sdphandler.cxx

Vovida 社区开源的 SIP 协议源码

bool validUlaw(SipSdp& sdp)
{

    assert(0);
    return false;
}


void
setupMedia(SipSdp& sipSdp, Data hostAddr, int port)
{
    setHost(sipSdp, hostAddr);


    // guarantees a valid SDP, at least vaguely.

    // do the media stuff
    SdpSession sdpDesc = sipSdp.getSdpDescriptor();
    // declare the media list pointer
    list < SdpMedia* > mediaList;
    // get the media list pointer from the session object
    mediaList = sdpDesc.getMediaList();

    list < SdpMedia* > ::iterator mediaIterator = mediaList.begin();

    SdpMedia* media = 0;

    if(mediaIterator != mediaList.end())
    {
        // xxx this should probably be more careful here
        // use the first one by default
        media = *mediaIterator;
    }
    else
    {
        media = new SdpMedia;
        media->setPort( port );
        sdpDesc.addMedia( media );
    }

    MediaAttributes* mediaAttrib;

    mediaAttrib = media->getMediaAttributes();
    if(!mediaAttrib)
    {
        mediaAttrib = new MediaAttributes();
        media->setMediaAttributes(mediaAttrib);
    }

    sipSdp.setSdpDescriptor( sdpDesc );
}


void
setAvt(SdpSession& sdpDesc, int payloadType)
{
   // declare the media list pointer
    list < SdpMedia* > mediaList;
    // get the media list pointer from the session object
    mediaList = sdpDesc.getMediaList();

    list < SdpMedia* > ::iterator mediaIterator = mediaList.begin();

    assert(mediaIterator != mediaList.end());

    SdpMedia* media = *mediaIterator;

    MediaAttributes* mediaAttrib;

    mediaAttrib = media->getMediaAttributes();

    assert(mediaAttrib != 0);

    media->addFormat(payloadType);
    SdpRtpMapAttribute* rtpMapAttribDtmf = new SdpRtpMapAttribute();
    rtpMapAttribDtmf->setPayloadType(payloadType);
    rtpMapAttribDtmf->setEncodingName("telephone-event");
    rtpMapAttribDtmf->setClockRate(8000);
    
    // add the value attribute for telephone-event
    ValueAttribute* eventAttrib = new ValueAttribute();
    eventAttrib->setAttribute( "fmtp" );

    char buf[256];

    sprintf(buf, "%d 0-15", payloadType);

    eventAttrib->setValue( buf );
    
    mediaAttrib->addmap(rtpMapAttribDtmf);
    mediaAttrib->addValueAttribute( eventAttrib );
}

void
setAvt(SipSdp& sipSdp, int payloadType)
{
    setupMedia(sipSdp,
               theSystem.gethostAddress(),
               UaDevice::instance()->getRtpPort());

    // do the media stuff
    SdpSession sdpDesc = sipSdp.getSdpDescriptor();
    setAvt(sdpDesc, payloadType);
    sipSdp.setSdpDescriptor(sdpDesc);
}


bool checkType(SipSdp& sdp, int& realPayloadType, Data eventName)
{
    list < SdpMedia* > remoteMediaList 
        = sdp.getSdpDescriptor().getMediaList();

    if (remoteMediaList.size() < 1)
    {
        cpLog(LOG_ERR, "did not find any media lists");
        return false;
    }

    if (remoteMediaList.size() > 1)
    {
        cpLog(LOG_WARNING, "Found %d media descriptors. Only using first one!",
              remoteMediaList.size());
    }

    MediaAttributes *mediaAttributes 
        = (*remoteMediaList.begin())->getMediaAttributes();
    if(mediaAttributes)
    {
        cpLog( LOG_DEBUG, "Has Media Attribute(s)" );
        vector < SdpRtpMapAttribute* > *rtpMaps = mediaAttributes->getmap();
        if(rtpMaps->size() == 0)
        {
            cpLog(LOG_ERR, "Empty RTP MAP attribute");
            return false;
        }
        vector < SdpRtpMapAttribute* > ::iterator rtpMap;
        //As suggested by Steve Frare, though this is just a hack
        //I am putting it till we investigate more and see why it
        //should throw an exception  in some cases on Windows
      try {
        rtpMap = rtpMaps->begin();
        while (rtpMap != rtpMaps->end())
        {
            Data encodingName = (*rtpMap)->getEncodingName();
            
            if (isEqualNoCase(encodingName, eventName))
            {
                realPayloadType = (*rtpMap)->getPayloadType();

                cpLog(LOG_DEBUG, 
                      "Has payload type %s, value %d",
                      encodingName.logData(),
                      realPayloadType);
                return true;
            }
            else
            {
                LocalScopeAllocator lo1;
                LocalScopeAllocator lo2;
                cpLog(LOG_DEBUG, "payload type %s != %s",
                      eventName.getData(lo1),
                      encodingName.getData(lo2));
            }
            ++rtpMap;
        }
    }
    catch(...)
    {
        cpLog( LOG_ERR, "SdpHandler::checkType exception.");
    }
    }
    else
    {
        cpLog( LOG_DEBUG, "No Media Attribute" );
        return false;
    }
    return false;
}


void
doAnswerStuff(SipSdp& local, Sptr<SipSdp> remote, Data hostAddr, int port)
{
    SdpSession sdpDesc;

    SdpMedia* media = new SdpMedia;

    media->setPort( port );   // which port to use when making a call
    // RTP doesn't have a suggested number

    sdpDesc.addMedia( media );

    //To further perform any other operation on that particular media
    //item, use loc.

    MediaAttributes* mediaAttrib = new MediaAttributes();
    assert(mediaAttrib);
    media->setMediaAttributes(mediaAttrib);

    // create the new value attribute object
    ValueAttribute* attrib = new ValueAttribute();
    // set the attribute and its value
    attrib->setAttribute("ptime");
    LocalScopeAllocator lo;
    attrib->setValue( Data( UaConfiguration::instance()->getNetworkRtpRate() ).getData(lo) );
    mediaAttrib->addValueAttribute(attrib);

    setUlaw(sdpDesc, 8000);

    if(remote != 0)
    {
        // Add RTP formats 'PCMU' and 'telephone-event' to local session
        // description, if supported by remote session description.

        int payloadType;
    
        if(checkType(*remote, payloadType, "telephone-event"))
        {
            cpLog(LOG_DEBUG, 
                  "Has DTMF NTE (RFC2833) in payload type: %d.", 
                  payloadType);
            if(payloadType == 100)
            {
                // there's ULAW in the remote
                setAvt(sdpDesc, payloadType);
            }
            else
            {
                // xxx fix this when dynamic payload types work.
                cpLog(LOG_DEBUG, "%d != 100, so will not respond", payloadType);
            }
        }
    }
    // always accept AVT tones.
    setAvt(sdpDesc, 100);

    local.setSdpDescriptor( sdpDesc );

    local.setUserName( theSystem.getUserName() );
    local.setSessionName( theSystem.getSessionName() );

    setHost(local, hostAddr);
}




void setAudioRequest( struct Vocal::HardwareAudioRequest* request,
                      SipSdp& remoteSdp,
                      SipSdp& localSdp)
{
    request->type = AudioStart;
    
    // Set remote host and port
    LocalScopeAllocator lo;
    strcpy( request->remoteHost,
            remoteSdp.getConnAddress().getData(lo) );
    request->remotePort = remoteSdp.getRtpPort();

    // set the local host and port
    request->localPort = localSdp.getRtpPort();
    strcpy( request->localHost,
            localSdp.getConnAddress().getData(lo) );
    
    // set the audio rate (should I use the remote rate?)
    request->rtpPacketSize 
        = UaConfiguration::instance()->getNetworkRtpRate();
    
    //TODO Why echo cancellation always true?
    request->echoCancellation = true;

    //TODO: this will be removed once 183 is integrated.
    request->sendRingback = false;
    
    cpLog( LOG_DEBUG, "Start audio( duplex ):" );
    cpLog( LOG_DEBUG, "localHost     = %s", request->localHost);
    cpLog( LOG_DEBUG, "localPort     = %d", request->localPort);
    cpLog( LOG_DEBUG, "remoteHost    = %s", request->remoteHost);
    cpLog( LOG_DEBUG, "remotePort    = %d", request->remotePort);
    cpLog( LOG_DEBUG, "rtpPacketSize = %d", request->rtpPacketSize);
}


void sendRemoteRingback(SipSdp& remoteSdp)
{
    LocalScopeAllocator lo;
    Sptr < UaHardwareEvent > ringbackSignal 
        = new UaHardwareEvent( UaDevice::getDeviceQueue() );
    ringbackSignal->type = HardwareAudioType;
    ringbackSignal->signalOrRequest.request.type = AudioStart;
    ringbackSignal->signalOrRequest.request.localPort = 0;
    ringbackSignal->signalOrRequest.request.remotePort 
        = remoteSdp.getRtpPort();

    strcpy(ringbackSignal->signalOrRequest.request.remoteHost, 
           remoteSdp.getConnAddress().getData(lo));

    ringbackSignal->signalOrRequest.request.echoCancellation = true;
    ringbackSignal->signalOrRequest.request.sendRingback = true;
    ringbackSignal->signalOrRequest.request.rtpPacketSize 
        = getRtpPacketSize(remoteSdp);
    if(ringbackSignal->signalOrRequest.request.rtpPacketSize <= 0)
    {
        ringbackSignal->signalOrRequest.request.rtpPacketSize 
            = UaConfiguration::instance()->getNetworkRtpRate();
    }
    UaDevice::getDeviceQueue()->add( ringbackSignal );
}


void startAudio( Sptr < SipSdp > localSdp, Sptr < SipSdp > remoteSdp )
{
    Sptr < UaHardwareEvent > signal = new UaHardwareEvent( UaDevice::getDeviceQueue() );
    signal->type = HardwareAudioType;

    struct HardwareAudioRequest* request = &(signal->signalOrRequest.request);


    if(remoteSdp->getSdpDescriptor().isHold())
    {
        cpLog( LOG_DEBUG, "Stopping audio (remote SDP == hold)");
        signal->signalOrRequest.request.type = AudioStop;
    }
    else
    {
        request->type = AudioStart;
        
        // Set remote host and port
        LocalScopeAllocator lo;
        strcpy( request->remoteHost,
                remoteSdp->getConnAddress().getData(lo) );
        request->remotePort = remoteSdp->getRtpPort();
        
        // Set local host and port
        request->localPort = localSdp->getRtpPort();
        strcpy( request->localHost,
                localSdp->getConnAddress().getData(lo) );
        
        /**
         * Echo cancellation is always true, since there is no code to
         * handle SDP echo cancellation negotiation.
         */
        request->echoCancellation = true;
        
        // Set rtpPacketSize to default in ua.cfg file
        int rtpPacketSize = getRtpPacketSize(*localSdp);
        
        if(rtpPacketSize <= 0)
        {
            rtpPacketSize = UaConfiguration::instance()->getNetworkRtpRate();
        }
        
        request->rtpPacketSize = rtpPacketSize;
        request->sendRingback = false;
        
        cpLog( LOG_DEBUG, "Start audio( duplex ):" );
        cpLog( LOG_DEBUG, "localHost     = %s", request->localHost);
        cpLog( LOG_DEBUG, "localPort     = %d", request->localPort);
        cpLog( LOG_DEBUG, "remoteHost    = %s", request->remoteHost);
        cpLog( LOG_DEBUG, "remotePort    = %d", request->remotePort);
        cpLog( LOG_DEBUG, "rtpPacketSize = %d", request->rtpPacketSize );
    }

    UaDevice::getDeviceQueue()->add( signal );
}


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