LibWebRTCMediaStackAdapter.cxx

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#if (_MSC_VER >= 1600)
#include <stdint.h>       // Use Visual Studio's stdint.h
#define _MSC_STDINT_H_    // This define will ensure that stdint.h in sipXport tree is not used
#endif
 
// resip includes
#include <rutil/Log.hxx>
#include <rutil/Logger.hxx>
#include <rutil/Lock.hxx>
#include <rutil/Random.hxx>
#include <resip/dum/DialogUsageManager.hxx>
#include <resip/dum/ClientInviteSession.hxx>
#include <resip/dum/ServerInviteSession.hxx>
#include <resip/dum/ClientSubscription.hxx>
#include <resip/dum/ServerOutOfDialogReq.hxx>
 
#include "ReconSubsystem.hxx"
#include "UserAgent.hxx"
#include "LibWebRTCMediaStackAdapter.hxx"
#include "ConversationManagerCmds.hxx"
#include "LibWebRTCConversation.hxx"
#include "Participant.hxx"
#include "LibWebRTCBridgeMixer.hxx"
#include "DtmfEvent.hxx"
#include "LibWebRTCRemoteParticipant.hxx"
#include "Conversation.hxx"
#include <rutil/WinLeakCheck.hxx>
 
#include "cajun/json/elements.h"
 
#if defined(WIN32) && !defined(__GNUC__)
#pragma warning( disable : 4355 )
#endif
 
using namespace recon;
using namespace resip;
using namespace std;
 
#define RESIPROCATE_SUBSYSTEM ReconSubsystem::RECON
 
LibWebRTCMediaStackAdapter::LibWebRTCMediaStackAdapter(ConversationManager& conversationManager)
: MediaStackAdapter(conversationManager),
  mLibWebRTCTOSValue(0) // FIXME - make this value configurable
{
   init();
}
 
LibWebRTCMediaStackAdapter::LibWebRTCMediaStackAdapter(ConversationManager& conversationManager, int defaultSampleRate, int maxSampleRate)
: MediaStackAdapter(conversationManager),
  mLibWebRTCTOSValue(0) // FIXME - make this value configurable
{
   init(defaultSampleRate, maxSampleRate);
}
 
void
LibWebRTCMediaStackAdapter::init(int defaultSampleRate, int maxSampleRate)
{
   // Connect to the LibWebRTC server
   DebugLog(<<"trying to connect to LibWebRTC host " << mLibWebRTCUri);
   mLibWebRTCConnection = mLibWebRTCManager.getLibWebRTCConnection(mLibWebRTCUri.c_str(), *this); // FIXME wait for connection
 
}
 
void
LibWebRTCMediaStackAdapter::onConnected()
{
   // FIXME
   // - if LibWebRTC connection was dropped but the same LibWebRTC instance is still
   // running then mPipeline remains valid and we don't need to replace it
   // - if it was a LibWebRTC restart then mPipeline does need to be replaced and all
   // calls need to be dropped at the SIP level so the clients reconnect
 
   mPipeline = make_shared<libwebrtc::MediaPipeline>(mLibWebRTCConnection);
   mPipeline->create([this]{
      DebugLog(<<"pipeline created with ID " << mPipeline->getId());
   });  // FIXME - wait for creation
}
 
void
LibWebRTCMediaStackAdapter::shutdown()
{
   InfoLog(<<"LibWebRTCMediaStackAdapter shutdown");
   if(mPipeline)
   {
      mPipeline->release([this](){
         InfoLog(<<"mPipeline has been released");
         mLibWebRTCConnection.reset();
      });
   }
   else
   {
      mLibWebRTCConnection.reset();
   }
   // FIXME - join / wait for any threads to stop
}
 
LibWebRTCMediaStackAdapter::~LibWebRTCMediaStackAdapter()
{
   // FIXME - move to ::shutdown() ?
   getBridgeMixer().reset();   // Make sure the mixer is destroyed before the media interface
}
 
void
LibWebRTCMediaStackAdapter::conversationManagerReady(ConversationManager* conversationManager)
{
}
 
void
LibWebRTCMediaStackAdapter::setUserAgent(UserAgent* userAgent)
{
   // FIXME for LibWebRTC
//   if (mMediaInterface)
//   {
//      // Enable/Disable DTMF digit logging according to UserAgentMasterProfile setting
//      mMediaInterface->allowLoggingDTMFDigits(userAgent->getUserAgentMasterProfile()->dtmfDigitLoggingEnabled());
//   }
 
}
 
ConversationHandle
LibWebRTCMediaStackAdapter::createSharedMediaInterfaceConversation(ConversationHandle sharedMediaInterfaceConvHandle, ConversationManager::AutoHoldMode autoHoldMode)
{
   if (isShuttingDown()) return 0;  // Don't allow new things to be created when we are shutting down
 
   ConversationHandle convHandle = getNewConversationHandle();
 
   CreateConversationCmd* cmd = new CreateConversationCmd(&getConversationManager(), convHandle, autoHoldMode, sharedMediaInterfaceConvHandle);
   post(cmd);
   return convHandle;
}
 
void
LibWebRTCMediaStackAdapter::outputBridgeMatrixImpl(ConversationHandle convHandle)
{
   // Note: convHandle of 0 only makes sense if sipXGlobalMediaInterfaceMode is enabled
   if (convHandle == 0)
   {
      if (getBridgeMixer() != 0)
      {
         getBridgeMixer()->outputBridgeMixWeights();
      }
      else
      {
         WarningLog(<< "ConversationManager::outputBridgeMatrix request with no conversation handle is not appropriate for current MediaInterfaceMode");
      }
   }
   else
   {
      Conversation* conversation = getConversation(convHandle);
      if (conversation)
      {
         if (conversation->getBridgeMixer() != 0)
         {
            conversation->getBridgeMixer()->outputBridgeMixWeights();
         }
         else
         {
            WarningLog(<< "ConversationManager::outputBridgeMatrix requested conversation wihtout a mixer/media interface, conversationHandle=" << convHandle);
         }
      }
      else
      {
         WarningLog(<< "ConversationManager::outputBridgeMatrix requested for non-existing conversationHandle=" << convHandle);
      }
   }
}
 
void
LibWebRTCMediaStackAdapter::buildSessionCapabilities(const resip::Data& ipaddress,
                               const std::vector<unsigned int>& codecIds, resip::SdpContents& sessionCaps)
{
   // FIXME LibWebRTC:
   // - check who calls this (called from testUA, reConServer, ...
   // - adapt for LibWebRTC
 
   // as a hack, we populate it with a static copy of the SDP from LibWebRTC
 
   // FIXME: ipaddress is assumed to be IP4
   Data caps("v=0\r\n"
             "o=- 3843572881 3843572881 IN IP4 " + ipaddress + "\r\n"
             "s=LibWebRTC Media Server\r\n"
             "c=IN IP4 " + ipaddress + "\r\n"
             "t=0 0\r\n"
             "m=audio 34712 RTP/AVP 0\r\n"
             "a=sendrecv\r\n"
             "a=rtcp:34713\r\n"
             "a=rtpmap:0 PCMU/8000\r\n"
             "a=ssrc:866007149 cname:user2141419334@host-99aef493\r\n"
             "m=video 2962 RTP/AVP 97 126\r\n"
             "a=sendrecv\r\n"
             "a=rtcp:2963\r\n"
             "a=rtpmap:97 H264/90000\r\n"
             "a=rtpmap:126 H264/90000\r\n"
             "a=fmtp:97 packetization-mode=0;profile-level-id=420016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000;sar-supported=16\r\n"
             "a=fmtp:126 packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000;sar-supported=16\r\n"
             "a=ssrc:1014502683 cname:user2141419334@host-99aef493\r\n");
 
   HeaderFieldValue hfv(caps.data(), caps.size());
   Mime type("application", "sdp");
   SdpContents sdp(hfv, type);
   sessionCaps = sdp;
}
 
bool
LibWebRTCMediaStackAdapter::supportsMultipleMediaInterfaces()
{
   return true;
}
 
bool
LibWebRTCMediaStackAdapter::canConversationsShareParticipants(Conversation* conversation1, Conversation* conversation2)
{
   return false;
}
 
Conversation *
LibWebRTCMediaStackAdapter::createConversationInstance(ConversationHandle handle,
      RelatedConversationSet* relatedConversationSet,  // Pass NULL to create new RelatedConversationSet
      ConversationHandle sharedMediaInterfaceConvHandle,
      ConversationManager::AutoHoldMode autoHoldMode)
{
   return new LibWebRTCConversation(handle, getConversationManager(), *this, relatedConversationSet, sharedMediaInterfaceConvHandle, autoHoldMode);
}
 
LocalParticipant *
LibWebRTCMediaStackAdapter::createLocalParticipantInstance(ParticipantHandle partHandle)
{
   return 0;
}
 
MediaResourceParticipant *
LibWebRTCMediaStackAdapter::createMediaResourceParticipantInstance(ParticipantHandle partHandle, const resip::Uri& mediaUrl, const std::shared_ptr<resip::Data>& audioBuffer, void* recordingCircularBuffer)
{
   return 0; // FIXME LibWebRTC - implement MediaResourceParticipant in LibWebRTC
}
 
void
LibWebRTCMediaStackAdapter::configureRemoteParticipantInstance(LibWebRTCRemoteParticipant* krp)
{
   std::shared_ptr<ConfigParse> cfg = getConfig();
   if(cfg)
   {
      krp->mRemoveExtraMediaDescriptors = cfg->getConfigBool("LibWebRTCRemoveExtraMediaDescriptors", false);
      krp->mSipRtpEndpoint = cfg->getConfigBool("LibWebRTCSipRtpEndpoint", true);
      krp->mReuseSdpAnswer = cfg->getConfigBool("LibWebRTCReuseSdpAnswer", false);
      krp->mWSAcceptsKeyframeRequests = cfg->getConfigBool("LibWebRTCWebSocketAcceptsKeyframeRequests", true);
   }
}
 
RemoteParticipant *
LibWebRTCMediaStackAdapter::createRemoteParticipantInstance(DialogUsageManager& dum, RemoteParticipantDialogSet& rpds)
{
   LibWebRTCRemoteParticipant *rp = new LibWebRTCRemoteParticipant(getConversationManager(), *this, dum, rpds);
   configureRemoteParticipantInstance(rp);
   return rp;
}
 
RemoteParticipant *
LibWebRTCMediaStackAdapter::createRemoteParticipantInstance(ParticipantHandle partHandle, DialogUsageManager& dum, RemoteParticipantDialogSet& rpds)
{
   LibWebRTCRemoteParticipant *rp = new LibWebRTCRemoteParticipant(partHandle, getConversationManager(), *this, dum, rpds);
   configureRemoteParticipantInstance(rp);
   return rp;
}
 
RemoteParticipantDialogSet *
LibWebRTCMediaStackAdapter::createRemoteParticipantDialogSetInstance(
      ConversationManager::ParticipantForkSelectMode forkSelectMode,
      std::shared_ptr<ConversationProfile> conversationProfile)
{
   return new LibWebRTCRemoteParticipantDialogSet(getConversationManager(), *this, forkSelectMode, conversationProfile);
}
 
void
LibWebRTCMediaStackAdapter::process()
{
   mLibWebRTCManager.process();
}
 
void
LibWebRTCMediaStackAdapter::initializeDtlsFactory(const resip::Data& defaultAoR)
{
   // FIXME LibWebRTC
}
 
 
/* ====================================================================
 
 Copyright (c) 2022, Software Freedom Institute https://softwarefreedom.institute
 Copyright (c) 2022, Daniel Pocock https://danielpocock.com
 Copyright (c) 2021-2022, SIP Spectrum, Inc. www.sipspectrum.com
 Copyright (c) 2007-2008, Plantronics, Inc.
 All rights reserved.
 
 Redistribution and use in source and binary forms, with or without
 modification, are permitted provided that the following conditions are 
 met:
 
 1. Redistributions of source code must retain the above copyright 
    notice, this list of conditions and the following disclaimer. 
 
 2. Redistributions in binary form must reproduce the above copyright
    notice, this list of conditions and the following disclaimer in the
    documentation and/or other materials provided with the distribution. 
 
 3. Neither the name of Plantronics nor the names of its contributors 
    may be used to endorse or promote products derived from this 
    software without specific prior written permission. 
 
 THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS 
 "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT 
 LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR 
 A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT 
 OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, 
 SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT 
 LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, 
 DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY 
 THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT 
 (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE 
 OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 
 ==================================================================== */