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collectioneditor.cpp
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Edric Milaret authored
- Happy new year - lower f fix Change-Id: I9b6d2ef340f4a33185dae75aef15538209749788 Tuleap: #218
Edric Milaret authored- Happy new year - lower f fix Change-Id: I9b6d2ef340f4a33185dae75aef15538209749788 Tuleap: #218
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sipcall.cpp 100.93 KiB
/*
* Copyright (C) 2004-2021 Savoir-faire Linux Inc.
*
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
* Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
* Author: Guillaume Roguez <guillaume.roguez@savoirfairelinux.com>
* Author: Philippe Gorley <philippe.gorley@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "call_factory.h"
#include "sipcall.h"
#include "sipaccount.h"
#include "sipaccountbase.h"
#include "sipvoiplink.h"
#include "logger.h"
#include "sdp.h"
#include "manager.h"
#include "string_utils.h"
#include "upnp/upnp_control.h"
#include "sip_utils.h"
#include "audio/audio_rtp_session.h"
#include "system_codec_container.h"
#include "im/instant_messaging.h"
#include "jami/call_const.h"
#include "jami/media_const.h"
#include "client/ring_signal.h"
#include "ice_transport.h"
#include "pjsip-ua/sip_inv.h"
#ifdef ENABLE_PLUGIN
#include "plugin/jamipluginmanager.h"
#endif
#ifdef ENABLE_VIDEO
#include "client/videomanager.h"
#include "video/video_rtp_session.h"
#include "jami/videomanager_interface.h"
#include <chrono>
#include <libavutil/display.h>
#include <video/sinkclient.h>
#endif
#include "audio/ringbufferpool.h"
#include "jamidht/channeled_transport.h"
#include "errno.h"
#include <opendht/crypto.h>
#include <opendht/thread_pool.h>
#include <fmt/ranges.h>
namespace jami {
using sip_utils::CONST_PJ_STR;
using namespace DRing::Call;
#ifdef ENABLE_VIDEO
static DeviceParams
getVideoSettings()
{
const auto& videomon = jami::getVideoDeviceMonitor();
return videomon.getDeviceParams(videomon.getDefaultDevice());
}
#endif
static constexpr std::chrono::seconds DEFAULT_ICE_INIT_TIMEOUT {35}; // seconds
static constexpr std::chrono::milliseconds EXPECTED_ICE_INIT_MAX_TIME {5000};
static constexpr std::chrono::seconds DEFAULT_ICE_NEGO_TIMEOUT {60}; // seconds
static constexpr std::chrono::milliseconds MS_BETWEEN_2_KEYFRAME_REQUEST {1000};
static constexpr int ICE_COMP_ID_RTP {1};
static constexpr int ICE_COMP_COUNT_PER_STREAM {2};
static constexpr auto MULTISTREAM_REQUIRED_VERSION_STR = "10.0.2"sv;
static const std::vector<unsigned> MULTISTREAM_REQUIRED_VERSION
= split_string_to_unsigned(MULTISTREAM_REQUIRED_VERSION_STR, '.');
SIPCall::SIPCall(const std::shared_ptr<SIPAccountBase>& account,
const std::string& callId,
Call::CallType type,
const std::map<std::string, std::string>& details)
: Call(account, callId, type, details)
, sdp_(new Sdp(callId))
, enableIce_(account->isIceForMediaEnabled())
, srtpEnabled_(account->isSrtpEnabled())
{
if (account->getUPnPActive())
upnp_.reset(new upnp::Controller());
setCallMediaLocal();
// Set the media caps.
sdp_->setLocalMediaCapabilities(MediaType::MEDIA_AUDIO,
account->getActiveAccountCodecInfoList(MEDIA_AUDIO));
#ifdef ENABLE_VIDEO
sdp_->setLocalMediaCapabilities(MediaType::MEDIA_VIDEO,
account->getActiveAccountCodecInfoList(MEDIA_VIDEO));
#endif
auto mediaAttrList = getSIPAccount()->createDefaultMediaList(getSIPAccount()->isVideoEnabled()
and not isAudioOnly(),
getState() == CallState::HOLD);
JAMI_DBG("[call:%s] Create a new [%s] SIP call with %lu media",
getCallId().c_str(),
type == Call::CallType::INCOMING
? "INCOMING"
: (type == Call::CallType::OUTGOING ? "OUTGOING" : "MISSED"),
mediaAttrList.size());
initMediaStreams(mediaAttrList);
}
SIPCall::SIPCall(const std::shared_ptr<SIPAccountBase>& account,
const std::string& callId,
Call::CallType type,
const std::vector<DRing::MediaMap>& mediaList)
: Call(account, callId, type)
, peerSupportMultiStream_(false)
, sdp_(new Sdp(callId))
, enableIce_(account->isIceForMediaEnabled())
, srtpEnabled_(account->isSrtpEnabled())
{
if (account->getUPnPActive())
upnp_.reset(new upnp::Controller());
setCallMediaLocal();
// Set the media caps. sdp_->setLocalMediaCapabilities(MediaType::MEDIA_AUDIO,
account->getActiveAccountCodecInfoList(MEDIA_AUDIO));
#ifdef ENABLE_VIDEO
sdp_->setLocalMediaCapabilities(MediaType::MEDIA_VIDEO,
account->getActiveAccountCodecInfoList(MEDIA_VIDEO));
#endif
auto mediaAttrList = MediaAttribute::buildMediaAttributesList(mediaList, isSrtpEnabled());
if (mediaAttrList.size() == 0) {
if (type_ == Call::CallType::INCOMING) {
// Handle incoming call without media offer.
JAMI_WARN(
"[call:%s] No media offered in the incoming invite. An offer will be provided in "
"the answer",
getCallId().c_str());
mediaAttrList = getSIPAccount()->createDefaultMediaList(false,
getState() == CallState::HOLD);
} else {
JAMI_WARN("[call:%s] Creating an outgoing call with empty offer", getCallId().c_str());
}
}
JAMI_DBG("[call:%s] Create a new [%s] SIP call with %lu media",
getCallId().c_str(),
type == Call::CallType::INCOMING
? "INCOMING"
: (type == Call::CallType::OUTGOING ? "OUTGOING" : "MISSED"),
mediaList.size());
initMediaStreams(mediaAttrList);
}
SIPCall::~SIPCall()
{
std::lock_guard<std::recursive_mutex> lk {callMutex_};
setTransport({});
setInviteSession(); // prevents callback usage
}
size_t
SIPCall::findRtpStreamIndex(const std::string& label) const
{
const auto iter = std::find_if(rtpStreams_.begin(),
rtpStreams_.end(),
[&label](const RtpStream& rtp) {
return label == rtp.mediaAttribute_->label_;
});
// Return the index if there is a match.
if (iter != rtpStreams_.end())
return std::distance(rtpStreams_.begin(), iter);
// No match found.
return rtpStreams_.size();
}
void
SIPCall::createRtpSession(RtpStream& stream)
{
if (not stream.mediaAttribute_)
throw std::runtime_error("Missing media attribute");
if (stream.mediaAttribute_->type_ == MediaType::MEDIA_AUDIO) {
stream.rtpSession_ = std::make_shared<AudioRtpSession>(id_);
}
#ifdef ENABLE_VIDEO
else if (stream.mediaAttribute_->type_ == MediaType::MEDIA_VIDEO) {
stream.rtpSession_ = std::make_shared<video::VideoRtpSession>(id_, getVideoSettings()); }
#endif
else {
throw std::runtime_error("Unsupported media type");
}
// Must be valid at this point.
if (not stream.rtpSession_)
throw std::runtime_error("Failed to create RTP Session");
;
}
void
SIPCall::configureRtpSession(const std::shared_ptr<RtpSession>& rtpSession,
const std::shared_ptr<MediaAttribute>& mediaAttr,
const MediaDescription& localMedia,
const MediaDescription& remoteMedia)
{
JAMI_DBG("[call:%s] Configuring [%s] rtp session",
getCallId().c_str(),
MediaAttribute::mediaTypeToString(mediaAttr->type_));
if (not rtpSession)
throw std::runtime_error("Must have a valid RTP Session");
// Configure the media stream
auto new_mtu = transport_->getTlsMtu();
rtpSession->setMtu(new_mtu);
rtpSession->updateMedia(remoteMedia, localMedia);
// Mute/un-mute media
if (mediaAttr->muted_) {
rtpSession->setMuted(true);
// TODO. Setting mute to true should be enough to mute.
// Kept for backward compatiblity.
rtpSession->setMediaSource("");
} else {
rtpSession->setMuted(false);
rtpSession->setMediaSource(mediaAttr->sourceUri_);
}
rtpSession->setSuccessfulSetupCb([w = weak()](MediaType type, bool isRemote) {
if (auto thisPtr = w.lock())
thisPtr->rtpSetupSuccess(type, isRemote);
});
#ifdef ENABLE_VIDEO
if (localMedia.type == MediaType::MEDIA_VIDEO) {
auto videoRtp = std::dynamic_pointer_cast<video::VideoRtpSession>(rtpSession);
assert(videoRtp);
videoRtp->setRequestKeyFrameCallback([w = weak()] {
runOnMainThread([w] {
if (auto thisPtr = w.lock())
thisPtr->requestKeyframe();
});
});
videoRtp->setChangeOrientationCallback([w = weak()](int angle) {
runOnMainThread([w, angle] {
if (auto thisPtr = w.lock())
thisPtr->setVideoOrientation(angle);
});
});
}
#endif
}
std::shared_ptr<SIPAccountBase>
SIPCall::getSIPAccount() const
{
return std::static_pointer_cast<SIPAccountBase>(getAccount().lock());}
#ifdef ENABLE_PLUGIN
void
SIPCall::createCallAVStreams()
{
if (hasVideo()) {
auto videoRtp = getVideoRtp();
if (not videoRtp)
return;
if (videoRtp->hasConference()) {
clearCallAVStreams();
return;
}
}
auto baseId = getCallId();
/**
* Map: maps the AudioFrame to an AVFrame
**/
auto audioMap = [](const std::shared_ptr<jami::MediaFrame>& m) -> AVFrame* {
return std::static_pointer_cast<AudioFrame>(m)->pointer();
};
auto const& audioRtp = getAudioRtp();
if (not audioRtp) {
throw std::runtime_error("Must have a valid Audio RTP Session");
}
// Preview
if (auto& localAudio = audioRtp->getAudioLocal()) {
auto previewSubject = std::make_shared<MediaStreamSubject>(audioMap);
StreamData microStreamData {baseId, false, StreamType::audio, getPeerNumber()};
createCallAVStream(microStreamData, *localAudio, previewSubject);
}
// Receive
if (auto& audioReceive = audioRtp->getAudioReceive()) {
auto receiveSubject = std::make_shared<MediaStreamSubject>(audioMap);
StreamData phoneStreamData {baseId, true, StreamType::audio, getPeerNumber()};
createCallAVStream(phoneStreamData, (AVMediaStream&) *audioReceive, receiveSubject);
}
#ifdef ENABLE_VIDEO
if (hasVideo()) {
auto videoRtp = getVideoRtp();
if (not videoRtp)
return;
// Map: maps the VideoFrame to an AVFrame
auto map = [](const std::shared_ptr<jami::MediaFrame> m) -> AVFrame* {
return std::static_pointer_cast<VideoFrame>(m)->pointer();
};
// Preview
if (auto& videoPreview = videoRtp->getVideoLocal()) {
auto previewSubject = std::make_shared<MediaStreamSubject>(map);
StreamData previewStreamData {getCallId(), false, StreamType::video, getPeerNumber()};
createCallAVStream(previewStreamData, *videoPreview, previewSubject);
}
// Receive
if (auto& videoReceive = videoRtp->getVideoReceive()) {
auto receiveSubject = std::make_shared<MediaStreamSubject>(map);
StreamData receiveStreamData {getCallId(), true, StreamType::video, getPeerNumber()};
createCallAVStream(receiveStreamData, *videoReceive, receiveSubject);
}
}
#endif
}
voidSIPCall::createCallAVStream(const StreamData& StreamData,
AVMediaStream& streamSource,
const std::shared_ptr<MediaStreamSubject>& mediaStreamSubject)
{
const std::string AVStreamId = StreamData.id + std::to_string(static_cast<int>(StreamData.type))
+ std::to_string(StreamData.direction);
std::lock_guard<std::mutex> lk(avStreamsMtx_);
auto it = callAVStreams.find(AVStreamId);
if (it != callAVStreams.end())
return;
it = callAVStreams.insert(it, {AVStreamId, mediaStreamSubject});
streamSource.attachPriorityObserver(it->second);
jami::Manager::instance()
.getJamiPluginManager()
.getCallServicesManager()
.createAVSubject(StreamData, it->second);
}
void
SIPCall::clearCallAVStreams()
{
std::lock_guard<std::mutex> lk(avStreamsMtx_);
callAVStreams.clear();
}
#endif // ENABLE_PLUGIN
void
SIPCall::setCallMediaLocal()
{
if (localAudioPort_ == 0
#ifdef ENABLE_VIDEO
|| localVideoPort_ == 0
#endif
)
generateMediaPorts();
}
void
SIPCall::generateMediaPorts()
{
auto account = getSIPAccount();
if (!account) {
JAMI_ERR("No account detected");
return;
}
// TODO. Setting specfic range for RTP ports is obsolete, in
// particular in the context of ICE.
// Reference: http://www.cs.columbia.edu/~hgs/rtp/faq.html#ports
// We only want to set ports to new values if they haven't been set
const unsigned callLocalAudioPort = account->generateAudioPort();
if (localAudioPort_ != 0)
account->releasePort(localAudioPort_);
localAudioPort_ = callLocalAudioPort;
sdp_->setLocalPublishedAudioPorts(callLocalAudioPort,
rtcpMuxEnabled_ ? 0 : callLocalAudioPort + 1);
#ifdef ENABLE_VIDEO
// https://projects.savoirfairelinux.com/issues/17498
const unsigned int callLocalVideoPort = account->generateVideoPort();
if (localVideoPort_ != 0)
account->releasePort(localVideoPort_);
// this should already be guaranteed by SIPAccount
assert(localAudioPort_ != callLocalVideoPort);
localVideoPort_ = callLocalVideoPort;
sdp_->setLocalPublishedVideoPorts(callLocalVideoPort,
rtcpMuxEnabled_ ? 0 : callLocalVideoPort + 1);
#endif
}
void
SIPCall::setContactHeader(pj_str_t contact)
{
pj_strcpy(&contactHeader_, &contact);
}
void
SIPCall::setTransport(const std::shared_ptr<SipTransport>& t)
{
if (t != transport_) {
JAMI_WARN("[call:%s] Setting tranport to [%p]", getCallId().c_str(), t.get());
}
transport_ = t;
if (not t) {
return;
}
if (isSrtpEnabled() and not transport_->isSecure()) {
JAMI_WARN("[call:%s] Crypto (SRTP) is negotiated over an un-encrypted signaling channel",
getCallId().c_str());
}
if (not isSrtpEnabled() and transport_->isSecure()) {
JAMI_WARN("[call:%s] The signaling channel is encrypted but the media is not encrypted",
getCallId().c_str());
}
const auto list_id = reinterpret_cast<uintptr_t>(this);
transport_->removeStateListener(list_id);
// listen for transport destruction
transport_->addStateListener(
list_id, [wthis_ = weak()](pjsip_transport_state state, const pjsip_transport_state_info*) {
if (auto this_ = wthis_.lock()) {
JAMI_DBG("[call:%s] SIP transport state [%i] - connection state [%u]",
this_->getCallId().c_str(),
state,
static_cast<unsigned>(this_->getConnectionState()));
// End the call if the SIP transport was shut down
auto isAlive = SipTransport::isAlive(state);
if (not isAlive and this_->getConnectionState() != ConnectionState::DISCONNECTED) {
JAMI_WARN("[call:%s] Ending call because underlying SIP transport was closed",
this_->getCallId().c_str());
this_->stopAllMedia();
this_->onFailure(ECONNRESET);
}
}
});
}
void
SIPCall::requestReinvite()
{
JAMI_DBG("[call:%s] Sending a SIP re-invite to request media change", getCallId().c_str());
if (isWaitingForIceAndMedia_) {
remainingRequest_ = Request::SwitchInput;
} else {
auto mediaList = getMediaAttributeList();
assert(not mediaList.empty());
// TODO. We should erase existing streams only after the new
// ones were successfully negotiated, and make a live switch. But
// for now, we reset all streams before creating new ones.
rtpStreams_.clear();
initMediaStreams(mediaList);
if (SIPSessionReinvite(mediaList) == PJ_SUCCESS) {
isWaitingForIceAndMedia_ = true;
}
}
}
/**
* Send a reINVITE inside an active dialog to modify its state
* Local SDP session should be modified before calling this method
*/
int
SIPCall::SIPSessionReinvite(const std::vector<MediaAttribute>& mediaAttrList)
{
assert(not mediaAttrList.empty());
std::lock_guard<std::recursive_mutex> lk {callMutex_};
// Do nothing if no invitation processed yet
if (not inviteSession_ or inviteSession_->invite_tsx)
return PJ_SUCCESS;
JAMI_DBG("[call:%s] Preparing and sending a re-invite (state=%s)",
getCallId().c_str(),
pjsip_inv_state_name(inviteSession_->state));
// Generate new ports to receive the new media stream
// LibAV doesn't discriminate SSRCs and will be confused about Seq changes on a given port
generateMediaPorts();
sdp_->clearIce();
auto acc = getSIPAccount();
if (not acc) {
JAMI_ERR("No account detected");
return !PJ_SUCCESS;
}
if (not sdp_->createOffer(mediaAttrList))
return !PJ_SUCCESS;
if (isIceEnabled()) {
createIceMediaTransport();
if (initIceMediaTransport(true))
addLocalIceAttributes();
}
pjsip_tx_data* tdata;
auto local_sdp = sdp_->getLocalSdpSession();
auto result = pjsip_inv_reinvite(inviteSession_.get(), nullptr, local_sdp, &tdata);
if (result == PJ_SUCCESS) {
if (!tdata)
return PJ_SUCCESS;
// Add user-agent header
sip_utils::addUserAgentHeader(acc->getUserAgentName(), tdata);
result = pjsip_inv_send_msg(inviteSession_.get(), tdata);
if (result == PJ_SUCCESS)
return PJ_SUCCESS;
JAMI_ERR("[call:%s] Failed to send REINVITE msg (pjsip: %s)",
getCallId().c_str(),
sip_utils::sip_strerror(result).c_str());
// Canceling internals without sending (anyways the send has just failed!)
pjsip_inv_cancel_reinvite(inviteSession_.get(), &tdata);
} else
JAMI_ERR("[call:%s] Failed to create REINVITE msg (pjsip: %s)",
getCallId().c_str(),
sip_utils::sip_strerror(result).c_str());
return !PJ_SUCCESS;
}
int
SIPCall::SIPSessionReinvite()
{
auto mediaList = getMediaAttributeList();
return SIPSessionReinvite(mediaList);
}
void
SIPCall::sendSIPInfo(std::string_view body, std::string_view subtype)
{
std::lock_guard<std::recursive_mutex> lk {callMutex_};
if (not inviteSession_ or not inviteSession_->dlg)
throw VoipLinkException("Couldn't get invite dialog");
constexpr pj_str_t methodName = CONST_PJ_STR("INFO");
constexpr pj_str_t type = CONST_PJ_STR("application");
pjsip_method method;
pjsip_method_init_np(&method, (pj_str_t*) &methodName);
/* Create request message. */
pjsip_tx_data* tdata;
if (pjsip_dlg_create_request(inviteSession_->dlg, &method, -1, &tdata) != PJ_SUCCESS) {
JAMI_ERR("[call:%s] Could not create dialog", getCallId().c_str());
return;
}
/* Create "application/<subtype>" message body. */
pj_str_t content = CONST_PJ_STR(body);
pj_str_t pj_subtype = CONST_PJ_STR(subtype);
tdata->msg->body = pjsip_msg_body_create(tdata->pool, &type, &pj_subtype, &content);
if (tdata->msg->body == NULL)
pjsip_tx_data_dec_ref(tdata);
else
pjsip_dlg_send_request(inviteSession_->dlg,
tdata,
Manager::instance().sipVoIPLink().getModId(),
NULL);
}
void
SIPCall::updateRecState(bool state)
{
std::string BODY = "<?xml version=\"1.0\" encoding=\"utf-8\" ?>"
"<media_control><vc_primitive><to_encoder>"
"<recording_state="
+ std::to_string(state)
+ "/>"
"</to_encoder></vc_primitive></media_control>";
// see https://tools.ietf.org/html/rfc5168 for XML Schema for Media Control details
JAMI_DBG("Sending recording state via SIP INFO");
try {
sendSIPInfo(BODY, "media_control+xml");
} catch (const std::exception& e) {
JAMI_ERR("Error sending recording state: %s", e.what());
}
}
void
SIPCall::requestKeyframe()
{
auto now = clock::now();
if ((now - lastKeyFrameReq_) < MS_BETWEEN_2_KEYFRAME_REQUEST
and lastKeyFrameReq_ != time_point::min())
return;
constexpr auto BODY = "<?xml version=\"1.0\" encoding=\"utf-8\" ?>" "<media_control><vc_primitive><to_encoder>"
"<picture_fast_update/>"
"</to_encoder></vc_primitive></media_control>"sv;
JAMI_DBG("Sending video keyframe request via SIP INFO");
try {
sendSIPInfo(BODY, "media_control+xml");
} catch (const std::exception& e) {
JAMI_ERR("Error sending video keyframe request: %s", e.what());
}
lastKeyFrameReq_ = now;
}
void
SIPCall::setMute(bool state)
{
std::string BODY = "<?xml version=\"1.0\" encoding=\"utf-8\" ?>"
"<media_control><vc_primitive><to_encoder>"
"<mute_state="
+ std::to_string(state)
+ "/>"
"</to_encoder></vc_primitive></media_control>";
// see https://tools.ietf.org/html/rfc5168 for XML Schema for Media Control details
JAMI_DBG("Sending mute state via SIP INFO");
try {
sendSIPInfo(BODY, "media_control+xml");
} catch (const std::exception& e) {
JAMI_ERR("Error sending mute state: %s", e.what());
}
}
void
SIPCall::setInviteSession(pjsip_inv_session* inviteSession)
{
std::lock_guard<std::recursive_mutex> lk {callMutex_};
if (inviteSession == nullptr and inviteSession_) {
JAMI_DBG("[call:%s] Delete current invite session", getCallId().c_str());
} else if (inviteSession != nullptr) {
// NOTE: The first reference of the invite session is owned by pjsip. If
// that counter goes down to zero the invite will be destroyed, and the
// unique_ptr will point freed datas. To avoid this, we increment the
// ref counter and let our unique_ptr share the ownership of the session
// with pjsip.
if (PJ_SUCCESS != pjsip_inv_add_ref(inviteSession)) {
JAMI_WARN("[call:%s] trying to set invalid invite session [%p]",
getCallId().c_str(),
inviteSession);
inviteSession_.reset(nullptr);
return;
}
JAMI_DBG("[call:%s] Set new invite session [%p]", getCallId().c_str(), inviteSession);
} else {
// Nothing to do.
return;
}
inviteSession_.reset(inviteSession);
}
void
SIPCall::terminateSipSession(int status)
{
JAMI_DBG("[call:%s] Terminate SIP session", getCallId().c_str());
std::lock_guard<std::recursive_mutex> lk {callMutex_};
if (inviteSession_ and inviteSession_->state != PJSIP_INV_STATE_DISCONNECTED) {
pjsip_tx_data* tdata = nullptr;
auto ret = pjsip_inv_end_session(inviteSession_.get(), status, nullptr, &tdata);
if (ret == PJ_SUCCESS) { if (tdata) {
auto account = getSIPAccount();
if (account) {
sip_utils::addContactHeader(account->getContactHeader(
transport_ ? transport_->get() : nullptr),
tdata);
// Add user-agent header
sip_utils::addUserAgentHeader(account->getUserAgentName(), tdata);
} else {
JAMI_ERR("No account detected");
std::ostringstream msg;
msg << "[call:" << getCallId().c_str() << "] "
<< "The account owning this call is invalid";
throw std::runtime_error(msg.str());
}
ret = pjsip_inv_send_msg(inviteSession_.get(), tdata);
if (ret != PJ_SUCCESS)
JAMI_ERR("[call:%s] failed to send terminate msg, SIP error (%s)",
getCallId().c_str(),
sip_utils::sip_strerror(ret).c_str());
}
} else
JAMI_ERR("[call:%s] failed to terminate INVITE@%p, SIP error (%s)",
getCallId().c_str(),
inviteSession_.get(),
sip_utils::sip_strerror(ret).c_str());
}
setInviteSession();
}
void
SIPCall::answer()
{
std::lock_guard<std::recursive_mutex> lk {callMutex_};
auto account = getSIPAccount();
if (!account) {
JAMI_ERR("No account detected");
return;
}
if (not inviteSession_)
throw VoipLinkException("[call:" + getCallId()
+ "] answer: no invite session for this call");
if (!inviteSession_->neg) {
JAMI_WARN("[call:%s] Negotiator is NULL, we've received an INVITE without an SDP",
getCallId().c_str());
Manager::instance().sipVoIPLink().createSDPOffer(inviteSession_.get());
}
setContactHeader(account->getContactHeader(transport_ ? transport_->get() : nullptr));
pjsip_tx_data* tdata;
if (!inviteSession_->last_answer)
throw std::runtime_error("Should only be called for initial answer");
// answer with SDP if no SDP was given in initial invite (i.e. inv->neg is NULL)
if (pjsip_inv_answer(inviteSession_.get(),
PJSIP_SC_OK,
NULL,
!inviteSession_->neg ? sdp_->getLocalSdpSession() : NULL,
&tdata)
!= PJ_SUCCESS)
throw std::runtime_error("Could not init invite request answer (200 OK)");
// contactStr must stay in scope as long as tdata
if (contactHeader_.slen) {
JAMI_DBG("[call:%s] Answering with contact header: %.*s", getCallId().c_str(),
(int) contactHeader_.slen,
contactHeader_.ptr);
sip_utils::addContactHeader(contactHeader_, tdata);
}
// Add user-agent header
sip_utils::addUserAgentHeader(account->getUserAgentName(), tdata);
if (pjsip_inv_send_msg(inviteSession_.get(), tdata) != PJ_SUCCESS) {
setInviteSession();
throw std::runtime_error("Could not send invite request answer (200 OK)");
}
setState(CallState::ACTIVE, ConnectionState::CONNECTED);
}
void
SIPCall::answer(const std::vector<DRing::MediaMap>& mediaList)
{
std::lock_guard<std::recursive_mutex> lk {callMutex_};
auto account = getSIPAccount();
if (not account) {
JAMI_ERR("No account detected");
return;
}
auto mediaAttrList = MediaAttribute::buildMediaAttributesList(mediaList, isSrtpEnabled());
if (mediaAttrList.empty()) {
JAMI_DBG("[call:%s] Media list must not be empty!", getCallId().c_str());
return;
}
if (not inviteSession_)
JAMI_DBG("[call:%s] No invite session for this call", getCallId().c_str());
JAMI_DBG("[call:%s] Answering incoming call with %lu media:",
getCallId().c_str(),
mediaAttrList.size());
if (mediaAttrList.size() != rtpStreams_.size()) {
JAMI_ERR("[call:%s] Media list size %lu in answer does not match. Expected %lu",
getCallId().c_str(),
mediaAttrList.size(),
rtpStreams_.size());
return;
}
for (size_t idx = 0; idx < mediaAttrList.size(); idx++) {
auto const& mediaAttr = mediaAttrList.at(idx);
JAMI_DBG("[call:%s] Media @%lu: %s",
getCallId().c_str(),
idx,
mediaAttr.toString(true).c_str());
}
// Apply the media attributes provided by the user.
for (size_t idx = 0; idx < mediaAttrList.size(); idx++) {
updateMediaStream(mediaAttrList[idx], idx);
}
if (not inviteSession_)
throw VoipLinkException("[call:" + getCallId()
+ "] answer: no invite session for this call");
if (not inviteSession_->neg) {
// We are answering to an INVITE that did not include a media offer (SDP).
// The SIP specification (RFCs 3261/6337) requires that if a UA wishes to
// proceed with the call, it must provide a media offer (SDP) if the initial // INVITE did not offer one. In this case, the SDP offer will be included in
// the SIP OK (200) answer. The peer UA will then include its SDP answer in
// the SIP ACK message.
// TODO. This code should be unified with the code used by accounts to create
// SDP offers.
JAMI_WARN("[call:%s] No negotiator session, peer sent an empty INVITE (without SDP)",
getCallId().c_str());
Manager::instance().sipVoIPLink().createSDPOffer(inviteSession_.get());
generateMediaPorts();
// Setup and create ICE offer
if (isIceEnabled()) {
sdp_->clearIce();
auto opts = account->getIceOptions();
auto publicAddr = account->getPublishedIpAddress();
if (publicAddr) {
opts.accountPublicAddr = publicAddr;
if (auto interfaceAddr = ip_utils::getInterfaceAddr(account->getLocalInterface(),
publicAddr.getFamily())) {
opts.accountLocalAddr = interfaceAddr;
createIceMediaTransport();
if (initIceMediaTransport(true, std::move(opts)))
addLocalIceAttributes();
} else {
JAMI_WARN("[call:%s] Cant init ICE transport, missing local address",
getCallId().c_str());
}
} else {
JAMI_WARN("[call:%s] Cant init ICE transport, missing public address",
getCallId().c_str());
}
}
}
setContactHeader(account->getContactHeader(transport_ ? transport_->get() : nullptr));
if (!inviteSession_->last_answer)
throw std::runtime_error("Should only be called for initial answer");
// Answer with an SDP offer if the initial invite was empty,
// otherwise, set the local_sdp session to null to use the
// current SDP session.
pjsip_tx_data* tdata;
if (pjsip_inv_answer(inviteSession_.get(),
PJSIP_SC_OK,
NULL,
not inviteSession_->neg ? sdp_->getLocalSdpSession() : NULL,
&tdata)
!= PJ_SUCCESS)
throw std::runtime_error("Could not init invite request answer (200 OK)");
if (contactHeader_.slen) {
JAMI_DBG("[call:%s] Answering with contact header: %.*s",
getCallId().c_str(),
(int) contactHeader_.slen,
contactHeader_.ptr);
sip_utils::addContactHeader(contactHeader_, tdata);
}
// Add user-agent header
sip_utils::addUserAgentHeader(account->getUserAgentName(), tdata);
if (pjsip_inv_send_msg(inviteSession_.get(), tdata) != PJ_SUCCESS) { setInviteSession();
throw std::runtime_error("Could not send invite request answer (200 OK)");
}
setState(CallState::ACTIVE, ConnectionState::CONNECTED);
}
void
SIPCall::answerMediaChangeRequest(const std::vector<DRing::MediaMap>& mediaList)
{
std::lock_guard<std::recursive_mutex> lk {callMutex_};
auto account = getSIPAccount();
if (not account) {
JAMI_ERR("[call:%s] No account detected", getCallId().c_str());
return;
}
auto mediaAttrList = MediaAttribute::buildMediaAttributesList(mediaList, isSrtpEnabled());
if (mediaAttrList.empty()) {
JAMI_DBG("[call:%s] Media list size is empty. Ignoring the media change request",
getCallId().c_str());
return;
}
if (not sdp_) {
JAMI_ERR("[call:%s] No valid SDP session", getCallId().c_str());
return;
}
JAMI_DBG("[call:%s] Current media", getCallId().c_str());
unsigned idx = 0;
for (auto const& rtp : rtpStreams_) {
JAMI_DBG("[call:%s] Media @%u: %s",
getCallId().c_str(),
idx++,
rtp.mediaAttribute_->toString(true).c_str());
}
JAMI_DBG("[call:%s] Answering to media change request with new media", getCallId().c_str());
idx = 0;
for (auto const& newMediaAttr : mediaAttrList) {
JAMI_DBG("[call:%s] Media @%u: %s",
getCallId().c_str(),
idx++,
newMediaAttr.toString(true).c_str());
}
updateAllMediaStreams(mediaAttrList);
if (not sdp_->processIncomingOffer(mediaAttrList)) {
JAMI_WARN("[call:%s] Could not process the new offer, ignoring", getCallId().c_str());
return;
}
if (not sdp_->getRemoteSdpSession()) {
JAMI_ERR("[call:%s] No valid remote SDP session", getCallId().c_str());
return;
}
if (isIceEnabled())
setupIceResponse();
if (not sdp_->startNegotiation()) {
JAMI_ERR("[call:%s] Could not start media negotiation for a re-invite request",
getCallId().c_str());
return;
}
if (pjsip_inv_set_sdp_answer(inviteSession_.get(), sdp_->getLocalSdpSession()) != PJ_SUCCESS) {
JAMI_ERR("[call:%s] Could not start media negotiation for a re-invite request",
getCallId().c_str());
return;
}
pjsip_tx_data* tdata;
if (pjsip_inv_answer(inviteSession_.get(), PJSIP_SC_OK, NULL, NULL, &tdata) != PJ_SUCCESS) {
JAMI_ERR("[call:%s] Could not init answer to a re-invite request", getCallId().c_str());
return;
}
if (contactHeader_.slen) {
sip_utils::addContactHeader(contactHeader_, tdata);
}
// Add user-agent header
sip_utils::addUserAgentHeader(account->getUserAgentName(), tdata);
if (pjsip_inv_send_msg(inviteSession_.get(), tdata) != PJ_SUCCESS) {
JAMI_ERR("[call:%s] Could not send answer to a re-invite request", getCallId().c_str());
setInviteSession();
return;
}
JAMI_DBG("[call:%s] Successfully answered the media change request", getCallId().c_str());
}
void
SIPCall::hangup(int reason)
{
std::lock_guard<std::recursive_mutex> lk {callMutex_};
pendingRecord_ = false;
if (inviteSession_ and inviteSession_->dlg) {
pjsip_route_hdr* route = inviteSession_->dlg->route_set.next;
while (route and route != &inviteSession_->dlg->route_set) {
char buf[1024];
int printed = pjsip_hdr_print_on(route, buf, sizeof(buf));
if (printed >= 0) {
buf[printed] = '\0';
JAMI_DBG("[call:%s] Route header %s", getCallId().c_str(), buf);
}
route = route->next;
}
int status = PJSIP_SC_OK;
if (reason)
status = reason;
else if (inviteSession_->state <= PJSIP_INV_STATE_EARLY
and inviteSession_->role != PJSIP_ROLE_UAC)
status = PJSIP_SC_CALL_TSX_DOES_NOT_EXIST;
else if (inviteSession_->state >= PJSIP_INV_STATE_DISCONNECTED)
status = PJSIP_SC_DECLINE;
// Notify the peer
terminateSipSession(status);
}
// Stop all RTP streams
stopAllMedia();
setState(Call::ConnectionState::DISCONNECTED, reason);
dht::ThreadPool::io().run([w = weak()] {
if (auto shared = w.lock())
shared->removeCall();
});
}
void
SIPCall::refuse()
{ if (!isIncoming() or getConnectionState() == ConnectionState::CONNECTED or !inviteSession_)
return;
stopAllMedia();
// Notify the peer
terminateSipSession(PJSIP_SC_BUSY_HERE);
setState(Call::ConnectionState::DISCONNECTED, ECONNABORTED);
removeCall();
}
static void
transfer_client_cb(pjsip_evsub* sub, pjsip_event* event)
{
auto mod_ua_id = Manager::instance().sipVoIPLink().getModId();
switch (pjsip_evsub_get_state(sub)) {
case PJSIP_EVSUB_STATE_ACCEPTED:
if (!event)
return;
pj_assert(event->type == PJSIP_EVENT_TSX_STATE
&& event->body.tsx_state.type == PJSIP_EVENT_RX_MSG);
break;
case PJSIP_EVSUB_STATE_TERMINATED:
pjsip_evsub_set_mod_data(sub, mod_ua_id, NULL);
break;
case PJSIP_EVSUB_STATE_ACTIVE: {
if (!event)
return;
pjsip_rx_data* r_data = event->body.rx_msg.rdata;
if (!r_data)
return;
std::string request(pjsip_rx_data_get_info(r_data));
pjsip_status_line status_line = {500, *pjsip_get_status_text(500)};
if (!r_data->msg_info.msg)
return;
if (r_data->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD
and request.find("NOTIFY") != std::string::npos) {
pjsip_msg_body* body = r_data->msg_info.msg->body;
if (!body)
return;
if (pj_stricmp2(&body->content_type.type, "message")
or pj_stricmp2(&body->content_type.subtype, "sipfrag"))
return;
if (pjsip_parse_status_line((char*) body->data, body->len, &status_line) != PJ_SUCCESS)
return;
}
if (!r_data->msg_info.cid)
return;
auto call = static_cast<SIPCall*>(pjsip_evsub_get_mod_data(sub, mod_ua_id));
if (!call)
return;
if (status_line.code / 100 == 2) {
if (call->inviteSession_) call->terminateSipSession(PJSIP_SC_GONE);
Manager::instance().hangupCall(call->getCallId());
pjsip_evsub_set_mod_data(sub, mod_ua_id, NULL);
}
break;
}
case PJSIP_EVSUB_STATE_NULL:
case PJSIP_EVSUB_STATE_SENT:
case PJSIP_EVSUB_STATE_PENDING:
case PJSIP_EVSUB_STATE_UNKNOWN:
default:
break;
}
}
bool
SIPCall::transferCommon(const pj_str_t* dst)
{
if (not inviteSession_ or not inviteSession_->dlg)
return false;
pjsip_evsub_user xfer_cb;
pj_bzero(&xfer_cb, sizeof(xfer_cb));
xfer_cb.on_evsub_state = &transfer_client_cb;
pjsip_evsub* sub;
if (pjsip_xfer_create_uac(inviteSession_->dlg, &xfer_cb, &sub) != PJ_SUCCESS)
return false;
/* Associate this voiplink of call with the client subscription
* We can not just associate call with the client subscription
* because after this function, we can no find the cooresponding
* voiplink from the call any more. But the voiplink is useful!
*/
pjsip_evsub_set_mod_data(sub, Manager::instance().sipVoIPLink().getModId(), this);
/*
* Create REFER request.
*/
pjsip_tx_data* tdata;
if (pjsip_xfer_initiate(sub, dst, &tdata) != PJ_SUCCESS)
return false;
/* Send. */
if (pjsip_xfer_send_request(sub, tdata) != PJ_SUCCESS)
return false;
return true;
}
void
SIPCall::transfer(const std::string& to)
{
auto account = getSIPAccount();
if (!account) {
JAMI_ERR("No account detected");
return;
}
if (Recordable::isRecording()) {
deinitRecorder();
stopRecording();
}
std::string toUri = account->getToUri(to);
const pj_str_t dst(CONST_PJ_STR(toUri)); JAMI_DBG("[call:%s] Transferring to %.*s", getCallId().c_str(), (int) dst.slen, dst.ptr);
if (!transferCommon(&dst))
throw VoipLinkException("Couldn't transfer");
}
bool
SIPCall::attendedTransfer(const std::string& to)
{
auto toCall = Manager::instance().callFactory.getCall<SIPCall>(to);
if (!toCall)
return false;
if (not toCall->inviteSession_ or not toCall->inviteSession_->dlg)
return false;
pjsip_dialog* target_dlg = toCall->inviteSession_->dlg;
pjsip_uri* uri = (pjsip_uri*) pjsip_uri_get_uri(target_dlg->remote.info->uri);
char str_dest_buf[PJSIP_MAX_URL_SIZE * 2] = {'<'};
pj_str_t dst = {str_dest_buf, 1};
dst.slen += pjsip_uri_print(PJSIP_URI_IN_REQ_URI,
uri,
str_dest_buf + 1,
sizeof(str_dest_buf) - 1);
dst.slen += pj_ansi_snprintf(str_dest_buf + dst.slen,
sizeof(str_dest_buf) - dst.slen,
"?"
"Replaces=%.*s"
"%%3Bto-tag%%3D%.*s"
"%%3Bfrom-tag%%3D%.*s>",
(int) target_dlg->call_id->id.slen,
target_dlg->call_id->id.ptr,
(int) target_dlg->remote.info->tag.slen,
target_dlg->remote.info->tag.ptr,
(int) target_dlg->local.info->tag.slen,
target_dlg->local.info->tag.ptr);
return transferCommon(&dst);
}
bool
SIPCall::onhold(OnReadyCb&& cb)
{
// If ICE is currently negotiating, we must wait before hold the call
if (isWaitingForIceAndMedia_) {
holdCb_ = std::move(cb);
remainingRequest_ = Request::HoldingOn;
return false;
}
auto result = hold();
if (cb)
cb(result);
return result;
}
bool
SIPCall::hold()
{
if (not setState(CallState::HOLD))
return false;
stopAllMedia();
if (getConnectionState() == ConnectionState::CONNECTED) {
if (SIPSessionReinvite() != PJ_SUCCESS) { JAMI_WARN("[call:%s] Reinvite failed", getCallId().c_str());
return true;
}
}
isWaitingForIceAndMedia_ = true;
return true;
}
bool
SIPCall::offhold(OnReadyCb&& cb)
{
// If ICE is currently negotiating, we must wait before unhold the call
if (isWaitingForIceAndMedia_) {
offHoldCb_ = std::move(cb);
remainingRequest_ = Request::HoldingOff;
return false;
}
auto result = unhold();
if (cb)
cb(result);
return result;
}
bool
SIPCall::unhold()
{
auto account = getSIPAccount();
if (!account) {
JAMI_ERR("No account detected");
return false;
}
bool success = false;
try {
success = internalOffHold([] {});
} catch (const SdpException& e) {
JAMI_ERR("[call:%s] %s", getCallId().c_str(), e.what());
throw VoipLinkException("SDP issue in offhold");
}
if (success)
isWaitingForIceAndMedia_ = true;
return success;
}
bool
SIPCall::internalOffHold(const std::function<void()>& sdp_cb)
{
if (not setState(CallState::ACTIVE))
return false;
sdp_cb();
if (getConnectionState() == ConnectionState::CONNECTED) {
if (SIPSessionReinvite() != PJ_SUCCESS) {
JAMI_WARN("[call:%s] resuming hold", getCallId().c_str());
if (isWaitingForIceAndMedia_) {
remainingRequest_ = Request::HoldingOn;
} else {
hold();
}
return false;
}
}
return true;
}
void
SIPCall::switchInput(const std::string& source)
{
JAMI_DBG("[call:%s] Set selected source to %s", getCallId().c_str(), source.c_str());
for (auto const& stream : rtpStreams_) {
auto mediaAttr = stream.mediaAttribute_;
mediaAttr->sourceUri_ = source;
}
// Check if the call is being recorded in order to continue
// ... the recording after the switch
bool isRec = Call::isRecording();
if (isWaitingForIceAndMedia_) {
remainingRequest_ = Request::SwitchInput;
} else {
if (SIPSessionReinvite() == PJ_SUCCESS) {
isWaitingForIceAndMedia_ = true;
}
}
if (isRec) {
readyToRecord_ = false;
resetMediaReady();
pendingRecord_ = true;
}
}
void
SIPCall::peerHungup()
{
pendingRecord_ = false;
// Stop all RTP streams
stopAllMedia();
if (inviteSession_)
terminateSipSession(PJSIP_SC_NOT_FOUND);
Call::peerHungup();
}
void
SIPCall::carryingDTMFdigits(char code)
{
int duration = Manager::instance().voipPreferences.getPulseLength();
char dtmf_body[1000];
int ret;
// handle flash code
if (code == '!') {
ret = snprintf(dtmf_body, sizeof dtmf_body - 1, "Signal=16\r\nDuration=%d\r\n", duration);
} else {
ret = snprintf(dtmf_body,
sizeof dtmf_body - 1,
"Signal=%c\r\nDuration=%d\r\n",
code,
duration);
}
try {
sendSIPInfo({dtmf_body, (size_t) ret}, "dtmf-relay");
} catch (const std::exception& e) {
JAMI_ERR("Error sending DTMF: %s", e.what());
}
}
voidSIPCall::setVideoOrientation(int rotation)
{
std::string sip_body = "<?xml version=\"1.0\" encoding=\"utf-8\" ?>"
"<media_control><vc_primitive><to_encoder>"
"<device_orientation="
+ std::to_string(-rotation)
+ "/>"
"</to_encoder></vc_primitive></media_control>";
JAMI_DBG("Sending device orientation via SIP INFO %d", rotation);
sendSIPInfo(sip_body, "media_control+xml");
}
void
SIPCall::sendTextMessage(const std::map<std::string, std::string>& messages, const std::string& from)
{
std::lock_guard<std::recursive_mutex> lk {callMutex_};
// TODO: for now we ignore the "from" (the previous implementation for sending this info was
// buggy and verbose), another way to send the original message sender will be implemented
// in the future
if (not subcalls_.empty()) {
pendingOutMessages_.emplace_back(messages, from);
for (auto& c : subcalls_)
c->sendTextMessage(messages, from);
} else {
if (inviteSession_) {
try {
im::sendSipMessage(inviteSession_.get(), messages);
} catch (...) {
}
} else {
pendingOutMessages_.emplace_back(messages, from);
JAMI_ERR("[call:%s] sendTextMessage: no invite session for this call",
getCallId().c_str());
}
}
}
void
SIPCall::removeCall()
{
#ifdef ENABLE_PLUGIN
jami::Manager::instance().getJamiPluginManager().getCallServicesManager().clearCallHandlerMaps(
getCallId());
#endif
std::lock_guard<std::recursive_mutex> lk {callMutex_};
JAMI_DBG("[call:%s] removeCall()", getCallId().c_str());
if (sdp_) {
sdp_->setActiveLocalSdpSession(nullptr);
sdp_->setActiveRemoteSdpSession(nullptr);
}
Call::removeCall();
{
std::lock_guard<std::mutex> lk(transportMtx_);
resetTransport(std::move(mediaTransport_));
}
setInviteSession();
setTransport({});
}
void
SIPCall::onFailure(signed cause)
{
if (setState(CallState::MERROR, ConnectionState::DISCONNECTED, cause)) {
runOnMainThread([w = weak()] {
if (auto shared = w.lock()) {
auto& call = *shared; Manager::instance().callFailure(call);
call.removeCall();
}
});
}
}
void
SIPCall::onBusyHere()
{
if (getCallType() == CallType::OUTGOING)
setState(CallState::PEER_BUSY, ConnectionState::DISCONNECTED);
else
setState(CallState::BUSY, ConnectionState::DISCONNECTED);
runOnMainThread([w = weak()] {
if (auto shared = w.lock()) {
auto& call = *shared;
Manager::instance().callBusy(call);
call.removeCall();
}
});
}
void
SIPCall::onClosed()
{
runOnMainThread([w = weak()] {
if (auto shared = w.lock()) {
auto& call = *shared;
Manager::instance().peerHungupCall(call);
call.removeCall();
}
});
}
void
SIPCall::onAnswered()
{
JAMI_WARN("[call:%s] onAnswered()", getCallId().c_str());
runOnMainThread([w = weak()] {
if (auto shared = w.lock()) {
if (shared->getConnectionState() != ConnectionState::CONNECTED) {
shared->setState(CallState::ACTIVE, ConnectionState::CONNECTED);
if (not shared->isSubcall()) {
Manager::instance().peerAnsweredCall(*shared);
}
}
}
});
}
void
SIPCall::sendKeyframe()
{
#ifdef ENABLE_VIDEO
dht::ThreadPool::computation().run([w = weak()] {
if (auto sthis = w.lock()) {
JAMI_DBG("handling picture fast update request");
if (auto const& videoRtp = sthis->getVideoRtp()) {
videoRtp->forceKeyFrame();
}
}
});
#endif
}
bool
SIPCall::isIceEnabled() const
{ return enableIce_;
}
void
SIPCall::setPeerUaVersion(std::string_view ua)
{
if (peerUserAgent_ == ua or ua.empty()) {
// Silently ignore if it did not change or empty.
return;
}
if (peerUserAgent_.empty()) {
JAMI_DBG("[call:%s] Set peer's User-Agent to [%.*s]",
getCallId().c_str(),
(int) ua.size(),
ua.data());
} else if (not peerUserAgent_.empty()) {
// Unlikely, but should be handled since we dont have control over the peer.
// Even if it's unexpected, we still try to parse the UA version.
JAMI_WARN("[call:%s] Peer's User-Agent unexpectedly changed from [%s] to [%.*s]",
getCallId().c_str(),
peerUserAgent_.c_str(),
(int) ua.size(),
ua.data());
}
peerUserAgent_ = ua;
// User-agent parsing
constexpr std::string_view PACK_NAME(PACKAGE_NAME " ");
auto pos = ua.find(PACK_NAME);
if (pos == std::string_view::npos) {
// Must have the expected package name.
JAMI_WARN("Could not find the expected package name in peer's User-Agent");
return;
}
ua = ua.substr(pos + PACK_NAME.length());
std::string_view version;
// Unstable (un-released) versions has a hiphen + commit Id after
// the version number. Find the commit Id if any, and ignore it.
pos = ua.find('-');
if (pos != std::string_view::npos) {
// Get the version and ignore the commit ID.
version = ua.substr(0, pos);
} else {
// Extract the version number.
pos = ua.find(' ');
if (pos != std::string_view::npos) {
version = ua.substr(0, pos);
}
}
if (version.empty()) {
JAMI_DBG("[call:%s] Could not parse peer's version", getCallId().c_str());
return;
}
auto peerVersion = split_string_to_unsigned(version, '.');
if (peerVersion.size() > 4u) {
JAMI_WARN("[call:%s] Could not parse peer's version", getCallId().c_str());
return;
}
// Check if peer's version is at least 10.0.2 to enable multi-stream.
peerSupportMultiStream_ = Account::meetMinimumRequiredVersion(peerVersion,
MULTISTREAM_REQUIRED_VERSION);
if (not peerSupportMultiStream_) {
JAMI_DBG( "Peer's version [%.*s] does not support multi-stream. Min required version: [%.*s]",
(int) version.size(),
version.data(),
(int) MULTISTREAM_REQUIRED_VERSION_STR.size(),
MULTISTREAM_REQUIRED_VERSION_STR.data());
}
}
void
SIPCall::onPeerRinging()
{
JAMI_DBG("[call:%s] Peer ringing", getCallId().c_str());
setState(ConnectionState::RINGING);
}
void
SIPCall::addLocalIceAttributes()
{
if (not isIceEnabled())
return;
auto mediaTransport = getIceMedia();
if (not mediaTransport) {
JAMI_ERR("[call:%s] Invalid ICE instance", getCallId().c_str());
return;
}
auto start = std::chrono::steady_clock::now();
if (not mediaTransport->isInitialized()) {
JAMI_DBG("[call:%s] Waiting for ICE initialization", getCallId().c_str());
// we need an initialized ICE to progress further
if (mediaTransport->waitForInitialization(DEFAULT_ICE_INIT_TIMEOUT) <= 0) {
JAMI_ERR("[call:%s] ICE initialization timed out", getCallId().c_str());
return;
}
// ICE initialization may take longer than usual in some cases,
// for instance when TURN servers do not respond in time (DNS
// resolution or other issues).
auto duration = std::chrono::steady_clock::now() - start;
if (duration > EXPECTED_ICE_INIT_MAX_TIME) {
JAMI_WARN("[call:%s] ICE initialization time was unexpectedly high (%ld ms)",
getCallId().c_str(),
std::chrono::duration_cast<std::chrono::milliseconds>(duration).count());
}
}
// Check the state of ICE instance, the initialization may have failed.
if (not mediaTransport->isInitialized()) {
JAMI_ERR("[call:%s] ICE session is not initialized", getCallId().c_str());
return;
}
// Check the state, the call might have been canceled while waiting.
// for initialization.
if (getState() == Call::CallState::OVER) {
JAMI_WARN("[call:%s] The call was terminated while waiting for ICE initialization",
getCallId().c_str());
return;
}
auto account = getSIPAccount();
if (not account) {
JAMI_ERR("No account detected");
return;
}
if (not sdp_) {
JAMI_ERR("No sdp detected");
return;
}
JAMI_DBG("[call:%s] Add local attributes for ICE instance [%p]",
getCallId().c_str(),
mediaTransport.get());
sdp_->addIceAttributes(mediaTransport->getLocalAttributes());
if (account->isIceCompIdRfc5245Compliant()) {
unsigned streamIdx = 0;
for (auto const& stream : rtpStreams_) {
if (not stream.mediaAttribute_->enabled_) {
// Dont add ICE candidates if the media is disabled
JAMI_DBG("[call:%s] media [%s] @ %u is disabled, dont add local candidates",
getCallId().c_str(),
stream.mediaAttribute_->toString().c_str(),
streamIdx);
continue;
}
JAMI_DBG("[call:%s] add ICE local candidates for media [%s] @ %u",
getCallId().c_str(),
stream.mediaAttribute_->toString().c_str(),
streamIdx);
// RTP
sdp_->addIceCandidates(streamIdx,
mediaTransport->getLocalCandidates(streamIdx, ICE_COMP_ID_RTP));
// RTCP if it has its own port
if (not rtcpMuxEnabled_) {
sdp_->addIceCandidates(streamIdx,
mediaTransport->getLocalCandidates(streamIdx,
ICE_COMP_ID_RTP + 1));
}
streamIdx++;
}
} else {
unsigned idx = 0;
unsigned compId = 1;
for (auto const& stream : rtpStreams_) {
if (not stream.mediaAttribute_->enabled_) {
// Skipping local ICE candidates if the media is disabled
continue;
}
JAMI_DBG("[call:%s] add ICE local candidates for media [%s] @ %u",
getCallId().c_str(),
stream.mediaAttribute_->toString().c_str(),
idx);
// RTP
sdp_->addIceCandidates(idx, mediaTransport->getLocalCandidates(compId));
compId++;
// RTCP if it has its own port
if (not rtcpMuxEnabled_) {
sdp_->addIceCandidates(idx, mediaTransport->getLocalCandidates(compId));
compId++;
}
idx++;
}
}
}
std::vector<IceCandidate>
SIPCall::getAllRemoteCandidates(IceTransport& transport) const
{
std::vector<IceCandidate> rem_candidates;
for (unsigned mediaIdx = 0; mediaIdx < static_cast<unsigned>(rtpStreams_.size()); mediaIdx++) {
IceCandidate cand;
for (auto& line : sdp_->getIceCandidates(mediaIdx)) {
if (transport.parseIceAttributeLine(mediaIdx, line, cand)) {
JAMI_DBG("[call:%s] Add remote ICE candidate: %s", getCallId().c_str(),
line.c_str());
rem_candidates.emplace_back(std::move(cand));
}
}
}
return rem_candidates;
}
std::shared_ptr<AccountCodecInfo>
SIPCall::getVideoCodec() const
{
#ifdef ENABLE_VIDEO
if (auto const& videoRtp = getVideoRtp())
return videoRtp->getCodec();
#endif
return {};
}
std::shared_ptr<AccountCodecInfo>
SIPCall::getAudioCodec() const
{
if (auto const& audioRtp = getAudioRtp())
return audioRtp->getCodec();
return {};
}
void
SIPCall::addMediaStream(const MediaAttribute& mediaAttr)
{
// Create and add the media stream with the provided attribute.
// Do not create the RTP sessions yet.
RtpStream stream;
stream.mediaAttribute_ = std::make_shared<MediaAttribute>(mediaAttr);
// Set default media source if empty. Kept for backward compatibility.
#ifdef ENABLE_VIDEO
if (stream.mediaAttribute_->sourceUri_.empty()) {
stream.mediaAttribute_->sourceUri_
= Manager::instance().getVideoManager().videoDeviceMonitor.getMRLForDefaultDevice();
}
#endif
if (stream.mediaAttribute_->sourceType_ == MediaSourceType::NONE) {
stream.mediaAttribute_->sourceType_ = MediaSourceType::CAPTURE_DEVICE;
}
rtpStreams_.emplace_back(std::move(stream));
}
size_t
SIPCall::initMediaStreams(const std::vector<MediaAttribute>& mediaAttrList)
{
for (size_t idx = 0; idx < mediaAttrList.size(); idx++) {
auto const& mediaAttr = mediaAttrList.at(idx);
if (mediaAttr.type_ != MEDIA_AUDIO && mediaAttr.type_ != MEDIA_VIDEO) {
JAMI_ERR("[call:%s] Unexpected media type %u", getCallId().c_str(), mediaAttr.type_);
assert(false);
}
addMediaStream(mediaAttr);
auto& stream = rtpStreams_.back();
createRtpSession(stream);
JAMI_DBG("[call:%s] Added media @%lu: %s",
getCallId().c_str(),
idx,
stream.mediaAttribute_->toString(true).c_str());
}
JAMI_DBG("[call:%s] Created %lu Media streams", getCallId().c_str(), rtpStreams_.size());
return rtpStreams_.size();
}
bool
SIPCall::hasVideo() const
{
#ifdef ENABLE_VIDEO
std::function<bool(const RtpStream& stream)> videoCheck = [](auto const& stream) {
return stream.mediaAttribute_->type_ == MediaType::MEDIA_VIDEO;
};
const auto iter = std::find_if(rtpStreams_.begin(), rtpStreams_.end(), videoCheck);
return iter != rtpStreams_.end();
#else
return false;
#endif
}
bool
SIPCall::isCaptureDeviceMuted(const MediaType& mediaType) const
{
// Return true only if all media of type 'mediaType' that use capture devices
// source, are muted.
std::function<bool(const RtpStream& stream)> mutedCheck = [&mediaType](auto const& stream) {
return (stream.mediaAttribute_->type_ == mediaType
and stream.mediaAttribute_->sourceType_ == MediaSourceType::CAPTURE_DEVICE
and not stream.mediaAttribute_->muted_);
};
const auto iter = std::find_if(rtpStreams_.begin(), rtpStreams_.end(), mutedCheck);
return iter == rtpStreams_.end();
}
void
SIPCall::updateNegotiatedMedia()
{
JAMI_DBG("[call:%s] updating negotiated media", getCallId().c_str());
if (not transport_ or not sdp_) {
JAMI_ERR("[call:%s] the call is in invalid state", getCallId().c_str());
return;
}
auto slots = sdp_->getMediaSlots();
bool peer_holding {true};
int streamIdx = -1;
for (const auto& slot : slots) {
streamIdx++;
const auto& local = slot.first;
const auto& remote = slot.second;
// Skip disabled media
if (not local.enabled) {
JAMI_DBG("[call:%s] [SDP:slot#%u] The media is disabled, skipping",
getCallId().c_str(),
streamIdx);
continue;
}
if (static_cast<size_t>(streamIdx) >= rtpStreams_.size()) {
throw std::runtime_error("Stream index is out-of-range");
}
auto const& rtpStream = rtpStreams_[streamIdx];
if (not rtpStream.mediaAttribute_) {
throw std::runtime_error("Missing media attribute"); }
// To enable a media, it must be enabled on both sides.
rtpStream.mediaAttribute_->enabled_ = local.enabled and remote.enabled;
if (not rtpStream.rtpSession_)
throw std::runtime_error("Must have a valid RTP Session");
if (local.type != MEDIA_AUDIO && local.type != MEDIA_VIDEO) {
JAMI_ERR("[call:%s] Unexpected media type %u", getCallId().c_str(), local.type);
throw std::runtime_error("Invalid media attribute");
}
if (local.type != remote.type) {
JAMI_ERR("[call:%s] [SDP:slot#%u] Inconsistent media type between local and remote",
getCallId().c_str(),
streamIdx);
continue;
}
if (local.enabled and not local.codec) {
JAMI_WARN("[call:%s] [SDP:slot#%u] Missing local codec", getCallId().c_str(), streamIdx);
continue;
}
if (remote.enabled and not remote.codec) {
JAMI_WARN("[call:%s] [SDP:slot#%u] Missing remote codec",
getCallId().c_str(),
streamIdx);
continue;
}
if (isSrtpEnabled() and local.enabled and not local.crypto) {
JAMI_WARN("[call:%s] [SDP:slot#%u] Secure mode but no local crypto attributes. "
"Ignoring the media",
getCallId().c_str(),
streamIdx);
continue;
}
if (isSrtpEnabled() and remote.enabled and not remote.crypto) {
JAMI_WARN("[call:%s] [SDP:slot#%u] Secure mode but no crypto remote attributes. "
"Ignoring the media",
getCallId().c_str(),
streamIdx);
continue;
}
// Aggregate holding info over all remote streams
peer_holding &= remote.onHold;
configureRtpSession(rtpStream.rtpSession_, rtpStream.mediaAttribute_, local, remote);
}
if (not isSubcall() and peerHolding_ != peer_holding) {
peerHolding_ = peer_holding;
emitSignal<DRing::CallSignal::PeerHold>(getCallId(), peerHolding_);
}
// Notify using the parent Id if it's a subcall.
auto callId = isSubcall() ? parent_->getCallId() : getCallId();
emitSignal<DRing::CallSignal::MediaNegotiationStatus>(
callId,
DRing::Media::MediaNegotiationStatusEvents::NEGOTIATION_SUCCESS,
MediaAttribute::mediaAttributesToMediaMaps(getMediaAttributeList()));
}
void
SIPCall::startAllMedia()
{ JAMI_DBG("[call:%s] starting the media", getCallId().c_str());
if (not transport_ or not sdp_) {
JAMI_ERR("[call:%s] the call is in invalid state", getCallId().c_str());
return;
}
if (isSrtpEnabled() && not transport_->isSecure()) {
JAMI_WARN("[call:%s] Crypto (SRTP) is negotiated over an insecure signaling transport",
getCallId().c_str());
}
// reset
readyToRecord_ = false;
resetMediaReady();
#ifdef ENABLE_VIDEO
bool isVideoEnabled = false;
#endif
int currentCompId = 1;
for (auto iter = rtpStreams_.begin(); iter != rtpStreams_.end(); iter++) {
if (not iter->mediaAttribute_) {
throw std::runtime_error("Missing media attribute");
}
#ifdef ENABLE_VIDEO
if (iter->mediaAttribute_->type_ == MEDIA_VIDEO)
isVideoEnabled |= iter->mediaAttribute_->enabled_;
#endif
// Not restarting media loop on hold as it's a huge waste of CPU ressources
// because of the audio loop
if (getState() != CallState::HOLD) {
if (isIceRunning()) {
// Create sockets for RTP and RTCP, and start the session.
auto iceRtpSocket = newIceSocket(currentCompId++);
std::unique_ptr<IceSocket> iceRtcpSocket;
if (not rtcpMuxEnabled_) {
iceRtcpSocket = newIceSocket(currentCompId++);
}
iter->rtpSession_->start(std::move(iceRtpSocket), std::move(iceRtcpSocket));
} else {
iter->rtpSession_->start(nullptr, nullptr);
}
}
}
#ifdef ENABLE_VIDEO
// TODO. Move this elsewhere (when adding participant to conf?)
if (!isVideoEnabled && !getConfId().empty()) {
auto conference = Manager::instance().getConferenceFromID(getConfId());
conference->attachVideo(getReceiveVideoFrameActiveWriter().get(), getCallId());
}
#endif
// Media is restarted, we can process the last holding request.
isWaitingForIceAndMedia_ = false;
if (remainingRequest_ != Request::NoRequest) {
bool result = true;
switch (remainingRequest_) {
case Request::HoldingOn:
result = hold();
if (holdCb_) {
holdCb_(result);
holdCb_ = nullptr;
}
break;
case Request::HoldingOff: result = unhold();
if (offHoldCb_) {
offHoldCb_(result);
offHoldCb_ = nullptr;
}
break;
case Request::SwitchInput:
SIPSessionReinvite();
break;
default:
break;
}
remainingRequest_ = Request::NoRequest;
}
#ifdef ENABLE_PLUGIN
// Create AVStreams associated with the call
createCallAVStreams();
#endif
}
void
SIPCall::restartMediaSender()
{
JAMI_DBG("[call:%s] restarting TX media streams", getCallId().c_str());
auto const& audioRtp = getAudioRtp();
if (audioRtp)
audioRtp->restartSender();
#ifdef ENABLE_VIDEO
if (hasVideo()) {
auto const& videoRtp = getVideoRtp();
if (videoRtp)
videoRtp->restartSender();
}
#endif
}
void
SIPCall::stopAllMedia()
{
JAMI_DBG("[call:%s] stopping all medias", getCallId().c_str());
if (Recordable::isRecording()) {
deinitRecorder();
stopRecording(); // if call stops, finish recording
}
auto const& audioRtp = getAudioRtp();
if (audioRtp)
audioRtp->stop();
#ifdef ENABLE_VIDEO
auto const& videoRtp = getVideoRtp();
{
std::lock_guard<std::mutex> lk(sinksMtx_);
for (auto it = callSinksMap_.begin(); it != callSinksMap_.end();) {
auto& videoReceive = videoRtp->getVideoReceive();
if (videoReceive) {
videoReceive->detach(it->second.get());
}
it->second->stop();
it = callSinksMap_.erase(it);
}
}
if (videoRtp)
videoRtp->stop();
#endif
#ifdef ENABLE_PLUGIN {
clearCallAVStreams();
std::lock_guard<std::mutex> lk(avStreamsMtx_);
Manager::instance().getJamiPluginManager().getCallServicesManager().clearAVSubject(
getCallId());
}
#endif
}
void
SIPCall::muteMedia(const std::string& mediaType, bool mute)
{
auto type = MediaAttribute::stringToMediaType(mediaType);
if (type == MediaType::MEDIA_AUDIO) {
JAMI_WARN("[call:%s] %s all audio medias",
getCallId().c_str(),
mute ? "muting " : "un-muting ");
} else if (type == MediaType::MEDIA_VIDEO) {
JAMI_WARN("[call:%s] %s all video medias",
getCallId().c_str(),
mute ? "muting" : "un-muting");
} else {
JAMI_ERR("[call:%s] invalid media type %s", getCallId().c_str(), mediaType.c_str());
assert(false);
}
// Get the current media attributes.
auto mediaList = getMediaAttributeList();
// Mute/Un-mute all medias with matching type.
for (auto& mediaAttr : mediaList) {
if (mediaAttr.type_ == type) {
mediaAttr.muted_ = mute;
}
}
// Apply
requestMediaChange(MediaAttribute::mediaAttributesToMediaMaps(mediaList));
}
void
SIPCall::updateMediaStream(const MediaAttribute& newMediaAttr, size_t streamIdx)
{
assert(streamIdx < rtpStreams_.size());
auto const& rtpStream = rtpStreams_[streamIdx];
assert(rtpStream.rtpSession_);
auto const& mediaAttr = rtpStream.mediaAttribute_;
assert(mediaAttr);
bool notify = false;
if (newMediaAttr.muted_ == mediaAttr->muted_) {
// Nothing to do. Already in the desired state.
JAMI_DBG("[call:%s] [%s] already %s",
getCallId().c_str(),
mediaAttr->label_.c_str(),
mediaAttr->muted_ ? "muted " : "un-muted ");
} else {
notify = true;
}
// Update
mediaAttr->muted_ = newMediaAttr.muted_;
// Only update source and type if actually set.
if (not newMediaAttr.sourceUri_.empty()) mediaAttr->sourceUri_ = newMediaAttr.sourceUri_;
if (newMediaAttr.sourceType_ != MediaSourceType::NONE)
mediaAttr->sourceType_ = newMediaAttr.sourceType_;
JAMI_DBG("[call:%s] %s [%s]",
getCallId().c_str(),
mediaAttr->muted_ ? "muting" : "un-muting",
mediaAttr->label_.c_str());
if (notify and mediaAttr->type_ == MediaType::MEDIA_AUDIO) {
rtpStream.rtpSession_->setMuted(mediaAttr->muted_);
setMute(mediaAttr->muted_);
if (not isSubcall())
emitSignal<DRing::CallSignal::AudioMuted>(getCallId(), mediaAttr->muted_);
return;
}
#ifdef ENABLE_VIDEO
if (notify and mediaAttr->type_ == MediaType::MEDIA_VIDEO and not isSubcall()) {
emitSignal<DRing::CallSignal::VideoMuted>(getCallId(), mediaAttr->muted_);
}
#endif
}
void
SIPCall::updateAllMediaStreams(const std::vector<MediaAttribute>& mediaAttrList)
{
JAMI_DBG("[call:%s] New local medias", getCallId().c_str());
unsigned idx = 0;
for (auto const& newMediaAttr : mediaAttrList) {
JAMI_DBG("[call:%s] Media @%u: %s",
getCallId().c_str(),
idx++,
newMediaAttr.toString(true).c_str());
}
JAMI_DBG("[call:%s] Updating local media streams", getCallId().c_str());
for (auto const& newAttr : mediaAttrList) {
auto streamIdx = findRtpStreamIndex(newAttr.label_);
if (streamIdx == rtpStreams_.size()) {
// Media does not exist, add a new one.
addMediaStream(newAttr);
auto& stream = rtpStreams_.back();
createRtpSession(stream);
JAMI_DBG("[call:%s] Added a new media stream [%s] @ index %lu",
getCallId().c_str(),
stream.mediaAttribute_->label_.c_str(),
streamIdx);
} else {
updateMediaStream(newAttr, streamIdx);
}
}
}
bool
SIPCall::isReinviteRequired(const std::vector<MediaAttribute>& mediaAttrList)
{
if (mediaAttrList.size() != rtpStreams_.size())
return true;
for (auto const& newAttr : mediaAttrList) {
auto streamIdx = findRtpStreamIndex(newAttr.label_);
if (streamIdx == rtpStreams_.size()) {
// Always needs a reinvite when a new media is added.
return true;
}
#ifdef ENABLE_VIDEO
if (newAttr.type_ == MediaType::MEDIA_VIDEO) {
assert(rtpStreams_[streamIdx].mediaAttribute_);
// Changes in video attributes always trigger a re-invite.
return newAttr.muted_ != rtpStreams_[streamIdx].mediaAttribute_->muted_;
}
#endif
}
return false;
}
bool
SIPCall::requestMediaChange(const std::vector<DRing::MediaMap>& mediaList)
{
auto mediaAttrList = MediaAttribute::buildMediaAttributesList(mediaList, isSrtpEnabled());
// If the peer does not support multi-stream and the size of the new
// media list is different from the current media list, the media
// change request will be ignored.
if (not peerSupportMultiStream_ and rtpStreams_.size() != mediaAttrList.size()) {
JAMI_WARN("[call:%s] Peer does not support multi-stream. Media change request ignored",
getCallId().c_str());
return false;
}
JAMI_DBG("[call:%s] Requesting media change. List of new media:", getCallId().c_str());
unsigned idx = 0;
for (auto const& newMediaAttr : mediaAttrList) {
JAMI_DBG("[call:%s] Media @%u: %s",
getCallId().c_str(),
idx++,
newMediaAttr.toString(true).c_str());
}
auto needReinvite = isReinviteRequired(mediaAttrList);
updateAllMediaStreams(mediaAttrList);
if (needReinvite) {
JAMI_DBG("[call:%s] Media change requires a new negotiation (re-invite)",
getCallId().c_str());
requestReinvite();
}
return true;
}
std::vector<MediaAttribute>
SIPCall::getMediaAttributeList() const
{
std::vector<MediaAttribute> mediaList;
mediaList.reserve(rtpStreams_.size());
for (auto const& stream : rtpStreams_) {
mediaList.emplace_back(*stream.mediaAttribute_);
}
return mediaList;
}
/// \brief Prepare media transport and launch media stream based on negotiated SDP
///
/// This method has to be called by link (ie SipVoIpLink) when SDP is negotiated and
/// media streams structures are knows.
/// In case of ICE transport used, the medias streams are launched asynchronously when
/// the transport is negotiated.
void
SIPCall::onMediaNegotiationComplete()
{ JAMI_WARN("[call:%s] Media negotiation complete", getCallId().c_str());
// Main call (no subcalls) must wait for ICE now, the rest of code needs to access
// to a negotiated transport.
runOnMainThread([w = weak()] {
if (auto this_ = w.lock()) {
std::lock_guard<std::recursive_mutex> lk {this_->callMutex_};
JAMI_WARN("[call:%s] media changed", this_->getCallId().c_str());
// The call is already ended, so we don't need to restart medias
if (not this_->inviteSession_
or this_->inviteSession_->state == PJSIP_INV_STATE_DISCONNECTED
or not this_->sdp_) {
return;
}
bool hasIce = this_->isIceEnabled();
if (hasIce) {
// If ICE is not used, start medias now
auto rem_ice_attrs = this_->sdp_->getIceAttributes();
hasIce = not rem_ice_attrs.ufrag.empty() and not rem_ice_attrs.pwd.empty();
}
if (hasIce) {
if (not this_->isSubcall()) {
// Start ICE checks. Media will be started once ICE checks complete.
this_->startIceMedia();
}
} else {
// No ICE, start media now.
JAMI_WARN("[call:%s] ICE media disabled, using default media ports",
this_->getCallId().c_str());
// Update the negotiated media.
this_->updateNegotiatedMedia();
// Start the media.
this_->stopAllMedia();
this_->startAllMedia();
}
}
});
}
void
SIPCall::startIceMedia()
{
JAMI_DBG("[call:%s] Starting ICE", getCallId().c_str());
auto mediaTransport = getIceMedia();
if (not mediaTransport or mediaTransport->isFailed()) {
JAMI_ERR("[call:%s] Media ICE init failed", getCallId().c_str());
onFailure(EIO);
return;
}
if (mediaTransport->isStarted()) {
// NOTE: for incoming calls, the ice is already there and running
if (mediaTransport->isRunning())
onIceNegoSucceed();
return;
}
if (not mediaTransport->isInitialized()) {
// In this case, onInitDone will occurs after the startIceMedia
waitForIceInit_ = true;
return;
}
// Start transport on SDP data and wait for negotiation
if (!sdp_)
return;
auto rem_ice_attrs = sdp_->getIceAttributes(); if (rem_ice_attrs.ufrag.empty() or rem_ice_attrs.pwd.empty()) {
JAMI_ERR("[call:%s] Media ICE attributes empty", getCallId().c_str());
onFailure(EIO);
return;
}
if (not mediaTransport->startIce(rem_ice_attrs, getAllRemoteCandidates(*mediaTransport))) {
JAMI_ERR("[call:%s] Media ICE start failed", getCallId().c_str());
onFailure(EIO);
}
}
void
SIPCall::onIceNegoSucceed()
{
JAMI_DBG("[call:%s] ICE negotiation succeeded", getCallId().c_str());
// Check if the call is already ended, so we don't need to restart medias
// This is typically the case in a multi-device context where one device
// can stop a call. So do not start medias
if (not inviteSession_ or inviteSession_->state == PJSIP_INV_STATE_DISCONNECTED or not sdp_) {
JAMI_ERR("[call:%s] ICE negotiation succeeded, but call is in invalid state",
getCallId().c_str());
return;
}
// Update the negotiated media.
updateNegotiatedMedia();
// Nego succeed: move to the new media transport
stopAllMedia();
startAllMedia();
}
pj_status_t
SIPCall::onReceiveReinvite(const pjmedia_sdp_session* offer, pjsip_rx_data* rdata)
{
JAMI_DBG("[call:%s] Received a re-invite", getCallId().c_str());
pj_status_t res = PJ_SUCCESS;
if (not sdp_) {
JAMI_ERR("SDP session is invalid");
return res;
}
sdp_->clearIce();
auto acc = getSIPAccount();
if (not acc) {
JAMI_ERR("No account detected");
return res;
}
Sdp::printSession(offer, "Remote session (media change request)", SdpDirection::OFFER);
sdp_->setReceivedOffer(offer);
auto const& mediaAttrList = Sdp::getMediaAttributeListFromSdp(offer);
if (mediaAttrList.empty()) {
JAMI_WARN("[call:%s] Media list is empty, ignoring", getCallId().c_str());
return res;
}
if (upnp_) {
openPortsUPnP();
}
pjsip_tx_data* tdata = nullptr;
if (pjsip_inv_initial_answer(inviteSession_.get(), rdata, PJSIP_SC_TRYING, NULL, NULL, &tdata)
!= PJ_SUCCESS) { JAMI_ERR("Could not create answer TRYING");
return res;
}
// Report the change request.
auto const& remoteMediaList = MediaAttribute::mediaAttributesToMediaMaps(mediaAttrList);
// TODO_MC. Validate this assessment.
// Report re-invites only if the number of media changed, otherwise answer
// using the current local attributes.
if (acc->isMultiStreamEnabled() and remoteMediaList.size() != rtpStreams_.size()) {
Manager::instance().mediaChangeRequested(getCallId(), getAccountId(), remoteMediaList);
} else {
auto localMediaList = MediaAttribute::mediaAttributesToMediaMaps(getMediaAttributeList());
answerMediaChangeRequest(localMediaList);
}
return res;
}
int
SIPCall::onReceiveOffer(const pjmedia_sdp_session* offer, const pjsip_rx_data* rdata)
{
if (!sdp_)
return !PJ_SUCCESS;
sdp_->clearIce();
auto acc = getSIPAccount();
if (!acc) {
JAMI_ERR("No account detected");
return !PJ_SUCCESS;
}
JAMI_DBG("[call:%s] Received a new offer (re-invite)", getCallId().c_str());
sdp_->setReceivedOffer(offer);
// Use current media list.
sdp_->processIncomingOffer(getMediaAttributeList());
if (isIceEnabled() and offer != nullptr) {
setupIceResponse();
}
sdp_->startNegotiation();
pjsip_tx_data* tdata = nullptr;
if (pjsip_inv_initial_answer(inviteSession_.get(),
const_cast<pjsip_rx_data*>(rdata),
PJSIP_SC_OK,
NULL,
NULL,
&tdata)
!= PJ_SUCCESS) {
JAMI_ERR("Could not create initial answer OK");
return !PJ_SUCCESS;
}
// Add user-agent header
sip_utils::addUserAgentHeader(getSIPAccount()->getUserAgentName(), tdata);
if (pjsip_inv_answer(inviteSession_.get(), PJSIP_SC_OK, NULL, sdp_->getLocalSdpSession(), &tdata)
!= PJ_SUCCESS) {
JAMI_ERR("Could not create answer OK");
return !PJ_SUCCESS;
}
// ContactStr must stay in scope as long as tdata
sip_utils::addContactHeader(getSIPAccount()->getContactHeader(getTransport()->get()), tdata);
if (pjsip_inv_send_msg(inviteSession_.get(), tdata) != PJ_SUCCESS) { JAMI_ERR("Could not send msg OK");
return !PJ_SUCCESS;
}
if (upnp_) {
openPortsUPnP();
}
return PJ_SUCCESS;
}
void
SIPCall::onReceiveOfferIn200OK(const pjmedia_sdp_session* offer)
{
if (not rtpStreams_.empty()) {
JAMI_ERR("[call:%s] Unexpected offer in '200 OK' answer", getCallId().c_str());
return;
}
auto acc = getSIPAccount();
if (not acc) {
JAMI_ERR("No account detected");
return;
}
if (not sdp_) {
JAMI_ERR("invalid SDP session");
return;
}
JAMI_DBG("[call:%s] Received an offer in '200 OK' answer", getCallId().c_str());
auto mediaList = Sdp::getMediaAttributeListFromSdp(offer);
// If this method is called, it means we are expecting an offer
// in the 200OK answer.
if (mediaList.empty()) {
JAMI_WARN("[call:%s] Remote media list is empty, ignoring", getCallId().c_str());
return;
}
Sdp::printSession(offer, "Remote session (offer in 200 OK answer)", SdpDirection::OFFER);
sdp_->clearIce();
sdp_->setReceivedOffer(offer);
// If we send an empty offer, video will be accepted only if locally
// enabled by the user.
for (auto& mediaAttr : mediaList) {
if (mediaAttr.type_ == MediaType::MEDIA_VIDEO and not acc->isVideoEnabled()) {
mediaAttr.enabled_ = false;
}
}
initMediaStreams(mediaList);
sdp_->processIncomingOffer(mediaList);
if (upnp_) {
openPortsUPnP();
}
if (isIceEnabled()) {
setupIceResponse();
}
sdp_->startNegotiation();
if (pjsip_inv_set_sdp_answer(inviteSession_.get(), sdp_->getLocalSdpSession()) != PJ_SUCCESS) {
JAMI_ERR("[call:%s] Could not start media negotiation for a re-invite request",
getCallId().c_str());
}}
void
SIPCall::openPortsUPnP()
{
if (not sdp_) {
JAMI_ERR("[call:%s] Current SDP instance is invalid", getCallId().c_str());
return;
}
/**
* Try to open the desired ports with UPnP,
* if they are used, use the alternative port and update the SDP session with the newly
* chosen port(s)
*
* TODO:
* No need to request mappings for specfic port numbers. Set the port to '0' to
* request the first available port (faster and more likely to succeed).
*/
JAMI_DBG("[call:%s] opening ports via UPNP for SDP session", getCallId().c_str());
// RTP port.
upnp_->reserveMapping(sdp_->getLocalAudioPort(), upnp::PortType::UDP);
// RTCP port.
upnp_->reserveMapping(sdp_->getLocalAudioControlPort(), upnp::PortType::UDP);
#ifdef ENABLE_VIDEO
// RTP port.
upnp_->reserveMapping(sdp_->getLocalVideoPort(), upnp::PortType::UDP);
// RTCP port.
upnp_->reserveMapping(sdp_->getLocalVideoControlPort(), upnp::PortType::UDP);
#endif
}
std::map<std::string, std::string>
SIPCall::getDetails() const
{
auto acc = getSIPAccount();
if (!acc) {
JAMI_ERR("No account detected");
return {};
}
auto details = Call::getDetails();
details.emplace(DRing::Call::Details::PEER_HOLDING, peerHolding_ ? TRUE_STR : FALSE_STR);
#ifdef ENABLE_VIDEO
for (auto const& stream : rtpStreams_) {
if (stream.mediaAttribute_->type_ != MediaType::MEDIA_VIDEO)
continue;
details.emplace(DRing::Call::Details::VIDEO_SOURCE, stream.mediaAttribute_->sourceUri_);
if (auto const& rtpSession = stream.rtpSession_) {
if (auto codec = rtpSession->getCodec())
details.emplace(DRing::Call::Details::VIDEO_CODEC, codec->systemCodecInfo.name);
else
details.emplace(DRing::Call::Details::VIDEO_CODEC, "");
}
}
#endif
#if HAVE_RINGNS
if (not peerRegisteredName_.empty())
details.emplace(DRing::Call::Details::REGISTERED_NAME, peerRegisteredName_);
#endif
#ifdef ENABLE_CLIENT_CERT
std::lock_guard<std::recursive_mutex> lk {callMutex_};
if (transport_ and transport_->isSecure()) {
const auto& tlsInfos = transport_->getTlsInfos(); if (tlsInfos.cipher != PJ_TLS_UNKNOWN_CIPHER) {
const auto& cipher = pj_ssl_cipher_name(tlsInfos.cipher);
details.emplace(DRing::TlsTransport::TLS_CIPHER, cipher ? cipher : "");
} else {
details.emplace(DRing::TlsTransport::TLS_CIPHER, "");
}
if (tlsInfos.peerCert) {
details.emplace(DRing::TlsTransport::TLS_PEER_CERT, tlsInfos.peerCert->toString());
auto ca = tlsInfos.peerCert->issuer;
unsigned n = 0;
while (ca) {
std::ostringstream name_str;
name_str << DRing::TlsTransport::TLS_PEER_CA_ << n++;
details.emplace(name_str.str(), ca->toString());
ca = ca->issuer;
}
details.emplace(DRing::TlsTransport::TLS_PEER_CA_NUM, std::to_string(n));
} else {
details.emplace(DRing::TlsTransport::TLS_PEER_CERT, "");
details.emplace(DRing::TlsTransport::TLS_PEER_CA_NUM, "");
}
}
#endif
return details;
}
void
SIPCall::enterConference(const std::string& confId)
{
#ifdef ENABLE_VIDEO
auto conf = Manager::instance().getConferenceFromID(confId);
if (conf == nullptr) {
JAMI_ERR("Unknown conference [%s]", confId.c_str());
return;
}
auto videoRtp = getVideoRtp();
if (not videoRtp) {
// In conference, we need to have a video RTP session even
// if it's an audio only call
videoRtp = addDummyVideoRtpSession();
if (not videoRtp) {
throw std::runtime_error("Failed to create dummy RTP video session");
}
}
videoRtp->enterConference(conf.get());
#endif
#ifdef ENABLE_PLUGIN
clearCallAVStreams();
#endif
}
void
SIPCall::exitConference()
{
auto const& audioRtp = getAudioRtp();
if (audioRtp && !isCaptureDeviceMuted(MediaType::MEDIA_AUDIO)) {
auto& rbPool = Manager::instance().getRingBufferPool();
rbPool.bindCallID(getCallId(), RingBufferPool::DEFAULT_ID);
rbPool.flush(RingBufferPool::DEFAULT_ID);
}
#ifdef ENABLE_VIDEO
auto const& videoRtp = getVideoRtp();
if (videoRtp)
videoRtp->exitConference();
#endif
#ifdef ENABLE_PLUGIN
createCallAVStreams();#endif
}
std::shared_ptr<Observable<std::shared_ptr<MediaFrame>>>
SIPCall::getReceiveVideoFrameActiveWriter()
{
#ifdef ENABLE_VIDEO
auto videoRtp = getVideoRtp();
if (videoRtp)
return videoRtp->getReceiveVideoFrameActiveWriter();
#endif
return {};
}
std::shared_ptr<video::VideoRtpSession>
SIPCall::addDummyVideoRtpSession()
{
#ifdef ENABLE_VIDEO
MediaAttribute mediaAttr(MediaType::MEDIA_VIDEO, true, true, false, "", "dummy video session");
addMediaStream(mediaAttr);
auto& stream = rtpStreams_.back();
createRtpSession(stream);
if (stream.rtpSession_) {
return std::dynamic_pointer_cast<video::VideoRtpSession>(stream.rtpSession_);
}
#endif
return {};
}
void
SIPCall::createSinks(const ConfInfo& infos)
{
#ifdef ENABLE_VIDEO
if (!hasVideo())
return;
std::lock_guard<std::mutex> lk(sinksMtx_);
auto videoRtp = getVideoRtp();
auto& videoReceive = videoRtp->getVideoReceive();
if (!videoReceive)
return;
auto id = getConfId().empty() ? getCallId() : getConfId();
Manager::instance().createSinkClients(id,
infos,
std::static_pointer_cast<video::VideoGenerator>(
videoReceive),
callSinksMap_);
#endif
}
std::shared_ptr<AudioRtpSession>
SIPCall::getAudioRtp() const
{
// For the moment, the clients support only one audio stream, so we
// return the first audio stream.
for (auto const& stream : rtpStreams_) {
auto rtp = stream.rtpSession_;
if (rtp->getMediaType() == MediaType::MEDIA_AUDIO) {
return std::dynamic_pointer_cast<AudioRtpSession>(rtp);
}
}
return nullptr;
}
#ifdef ENABLE_VIDEO
std::shared_ptr<video::VideoRtpSession>
SIPCall::getVideoRtp() const{
for (auto const& stream : rtpStreams_) {
auto rtp = stream.rtpSession_;
if (rtp->getMediaType() == MediaType::MEDIA_VIDEO) {
return std::dynamic_pointer_cast<video::VideoRtpSession>(rtp);
}
}
return nullptr;
}
#endif
std::vector<std::shared_ptr<RtpSession>>
SIPCall::getRtpSessionList() const
{
std::vector<std::shared_ptr<RtpSession>> rtpList;
rtpList.reserve(rtpStreams_.size());
for (auto const& stream : rtpStreams_) {
rtpList.emplace_back(stream.rtpSession_);
}
return rtpList;
}
void
SIPCall::monitor() const
{
if (isSubcall())
return;
auto acc = getSIPAccount();
if (!acc) {
JAMI_ERR("No account detected");
return;
}
JAMI_DBG("- Call %s with %s:", getCallId().c_str(), getPeerNumber().c_str());
JAMI_DBG("\t- Duration: %s", dht::print_duration(getCallDuration()).c_str());
for (const auto& stream : rtpStreams_)
JAMI_DBG("\t- Media: %s", stream.mediaAttribute_->toString(true).c_str());
#ifdef ENABLE_VIDEO
if (auto codec = getVideoCodec())
JAMI_DBG("\t- Video codec: %s", codec->systemCodecInfo.name.c_str());
#endif
if (auto transport = getIceMedia()) {
JAMI_DBG("\t- Medias: %s", transport->link().c_str());
}
}
bool
SIPCall::toggleRecording()
{
pendingRecord_ = true;
if (not readyToRecord_)
return true;
// add streams to recorder before starting the record
if (not Call::isRecording()) {
updateRecState(true);
auto account = getSIPAccount();
if (!account) {
JAMI_ERR("No account detected");
return false;
}
auto title = fmt::format("Conversation at %TIMESTAMP between {} and {}",
account->getUserUri(),
peerUri_);
recorder_->setMetadata(title, ""); // use default description
auto const& audioRtp = getAudioRtp();
if (audioRtp)
audioRtp->initRecorder(recorder_);
#ifdef ENABLE_VIDEO
if (hasVideo()) {
auto const& videoRtp = getVideoRtp(); if (videoRtp)
videoRtp->initRecorder(recorder_);
}
#endif
} else {
updateRecState(false);
deinitRecorder();
}
pendingRecord_ = false;
return Call::toggleRecording();
}
void
SIPCall::deinitRecorder()
{
if (Call::isRecording()) {
auto const& audioRtp = getAudioRtp();
if (audioRtp)
audioRtp->deinitRecorder(recorder_);
#ifdef ENABLE_VIDEO
if (hasVideo()) {
auto const& videoRtp = getVideoRtp();
if (videoRtp)
videoRtp->deinitRecorder(recorder_);
}
#endif
}
}
void
SIPCall::InvSessionDeleter::operator()(pjsip_inv_session* inv) const noexcept
{
// prevent this from getting accessed in callbacks
// JAMI_WARN: this is not thread-safe!
if (!inv)
return;
inv->mod_data[Manager::instance().sipVoIPLink().getModId()] = nullptr;
// NOTE: the counter is incremented by sipvoiplink (transaction_request_cb)
pjsip_inv_dec_ref(inv);
}
void
SIPCall::createIceMediaTransport()
{
auto& iceTransportFactory = Manager::instance().getIceTransportFactory();
std::lock_guard<std::mutex> lk(transportMtx_);
resetTransport(std::move(mediaTransport_));
mediaTransport_ = iceTransportFactory.createTransport(getCallId().c_str());
if (mediaTransport_) {
JAMI_DBG("[call:%s] Successfully created media ICE transport [ice:%p]",
getCallId().c_str(),
mediaTransport_.get());
} else {
JAMI_ERR("[call:%s] Failed to create media ICE transport", getCallId().c_str());
}
}
bool
SIPCall::initIceMediaTransport(bool master, std::optional<IceTransportOptions> options)
{
auto acc = getSIPAccount();
if (!acc) {
JAMI_ERR("No account detected");
return false;
}
JAMI_DBG("[call:%s] Init media ICE transport", getCallId().c_str());
auto mediaTransport = getIceMedia();
if (not mediaTransport) {
JAMI_ERR("[call:%s] Failed to create media ICE transport", getCallId().c_str()); return false;
}
auto iceOptions = options == std::nullopt ? acc->getIceOptions() : *options;
auto optOnInitDone = std::move(iceOptions.onInitDone);
auto optOnNegoDone = std::move(iceOptions.onNegoDone);
iceOptions.onInitDone = [w = weak(), cb = std::move(optOnInitDone)](bool ok) {
runOnMainThread([w = std::move(w), cb = std::move(cb), ok] {
auto call = w.lock();
if (cb)
cb(ok);
if (!ok or !call or !call->waitForIceInit_.exchange(false))
return;
std::lock_guard<std::recursive_mutex> lk {call->callMutex_};
auto rem_ice_attrs = call->sdp_->getIceAttributes();
// Init done but no remote_ice_attributes, the ice->start will be triggered later
if (rem_ice_attrs.ufrag.empty() or rem_ice_attrs.pwd.empty())
return;
call->startIceMedia();
});
};
iceOptions.onNegoDone = [w = weak(), cb = std::move(optOnNegoDone)](bool ok) {
runOnMainThread([w = std::move(w), cb = std::move(cb), ok] {
if (cb)
cb(ok);
if (auto call = w.lock()) {
// The ICE is related to subcalls, but medias are handled by parent call
std::lock_guard<std::recursive_mutex> lk {call->callMutex_};
call = call->isSubcall() ? std::dynamic_pointer_cast<SIPCall>(call->parent_) : call;
if (!ok) {
JAMI_ERR("[call:%s] Media ICE negotiation failed", call->getCallId().c_str());
call->onFailure(EIO);
return;
}
call->onIceNegoSucceed();
}
});
};
iceOptions.master = master;
iceOptions.streamsCount = static_cast<unsigned>(rtpStreams_.size());
// Each RTP stream requires a pair of ICE components (RTP + RTCP).
iceOptions.compCountPerStream = ICE_COMP_COUNT_PER_STREAM;
// Init ICE.
mediaTransport->initIceInstance(iceOptions);
return true;
}
std::vector<std::string>
SIPCall::getLocalIceCandidates(unsigned compId) const
{
std::lock_guard<std::mutex> lk(transportMtx_);
if (not mediaTransport_) {
JAMI_WARN("[call:%s] no media ICE transport", getCallId().c_str());
return {};
}
return mediaTransport_->getLocalCandidates(compId);
}
void
SIPCall::resetTransport(std::shared_ptr<IceTransport>&& transport)
{
// Move the transport to another thread and destroy it there if possible
if (transport) {
dht::ThreadPool::io().run(
[transport = std::move(transport)]() mutable { transport.reset(); }); }
}
void
SIPCall::merge(Call& call)
{
JAMI_DBG("[call:%s] merge subcall %s", getCallId().c_str(), call.getCallId().c_str());
// This static cast is safe as this method is private and overload Call::merge
auto& subcall = static_cast<SIPCall&>(call);
std::lock(callMutex_, subcall.callMutex_);
std::lock_guard<std::recursive_mutex> lk1 {callMutex_, std::adopt_lock};
std::lock_guard<std::recursive_mutex> lk2 {subcall.callMutex_, std::adopt_lock};
inviteSession_ = std::move(subcall.inviteSession_);
if (inviteSession_)
inviteSession_->mod_data[Manager::instance().sipVoIPLink().getModId()] = this;
setTransport(std::move(subcall.transport_));
sdp_ = std::move(subcall.sdp_);
peerHolding_ = subcall.peerHolding_;
upnp_ = std::move(subcall.upnp_);
std::copy_n(subcall.contactBuffer_, PJSIP_MAX_URL_SIZE, contactBuffer_);
pj_strcpy(&contactHeader_, &subcall.contactHeader_);
localAudioPort_ = subcall.localAudioPort_;
localVideoPort_ = subcall.localVideoPort_;
peerUserAgent_ = subcall.peerUserAgent_;
peerSupportMultiStream_ = subcall.peerSupportMultiStream_;
Call::merge(subcall);
if (isIceEnabled())
startIceMedia();
}
bool
SIPCall::remoteHasValidIceAttributes()
{
if (not sdp_) {
throw std::runtime_error("Must have a valid SDP Session");
}
auto rem_ice_attrs = sdp_->getIceAttributes();
if (rem_ice_attrs.ufrag.empty()) {
JAMI_WARN("[call:%s] Missing ICE username fragment attribute in remote SDP",
getCallId().c_str());
return false;
}
if (rem_ice_attrs.pwd.empty()) {
JAMI_WARN("[call:%s] Missing ICE password attribute in remote SDP", getCallId().c_str());
return false;
}
return true;
}
void
SIPCall::setIceMedia(std::shared_ptr<IceTransport> ice)
{
JAMI_DBG("[call:%s] Setting ICE session [%p]", getCallId().c_str(), ice.get());
std::lock_guard<std::mutex> lk(transportMtx_);
if (not isSubcall()) {
JAMI_ERR("[call:%s] The call is expected to be a sub-call", getCallId().c_str());
}
mediaTransport_ = std::move(ice);
}
void
SIPCall::setupIceResponse()
{ JAMI_DBG("[call:%s] Setup ICE response", getCallId().c_str());
auto account = getSIPAccount();
if (not account) {
JAMI_ERR("No account detected");
}
if (not remoteHasValidIceAttributes()) {
// If ICE attributes are not present, skip the ICE initialization
// step (most likely ICE is not used).
JAMI_ERR("[call:%s] no ICE data in remote SDP", getCallId().c_str());
return;
}
auto opt = account->getIceOptions();
// Try to use the discovered public address. If not available,
// fallback on local address.
opt.accountPublicAddr = account->getPublishedIpAddress();
if (opt.accountLocalAddr) {
opt.accountLocalAddr = ip_utils::getInterfaceAddr(account->getLocalInterface(),
opt.accountPublicAddr.getFamily());
} else {
// Just set the local address for both, most likely the account is not
// registered.
opt.accountLocalAddr = ip_utils::getInterfaceAddr(account->getLocalInterface(), AF_INET);
opt.accountPublicAddr = opt.accountLocalAddr;
}
if (not opt.accountLocalAddr) {
JAMI_ERR("[call:%s] No local address, ICE can't be initialized", getCallId().c_str());
onFailure(EIO);
return;
}
createIceMediaTransport();
if (not initIceMediaTransport(false, opt)) {
JAMI_ERR("[call:%s] ICE initialization failed", getCallId().c_str());
// Fatal condition
// TODO: what's SIP rfc says about that?
// (same question in startIceMedia)
onFailure(EIO);
return;
}
// WARNING: This call blocks! (need ice init done)
addLocalIceAttributes();
}
bool
SIPCall::isIceRunning() const
{
std::lock_guard<std::mutex> lk(transportMtx_);
return mediaTransport_ and mediaTransport_->isRunning();
}
std::unique_ptr<IceSocket>
SIPCall::newIceSocket(unsigned compId)
{
return std::unique_ptr<IceSocket> {new IceSocket(mediaTransport_, compId)};
}
void
SIPCall::rtpSetupSuccess(MediaType type, bool isRemote)
{
std::lock_guard<std::mutex> lk {setupSuccessMutex_};
if (type == MEDIA_AUDIO) {
if (isRemote)
mediaReady_.at("a:remote") = true;
else
mediaReady_.at("a:local") = true; } else {
if (isRemote)
mediaReady_.at("v:remote") = true;
else
mediaReady_.at("v:local") = true;
}
if (mediaReady_.at("a:local") and mediaReady_.at("a:remote") and mediaReady_.at("v:remote")) {
if (Manager::instance().videoPreferences.getRecordPreview() or mediaReady_.at("v:local"))
readyToRecord_ = true;
}
if (pendingRecord_ && readyToRecord_)
toggleRecording();
}
void
SIPCall::peerRecording(bool state)
{
const std::string& id = getConfId().empty() ? getCallId() : getConfId();
if (state) {
JAMI_WARN("Peer is recording");
emitSignal<DRing::CallSignal::RemoteRecordingChanged>(id, getPeerNumber(), true);
} else {
JAMI_WARN("Peer stopped recording");
emitSignal<DRing::CallSignal::RemoteRecordingChanged>(id, getPeerNumber(), false);
}
peerRecording_ = state;
}
void
SIPCall::peerMuted(bool muted)
{
if (muted) {
JAMI_WARN("Peer muted");
} else {
JAMI_WARN("Peer un-muted");
}
peerMuted_ = muted;
if (auto conf = Manager::instance().getConferenceFromID(getConfId())) {
conf->updateMuted();
}
}
void
SIPCall::resetMediaReady()
{
for (auto& m : mediaReady_)
m.second = false;
}
} // namespace jami