diff --git a/daemon/README b/daemon/README
index df9de2a4d7e8be0e353f42fd9a5e7248e40807ba..46befaa292c224b46f2ee2ff3a7d7f4a3e96c1f4 100644
--- a/daemon/README
+++ b/daemon/README
@@ -175,9 +175,6 @@ interaction between daemon and client will not work; for each platform where
dbus is not available the client should implement all the methods in the
*_stub.cpp files.
-Note that you need a recent version of ucommon (>= 6.0.0) and of ccrtp
-compiled against it to use clang.
-
SIP/IAx2 accounts
---------------------
diff --git a/daemon/src/client/callmanager.cpp b/daemon/src/client/callmanager.cpp
index 42450ed2bf9252ff4409e36e00d1cd8adf85d918..8fad82a7c4effcb25bf9a0c3532a92d238499f1c 100644
--- a/daemon/src/client/callmanager.cpp
+++ b/daemon/src/client/callmanager.cpp
@@ -280,28 +280,6 @@ CallManager::startTone(int32_t start, int32_t type)
Manager::instance().stopTone();
}
-// TODO: this will have to be adapted
-// for conferencing in order to get
-// the right pointer for the given
-// callID.
-#if USE_CCRTP && HAVE_ZRTP
-ring::AudioZrtpSession *
-CallManager::getAudioZrtpSession(const std::string& callID)
-{
- // TODO: remove SIP dependency
- const auto call = Manager::instance().callFactory.getCall<SIPCall>(callID);
- if (!call)
- throw CallManagerException("Call id " + callID + " is not valid");
-
- ring::AudioZrtpSession * zSession = call->getAudioRtp().getAudioZrtpSession();
-
- if (!zSession)
- throw CallManagerException("Failed to get AudioZrtpSession");
-
- return zSession;
-}
-#endif
-
void
CallManager::setSASVerified(const std::string& callID)
{
diff --git a/daemon/src/client/callmanager.h b/daemon/src/client/callmanager.h
index 004321a035611e4a29a69c17b555fce53843b7c4..c8ebec64ca21d95a7bd84b10667d5d084ff21d88 100644
--- a/daemon/src/client/callmanager.h
+++ b/daemon/src/client/callmanager.h
@@ -156,12 +156,6 @@ class CallManager
void onRtcpReportReceived(const std::string& callID, const std::map<std::string, int>& stats);
private:
-
-#if USE_CCRTP && HAVE_ZRTP
- ring::AudioZrtpSession * getAudioZrtpSession(const std::string& callID);
-#endif
-
- private:
#pragma GCC diagnostic ignored "-Wmissing-field-initializers"
// Event handlers; needed by the library API
ring_call_ev_handlers evHandlers_{};
diff --git a/daemon/src/media/audio/audiolayer.h b/daemon/src/media/audio/audiolayer.h
index eff45116c5d6edc947f0ea2c4f2db261f85bf76f..98afab5695e5256801cf4792b88e1c296a34b1f6 100644
--- a/daemon/src/media/audio/audiolayer.h
+++ b/daemon/src/media/audio/audiolayer.h
@@ -60,10 +60,6 @@
class AudioPreference;
-namespace ost {
-class Time;
-}
-
namespace ring {
class Resampler;
diff --git a/daemon/src/ringdht/ringaccount.cpp b/daemon/src/ringdht/ringaccount.cpp
index fc001770febf62df724ed99d7f45157ddec91b22..338aaccb7a0dc3a0f19889069767377f75522843 100644
--- a/daemon/src/ringdht/ringaccount.cpp
+++ b/daemon/src/ringdht/ringaccount.cpp
@@ -239,21 +239,6 @@ RingAccount::createOutgoingCall(const std::shared_ptr<SIPCall>& call, const std:
std::vector<ring::AudioCodec *> audioCodecs;
audioCodecs.push_back(ac);
-#if USE_CCRTP
- try {
- call->getAudioRtp().initConfig();
- call->getAudioRtp().initSession();
-
- if (isStunEnabled())
- call->updateSDPFromSTUN();
-
- call->getAudioRtp().initLocalCryptoInfo();
- call->getAudioRtp().start(audioCodecs);
- } catch (...) {
- throw VoipLinkException("Could not start rtp session for early media");
- }
-#endif
-
// Building the local SDP offer
auto& sdp = call->getSDP();
diff --git a/daemon/src/sip/sipaccount.cpp b/daemon/src/sip/sipaccount.cpp
index b1b8ff4c092a6ce23b133d0d19bdb68de44d86f1..e51b160f69213595e5260dbbc4001f0f42ea7be2 100644
--- a/daemon/src/sip/sipaccount.cpp
+++ b/daemon/src/sip/sipaccount.cpp
@@ -235,21 +235,6 @@ SIPAccount::newOutgoingCall(const std::string& id, const std::string& toUrl)
std::vector<ring::AudioCodec *> audioCodecs;
audioCodecs.push_back(ac);
-#if USE_CCRTP
- try {
- call->getAudioRtp().initConfig();
- call->getAudioRtp().initSession();
-
- if (isStunEnabled())
- call->updateSDPFromSTUN();
-
- call->getAudioRtp().initLocalCryptoInfo();
- call->getAudioRtp().start(audioCodecs);
- } catch (...) {
- throw VoipLinkException("Could not start rtp session for early media");
- }
-#endif
-
// Building the local SDP offer
auto& sdp = call->getSDP();
diff --git a/daemon/src/sip/sipcall.cpp b/daemon/src/sip/sipcall.cpp
index 33dc2c3ab78570f9778928f0f601ec0e68bcc37e..31575639a060abda2d542260573e148a99bd400e 100644
--- a/daemon/src/sip/sipcall.cpp
+++ b/daemon/src/sip/sipcall.cpp
@@ -91,9 +91,7 @@ static void
dtmfSend(SIPCall &call, char code, const std::string &dtmf)
{
if (dtmf == SIPAccount::OVERRTP_STR) {
-#if USE_CCRTP
- call.getAudioRtp().sendDtmfDigit(code);
-#endif
+ RING_WARN("DTMF over RTP not supported yet");
return;
} else if (dtmf != SIPAccount::SIPINFO_STR) {
RING_WARN("Unknown DTMF type %s, defaulting to %s instead",
@@ -246,24 +244,7 @@ SIPCall::sendSIPInfo(const char *const body, const char *const subtype)
void
SIPCall::updateSDPFromSTUN()
{
-#if USE_CCRTP
- auto& account = getSIPAccount();
- std::vector<long> socketDescriptors(getAudioRtp().getSocketDescriptors());
-
- try {
- std::vector<pj_sockaddr> stunPorts(getSIPVoIPLink()->sipTransport->getSTUNAddresses(account.getStunServerName(), account.getStunPort(), socketDescriptors));
-
- // FIXME: get video sockets
- //stunPorts.resize(4);
-
- account.setPublishedAddress(stunPorts[0]);
- // published IP MUST be updated first, since RTCP depends on it
- sdp_->setPublishedIP(account.getPublishedAddress());
- sdp_->updatePorts(stunPorts);
- } catch (const std::runtime_error &e) {
- RING_ERR("%s", e.what());
- }
-#endif
+ RING_WARN("SIPCall::updateSDPFromSTUN() not implemented", __func__);
}
void SIPCall::answer()
@@ -598,13 +579,6 @@ SIPCall::offhold()
RING_ERR("%s", e.what());
throw VoipLinkException("SDP issue in offhold");
}
-#if USE_CCRTP
- catch (const ost::Socket *) {
- throw VoipLinkException("Socket problem in offhold");
- } catch (const ring::AudioRtpFactoryException &) {
- throw VoipLinkException("Socket problem in offhold");
- }
-#endif
}
void
@@ -642,18 +616,6 @@ SIPCall::internalOffHold(const std::function<void()> &SDPUpdateFunc)
throw std::runtime_error("Could not instantiate any codecs");
}
-#if USE_CCRTP
- audiortp_.initConfig();
- audiortp_.initSession();
-
- // Invoke closure
- SDPUpdateFunc();
-
- audiortp_.restoreLocalContext();
- audiortp_.initLocalCryptoInfoOnOffHold();
- audiortp_.start(audioCodecs);
-#endif
-
sdp_->removeAttributeFromLocalAudioMedia("sendrecv");
sdp_->removeAttributeFromLocalAudioMedia("sendonly");
sdp_->addAttributeToLocalAudioMedia("sendrecv");
@@ -842,52 +804,6 @@ SIPCall::startAllMedia()
CryptoOffer crypto_offer;
getSDP().getRemoteSdpCryptoFromOffer(sdp_->getActiveRemoteSdpSession(), crypto_offer);
-#if USE_CCRTP && HAVE_SDES
- bool nego_success = false;
-
- if (!crypto_offer.empty()) {
- std::vector<ring::CryptoSuiteDefinition> localCapabilities;
-
- for (size_t i = 0; i < RING_ARRAYSIZE(ring::CryptoSuites); ++i)
- localCapabilities.push_back(ring::CryptoSuites[i]);
-
- ring::SdesNegotiator sdesnego(localCapabilities, crypto_offer);
- auto callMgr = Manager::instance().getClient()->getCallManager();
-
- if (sdesnego.negotiate()) {
- nego_success = true;
-
- try {
- audiortp_.setRemoteCryptoInfo(sdesnego);
- callMgr->secureSdesOn(getCallId());
- } catch (const AudioRtpFactoryException &e) {
- RING_ERR("%s", e.what());
- callMgr->secureSdesOff(getCallId());
- }
- } else {
- RING_ERR("SDES negotiation failure");
- callMgr->secureSdesOff(getCallId());
- }
- } else {
- RING_DBG("No crypto offer available");
- }
-
- // We did not find any crypto context for this media, RTP fallback
- if (!nego_success && audiortp_.isSdesEnabled()) {
- RING_ERR("Negotiation failed but SRTP is enabled, fallback on RTP");
- audiortp_.stop();
- audiortp_.setSrtpEnabled(false);
-
- const auto& account = getSIPAccount();
- if (account.getSrtpFallback()) {
- audiortp_.initSession();
-
- if (account.isStunEnabled())
- updateSDPFromSTUN();
- }
- }
-#endif // USE_CCRTP && HAVE_SDES
-
std::vector<ring::AudioCodec*> sessionMedia(sdp_->getSessionAudioMedia());
if (sessionMedia.empty()) {
@@ -914,11 +830,6 @@ SIPCall::startAllMedia()
audioCodecs.push_back(ac);
}
}
-
-#if USE_CCRTP
- if (not audioCodecs.empty())
- getAudioRtp().updateSessionMedia(audioCodecs);
-#endif
} catch (const SdpException &e) {
RING_ERR("%s", e.what());
} catch (const std::exception &rtpException) {
diff --git a/daemon/src/sip/sipcall.h b/daemon/src/sip/sipcall.h
index fba24c03c33f7e18db58d426e89fc073f2a79c9a..e8dfbecc9b8eb384ec3250cd7e6b49cc19637a26 100644
--- a/daemon/src/sip/sipcall.h
+++ b/daemon/src/sip/sipcall.h
@@ -39,9 +39,7 @@
#endif
#include "call.h"
-#if USE_CCRTP
-#include "audio/audiortp/audio_rtp_factory.h"
-#endif
+
#ifdef RING_VIDEO
#include "video/video_rtp_session.h"
#endif
@@ -94,22 +92,12 @@ class SIPCall : public Call
return *sdp_;
}
-#if USE_CCRTP
- /**
- * Returns a pointer to the AudioRtp object
- */
- ring::AudioRtpFactory& getAudioRtp() {
- return audiortp_;
- }
-#else
-
/**
* Returns a pointer to the AVFormatRtpSession object
*/
ring::AVFormatRtpSession& getAVFormatRTP() const {
return *avformatrtp_;
}
-#endif
#ifdef RING_VIDEO
/**
@@ -211,14 +199,7 @@ class SIPCall : public Call
std::vector<ring::IceCandidate> getAllRemoteCandidates();
-#if USE_CCRTP
- /**
- * Audio Rtp Session factory
- */
- ring::AudioRtpFactory audiortp_;
-#else
std::unique_ptr<ring::AVFormatRtpSession> avformatrtp_;
-#endif
#ifdef RING_VIDEO
/**
diff --git a/daemon/src/sip/sipvoiplink.cpp b/daemon/src/sip/sipvoiplink.cpp
index 5d240ccaf39c21b4d02ae3b9f7bcf25fbbcb88f6..6e65b611f46963f3402547c9e96f8d72b541a329 100644
--- a/daemon/src/sip/sipvoiplink.cpp
+++ b/daemon/src/sip/sipvoiplink.cpp
@@ -326,53 +326,10 @@ transaction_request_cb(pjsip_rx_data *rdata)
call->initRecFilename(peerNumber);
call->setCallMediaLocal(addrToUse);
call->getSDP().setPublishedIP(addrSdp);
-#if USE_CCRTP
- call->getAudioRtp().initConfig();
- try {
- call->getAudioRtp().initSession();
- } catch (const ost::Socket::Error &err) {
- RING_ERR("AudioRtp socket error");
- return PJ_FALSE;
- }
-#endif
if (account->isStunEnabled())
call->updateSDPFromSTUN();
-#if USE_CCRTP
- if (body and body->len > 0 and call->getAudioRtp().isSdesEnabled()) {
- std::string sdpOffer(static_cast<const char*>(body->data), body->len);
- size_t start = sdpOffer.find("a=crypto:");
-
- // Found crypto header in SDP
- if (start != std::string::npos) {
- CryptoOffer crypto_offer;
- crypto_offer.push_back(std::string(sdpOffer.substr(start, (sdpOffer.size() - start) - 1)));
-
- const size_t size = RING_ARRAYSIZE(ring::CryptoSuites);
- std::vector<ring::CryptoSuiteDefinition> localCapabilities(size);
-
- std::copy(ring::CryptoSuites, ring::CryptoSuites + size,
- localCapabilities.begin());
-
-#if HAVE_SDES
- ring::SdesNegotiator sdesnego(localCapabilities, crypto_offer);
-
- if (sdesnego.negotiate()) {
- try {
- call->getAudioRtp().setRemoteCryptoInfo(sdesnego);
- call->getAudioRtp().initLocalCryptoInfo();
- } catch (const AudioRtpFactoryException &e) {
- RING_ERR("%s", e.what());
- return PJ_FALSE;
- }
- }
-
-#endif
- }
- }
-#endif
-
call->getSDP().receiveOffer(r_sdp, account->getActiveAudioCodecs(), account->getActiveVideoCodecs());
if (not call->getIceTransport()) {
RING_DBG("Initializing ICE transport");
@@ -390,9 +347,6 @@ transaction_request_cb(pjsip_rx_data *rdata)
std::vector<ring::AudioCodec *> audioCodecs;
audioCodecs.push_back(ac);
-#if USE_CCRTP
- call->getAudioRtp().start(audioCodecs);
-#endif
pjsip_dialog *dialog = 0;