建了两个rtp的套接字去发送和接受两个客户端的媒体数据，为什么会这样？为什么不是一个套接字来处理A到C的媒体数据另外一个套接字来处理C到A的媒体数据？

未完，待续......