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sipcall.cpp
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checkAudio must be called to close audio threads when a call is closed. Change-Id: Iccfdccee9bd9dab2226a2092cf62e9ca225cf2ef
checkAudio must be called to close audio threads when a call is closed. Change-Id: Iccfdccee9bd9dab2226a2092cf62e9ca225cf2ef
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sipcall.cpp 40.68 KiB
/*
* Copyright (C) 2004-2018 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>
*
* 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" // for SIPAccount::ACCOUNT_TYPE
#include "sipaccountbase.h"
#include "sipvoiplink.h"
#include "sdes_negotiator.h"
#include "logger.h" // for _debug
#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 "dring/call_const.h"
#include "dring/media_const.h"
#include "client/ring_signal.h"
#include "ice_transport.h"
#ifdef RING_VIDEO
#include "client/videomanager.h"
#include "video/video_rtp_session.h"
#include "dring/videomanager_interface.h"
#include <chrono>
#endif
#include "errno.h"
#include <opendht/crypto.h>
namespace ring {
using sip_utils::CONST_PJ_STR;
#ifdef RING_VIDEO
static DeviceParams
getVideoSettings()
{
const auto& videomon = ring::getVideoDeviceMonitor();
return videomon.getDeviceParams(videomon.getDefaultDevice());
}
#endif
static constexpr int DEFAULT_ICE_INIT_TIMEOUT {35}; // seconds
static constexpr int DEFAULT_ICE_NEGO_TIMEOUT {60}; // seconds
// SDP media Ids
static constexpr int SDP_AUDIO_MEDIA_ID {0};
static constexpr int SDP_VIDEO_MEDIA_ID {1};
// ICE components Id used on SIP
static constexpr int ICE_AUDIO_RTP_COMPID {0};
static constexpr int ICE_AUDIO_RTCP_COMPID {1};
static constexpr int ICE_VIDEO_RTP_COMPID {2};
static constexpr int ICE_VIDEO_RTCP_COMPID {3};
const char* const SIPCall::LINK_TYPE = SIPAccount::ACCOUNT_TYPE;
SIPCall::SIPCall(SIPAccountBase& account, const std::string& id, Call::CallType type,
const std::map<std::string, std::string>& details)
: Call(account, id, type, details)
, avformatrtp_(new AudioRtpSession(id))
#ifdef RING_VIDEO
// The ID is used to associate video streams to calls
, videortp_(new video::VideoRtpSession(id, getVideoSettings()))
, videoInput_(Manager::instance().getVideoManager().videoDeviceMonitor.getMRLForDefaultDevice())
#endif
, sdp_(new Sdp(id))
{
if (account.getUPnPActive())
upnp_.reset(new upnp::Controller());
setCallMediaLocal();
}
SIPCall::~SIPCall()
{
std::lock_guard<std::recursive_mutex> lk {callMutex_};
setTransport({});
inv.reset(); // prevents callback usage
}
SIPAccountBase&
SIPCall::getSIPAccount() const
{
return static_cast<SIPAccountBase&>(getAccount());
}
void
SIPCall::setCallMediaLocal()
{
if (localAudioPort_ == 0
#ifdef RING_VIDEO
|| localVideoPort_ == 0
#endif
)
generateMediaPorts();
}
void
SIPCall::generateMediaPorts()
{
auto& account = getSIPAccount();
// 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_->setLocalPublishedAudioPort(callLocalAudioPort);
#ifdef RING_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_->setLocalPublishedVideoPort(callLocalVideoPort);
#endif
}
void SIPCall::setContactHeader(pj_str_t *contact)
{
pj_strcpy(&contactHeader_, contact);
}
void
SIPCall::setTransport(const std::shared_ptr<SipTransport>& t)
{
if (isSecure() and t and not t->isSecure()) {
RING_ERR("Can't set unsecure transport to secure call.");
return;
}
const auto list_id = reinterpret_cast<uintptr_t>(this);
if (transport_)
transport_->removeStateListener(list_id);
transport_ = t;
if (transport_) {
setSecure(transport_->isSecure());
std::weak_ptr<SIPCall> wthis_ = std::static_pointer_cast<SIPCall>(shared_from_this());
// listen for transport destruction
transport_->addStateListener(list_id,
[wthis_] (pjsip_transport_state state, const pjsip_transport_state_info*) {
if (auto this_ = wthis_.lock()) {
// end the call if the SIP transport is shut down
if (not SipTransport::isAlive(this_->transport_, state) and this_->getConnectionState() != ConnectionState::DISCONNECTED) {
RING_WARN("[call:%s] Ending call because underlying SIP transport was closed",
this_->getCallId().c_str());
this_->onFailure(ECONNRESET);
}
}
});
}
}
/**
* Send a reINVITE inside an active dialog to modify its state
* Local SDP session should be modified before calling this method
*/
int
SIPCall::SIPSessionReinvite()
{
std::lock_guard<std::recursive_mutex> lk {callMutex_};
// Do nothing if no invitation processed yet
if (not inv or inv->invite_tsx)
return PJ_SUCCESS;
RING_DBG("[call:%s] Processing reINVITE (state=%s)", getCallId().c_str(),
pjsip_inv_state_name(inv->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 sdp_->createOffer(acc.getActiveAccountCodecInfoList(MEDIA_AUDIO),
acc.getActiveAccountCodecInfoList(acc.isVideoEnabled() and not isAudioOnly() ? MEDIA_VIDEO
: MEDIA_NONE), acc.getSrtpKeyExchange(),
getState() == CallState::HOLD))
return !PJ_SUCCESS;
if (initIceMediaTransport(true))
setupLocalSDPFromIce();
pjsip_tx_data* tdata;
auto local_sdp = sdp_->getLocalSdpSession();
auto result = pjsip_inv_reinvite(inv.get(), nullptr, local_sdp, &tdata);
if (result == PJ_SUCCESS) {
if (!tdata)
return PJ_SUCCESS;
result = pjsip_inv_send_msg(inv.get(), tdata);
if (result == PJ_SUCCESS)
return PJ_SUCCESS;
RING_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(inv.get(), &tdata);
} else
RING_ERR("[call:%s] Failed to create REINVITE msg (pjsip: %s)", getCallId().c_str(),
sip_utils::sip_strerror(result).c_str());
return !PJ_SUCCESS;
}
void
SIPCall::sendSIPInfo(const char *const body, const char *const subtype)
{
std::lock_guard<std::recursive_mutex> lk {callMutex_};
if (not inv or not inv->dlg)
throw VoipLinkException("Couldn't get invite dialog");
pj_str_t methodName = CONST_PJ_STR("INFO");
const pj_str_t type = CONST_PJ_STR("application");
pjsip_method method;
pjsip_method_init_np(&method, &methodName);
/* Create request message. */
pjsip_tx_data *tdata;
if (pjsip_dlg_create_request(inv->dlg, &method, -1, &tdata) != PJ_SUCCESS) {
RING_ERR("[call:%s] Could not create dialog", getCallId().c_str());
return;
}
/* Create "application/<subtype>" message body. */
pj_str_t content;
pj_cstr(&content, body);
pj_str_t pj_subtype;
pj_cstr(&pj_subtype, 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(inv->dlg, tdata, getSIPVoIPLink()->getModId(), NULL);
}
void
SIPCall::updateSDPFromSTUN()
{
RING_WARN("[call:%s] SIPCall::updateSDPFromSTUN() not implemented", getCallId().c_str());
}
void
SIPCall::terminateSipSession(int status)
{
RING_DBG("[call:%s] Terminate SIP session", getCallId().c_str()); std::lock_guard<std::recursive_mutex> lk {callMutex_};
if (inv and inv->state != PJSIP_INV_STATE_DISCONNECTED) {
pjsip_tx_data* tdata = nullptr;
auto ret = pjsip_inv_end_session(inv.get(), status, nullptr, &tdata);
if (ret == PJ_SUCCESS) {
if (tdata) {
auto contact = getSIPAccount().getContactHeader(transport_ ? transport_->get() : nullptr);
sip_utils::addContactHeader(&contact, tdata);
ret = pjsip_inv_send_msg(inv.get(), tdata);
if (ret != PJ_SUCCESS)
RING_ERR("[call:%s] failed to send terminate msg, SIP error (%s)",
getCallId().c_str(), sip_utils::sip_strerror(ret).c_str());
}
} else
RING_ERR("[call:%s] failed to terminate INVITE@%p, SIP error (%s)",
getCallId().c_str(), inv.get(), sip_utils::sip_strerror(ret).c_str());
}
inv.reset();
}
void
SIPCall::answer()
{
std::lock_guard<std::recursive_mutex> lk {callMutex_};
auto& account = getSIPAccount();
if (not inv)
throw VoipLinkException("[call:" + getCallId() + "] answer: no invite session for this call");
if (!inv->neg) {
RING_WARN("[call:%s] Negotiator is NULL, we've received an INVITE without an SDP",
getCallId().c_str());
pjmedia_sdp_session *dummy = 0;
getSIPVoIPLink()->createSDPOffer(inv.get(), &dummy);
if (account.isStunEnabled())
updateSDPFromSTUN();
}
pj_str_t contact(account.getContactHeader(transport_ ? transport_->get() : nullptr));
setContactHeader(&contact);
pjsip_tx_data *tdata;
if (!inv->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(inv.get(), PJSIP_SC_OK, NULL, !inv->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) {
RING_DBG("[call:%s] Answering with contact header: %.*s",
getCallId().c_str(), (int)contactHeader_.slen, contactHeader_.ptr);
sip_utils::addContactHeader(&contactHeader_, tdata);
}
if (pjsip_inv_send_msg(inv.get(), tdata) != PJ_SUCCESS) {
inv.reset();
throw std::runtime_error("Could not send invite request answer (200 OK)");
}
setState(CallState::ACTIVE, ConnectionState::CONNECTED);
}
void
SIPCall::hangup(int reason)
{
std::lock_guard<std::recursive_mutex> lk {callMutex_}; if (inv and inv->dlg) {
pjsip_route_hdr *route = inv->dlg->route_set.next;
while (route and route != &inv->dlg->route_set) {
char buf[1024];
int printed = pjsip_hdr_print_on(route, buf, sizeof(buf));
if (printed >= 0) {
buf[printed] = '\0';
RING_DBG("[call:%s] Route header %s", getCallId().c_str(), buf);
}
route = route->next;
}
const int status = reason ? reason :
inv->state <= PJSIP_INV_STATE_EARLY and inv->role != PJSIP_ROLE_UAC ?
PJSIP_SC_CALL_TSX_DOES_NOT_EXIST :
inv->state >= PJSIP_INV_STATE_DISCONNECTED ? PJSIP_SC_DECLINE : 0;
// Notify the peer
terminateSipSession(status);
}
// Stop all RTP streams
stopAllMedia();
setState(Call::ConnectionState::DISCONNECTED, reason);
removeCall();
}
void
SIPCall::refuse()
{
if (!isIncoming() or getConnectionState() == ConnectionState::CONNECTED or !inv)
return;
stopAllMedia();
// Notify the peer
terminateSipSession(PJSIP_SC_DECLINE);
setState(Call::ConnectionState::DISCONNECTED, ECONNABORTED);
removeCall();
}
static void
transfer_client_cb(pjsip_evsub *sub, pjsip_event *event)
{
auto link = getSIPVoIPLink();
if (not link) {
RING_ERR("no more VoIP link");
return;
}
auto mod_ua_id = link->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->inv)
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(pj_str_t *dst)
{
if (not inv or not inv->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(inv->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, getSIPVoIPLink()->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 (Recordable::isRecording())
stopRecording();
std::string toUri;
pj_str_t dst = { 0, 0 };
toUri = account.getToUri(to);
pj_cstr(&dst, toUri.c_str());
RING_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)
{
const auto toCall = Manager::instance().callFactory.getCall<SIPCall>(to);
if (!toCall)
return false;
if (not toCall->inv or not toCall->inv->dlg)
return false;
pjsip_dialog *target_dlg = toCall->inv->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(){
// If ICE is currently negotiating, we must wait before hold the call
if (isWaitingForIceAndMedia_) {
remainingRequest_ = Request::HoldingOn;
return false;
}
if (not setState(CallState::HOLD))
return false;
stopAllMedia();
if (getConnectionState() == ConnectionState::CONNECTED) {
if (SIPSessionReinvite() != PJ_SUCCESS) {
RING_WARN("[call:%s] Reinvite failed", getCallId().c_str());
return true;
}
}
isWaitingForIceAndMedia_ = true;
return true;
}
bool
SIPCall::offhold()
{
bool success = false;
// If ICE is currently negotiating, we must wait before unhold the call
if (isWaitingForIceAndMedia_) {
remainingRequest_ = Request::HoldingOff;
return false;
}
auto& account = getSIPAccount();
try {
if (account.isStunEnabled())
success = internalOffHold([&] { updateSDPFromSTUN(); });
else
success = internalOffHold([] {});
} catch (const SdpException &e) {
RING_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) {
RING_WARN("[call:%s] resuming hold", getCallId().c_str());
onhold();
return false;
}
}
return true;
}
void
SIPCall::switchInput(const std::string& resource)
{
#ifdef RING_VIDEO
videoInput_ = resource;
if (isWaitingForIceAndMedia_) {
remainingRequest_ = Request::SwitchInput;
} else {
if (SIPSessionReinvite() == PJ_SUCCESS) {
isWaitingForIceAndMedia_ = true;
}
}
#endif
}
void
SIPCall::peerHungup()
{
// Stop all RTP streams
stopAllMedia();
if (inv)
terminateSipSession(PJSIP_SC_NOT_FOUND);
else
RING_ERR("[call:%s] peerHungup: no invite session for this call", getCallId().c_str());
Call::peerHungup();
}
void
SIPCall::carryingDTMFdigits(char code)
{
int duration = Manager::instance().voipPreferences.getPulseLength();
char dtmf_body[1000];
// handle flash code
if (code == '!') {
snprintf(dtmf_body, sizeof dtmf_body - 1, "Signal=16\r\nDuration=%d\r\n", duration);
} else {
snprintf(dtmf_body, sizeof dtmf_body - 1, "Signal=%c\r\nDuration=%d\r\n", code, duration);
}
try {
sendSIPInfo(dtmf_body, "dtmf-relay");
} catch (const std::exception& e) {
RING_ERR("Error sending DTMF: %s", e.what());
}
}
void
SIPCall::sendTextMessage(const std::map<std::string, std::string>& messages,
const std::string& from)
{
//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 (inv) {
try {
im::sendSipMessage(inv.get(), messages);
} catch (...) {}
} else {
pendingOutMessages_.emplace_back(messages, from);
RING_ERR("[call:%s] sendTextMessage: no invite session for this call", getCallId().c_str());
}
}}
void
SIPCall::removeCall()
{
std::lock_guard<std::recursive_mutex> lk {callMutex_};
RING_WARN("[call:%s] removeCall()", getCallId().c_str());
Call::removeCall();
mediaTransport_.reset();
inv.reset();
setTransport({});
}
void
SIPCall::onFailure(signed cause)
{
setState(CallState::MERROR, ConnectionState::DISCONNECTED, cause);
runOnMainThread([w = std::weak_ptr<Call>(shared_from_this())] {
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 = std::weak_ptr<Call>(shared_from_this())] {
if (auto shared = w.lock()) {
auto& call = *shared;
Manager::instance().callBusy(call);
call.removeCall();
}
});
}
void
SIPCall::onClosed()
{
runOnMainThread([w = std::weak_ptr<Call>(shared_from_this())] {
if (auto shared = w.lock()) {
auto& call = *shared;
Manager::instance().peerHungupCall(call);
call.removeCall();
Manager::instance().checkAudio();
}
});
}
void
SIPCall::onAnswered()
{
RING_WARN("[call:%s] onAnswered()", getCallId().c_str());
if (getConnectionState() != ConnectionState::CONNECTED) {
setState(CallState::ACTIVE, ConnectionState::CONNECTED);
if (not isSubcall()) {
runOnMainThread([w = std::weak_ptr<Call>(shared_from_this())] {
if (auto shared = w.lock()) {
Manager::instance().peerAnsweredCall(*shared);
}
});
}
}}
void
SIPCall::onPeerRinging()
{
setState(ConnectionState::RINGING);
}
void
SIPCall::setupLocalSDPFromIce()
{
auto media_tr = getIceMediaTransport();
if (not media_tr) {
RING_WARN("[call:%s] no media ICE transport, SDP not changed", getCallId().c_str());
return;
}
// we need an initialized ICE to progress further
if (media_tr->waitForInitialization(DEFAULT_ICE_INIT_TIMEOUT) <= 0) {
RING_ERR("[call:%s] Medias' ICE init failed", getCallId().c_str());
return;
}
if (const auto& ip = getSIPAccount().getPublishedIpAddress()) {
for (unsigned compId = 1; compId <= media_tr->getComponentCount(); ++compId)
media_tr->registerPublicIP(compId, ip);
}
RING_WARN("[call:%s] fill SDP with ICE transport %p", getCallId().c_str(), media_tr);
sdp_->addIceAttributes(media_tr->getLocalAttributes());
// Add video and audio channels
sdp_->addIceCandidates(SDP_AUDIO_MEDIA_ID, media_tr->getLocalCandidates(ICE_AUDIO_RTP_COMPID));
sdp_->addIceCandidates(SDP_AUDIO_MEDIA_ID, media_tr->getLocalCandidates(ICE_AUDIO_RTCP_COMPID));
#ifdef RING_VIDEO
sdp_->addIceCandidates(SDP_VIDEO_MEDIA_ID, media_tr->getLocalCandidates(ICE_VIDEO_RTP_COMPID));
sdp_->addIceCandidates(SDP_VIDEO_MEDIA_ID, media_tr->getLocalCandidates(ICE_VIDEO_RTCP_COMPID));
#endif
}
std::vector<IceCandidate>
SIPCall::getAllRemoteCandidates()
{
std::vector<IceCandidate> rem_candidates;
auto media_tr = getIceMediaTransport();
auto addSDPCandidates = [&, this](unsigned sdpMediaId,
std::vector<IceCandidate>& out) {
IceCandidate cand;
for (auto& line : sdp_->getIceCandidates(sdpMediaId)) {
if (media_tr->getCandidateFromSDP(line, cand)) {
RING_DBG("[call:%s] add remote ICE candidate: %s", getCallId().c_str(), line.c_str());
out.emplace_back(cand);
}
}
};
addSDPCandidates(SDP_AUDIO_MEDIA_ID, rem_candidates);
#ifdef RING_VIDEO
addSDPCandidates(SDP_VIDEO_MEDIA_ID, rem_candidates);
#endif
return rem_candidates;
}
bool
SIPCall::useVideoCodec(const AccountVideoCodecInfo* codec) const
{
#ifdef RING_VIDEO if (videortp_->isSending())
return videortp_->useCodec(codec);
#endif
return false;
}
void
SIPCall::startAllMedia()
{
RING_WARN("[call:%s] startAllMedia()", getCallId().c_str());
if (isSecure() && not transport_->isSecure()) {
RING_ERR("[call:%s] Can't perform secure call over insecure SIP transport",
getCallId().c_str());
onFailure(EPROTONOSUPPORT);
return;
}
auto slots = sdp_->getMediaSlots();
unsigned ice_comp_id = 0;
bool peer_holding {true};
int slotN = -1;
for (const auto& slot : slots) {
++slotN;
const auto& local = slot.first;
const auto& remote = slot.second;
if (local.type != remote.type) {
RING_ERR("[call:%s] [SDP:slot#%u] Inconsistent media types between local and remote",
getCallId().c_str(), slotN);
continue;
}
RtpSession* rtp = local.type == MEDIA_AUDIO
? static_cast<RtpSession*>(avformatrtp_.get())
#ifdef RING_VIDEO
: static_cast<RtpSession*>(videortp_.get());
#else
: nullptr;
#endif
if (not rtp)
continue;
if (!local.codec) {
RING_WARN("[call:%s] [SDP:slot#%u] Missing local codec", getCallId().c_str(),
slotN);
continue;
}
if (!remote.codec) {
RING_WARN("[call:%s] [SDP:slot#%u] Missing remote codec", getCallId().c_str(),
slotN);
continue;
}
peer_holding &= remote.holding;
if (isSecure() && (not local.crypto || not remote.crypto)) {
RING_ERR("[call:%s] [SDP:slot#%u] Can't perform secure call over insecure RTP transport",
getCallId().c_str(), slotN);
continue;
}
auto new_mtu = transport_->getTlsMtu();
avformatrtp_->setMtu(new_mtu);
#ifdef RING_VIDEO
if (local.type == MEDIA_VIDEO)
videortp_->switchInput(videoInput_);
videortp_->setMtu(new_mtu);#endif
rtp->updateMedia(remote, local);
// 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()) {
rtp->start(newIceSocket(ice_comp_id + 0),
newIceSocket(ice_comp_id + 1));
ice_comp_id += 2;
} else
rtp->start(nullptr, nullptr);
}
switch (local.type) {
#ifdef RING_VIDEO
case MEDIA_VIDEO:
isVideoMuted_ = videoInput_.empty();
break;
#endif
case MEDIA_AUDIO:
isAudioMuted_ = not rtp->isSending();
break;
default: break;
}
}
if (not isSubcall() and peerHolding_ != peer_holding) {
peerHolding_ = peer_holding;
emitSignal<DRing::CallSignal::PeerHold>(getCallId(), peerHolding_);
}
// Media is restarted, we can process the last holding request.
isWaitingForIceAndMedia_ = false;
if (remainingRequest_ != Request::NoRequest) {
switch (remainingRequest_) {
case Request::HoldingOn:
onhold();
break;
case Request::HoldingOff:
offhold();
break;
case Request::SwitchInput:
SIPSessionReinvite();
break;
default:
break;
}
remainingRequest_ = Request::NoRequest;
}
}
void
SIPCall::restartMediaSender()
{
RING_DBG("[call:%s] restarting TX media streams", getCallId().c_str());
avformatrtp_->restartSender();
#ifdef RING_VIDEO
videortp_->restartSender();
#endif
}
void
SIPCall::stopAllMedia()
{
RING_DBG("[call:%s] stopping all medias", getCallId().c_str());
if (Recordable::isRecording())
Recordable::stopRecording(); // if call stops, finish recording
avformatrtp_->stop();
#ifdef RING_VIDEO videortp_->stop();
#endif
}
void
SIPCall::muteMedia(const std::string& mediaType, bool mute)
{
if (mediaType.compare(DRing::Media::Details::MEDIA_TYPE_VIDEO) == 0) {
#ifdef RING_VIDEO
if (mute == isVideoMuted_) return;
RING_WARN("[call:%s] video muting %s", getCallId().c_str(), bool_to_str(mute));
isVideoMuted_ = mute;
videoInput_ = isVideoMuted_ ? "" : Manager::instance().getVideoManager().videoDeviceMonitor.getMRLForDefaultDevice();
DRing::switchInput(getCallId(), videoInput_);
if (not isSubcall())
emitSignal<DRing::CallSignal::VideoMuted>(getCallId(), isVideoMuted_);
#endif
} else if (mediaType.compare(DRing::Media::Details::MEDIA_TYPE_AUDIO) == 0) {
if (mute == isAudioMuted_) return;
RING_WARN("[call:%s] audio muting %s", getCallId().c_str(), bool_to_str(mute));
isAudioMuted_ = mute;
avformatrtp_->setMuted(isAudioMuted_);
if (not isSubcall())
emitSignal<DRing::CallSignal::AudioMuted>(getCallId(), isAudioMuted_);
}
}
/// \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 asynchonously when
/// the transport is negotiated.
void
SIPCall::onMediaUpdate()
{
RING_WARN("[call:%s] medias changed", getCallId().c_str());
// If ICE is not used, start medias now
auto rem_ice_attrs = sdp_->getIceAttributes();
if (rem_ice_attrs.ufrag.empty() or rem_ice_attrs.pwd.empty()) {
RING_WARN("[call:%s] no remote ICE for medias", getCallId().c_str());
stopAllMedia();
startAllMedia();
return;
}
// Main call (no subcalls) must wait for ICE now, the rest of code needs to access
// to a negotiated transport.
if (not isSubcall())
waitForIceAndStartMedia();
}
void
SIPCall::waitForIceAndStartMedia()
{
// Initialization waiting task
auto weak_call = std::weak_ptr<SIPCall>(std::static_pointer_cast<SIPCall>(shared_from_this()));
Manager::instance().addTask([weak_call] {
// TODO: polling algo, to it by event
if (auto call = weak_call.lock()) {
auto ice = call->getIceMediaTransport();
if (ice->isFailed()) {
RING_ERR("[call:%s] Media ICE init failed", call->getCallId().c_str());
call->onFailure(EIO);
return false;
}
if (!ice->isInitialized()) return true;
// Start transport on SDP data and wait for negotiation
auto rem_ice_attrs = call->sdp_->getIceAttributes();
if (rem_ice_attrs.ufrag.empty() or rem_ice_attrs.pwd.empty()) {
RING_ERR("[call:%s] Media ICE attributes empty", call->getCallId().c_str());
call->onFailure(EIO);
return false;
}
if (not ice->start(rem_ice_attrs, call->getAllRemoteCandidates())) {
RING_ERR("[call:%s] Media ICE start failed", call->getCallId().c_str());
call->onFailure(EIO);
return false;
}
// Negotiation waiting task
Manager::instance().addTask([weak_call] {
if (auto call = weak_call.lock()) {
auto ice = call->getIceMediaTransport();
if (ice->isFailed()) {
RING_ERR("[call:%s] Media ICE negotiation failed", call->getCallId().c_str());
call->onFailure(EIO);
return false;
}
if (not ice->isRunning())
return true;
// Nego succeed: move to the new media transport
call->stopAllMedia();
if (call->tmpMediaTransport_)
call->mediaTransport_ = std::move(call->tmpMediaTransport_);
call->startAllMedia();
return false;
}
return false;
});
return false;
}
return false;
});
}
void
SIPCall::onReceiveOffer(const pjmedia_sdp_session* offer)
{
sdp_->clearIce();
auto& acc = getSIPAccount();
sdp_->receiveOffer(offer,
acc.getActiveAccountCodecInfoList(MEDIA_AUDIO),
acc.getActiveAccountCodecInfoList(acc.isVideoEnabled() ? MEDIA_VIDEO : MEDIA_NONE),
acc.getSrtpKeyExchange(),
getState() == CallState::HOLD);
setRemoteSdp(offer);
sdp_->startNegotiation();
pjsip_inv_set_sdp_answer(inv.get(), sdp_->getLocalSdpSession());
openPortsUPnP();
}
void
SIPCall::openPortsUPnP()
{
if (upnp_) {
/**
* 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: the initial ports were chosen from the list of available ports and were marked as used * the newly selected port should possibly be checked against the list of used ports and marked
* as used, the old port should be "released"
*/
RING_DBG("[call:%s] opening ports via UPNP for SDP session", getCallId().c_str());
uint16_t audio_port_used;
if (upnp_->addAnyMapping(sdp_->getLocalAudioPort(), upnp::PortType::UDP, true, &audio_port_used)) {
uint16_t control_port_used;
if (upnp_->addAnyMapping(sdp_->getLocalAudioControlPort(), upnp::PortType::UDP, true, &control_port_used)) {
sdp_->setLocalPublishedAudioPorts(audio_port_used, control_port_used);
}
}
#ifdef RING_VIDEO
uint16_t video_port_used;
if (upnp_->addAnyMapping(sdp_->getLocalVideoPort(), upnp::PortType::UDP, true, &video_port_used)) {
uint16_t control_port_used;
if (upnp_->addAnyMapping(sdp_->getLocalVideoControlPort(), upnp::PortType::UDP, true, &control_port_used)) {
sdp_->setLocalPublishedVideoPorts(video_port_used, control_port_used);
}
}
#endif
}
}
std::map<std::string, std::string>
SIPCall::getDetails() const
{
auto details = Call::getDetails();
details.emplace(DRing::Call::Details::PEER_HOLDING,
peerHolding_ ? TRUE_STR : FALSE_STR);
auto& acc = getSIPAccount();
#ifdef RING_VIDEO
// If Video is not enabled return an empty string
details.emplace(DRing::Call::Details::VIDEO_SOURCE, acc.isVideoEnabled() ? videoInput_ : "");
#endif
#if HAVE_RINGNS
if (not peerRegistredName_.empty())
details.emplace(DRing::Call::Details::REGISTERED_NAME, peerRegistredName_);
#endif
if (transport_ and transport_->isSecure()) {
const auto& tlsInfos = transport_->getTlsInfos();
const auto& cipher = pj_ssl_cipher_name(tlsInfos.cipher);
details.emplace(DRing::TlsTransport::TLS_CIPHER, cipher ? 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,
ring::to_string(n));
} else {
details.emplace(DRing::TlsTransport::TLS_PEER_CERT, "");
details.emplace(DRing::TlsTransport::TLS_PEER_CA_NUM, "");
}
}
return details;
}
bool
SIPCall::toggleRecording()
{ const bool startRecording = Call::toggleRecording();
if (startRecording) {
std::stringstream ss;
ss << "Conversation at %TIMESTAMP between "
<< getSIPAccount().getUserUri() << " and " << peerUri_;
recorder_->setMetadata(ss.str(), ""); // use default description
if (avformatrtp_)
avformatrtp_->initRecorder(recorder_);
#ifdef RING_VIDEO
if (!isAudioOnly_ && videortp_)
videortp_->initRecorder(recorder_);
#endif
}
return startRecording;
}
void
SIPCall::setSecure(bool sec)
{
if (srtpEnabled_)
return;
if (sec && getConnectionState() != ConnectionState::DISCONNECTED) {
throw std::runtime_error("Can't enable security since call is already connected");
}
srtpEnabled_ = sec;
}
void
SIPCall::InvSessionDeleter::operator ()(pjsip_inv_session* inv) const noexcept
{
// prevent this from getting accessed in callbacks
// RING_WARN: this is not thread-safe!
inv->mod_data[getSIPVoIPLink()->getModId()] = nullptr;
}
bool
SIPCall::initIceMediaTransport(bool master, unsigned channel_num)
{
RING_DBG("[call:%s] create media ICE transport", getCallId().c_str());
auto& iceTransportFactory = Manager::instance().getIceTransportFactory();
tmpMediaTransport_ = iceTransportFactory.createTransport(getCallId().c_str(),
channel_num, master,
getAccount().getIceOptions());
return static_cast<bool>(tmpMediaTransport_);
}
void
SIPCall::merge(Call& call)
{
RING_DBG("[sipcall:%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);
inv = std::move(subcall.inv);
inv->mod_data[getSIPVoIPLink()->getModId()] = this;
setTransport(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_;
mediaTransport_ = std::move(subcall.mediaTransport_);
tmpMediaTransport_ = std::move(subcall.tmpMediaTransport_);
Call::merge(subcall);
waitForIceAndStartMedia();
}
void
SIPCall::setRemoteSdp(const pjmedia_sdp_session* sdp)
{
if (!sdp)
return;
// If ICE is not used, start medias now
auto rem_ice_attrs = sdp_->getIceAttributes();
if (rem_ice_attrs.ufrag.empty() or rem_ice_attrs.pwd.empty()) {
RING_WARN("[call:%s] no ICE data in remote SDP", getCallId().c_str());
return;
}
if (not initIceMediaTransport(false)) {
// Fatal condition
// TODO: what's SIP rfc says about that?
// (same question in waitForIceAndStartMedia)
onFailure(EIO);
return;
}
// WARNING: This call blocks! (need ice init done)
setupLocalSDPFromIce();
}
bool
SIPCall::isIceRunning() const
{
return mediaTransport_ and mediaTransport_->isRunning();
}
std::unique_ptr<IceSocket>
SIPCall::newIceSocket(unsigned compId)
{
return std::unique_ptr<IceSocket> {new IceSocket(mediaTransport_, compId)};
}
} // namespace ring