mirror of
https://github.com/mollyim/webrtc.git
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3df5dcac9b
Bug: webrtc:8222 Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad Reviewed-on: https://webrtc-review.googlesource.com/6640 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20364}
2515 lines
91 KiB
C++
2515 lines
91 KiB
C++
/*
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* Copyright 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/webrtcsession.h"
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#include <limits.h>
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#include <algorithm>
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#include <set>
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#include <utility>
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#include <vector>
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#include "api/call/audio_sink.h"
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#include "api/jsepicecandidate.h"
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#include "api/jsepsessiondescription.h"
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#include "api/peerconnectioninterface.h"
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#include "call/call.h"
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#include "media/base/mediaconstants.h"
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#include "media/sctp/sctptransportinternal.h"
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#include "p2p/base/portallocator.h"
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#include "pc/channel.h"
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#include "pc/channelmanager.h"
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#include "pc/mediasession.h"
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#include "pc/sctputils.h"
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#include "pc/webrtcsessiondescriptionfactory.h"
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#include "rtc_base/basictypes.h"
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#include "rtc_base/bind.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/helpers.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/stringencode.h"
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#include "rtc_base/stringutils.h"
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#ifdef HAVE_QUIC
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#include "p2p/quic/quictransportchannel.h"
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#endif // HAVE_QUIC
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using cricket::ContentInfo;
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using cricket::ContentInfos;
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using cricket::MediaContentDescription;
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using cricket::SessionDescription;
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using cricket::TransportInfo;
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using cricket::LOCAL_PORT_TYPE;
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using cricket::STUN_PORT_TYPE;
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using cricket::RELAY_PORT_TYPE;
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using cricket::PRFLX_PORT_TYPE;
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namespace webrtc {
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// Error messages
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const char kBundleWithoutRtcpMux[] = "RTCP-MUX must be enabled when BUNDLE "
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"is enabled.";
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const char kCreateChannelFailed[] = "Failed to create channels.";
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const char kInvalidCandidates[] = "Description contains invalid candidates.";
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const char kInvalidSdp[] = "Invalid session description.";
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const char kMlineMismatchInAnswer[] =
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"The order of m-lines in answer doesn't match order in offer. Rejecting "
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"answer.";
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const char kMlineMismatchInSubsequentOffer[] =
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"The order of m-lines in subsequent offer doesn't match order from "
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"previous offer/answer.";
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const char kPushDownTDFailed[] =
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"Failed to push down transport description:";
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const char kSdpWithoutDtlsFingerprint[] =
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"Called with SDP without DTLS fingerprint.";
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const char kSdpWithoutSdesCrypto[] =
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"Called with SDP without SDES crypto.";
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const char kSdpWithoutIceUfragPwd[] =
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"Called with SDP without ice-ufrag and ice-pwd.";
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const char kSessionError[] = "Session error code: ";
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const char kSessionErrorDesc[] = "Session error description: ";
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const char kDtlsSrtpSetupFailureRtp[] =
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"Couldn't set up DTLS-SRTP on RTP channel.";
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const char kDtlsSrtpSetupFailureRtcp[] =
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"Couldn't set up DTLS-SRTP on RTCP channel.";
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const char kEnableBundleFailed[] = "Failed to enable BUNDLE.";
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IceCandidatePairType GetIceCandidatePairCounter(
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const cricket::Candidate& local,
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const cricket::Candidate& remote) {
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const auto& l = local.type();
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const auto& r = remote.type();
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const auto& host = LOCAL_PORT_TYPE;
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const auto& srflx = STUN_PORT_TYPE;
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const auto& relay = RELAY_PORT_TYPE;
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const auto& prflx = PRFLX_PORT_TYPE;
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if (l == host && r == host) {
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bool local_private = IPIsPrivate(local.address().ipaddr());
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bool remote_private = IPIsPrivate(remote.address().ipaddr());
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if (local_private) {
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if (remote_private) {
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return kIceCandidatePairHostPrivateHostPrivate;
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} else {
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return kIceCandidatePairHostPrivateHostPublic;
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}
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} else {
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if (remote_private) {
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return kIceCandidatePairHostPublicHostPrivate;
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} else {
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return kIceCandidatePairHostPublicHostPublic;
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}
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}
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}
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if (l == host && r == srflx)
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return kIceCandidatePairHostSrflx;
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if (l == host && r == relay)
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return kIceCandidatePairHostRelay;
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if (l == host && r == prflx)
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return kIceCandidatePairHostPrflx;
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if (l == srflx && r == host)
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return kIceCandidatePairSrflxHost;
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if (l == srflx && r == srflx)
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return kIceCandidatePairSrflxSrflx;
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if (l == srflx && r == relay)
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return kIceCandidatePairSrflxRelay;
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if (l == srflx && r == prflx)
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return kIceCandidatePairSrflxPrflx;
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if (l == relay && r == host)
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return kIceCandidatePairRelayHost;
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if (l == relay && r == srflx)
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return kIceCandidatePairRelaySrflx;
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if (l == relay && r == relay)
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return kIceCandidatePairRelayRelay;
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if (l == relay && r == prflx)
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return kIceCandidatePairRelayPrflx;
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if (l == prflx && r == host)
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return kIceCandidatePairPrflxHost;
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if (l == prflx && r == srflx)
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return kIceCandidatePairPrflxSrflx;
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if (l == prflx && r == relay)
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return kIceCandidatePairPrflxRelay;
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return kIceCandidatePairMax;
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}
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// Verify that the order of media sections in |desc1| matches |desc2|. The
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// number of m= sections could be different.
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static bool MediaSectionsInSameOrder(const SessionDescription* desc1,
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const SessionDescription* desc2) {
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if (!desc1 || !desc2) {
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return false;
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}
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for (size_t i = 0;
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i < desc1->contents().size() && i < desc2->contents().size(); ++i) {
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if ((desc2->contents()[i].name) != desc1->contents()[i].name) {
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return false;
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}
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const MediaContentDescription* desc2_mdesc =
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static_cast<const MediaContentDescription*>(
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desc2->contents()[i].description);
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const MediaContentDescription* desc1_mdesc =
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static_cast<const MediaContentDescription*>(
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desc1->contents()[i].description);
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if (desc2_mdesc->type() != desc1_mdesc->type()) {
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return false;
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}
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}
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return true;
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}
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static bool MediaSectionsHaveSameCount(const SessionDescription* desc1,
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const SessionDescription* desc2) {
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if (!desc1 || !desc2) {
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return false;
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}
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return desc1->contents().size() == desc2->contents().size();
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}
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// Checks that each non-rejected content has SDES crypto keys or a DTLS
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// fingerprint, unless it's in a BUNDLE group, in which case only the
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// BUNDLE-tag section (first media section/description in the BUNDLE group)
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// needs a ufrag and pwd. Mismatches, such as replying with a DTLS fingerprint
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// to SDES keys, will be caught in JsepTransport negotiation, and backstopped
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// by Channel's |srtp_required| check.
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static bool VerifyCrypto(const SessionDescription* desc,
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bool dtls_enabled,
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std::string* error) {
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const cricket::ContentGroup* bundle =
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desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
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const ContentInfos& contents = desc->contents();
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for (size_t index = 0; index < contents.size(); ++index) {
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const ContentInfo* cinfo = &contents[index];
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if (cinfo->rejected) {
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continue;
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}
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if (bundle && bundle->HasContentName(cinfo->name) &&
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cinfo->name != *(bundle->FirstContentName())) {
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// This isn't the first media section in the BUNDLE group, so it's not
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// required to have crypto attributes, since only the crypto attributes
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// from the first section actually get used.
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continue;
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}
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// If the content isn't rejected or bundled into another m= section, crypto
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// must be present.
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const MediaContentDescription* media =
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static_cast<const MediaContentDescription*>(cinfo->description);
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const TransportInfo* tinfo = desc->GetTransportInfoByName(cinfo->name);
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if (!media || !tinfo) {
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// Something is not right.
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LOG(LS_ERROR) << kInvalidSdp;
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*error = kInvalidSdp;
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return false;
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}
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if (dtls_enabled) {
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if (!tinfo->description.identity_fingerprint) {
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LOG(LS_WARNING) <<
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"Session description must have DTLS fingerprint if DTLS enabled.";
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*error = kSdpWithoutDtlsFingerprint;
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return false;
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}
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} else {
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if (media->cryptos().empty()) {
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LOG(LS_WARNING) <<
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"Session description must have SDES when DTLS disabled.";
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*error = kSdpWithoutSdesCrypto;
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return false;
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}
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}
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}
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return true;
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}
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// Checks that each non-rejected content has ice-ufrag and ice-pwd set, unless
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// it's in a BUNDLE group, in which case only the BUNDLE-tag section (first
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// media section/description in the BUNDLE group) needs a ufrag and pwd.
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static bool VerifyIceUfragPwdPresent(const SessionDescription* desc) {
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const cricket::ContentGroup* bundle =
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desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
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const ContentInfos& contents = desc->contents();
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for (size_t index = 0; index < contents.size(); ++index) {
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const ContentInfo* cinfo = &contents[index];
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if (cinfo->rejected) {
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continue;
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}
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if (bundle && bundle->HasContentName(cinfo->name) &&
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cinfo->name != *(bundle->FirstContentName())) {
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// This isn't the first media section in the BUNDLE group, so it's not
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// required to have ufrag/password, since only the ufrag/password from
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// the first section actually get used.
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continue;
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}
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// If the content isn't rejected or bundled into another m= section,
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// ice-ufrag and ice-pwd must be present.
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const TransportInfo* tinfo = desc->GetTransportInfoByName(cinfo->name);
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if (!tinfo) {
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// Something is not right.
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LOG(LS_ERROR) << kInvalidSdp;
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return false;
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}
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if (tinfo->description.ice_ufrag.empty() ||
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tinfo->description.ice_pwd.empty()) {
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LOG(LS_ERROR) << "Session description must have ice ufrag and pwd.";
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return false;
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}
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}
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return true;
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}
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static bool GetTrackIdBySsrc(const SessionDescription* session_description,
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uint32_t ssrc,
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std::string* track_id) {
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RTC_DCHECK(track_id != NULL);
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const cricket::ContentInfo* audio_info =
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cricket::GetFirstAudioContent(session_description);
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if (audio_info) {
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const cricket::MediaContentDescription* audio_content =
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static_cast<const cricket::MediaContentDescription*>(
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audio_info->description);
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const auto* found =
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cricket::GetStreamBySsrc(audio_content->streams(), ssrc);
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if (found) {
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*track_id = found->id;
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return true;
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}
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}
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const cricket::ContentInfo* video_info =
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cricket::GetFirstVideoContent(session_description);
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if (video_info) {
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const cricket::MediaContentDescription* video_content =
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static_cast<const cricket::MediaContentDescription*>(
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video_info->description);
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const auto* found =
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cricket::GetStreamBySsrc(video_content->streams(), ssrc);
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if (found) {
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*track_id = found->id;
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return true;
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}
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}
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return false;
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}
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// Get the SCTP port out of a SessionDescription.
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// Return -1 if not found.
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static int GetSctpPort(const SessionDescription* session_description) {
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const ContentInfo* content_info = GetFirstDataContent(session_description);
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RTC_DCHECK(content_info);
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if (!content_info) {
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return -1;
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}
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const cricket::DataContentDescription* data =
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static_cast<const cricket::DataContentDescription*>(
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(content_info->description));
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std::string value;
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cricket::DataCodec match_pattern(cricket::kGoogleSctpDataCodecPlType,
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cricket::kGoogleSctpDataCodecName);
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for (const cricket::DataCodec& codec : data->codecs()) {
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if (!codec.Matches(match_pattern)) {
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continue;
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}
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if (codec.GetParam(cricket::kCodecParamPort, &value)) {
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return rtc::FromString<int>(value);
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}
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}
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return -1;
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}
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static bool BadSdp(const std::string& source,
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const std::string& type,
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const std::string& reason,
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std::string* err_desc) {
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std::ostringstream desc;
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desc << "Failed to set " << source;
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if (!type.empty()) {
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desc << " " << type;
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}
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desc << " SDP: " << reason;
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if (err_desc) {
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*err_desc = desc.str();
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}
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LOG(LS_ERROR) << desc.str();
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return false;
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}
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static bool BadSdp(cricket::ContentSource source,
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const std::string& type,
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const std::string& reason,
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std::string* err_desc) {
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if (source == cricket::CS_LOCAL) {
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return BadSdp("local", type, reason, err_desc);
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} else {
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return BadSdp("remote", type, reason, err_desc);
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}
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}
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static bool BadLocalSdp(const std::string& type,
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const std::string& reason,
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std::string* err_desc) {
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return BadSdp(cricket::CS_LOCAL, type, reason, err_desc);
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}
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static bool BadRemoteSdp(const std::string& type,
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const std::string& reason,
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std::string* err_desc) {
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return BadSdp(cricket::CS_REMOTE, type, reason, err_desc);
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}
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static bool BadOfferSdp(cricket::ContentSource source,
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const std::string& reason,
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std::string* err_desc) {
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return BadSdp(source, SessionDescriptionInterface::kOffer, reason, err_desc);
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}
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static bool BadPranswerSdp(cricket::ContentSource source,
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const std::string& reason,
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std::string* err_desc) {
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return BadSdp(source, SessionDescriptionInterface::kPrAnswer,
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reason, err_desc);
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}
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static bool BadAnswerSdp(cricket::ContentSource source,
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const std::string& reason,
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std::string* err_desc) {
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return BadSdp(source, SessionDescriptionInterface::kAnswer, reason, err_desc);
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}
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#define GET_STRING_OF_STATE(state) \
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case webrtc::WebRtcSession::state: \
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result = #state; \
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break;
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static std::string GetStateString(webrtc::WebRtcSession::State state) {
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std::string result;
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switch (state) {
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GET_STRING_OF_STATE(STATE_INIT)
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GET_STRING_OF_STATE(STATE_SENTOFFER)
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GET_STRING_OF_STATE(STATE_RECEIVEDOFFER)
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GET_STRING_OF_STATE(STATE_SENTPRANSWER)
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GET_STRING_OF_STATE(STATE_RECEIVEDPRANSWER)
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GET_STRING_OF_STATE(STATE_INPROGRESS)
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GET_STRING_OF_STATE(STATE_CLOSED)
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default:
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RTC_NOTREACHED();
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break;
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}
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return result;
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}
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#define GET_STRING_OF_ERROR_CODE(err) \
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case webrtc::WebRtcSession::err: \
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result = #err; \
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break;
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static std::string GetErrorCodeString(webrtc::WebRtcSession::Error err) {
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std::string result;
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switch (err) {
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GET_STRING_OF_ERROR_CODE(ERROR_NONE)
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GET_STRING_OF_ERROR_CODE(ERROR_CONTENT)
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GET_STRING_OF_ERROR_CODE(ERROR_TRANSPORT)
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default:
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RTC_NOTREACHED();
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break;
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}
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return result;
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}
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static std::string MakeErrorString(const std::string& error,
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const std::string& desc) {
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std::ostringstream ret;
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ret << error << " " << desc;
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return ret.str();
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}
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static std::string MakeTdErrorString(const std::string& desc) {
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return MakeErrorString(kPushDownTDFailed, desc);
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}
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|
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// Returns true if |new_desc| requests an ICE restart (i.e., new ufrag/pwd).
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bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc,
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const SessionDescriptionInterface* new_desc,
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const std::string& content_name) {
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if (!old_desc) {
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return false;
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}
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const SessionDescription* new_sd = new_desc->description();
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const SessionDescription* old_sd = old_desc->description();
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const ContentInfo* cinfo = new_sd->GetContentByName(content_name);
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if (!cinfo || cinfo->rejected) {
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return false;
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}
|
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// If the content isn't rejected, check if ufrag and password has changed.
|
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const cricket::TransportDescription* new_transport_desc =
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new_sd->GetTransportDescriptionByName(content_name);
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const cricket::TransportDescription* old_transport_desc =
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old_sd->GetTransportDescriptionByName(content_name);
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if (!new_transport_desc || !old_transport_desc) {
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// No transport description exists. This is not an ICE restart.
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return false;
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}
|
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if (cricket::IceCredentialsChanged(
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old_transport_desc->ice_ufrag, old_transport_desc->ice_pwd,
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new_transport_desc->ice_ufrag, new_transport_desc->ice_pwd)) {
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LOG(LS_INFO) << "Remote peer requests ICE restart for " << content_name
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<< ".";
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return true;
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}
|
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return false;
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}
|
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|
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WebRtcSession::WebRtcSession(
|
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Call* call,
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cricket::ChannelManager* channel_manager,
|
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const cricket::MediaConfig& media_config,
|
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RtcEventLog* event_log,
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rtc::Thread* network_thread,
|
|
rtc::Thread* worker_thread,
|
|
rtc::Thread* signaling_thread,
|
|
cricket::PortAllocator* port_allocator,
|
|
std::unique_ptr<cricket::TransportController> transport_controller,
|
|
std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory)
|
|
: network_thread_(network_thread),
|
|
worker_thread_(worker_thread),
|
|
signaling_thread_(signaling_thread),
|
|
// RFC 3264: The numeric value of the session id and version in the
|
|
// o line MUST be representable with a "64 bit signed integer".
|
|
// Due to this constraint session id |sid_| is max limited to LLONG_MAX.
|
|
sid_(rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX)),
|
|
transport_controller_(std::move(transport_controller)),
|
|
sctp_factory_(std::move(sctp_factory)),
|
|
media_config_(media_config),
|
|
event_log_(event_log),
|
|
call_(call),
|
|
channel_manager_(channel_manager),
|
|
ice_observer_(NULL),
|
|
ice_connection_state_(PeerConnectionInterface::kIceConnectionNew),
|
|
ice_connection_receiving_(true),
|
|
older_version_remote_peer_(false),
|
|
dtls_enabled_(false),
|
|
data_channel_type_(cricket::DCT_NONE),
|
|
metrics_observer_(NULL) {
|
|
transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLED);
|
|
transport_controller_->SignalConnectionState.connect(
|
|
this, &WebRtcSession::OnTransportControllerConnectionState);
|
|
transport_controller_->SignalReceiving.connect(
|
|
this, &WebRtcSession::OnTransportControllerReceiving);
|
|
transport_controller_->SignalGatheringState.connect(
|
|
this, &WebRtcSession::OnTransportControllerGatheringState);
|
|
transport_controller_->SignalCandidatesGathered.connect(
|
|
this, &WebRtcSession::OnTransportControllerCandidatesGathered);
|
|
transport_controller_->SignalCandidatesRemoved.connect(
|
|
this, &WebRtcSession::OnTransportControllerCandidatesRemoved);
|
|
transport_controller_->SignalDtlsHandshakeError.connect(
|
|
this, &WebRtcSession::OnTransportControllerDtlsHandshakeError);
|
|
}
|
|
|
|
WebRtcSession::~WebRtcSession() {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
// Destroy video channels first since they may have a pointer to a voice
|
|
// channel.
|
|
for (auto* channel : video_channels_) {
|
|
DestroyVideoChannel(channel);
|
|
}
|
|
for (auto* channel : voice_channels_) {
|
|
DestroyVoiceChannel(channel);
|
|
}
|
|
if (rtp_data_channel_) {
|
|
DestroyDataChannel();
|
|
}
|
|
if (sctp_transport_) {
|
|
SignalDataChannelDestroyed();
|
|
network_thread_->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&WebRtcSession::DestroySctpTransport_n, this));
|
|
}
|
|
#ifdef HAVE_QUIC
|
|
if (quic_data_transport_) {
|
|
quic_data_transport_.reset();
|
|
}
|
|
#endif
|
|
|
|
LOG(LS_INFO) << "Session: " << id() << " is destroyed.";
|
|
}
|
|
|
|
bool WebRtcSession::Initialize(
|
|
const PeerConnectionFactoryInterface::Options& options,
|
|
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
|
|
const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
|
|
bundle_policy_ = rtc_configuration.bundle_policy;
|
|
rtcp_mux_policy_ = rtc_configuration.rtcp_mux_policy;
|
|
transport_controller_->SetSslMaxProtocolVersion(options.ssl_max_version);
|
|
|
|
// Obtain a certificate from RTCConfiguration if any were provided (optional).
|
|
rtc::scoped_refptr<rtc::RTCCertificate> certificate;
|
|
if (!rtc_configuration.certificates.empty()) {
|
|
// TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
|
|
// just picking the first one. The decision should be made based on the DTLS
|
|
// handshake. The DTLS negotiations need to know about all certificates.
|
|
certificate = rtc_configuration.certificates[0];
|
|
}
|
|
|
|
SetIceConfig(ParseIceConfig(rtc_configuration));
|
|
|
|
if (options.disable_encryption) {
|
|
dtls_enabled_ = false;
|
|
} else {
|
|
// Enable DTLS by default if we have an identity store or a certificate.
|
|
dtls_enabled_ = (cert_generator || certificate);
|
|
// |rtc_configuration| can override the default |dtls_enabled_| value.
|
|
if (rtc_configuration.enable_dtls_srtp) {
|
|
dtls_enabled_ = *(rtc_configuration.enable_dtls_srtp);
|
|
}
|
|
}
|
|
|
|
// Enable creation of RTP data channels if the kEnableRtpDataChannels is set.
|
|
// It takes precendence over the disable_sctp_data_channels
|
|
// PeerConnectionFactoryInterface::Options.
|
|
if (rtc_configuration.enable_rtp_data_channel) {
|
|
data_channel_type_ = cricket::DCT_RTP;
|
|
}
|
|
#ifdef HAVE_QUIC
|
|
else if (rtc_configuration.enable_quic) {
|
|
// Use QUIC instead of DTLS when |enable_quic| is true.
|
|
data_channel_type_ = cricket::DCT_QUIC;
|
|
transport_controller_->use_quic();
|
|
if (dtls_enabled_) {
|
|
LOG(LS_INFO) << "Using QUIC instead of DTLS";
|
|
}
|
|
quic_data_transport_.reset(
|
|
new QuicDataTransport(signaling_thread(), worker_thread(),
|
|
network_thread(), transport_controller_.get()));
|
|
}
|
|
#endif // HAVE_QUIC
|
|
else {
|
|
// DTLS has to be enabled to use SCTP.
|
|
if (!options.disable_sctp_data_channels && dtls_enabled_) {
|
|
data_channel_type_ = cricket::DCT_SCTP;
|
|
}
|
|
}
|
|
|
|
video_options_.screencast_min_bitrate_kbps =
|
|
rtc_configuration.screencast_min_bitrate;
|
|
audio_options_.combined_audio_video_bwe =
|
|
rtc_configuration.combined_audio_video_bwe;
|
|
|
|
audio_options_.audio_jitter_buffer_max_packets =
|
|
rtc::Optional<int>(rtc_configuration.audio_jitter_buffer_max_packets);
|
|
|
|
audio_options_.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(
|
|
rtc_configuration.audio_jitter_buffer_fast_accelerate);
|
|
|
|
if (!dtls_enabled_) {
|
|
// Construct with DTLS disabled.
|
|
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
|
|
signaling_thread(), channel_manager_, this, id(),
|
|
std::unique_ptr<rtc::RTCCertificateGeneratorInterface>()));
|
|
} else {
|
|
// Construct with DTLS enabled.
|
|
if (!certificate) {
|
|
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
|
|
signaling_thread(), channel_manager_, this, id(),
|
|
std::move(cert_generator)));
|
|
} else {
|
|
// Use the already generated certificate.
|
|
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
|
|
signaling_thread(), channel_manager_, this, id(), certificate));
|
|
}
|
|
}
|
|
|
|
webrtc_session_desc_factory_->SignalCertificateReady.connect(
|
|
this, &WebRtcSession::OnCertificateReady);
|
|
|
|
if (options.disable_encryption) {
|
|
webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
|
|
}
|
|
|
|
webrtc_session_desc_factory_->set_enable_encrypted_rtp_header_extensions(
|
|
options.crypto_options.enable_encrypted_rtp_header_extensions);
|
|
|
|
return true;
|
|
}
|
|
|
|
void WebRtcSession::Close() {
|
|
SetState(STATE_CLOSED);
|
|
RemoveUnusedChannels(nullptr);
|
|
call_ = nullptr;
|
|
RTC_DCHECK(voice_channels_.empty());
|
|
RTC_DCHECK(video_channels_.empty());
|
|
RTC_DCHECK(!rtp_data_channel_);
|
|
RTC_DCHECK(!sctp_transport_);
|
|
}
|
|
|
|
cricket::BaseChannel* WebRtcSession::GetChannel(
|
|
const std::string& content_name) {
|
|
if (voice_channel() && voice_channel()->content_name() == content_name) {
|
|
return voice_channel();
|
|
}
|
|
if (video_channel() && video_channel()->content_name() == content_name) {
|
|
return video_channel();
|
|
}
|
|
if (rtp_data_channel() &&
|
|
rtp_data_channel()->content_name() == content_name) {
|
|
return rtp_data_channel();
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
bool WebRtcSession::GetSctpSslRole(rtc::SSLRole* role) {
|
|
if (!local_description() || !remote_description()) {
|
|
LOG(LS_INFO) << "Local and Remote descriptions must be applied to get the "
|
|
<< "SSL Role of the SCTP transport.";
|
|
return false;
|
|
}
|
|
if (!sctp_transport_) {
|
|
LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
|
|
<< "SSL Role of the SCTP transport.";
|
|
return false;
|
|
}
|
|
|
|
return transport_controller_->GetSslRole(*sctp_transport_name_, role);
|
|
}
|
|
|
|
bool WebRtcSession::GetSslRole(const std::string& content_name,
|
|
rtc::SSLRole* role) {
|
|
if (!local_description() || !remote_description()) {
|
|
LOG(LS_INFO) << "Local and Remote descriptions must be applied to get the "
|
|
<< "SSL Role of the session.";
|
|
return false;
|
|
}
|
|
|
|
return transport_controller_->GetSslRole(GetTransportName(content_name),
|
|
role);
|
|
}
|
|
|
|
void WebRtcSession::CreateOffer(
|
|
CreateSessionDescriptionObserver* observer,
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& options,
|
|
const cricket::MediaSessionOptions& session_options) {
|
|
webrtc_session_desc_factory_->CreateOffer(observer, options, session_options);
|
|
}
|
|
|
|
void WebRtcSession::CreateAnswer(
|
|
CreateSessionDescriptionObserver* observer,
|
|
const cricket::MediaSessionOptions& session_options) {
|
|
webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
|
|
}
|
|
|
|
bool WebRtcSession::SetLocalDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
std::string* err_desc) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
|
|
// Validate SDP.
|
|
if (!ValidateSessionDescription(desc.get(), cricket::CS_LOCAL, err_desc)) {
|
|
return false;
|
|
}
|
|
|
|
// Update the initial_offerer flag if this session is the initial_offerer.
|
|
Action action = GetAction(desc->type());
|
|
if (state() == STATE_INIT && action == kOffer) {
|
|
initial_offerer_ = true;
|
|
transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);
|
|
}
|
|
|
|
if (action == kAnswer) {
|
|
current_local_description_ = std::move(desc);
|
|
pending_local_description_ = nullptr;
|
|
current_remote_description_ = std::move(pending_remote_description_);
|
|
} else {
|
|
pending_local_description_ = std::move(desc);
|
|
}
|
|
|
|
// Transport and Media channels will be created only when offer is set.
|
|
if (action == kOffer && !CreateChannels(local_description()->description())) {
|
|
// TODO(mallinath) - Handle CreateChannel failure, as new local description
|
|
// is applied. Restore back to old description.
|
|
return BadLocalSdp(local_description()->type(), kCreateChannelFailed,
|
|
err_desc);
|
|
}
|
|
|
|
// Remove unused channels if MediaContentDescription is rejected.
|
|
RemoveUnusedChannels(local_description()->description());
|
|
|
|
if (!UpdateSessionState(action, cricket::CS_LOCAL, err_desc)) {
|
|
return false;
|
|
}
|
|
if (remote_description()) {
|
|
// Now that we have a local description, we can push down remote candidates.
|
|
UseCandidatesInSessionDescription(remote_description());
|
|
}
|
|
|
|
pending_ice_restarts_.clear();
|
|
if (error() != ERROR_NONE) {
|
|
return BadLocalSdp(local_description()->type(), GetSessionErrorMsg(),
|
|
err_desc);
|
|
}
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcSession::SetRemoteDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
std::string* err_desc) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
|
|
// Validate SDP.
|
|
if (!ValidateSessionDescription(desc.get(), cricket::CS_REMOTE, err_desc)) {
|
|
return false;
|
|
}
|
|
|
|
// Hold this pointer so candidates can be copied to it later in the method.
|
|
SessionDescriptionInterface* desc_ptr = desc.get();
|
|
|
|
const SessionDescriptionInterface* old_remote_description =
|
|
remote_description();
|
|
// Grab ownership of the description being replaced for the remainder of this
|
|
// method, since it's used below as |old_remote_description|.
|
|
std::unique_ptr<SessionDescriptionInterface> replaced_remote_description;
|
|
Action action = GetAction(desc->type());
|
|
if (action == kAnswer) {
|
|
replaced_remote_description = pending_remote_description_
|
|
? std::move(pending_remote_description_)
|
|
: std::move(current_remote_description_);
|
|
current_remote_description_ = std::move(desc);
|
|
pending_remote_description_ = nullptr;
|
|
current_local_description_ = std::move(pending_local_description_);
|
|
} else {
|
|
replaced_remote_description = std::move(pending_remote_description_);
|
|
pending_remote_description_ = std::move(desc);
|
|
}
|
|
|
|
// Transport and Media channels will be created only when offer is set.
|
|
if (action == kOffer &&
|
|
!CreateChannels(remote_description()->description())) {
|
|
// TODO(mallinath) - Handle CreateChannel failure, as new local description
|
|
// is applied. Restore back to old description.
|
|
return BadRemoteSdp(remote_description()->type(), kCreateChannelFailed,
|
|
err_desc);
|
|
}
|
|
|
|
// Remove unused channels if MediaContentDescription is rejected.
|
|
RemoveUnusedChannels(remote_description()->description());
|
|
|
|
// NOTE: Candidates allocation will be initiated only when SetLocalDescription
|
|
// is called.
|
|
if (!UpdateSessionState(action, cricket::CS_REMOTE, err_desc)) {
|
|
return false;
|
|
}
|
|
|
|
if (local_description() &&
|
|
!UseCandidatesInSessionDescription(remote_description())) {
|
|
return BadRemoteSdp(remote_description()->type(), kInvalidCandidates,
|
|
err_desc);
|
|
}
|
|
|
|
if (old_remote_description) {
|
|
for (const cricket::ContentInfo& content :
|
|
old_remote_description->description()->contents()) {
|
|
// Check if this new SessionDescription contains new ICE ufrag and
|
|
// password that indicates the remote peer requests an ICE restart.
|
|
// TODO(deadbeef): When we start storing both the current and pending
|
|
// remote description, this should reset pending_ice_restarts and compare
|
|
// against the current description.
|
|
if (CheckForRemoteIceRestart(old_remote_description, remote_description(),
|
|
content.name)) {
|
|
if (action == kOffer) {
|
|
pending_ice_restarts_.insert(content.name);
|
|
}
|
|
} else {
|
|
// We retain all received candidates only if ICE is not restarted.
|
|
// When ICE is restarted, all previous candidates belong to an old
|
|
// generation and should not be kept.
|
|
// TODO(deadbeef): This goes against the W3C spec which says the remote
|
|
// description should only contain candidates from the last set remote
|
|
// description plus any candidates added since then. We should remove
|
|
// this once we're sure it won't break anything.
|
|
WebRtcSessionDescriptionFactory::CopyCandidatesFromSessionDescription(
|
|
old_remote_description, content.name, desc_ptr);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (error() != ERROR_NONE) {
|
|
return BadRemoteSdp(remote_description()->type(), GetSessionErrorMsg(),
|
|
err_desc);
|
|
}
|
|
|
|
// Set the the ICE connection state to connecting since the connection may
|
|
// become writable with peer reflexive candidates before any remote candidate
|
|
// is signaled.
|
|
// TODO(pthatcher): This is a short-term solution for crbug/446908. A real fix
|
|
// is to have a new signal the indicates a change in checking state from the
|
|
// transport and expose a new checking() member from transport that can be
|
|
// read to determine the current checking state. The existing SignalConnecting
|
|
// actually means "gathering candidates", so cannot be be used here.
|
|
if (remote_description()->type() != SessionDescriptionInterface::kOffer &&
|
|
ice_connection_state_ == PeerConnectionInterface::kIceConnectionNew) {
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
|
|
}
|
|
return true;
|
|
}
|
|
|
|
// TODO(steveanton): Eventually it'd be nice to store the channels as a single
|
|
// vector of BaseChannel pointers instead of separate voice and video channel
|
|
// vectors. At that point, this will become a simple getter.
|
|
std::vector<cricket::BaseChannel*> WebRtcSession::Channels() const {
|
|
std::vector<cricket::BaseChannel*> channels;
|
|
channels.insert(channels.end(), voice_channels_.begin(),
|
|
voice_channels_.end());
|
|
channels.insert(channels.end(), video_channels_.begin(),
|
|
video_channels_.end());
|
|
if (rtp_data_channel_) {
|
|
channels.push_back(rtp_data_channel_.get());
|
|
}
|
|
return channels;
|
|
}
|
|
|
|
void WebRtcSession::LogState(State old_state, State new_state) {
|
|
LOG(LS_INFO) << "Session:" << id()
|
|
<< " Old state:" << GetStateString(old_state)
|
|
<< " New state:" << GetStateString(new_state);
|
|
}
|
|
|
|
void WebRtcSession::SetState(State state) {
|
|
RTC_DCHECK(signaling_thread_->IsCurrent());
|
|
if (state != state_) {
|
|
LogState(state_, state);
|
|
state_ = state;
|
|
SignalState(this, state_);
|
|
}
|
|
}
|
|
|
|
void WebRtcSession::SetError(Error error, const std::string& error_desc) {
|
|
RTC_DCHECK(signaling_thread_->IsCurrent());
|
|
if (error != error_) {
|
|
error_ = error;
|
|
error_desc_ = error_desc;
|
|
}
|
|
}
|
|
|
|
bool WebRtcSession::UpdateSessionState(
|
|
Action action, cricket::ContentSource source,
|
|
std::string* err_desc) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
|
|
// If there's already a pending error then no state transition should happen.
|
|
// But all call-sites should be verifying this before calling us!
|
|
RTC_DCHECK(error() == ERROR_NONE);
|
|
std::string td_err;
|
|
if (action == kOffer) {
|
|
if (!PushdownTransportDescription(source, cricket::CA_OFFER, &td_err)) {
|
|
return BadOfferSdp(source, MakeTdErrorString(td_err), err_desc);
|
|
}
|
|
SetState(source == cricket::CS_LOCAL ? STATE_SENTOFFER
|
|
: STATE_RECEIVEDOFFER);
|
|
if (!PushdownMediaDescription(cricket::CA_OFFER, source, err_desc)) {
|
|
SetError(ERROR_CONTENT, *err_desc);
|
|
}
|
|
if (error() != ERROR_NONE) {
|
|
return BadOfferSdp(source, GetSessionErrorMsg(), err_desc);
|
|
}
|
|
} else if (action == kPrAnswer) {
|
|
if (!PushdownTransportDescription(source, cricket::CA_PRANSWER, &td_err)) {
|
|
return BadPranswerSdp(source, MakeTdErrorString(td_err), err_desc);
|
|
}
|
|
EnableChannels();
|
|
SetState(source == cricket::CS_LOCAL ? STATE_SENTPRANSWER
|
|
: STATE_RECEIVEDPRANSWER);
|
|
if (!PushdownMediaDescription(cricket::CA_PRANSWER, source, err_desc)) {
|
|
SetError(ERROR_CONTENT, *err_desc);
|
|
}
|
|
if (error() != ERROR_NONE) {
|
|
return BadPranswerSdp(source, GetSessionErrorMsg(), err_desc);
|
|
}
|
|
} else if (action == kAnswer) {
|
|
const cricket::ContentGroup* local_bundle =
|
|
local_description()->description()->GetGroupByName(
|
|
cricket::GROUP_TYPE_BUNDLE);
|
|
const cricket::ContentGroup* remote_bundle =
|
|
remote_description()->description()->GetGroupByName(
|
|
cricket::GROUP_TYPE_BUNDLE);
|
|
if (local_bundle && remote_bundle) {
|
|
// The answerer decides the transport to bundle on.
|
|
const cricket::ContentGroup* answer_bundle =
|
|
(source == cricket::CS_LOCAL ? local_bundle : remote_bundle);
|
|
if (!EnableBundle(*answer_bundle)) {
|
|
LOG(LS_WARNING) << "Failed to enable BUNDLE.";
|
|
return BadAnswerSdp(source, kEnableBundleFailed, err_desc);
|
|
}
|
|
}
|
|
// Only push down the transport description after enabling BUNDLE; we don't
|
|
// want to push down a description on a transport about to be destroyed.
|
|
if (!PushdownTransportDescription(source, cricket::CA_ANSWER, &td_err)) {
|
|
return BadAnswerSdp(source, MakeTdErrorString(td_err), err_desc);
|
|
}
|
|
EnableChannels();
|
|
SetState(STATE_INPROGRESS);
|
|
if (!PushdownMediaDescription(cricket::CA_ANSWER, source, err_desc)) {
|
|
SetError(ERROR_CONTENT, *err_desc);
|
|
}
|
|
if (error() != ERROR_NONE) {
|
|
return BadAnswerSdp(source, GetSessionErrorMsg(), err_desc);
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
WebRtcSession::Action WebRtcSession::GetAction(const std::string& type) {
|
|
if (type == SessionDescriptionInterface::kOffer) {
|
|
return WebRtcSession::kOffer;
|
|
} else if (type == SessionDescriptionInterface::kPrAnswer) {
|
|
return WebRtcSession::kPrAnswer;
|
|
} else if (type == SessionDescriptionInterface::kAnswer) {
|
|
return WebRtcSession::kAnswer;
|
|
}
|
|
RTC_NOTREACHED() << "unknown action type";
|
|
return WebRtcSession::kOffer;
|
|
}
|
|
|
|
bool WebRtcSession::PushdownMediaDescription(
|
|
cricket::ContentAction action,
|
|
cricket::ContentSource source,
|
|
std::string* err) {
|
|
const SessionDescription* sdesc =
|
|
(source == cricket::CS_LOCAL ? local_description() : remote_description())
|
|
->description();
|
|
RTC_DCHECK(sdesc);
|
|
bool all_success = true;
|
|
for (auto* channel : Channels()) {
|
|
// TODO(steveanton): Add support for multiple channels of the same type.
|
|
const ContentInfo* content_info =
|
|
cricket::GetFirstMediaContent(sdesc->contents(), channel->media_type());
|
|
if (!content_info) {
|
|
continue;
|
|
}
|
|
const MediaContentDescription* content_desc =
|
|
static_cast<const MediaContentDescription*>(content_info->description);
|
|
if (content_desc && !content_info->rejected) {
|
|
bool success = (source == cricket::CS_LOCAL)
|
|
? channel->SetLocalContent(content_desc, action, err)
|
|
: channel->SetRemoteContent(content_desc, action, err);
|
|
if (!success) {
|
|
all_success = false;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
// Need complete offer/answer with an SCTP m= section before starting SCTP,
|
|
// according to https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-19
|
|
if (sctp_transport_ && local_description() && remote_description() &&
|
|
cricket::GetFirstDataContent(local_description()->description()) &&
|
|
cricket::GetFirstDataContent(remote_description()->description())) {
|
|
all_success &= network_thread_->Invoke<bool>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&WebRtcSession::PushdownSctpParameters_n, this, source));
|
|
}
|
|
return all_success;
|
|
}
|
|
|
|
bool WebRtcSession::PushdownSctpParameters_n(cricket::ContentSource source) {
|
|
RTC_DCHECK(network_thread_->IsCurrent());
|
|
RTC_DCHECK(local_description());
|
|
RTC_DCHECK(remote_description());
|
|
// Apply the SCTP port (which is hidden inside a DataCodec structure...)
|
|
// When we support "max-message-size", that would also be pushed down here.
|
|
return sctp_transport_->Start(
|
|
GetSctpPort(local_description()->description()),
|
|
GetSctpPort(remote_description()->description()));
|
|
}
|
|
|
|
bool WebRtcSession::PushdownTransportDescription(cricket::ContentSource source,
|
|
cricket::ContentAction action,
|
|
std::string* error_desc) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
|
|
if (source == cricket::CS_LOCAL) {
|
|
return PushdownLocalTransportDescription(local_description()->description(),
|
|
action, error_desc);
|
|
}
|
|
return PushdownRemoteTransportDescription(remote_description()->description(),
|
|
action, error_desc);
|
|
}
|
|
|
|
bool WebRtcSession::PushdownLocalTransportDescription(
|
|
const SessionDescription* sdesc,
|
|
cricket::ContentAction action,
|
|
std::string* err) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
|
|
if (!sdesc) {
|
|
return false;
|
|
}
|
|
|
|
for (const TransportInfo& tinfo : sdesc->transport_infos()) {
|
|
if (!transport_controller_->SetLocalTransportDescription(
|
|
tinfo.content_name, tinfo.description, action, err)) {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcSession::PushdownRemoteTransportDescription(
|
|
const SessionDescription* sdesc,
|
|
cricket::ContentAction action,
|
|
std::string* err) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
|
|
if (!sdesc) {
|
|
return false;
|
|
}
|
|
|
|
for (const TransportInfo& tinfo : sdesc->transport_infos()) {
|
|
if (!transport_controller_->SetRemoteTransportDescription(
|
|
tinfo.content_name, tinfo.description, action, err)) {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcSession::GetTransportDescription(
|
|
const SessionDescription* description,
|
|
const std::string& content_name,
|
|
cricket::TransportDescription* tdesc) {
|
|
if (!description || !tdesc) {
|
|
return false;
|
|
}
|
|
const TransportInfo* transport_info =
|
|
description->GetTransportInfoByName(content_name);
|
|
if (!transport_info) {
|
|
return false;
|
|
}
|
|
*tdesc = transport_info->description;
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcSession::EnableBundle(const cricket::ContentGroup& bundle) {
|
|
const std::string* first_content_name = bundle.FirstContentName();
|
|
if (!first_content_name) {
|
|
LOG(LS_WARNING) << "Tried to BUNDLE with no contents.";
|
|
return false;
|
|
}
|
|
const std::string& transport_name = *first_content_name;
|
|
|
|
#ifdef HAVE_QUIC
|
|
if (quic_data_transport_ &&
|
|
bundle.HasContentName(quic_data_transport_->content_name()) &&
|
|
quic_data_transport_->transport_name() != transport_name) {
|
|
LOG(LS_ERROR) << "Unable to BUNDLE " << quic_data_transport_->content_name()
|
|
<< " on " << transport_name << "with QUIC.";
|
|
}
|
|
#endif
|
|
auto maybe_set_transport = [this, bundle,
|
|
transport_name](cricket::BaseChannel* ch) {
|
|
if (!ch || !bundle.HasContentName(ch->content_name())) {
|
|
return true;
|
|
}
|
|
|
|
std::string old_transport_name = ch->transport_name();
|
|
if (old_transport_name == transport_name) {
|
|
LOG(LS_INFO) << "BUNDLE already enabled for " << ch->content_name()
|
|
<< " on " << transport_name << ".";
|
|
return true;
|
|
}
|
|
|
|
cricket::DtlsTransportInternal* rtp_dtls_transport =
|
|
transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
bool need_rtcp = (ch->rtcp_dtls_transport() != nullptr);
|
|
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
|
|
if (need_rtcp) {
|
|
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
|
|
ch->SetTransports(rtp_dtls_transport, rtcp_dtls_transport);
|
|
LOG(LS_INFO) << "Enabled BUNDLE for " << ch->content_name() << " on "
|
|
<< transport_name << ".";
|
|
transport_controller_->DestroyDtlsTransport(
|
|
old_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
// If the channel needs rtcp, it means that the channel used to have a
|
|
// rtcp transport which needs to be deleted now.
|
|
if (need_rtcp) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
old_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
return true;
|
|
};
|
|
|
|
if (!maybe_set_transport(voice_channel()) ||
|
|
!maybe_set_transport(video_channel()) ||
|
|
!maybe_set_transport(rtp_data_channel())) {
|
|
return false;
|
|
}
|
|
// For SCTP, transport creation/deletion happens here instead of in the
|
|
// object itself.
|
|
if (sctp_transport_) {
|
|
RTC_DCHECK(sctp_transport_name_);
|
|
RTC_DCHECK(sctp_content_name_);
|
|
if (transport_name != *sctp_transport_name_ &&
|
|
bundle.HasContentName(*sctp_content_name_)) {
|
|
network_thread_->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&WebRtcSession::ChangeSctpTransport_n, this,
|
|
transport_name));
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcSession::ProcessIceMessage(const IceCandidateInterface* candidate) {
|
|
if (!remote_description()) {
|
|
LOG(LS_ERROR) << "ProcessIceMessage: ICE candidates can't be added "
|
|
<< "without any remote session description.";
|
|
return false;
|
|
}
|
|
|
|
if (!candidate) {
|
|
LOG(LS_ERROR) << "ProcessIceMessage: Candidate is NULL.";
|
|
return false;
|
|
}
|
|
|
|
bool valid = false;
|
|
bool ready = ReadyToUseRemoteCandidate(candidate, NULL, &valid);
|
|
if (!valid) {
|
|
return false;
|
|
}
|
|
|
|
// Add this candidate to the remote session description.
|
|
if (!mutable_remote_description()->AddCandidate(candidate)) {
|
|
LOG(LS_ERROR) << "ProcessIceMessage: Candidate cannot be used.";
|
|
return false;
|
|
}
|
|
|
|
if (ready) {
|
|
return UseCandidate(candidate);
|
|
} else {
|
|
LOG(LS_INFO) << "ProcessIceMessage: Not ready to use candidate.";
|
|
return true;
|
|
}
|
|
}
|
|
|
|
bool WebRtcSession::RemoveRemoteIceCandidates(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
if (!remote_description()) {
|
|
LOG(LS_ERROR) << "RemoveRemoteIceCandidates: ICE candidates can't be "
|
|
<< "removed without any remote session description.";
|
|
return false;
|
|
}
|
|
|
|
if (candidates.empty()) {
|
|
LOG(LS_ERROR) << "RemoveRemoteIceCandidates: candidates are empty.";
|
|
return false;
|
|
}
|
|
|
|
size_t number_removed =
|
|
mutable_remote_description()->RemoveCandidates(candidates);
|
|
if (number_removed != candidates.size()) {
|
|
LOG(LS_ERROR) << "RemoveRemoteIceCandidates: Failed to remove candidates. "
|
|
<< "Requested " << candidates.size() << " but only "
|
|
<< number_removed << " are removed.";
|
|
}
|
|
|
|
// Remove the candidates from the transport controller.
|
|
std::string error;
|
|
bool res = transport_controller_->RemoveRemoteCandidates(candidates, &error);
|
|
if (!res && !error.empty()) {
|
|
LOG(LS_ERROR) << "Error when removing remote candidates: " << error;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
cricket::IceConfig WebRtcSession::ParseIceConfig(
|
|
const PeerConnectionInterface::RTCConfiguration& config) const {
|
|
cricket::ContinualGatheringPolicy gathering_policy;
|
|
// TODO(honghaiz): Add the third continual gathering policy in
|
|
// PeerConnectionInterface and map it to GATHER_CONTINUALLY_AND_RECOVER.
|
|
switch (config.continual_gathering_policy) {
|
|
case PeerConnectionInterface::GATHER_ONCE:
|
|
gathering_policy = cricket::GATHER_ONCE;
|
|
break;
|
|
case PeerConnectionInterface::GATHER_CONTINUALLY:
|
|
gathering_policy = cricket::GATHER_CONTINUALLY;
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
gathering_policy = cricket::GATHER_ONCE;
|
|
}
|
|
cricket::IceConfig ice_config;
|
|
ice_config.receiving_timeout = config.ice_connection_receiving_timeout;
|
|
ice_config.prioritize_most_likely_candidate_pairs =
|
|
config.prioritize_most_likely_ice_candidate_pairs;
|
|
ice_config.backup_connection_ping_interval =
|
|
config.ice_backup_candidate_pair_ping_interval;
|
|
ice_config.continual_gathering_policy = gathering_policy;
|
|
ice_config.presume_writable_when_fully_relayed =
|
|
config.presume_writable_when_fully_relayed;
|
|
ice_config.ice_check_min_interval = config.ice_check_min_interval;
|
|
ice_config.regather_all_networks_interval_range =
|
|
config.ice_regather_interval_range;
|
|
return ice_config;
|
|
}
|
|
|
|
void WebRtcSession::SetIceConfig(const cricket::IceConfig& config) {
|
|
transport_controller_->SetIceConfig(config);
|
|
}
|
|
|
|
void WebRtcSession::MaybeStartGathering() {
|
|
transport_controller_->MaybeStartGathering();
|
|
}
|
|
|
|
bool WebRtcSession::GetLocalTrackIdBySsrc(uint32_t ssrc,
|
|
std::string* track_id) {
|
|
if (!local_description()) {
|
|
return false;
|
|
}
|
|
return webrtc::GetTrackIdBySsrc(local_description()->description(), ssrc,
|
|
track_id);
|
|
}
|
|
|
|
bool WebRtcSession::GetRemoteTrackIdBySsrc(uint32_t ssrc,
|
|
std::string* track_id) {
|
|
if (!remote_description()) {
|
|
return false;
|
|
}
|
|
return webrtc::GetTrackIdBySsrc(remote_description()->description(), ssrc,
|
|
track_id);
|
|
}
|
|
|
|
std::string WebRtcSession::BadStateErrMsg(State state) {
|
|
std::ostringstream desc;
|
|
desc << "Called in wrong state: " << GetStateString(state);
|
|
return desc.str();
|
|
}
|
|
|
|
bool WebRtcSession::SendData(const cricket::SendDataParams& params,
|
|
const rtc::CopyOnWriteBuffer& payload,
|
|
cricket::SendDataResult* result) {
|
|
if (!rtp_data_channel_ && !sctp_transport_) {
|
|
LOG(LS_ERROR) << "SendData called when rtp_data_channel_ "
|
|
<< "and sctp_transport_ are NULL.";
|
|
return false;
|
|
}
|
|
return rtp_data_channel_
|
|
? rtp_data_channel_->SendData(params, payload, result)
|
|
: network_thread_->Invoke<bool>(
|
|
RTC_FROM_HERE,
|
|
Bind(&cricket::SctpTransportInternal::SendData,
|
|
sctp_transport_.get(), params, payload, result));
|
|
}
|
|
|
|
bool WebRtcSession::ConnectDataChannel(DataChannel* webrtc_data_channel) {
|
|
if (!rtp_data_channel_ && !sctp_transport_) {
|
|
// Don't log an error here, because DataChannels are expected to call
|
|
// ConnectDataChannel in this state. It's the only way to initially tell
|
|
// whether or not the underlying transport is ready.
|
|
return false;
|
|
}
|
|
if (rtp_data_channel_) {
|
|
rtp_data_channel_->SignalReadyToSendData.connect(
|
|
webrtc_data_channel, &DataChannel::OnChannelReady);
|
|
rtp_data_channel_->SignalDataReceived.connect(webrtc_data_channel,
|
|
&DataChannel::OnDataReceived);
|
|
} else {
|
|
SignalSctpReadyToSendData.connect(webrtc_data_channel,
|
|
&DataChannel::OnChannelReady);
|
|
SignalSctpDataReceived.connect(webrtc_data_channel,
|
|
&DataChannel::OnDataReceived);
|
|
SignalSctpStreamClosedRemotely.connect(
|
|
webrtc_data_channel, &DataChannel::OnStreamClosedRemotely);
|
|
}
|
|
return true;
|
|
}
|
|
|
|
void WebRtcSession::DisconnectDataChannel(DataChannel* webrtc_data_channel) {
|
|
if (!rtp_data_channel_ && !sctp_transport_) {
|
|
LOG(LS_ERROR) << "DisconnectDataChannel called when rtp_data_channel_ and "
|
|
"sctp_transport_ are NULL.";
|
|
return;
|
|
}
|
|
if (rtp_data_channel_) {
|
|
rtp_data_channel_->SignalReadyToSendData.disconnect(webrtc_data_channel);
|
|
rtp_data_channel_->SignalDataReceived.disconnect(webrtc_data_channel);
|
|
} else {
|
|
SignalSctpReadyToSendData.disconnect(webrtc_data_channel);
|
|
SignalSctpDataReceived.disconnect(webrtc_data_channel);
|
|
SignalSctpStreamClosedRemotely.disconnect(webrtc_data_channel);
|
|
}
|
|
}
|
|
|
|
void WebRtcSession::AddSctpDataStream(int sid) {
|
|
if (!sctp_transport_) {
|
|
LOG(LS_ERROR) << "AddSctpDataStream called when sctp_transport_ is NULL.";
|
|
return;
|
|
}
|
|
network_thread_->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::OpenStream,
|
|
sctp_transport_.get(), sid));
|
|
}
|
|
|
|
void WebRtcSession::RemoveSctpDataStream(int sid) {
|
|
if (!sctp_transport_) {
|
|
LOG(LS_ERROR) << "RemoveSctpDataStream called when sctp_transport_ is "
|
|
<< "NULL.";
|
|
return;
|
|
}
|
|
network_thread_->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::ResetStream,
|
|
sctp_transport_.get(), sid));
|
|
}
|
|
|
|
bool WebRtcSession::ReadyToSendData() const {
|
|
return (rtp_data_channel_ && rtp_data_channel_->ready_to_send_data()) ||
|
|
sctp_ready_to_send_data_;
|
|
}
|
|
|
|
std::unique_ptr<SessionStats> WebRtcSession::GetStats_s() {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
ChannelNamePairs channel_name_pairs;
|
|
if (voice_channel()) {
|
|
channel_name_pairs.voice = rtc::Optional<ChannelNamePair>(ChannelNamePair(
|
|
voice_channel()->content_name(), voice_channel()->transport_name()));
|
|
}
|
|
if (video_channel()) {
|
|
channel_name_pairs.video = rtc::Optional<ChannelNamePair>(ChannelNamePair(
|
|
video_channel()->content_name(), video_channel()->transport_name()));
|
|
}
|
|
if (rtp_data_channel()) {
|
|
channel_name_pairs.data = rtc::Optional<ChannelNamePair>(
|
|
ChannelNamePair(rtp_data_channel()->content_name(),
|
|
rtp_data_channel()->transport_name()));
|
|
}
|
|
if (sctp_transport_) {
|
|
RTC_DCHECK(sctp_content_name_);
|
|
RTC_DCHECK(sctp_transport_name_);
|
|
channel_name_pairs.data = rtc::Optional<ChannelNamePair>(
|
|
ChannelNamePair(*sctp_content_name_, *sctp_transport_name_));
|
|
}
|
|
return GetStats(channel_name_pairs);
|
|
}
|
|
|
|
std::unique_ptr<SessionStats> WebRtcSession::GetStats(
|
|
const ChannelNamePairs& channel_name_pairs) {
|
|
if (network_thread()->IsCurrent()) {
|
|
return GetStats_n(channel_name_pairs);
|
|
}
|
|
return network_thread()->Invoke<std::unique_ptr<SessionStats>>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&WebRtcSession::GetStats_n, this, channel_name_pairs));
|
|
}
|
|
|
|
bool WebRtcSession::GetLocalCertificate(
|
|
const std::string& transport_name,
|
|
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
|
|
return transport_controller_->GetLocalCertificate(transport_name,
|
|
certificate);
|
|
}
|
|
|
|
std::unique_ptr<rtc::SSLCertificate> WebRtcSession::GetRemoteSSLCertificate(
|
|
const std::string& transport_name) {
|
|
return transport_controller_->GetRemoteSSLCertificate(transport_name);
|
|
}
|
|
|
|
cricket::DataChannelType WebRtcSession::data_channel_type() const {
|
|
return data_channel_type_;
|
|
}
|
|
|
|
bool WebRtcSession::IceRestartPending(const std::string& content_name) const {
|
|
return pending_ice_restarts_.find(content_name) !=
|
|
pending_ice_restarts_.end();
|
|
}
|
|
|
|
void WebRtcSession::SetNeedsIceRestartFlag() {
|
|
transport_controller_->SetNeedsIceRestartFlag();
|
|
}
|
|
|
|
bool WebRtcSession::NeedsIceRestart(const std::string& content_name) const {
|
|
return transport_controller_->NeedsIceRestart(content_name);
|
|
}
|
|
|
|
void WebRtcSession::OnCertificateReady(
|
|
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
|
|
transport_controller_->SetLocalCertificate(certificate);
|
|
}
|
|
|
|
void WebRtcSession::OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp) {
|
|
SetError(ERROR_TRANSPORT,
|
|
rtcp ? kDtlsSrtpSetupFailureRtcp : kDtlsSrtpSetupFailureRtp);
|
|
}
|
|
|
|
bool WebRtcSession::waiting_for_certificate_for_testing() const {
|
|
return webrtc_session_desc_factory_->waiting_for_certificate_for_testing();
|
|
}
|
|
|
|
const rtc::scoped_refptr<rtc::RTCCertificate>&
|
|
WebRtcSession::certificate_for_testing() {
|
|
return transport_controller_->certificate_for_testing();
|
|
}
|
|
|
|
void WebRtcSession::SetIceConnectionState(
|
|
PeerConnectionInterface::IceConnectionState state) {
|
|
if (ice_connection_state_ == state) {
|
|
return;
|
|
}
|
|
|
|
LOG(LS_INFO) << "Changing IceConnectionState " << ice_connection_state_
|
|
<< " => " << state;
|
|
RTC_DCHECK(ice_connection_state_ !=
|
|
PeerConnectionInterface::kIceConnectionClosed);
|
|
ice_connection_state_ = state;
|
|
if (ice_observer_) {
|
|
ice_observer_->OnIceConnectionStateChange(ice_connection_state_);
|
|
}
|
|
}
|
|
|
|
void WebRtcSession::OnTransportControllerConnectionState(
|
|
cricket::IceConnectionState state) {
|
|
switch (state) {
|
|
case cricket::kIceConnectionConnecting:
|
|
// If the current state is Connected or Completed, then there were
|
|
// writable channels but now there are not, so the next state must
|
|
// be Disconnected.
|
|
// kIceConnectionConnecting is currently used as the default,
|
|
// un-connected state by the TransportController, so its only use is
|
|
// detecting disconnections.
|
|
if (ice_connection_state_ ==
|
|
PeerConnectionInterface::kIceConnectionConnected ||
|
|
ice_connection_state_ ==
|
|
PeerConnectionInterface::kIceConnectionCompleted) {
|
|
SetIceConnectionState(
|
|
PeerConnectionInterface::kIceConnectionDisconnected);
|
|
}
|
|
break;
|
|
case cricket::kIceConnectionFailed:
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionFailed);
|
|
break;
|
|
case cricket::kIceConnectionConnected:
|
|
LOG(LS_INFO) << "Changing to ICE connected state because "
|
|
<< "all transports are writable.";
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
|
|
break;
|
|
case cricket::kIceConnectionCompleted:
|
|
LOG(LS_INFO) << "Changing to ICE completed state because "
|
|
<< "all transports are complete.";
|
|
if (ice_connection_state_ !=
|
|
PeerConnectionInterface::kIceConnectionConnected) {
|
|
// If jumping directly from "checking" to "connected",
|
|
// signal "connected" first.
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
|
|
}
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
|
|
if (metrics_observer_) {
|
|
ReportTransportStats();
|
|
}
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
}
|
|
}
|
|
|
|
void WebRtcSession::OnTransportControllerReceiving(bool receiving) {
|
|
SetIceConnectionReceiving(receiving);
|
|
}
|
|
|
|
void WebRtcSession::SetIceConnectionReceiving(bool receiving) {
|
|
if (ice_connection_receiving_ == receiving) {
|
|
return;
|
|
}
|
|
ice_connection_receiving_ = receiving;
|
|
if (ice_observer_) {
|
|
ice_observer_->OnIceConnectionReceivingChange(receiving);
|
|
}
|
|
}
|
|
|
|
void WebRtcSession::OnTransportControllerCandidatesGathered(
|
|
const std::string& transport_name,
|
|
const cricket::Candidates& candidates) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
int sdp_mline_index;
|
|
if (!GetLocalCandidateMediaIndex(transport_name, &sdp_mline_index)) {
|
|
LOG(LS_ERROR) << "OnTransportControllerCandidatesGathered: content name "
|
|
<< transport_name << " not found";
|
|
return;
|
|
}
|
|
|
|
for (cricket::Candidates::const_iterator citer = candidates.begin();
|
|
citer != candidates.end(); ++citer) {
|
|
// Use transport_name as the candidate media id.
|
|
std::unique_ptr<JsepIceCandidate> candidate(
|
|
new JsepIceCandidate(transport_name, sdp_mline_index, *citer));
|
|
if (local_description()) {
|
|
mutable_local_description()->AddCandidate(candidate.get());
|
|
}
|
|
if (ice_observer_) {
|
|
ice_observer_->OnIceCandidate(std::move(candidate));
|
|
}
|
|
}
|
|
}
|
|
|
|
void WebRtcSession::OnTransportControllerCandidatesRemoved(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
// Sanity check.
|
|
for (const cricket::Candidate& candidate : candidates) {
|
|
if (candidate.transport_name().empty()) {
|
|
LOG(LS_ERROR) << "OnTransportControllerCandidatesRemoved: "
|
|
<< "empty content name in candidate "
|
|
<< candidate.ToString();
|
|
return;
|
|
}
|
|
}
|
|
|
|
if (local_description()) {
|
|
mutable_local_description()->RemoveCandidates(candidates);
|
|
}
|
|
if (ice_observer_) {
|
|
ice_observer_->OnIceCandidatesRemoved(candidates);
|
|
}
|
|
}
|
|
|
|
void WebRtcSession::OnTransportControllerDtlsHandshakeError(
|
|
rtc::SSLHandshakeError error) {
|
|
if (metrics_observer_) {
|
|
metrics_observer_->IncrementEnumCounter(
|
|
webrtc::kEnumCounterDtlsHandshakeError, static_cast<int>(error),
|
|
static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
|
|
}
|
|
}
|
|
|
|
// Enabling voice and video (and RTP data) channels.
|
|
void WebRtcSession::EnableChannels() {
|
|
for (cricket::VoiceChannel* voice_channel : voice_channels_) {
|
|
if (!voice_channel->enabled()) {
|
|
voice_channel->Enable(true);
|
|
}
|
|
}
|
|
|
|
for (cricket::VideoChannel* video_channel : video_channels_) {
|
|
if (!video_channel->enabled()) {
|
|
video_channel->Enable(true);
|
|
}
|
|
}
|
|
|
|
if (rtp_data_channel_ && !rtp_data_channel_->enabled())
|
|
rtp_data_channel_->Enable(true);
|
|
}
|
|
|
|
// Returns the media index for a local ice candidate given the content name.
|
|
bool WebRtcSession::GetLocalCandidateMediaIndex(const std::string& content_name,
|
|
int* sdp_mline_index) {
|
|
if (!local_description() || !sdp_mline_index) {
|
|
return false;
|
|
}
|
|
|
|
bool content_found = false;
|
|
const ContentInfos& contents = local_description()->description()->contents();
|
|
for (size_t index = 0; index < contents.size(); ++index) {
|
|
if (contents[index].name == content_name) {
|
|
*sdp_mline_index = static_cast<int>(index);
|
|
content_found = true;
|
|
break;
|
|
}
|
|
}
|
|
return content_found;
|
|
}
|
|
|
|
bool WebRtcSession::UseCandidatesInSessionDescription(
|
|
const SessionDescriptionInterface* remote_desc) {
|
|
if (!remote_desc) {
|
|
return true;
|
|
}
|
|
bool ret = true;
|
|
|
|
for (size_t m = 0; m < remote_desc->number_of_mediasections(); ++m) {
|
|
const IceCandidateCollection* candidates = remote_desc->candidates(m);
|
|
for (size_t n = 0; n < candidates->count(); ++n) {
|
|
const IceCandidateInterface* candidate = candidates->at(n);
|
|
bool valid = false;
|
|
if (!ReadyToUseRemoteCandidate(candidate, remote_desc, &valid)) {
|
|
if (valid) {
|
|
LOG(LS_INFO) << "UseCandidatesInSessionDescription: Not ready to use "
|
|
<< "candidate.";
|
|
}
|
|
continue;
|
|
}
|
|
ret = UseCandidate(candidate);
|
|
if (!ret) {
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
bool WebRtcSession::UseCandidate(const IceCandidateInterface* candidate) {
|
|
size_t mediacontent_index = static_cast<size_t>(candidate->sdp_mline_index());
|
|
size_t remote_content_size =
|
|
remote_description()->description()->contents().size();
|
|
if (mediacontent_index >= remote_content_size) {
|
|
LOG(LS_ERROR) << "UseCandidate: Invalid candidate media index.";
|
|
return false;
|
|
}
|
|
|
|
cricket::ContentInfo content =
|
|
remote_description()->description()->contents()[mediacontent_index];
|
|
std::vector<cricket::Candidate> candidates;
|
|
candidates.push_back(candidate->candidate());
|
|
// Invoking BaseSession method to handle remote candidates.
|
|
std::string error;
|
|
if (transport_controller_->AddRemoteCandidates(content.name, candidates,
|
|
&error)) {
|
|
// Candidates successfully submitted for checking.
|
|
if (ice_connection_state_ == PeerConnectionInterface::kIceConnectionNew ||
|
|
ice_connection_state_ ==
|
|
PeerConnectionInterface::kIceConnectionDisconnected) {
|
|
// If state is New, then the session has just gotten its first remote ICE
|
|
// candidates, so go to Checking.
|
|
// If state is Disconnected, the session is re-using old candidates or
|
|
// receiving additional ones, so go to Checking.
|
|
// If state is Connected, stay Connected.
|
|
// TODO(bemasc): If state is Connected, and the new candidates are for a
|
|
// newly added transport, then the state actually _should_ move to
|
|
// checking. Add a way to distinguish that case.
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
|
|
}
|
|
// TODO(bemasc): If state is Completed, go back to Connected.
|
|
} else {
|
|
if (!error.empty()) {
|
|
LOG(LS_WARNING) << error;
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
void WebRtcSession::RemoveUnusedChannels(const SessionDescription* desc) {
|
|
// TODO(steveanton): Add support for multiple audio/video channels.
|
|
// Destroy video channel first since it may have a pointer to the
|
|
// voice channel.
|
|
const cricket::ContentInfo* video_info = cricket::GetFirstVideoContent(desc);
|
|
if ((!video_info || video_info->rejected) && video_channel()) {
|
|
RemoveAndDestroyVideoChannel(video_channel());
|
|
}
|
|
|
|
const cricket::ContentInfo* voice_info = cricket::GetFirstAudioContent(desc);
|
|
if ((!voice_info || voice_info->rejected) && voice_channel()) {
|
|
RemoveAndDestroyVoiceChannel(voice_channel());
|
|
}
|
|
|
|
const cricket::ContentInfo* data_info =
|
|
cricket::GetFirstDataContent(desc);
|
|
if (!data_info || data_info->rejected) {
|
|
if (rtp_data_channel_) {
|
|
DestroyDataChannel();
|
|
}
|
|
if (sctp_transport_) {
|
|
SignalDataChannelDestroyed();
|
|
network_thread_->Invoke<void>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&WebRtcSession::DestroySctpTransport_n, this));
|
|
}
|
|
#ifdef HAVE_QUIC
|
|
// Clean up the existing QuicDataTransport and its QuicTransportChannels.
|
|
if (quic_data_transport_) {
|
|
quic_data_transport_.reset();
|
|
}
|
|
#endif
|
|
}
|
|
}
|
|
|
|
// Returns the name of the transport channel when BUNDLE is enabled, or nullptr
|
|
// if the channel is not part of any bundle.
|
|
const std::string* WebRtcSession::GetBundleTransportName(
|
|
const cricket::ContentInfo* content,
|
|
const cricket::ContentGroup* bundle) {
|
|
if (!bundle) {
|
|
return nullptr;
|
|
}
|
|
const std::string* first_content_name = bundle->FirstContentName();
|
|
if (!first_content_name) {
|
|
LOG(LS_WARNING) << "Tried to BUNDLE with no contents.";
|
|
return nullptr;
|
|
}
|
|
if (!bundle->HasContentName(content->name)) {
|
|
LOG(LS_WARNING) << content->name << " is not part of any bundle group";
|
|
return nullptr;
|
|
}
|
|
LOG(LS_INFO) << "Bundling " << content->name << " on " << *first_content_name;
|
|
return first_content_name;
|
|
}
|
|
|
|
bool WebRtcSession::CreateChannels(const SessionDescription* desc) {
|
|
// TODO(steveanton): Add support for multiple audio/video channels.
|
|
const cricket::ContentGroup* bundle_group = nullptr;
|
|
if (bundle_policy_ == PeerConnectionInterface::kBundlePolicyMaxBundle) {
|
|
bundle_group = desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
|
|
if (!bundle_group) {
|
|
LOG(LS_WARNING) << "max-bundle specified without BUNDLE specified";
|
|
return false;
|
|
}
|
|
}
|
|
// Creating the media channels and transport proxies.
|
|
const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(desc);
|
|
if (voice && !voice->rejected && !voice_channel()) {
|
|
if (!CreateVoiceChannel(voice,
|
|
GetBundleTransportName(voice, bundle_group))) {
|
|
LOG(LS_ERROR) << "Failed to create voice channel.";
|
|
return false;
|
|
}
|
|
}
|
|
|
|
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);
|
|
if (video && !video->rejected && !video_channel()) {
|
|
if (!CreateVideoChannel(video,
|
|
GetBundleTransportName(video, bundle_group))) {
|
|
LOG(LS_ERROR) << "Failed to create video channel.";
|
|
return false;
|
|
}
|
|
}
|
|
|
|
const cricket::ContentInfo* data = cricket::GetFirstDataContent(desc);
|
|
if (data_channel_type_ != cricket::DCT_NONE && data && !data->rejected &&
|
|
!rtp_data_channel_ && !sctp_transport_) {
|
|
if (!CreateDataChannel(data, GetBundleTransportName(data, bundle_group))) {
|
|
LOG(LS_ERROR) << "Failed to create data channel.";
|
|
return false;
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcSession::CreateVoiceChannel(const cricket::ContentInfo* content,
|
|
const std::string* bundle_transport) {
|
|
// TODO(steveanton): Check to see if it's safe to create multiple voice
|
|
// channels.
|
|
RTC_DCHECK(voice_channels_.empty());
|
|
|
|
bool require_rtcp_mux =
|
|
rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
|
|
|
|
std::string transport_name =
|
|
bundle_transport ? *bundle_transport : content->name;
|
|
|
|
cricket::DtlsTransportInternal* rtp_dtls_transport =
|
|
transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
|
|
if (!require_rtcp_mux) {
|
|
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
|
|
cricket::VoiceChannel* voice_channel = channel_manager_->CreateVoiceChannel(
|
|
call_, media_config_, rtp_dtls_transport, rtcp_dtls_transport,
|
|
transport_controller_->signaling_thread(), content->name, SrtpRequired(),
|
|
audio_options_);
|
|
if (!voice_channel) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
if (rtcp_dtls_transport) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
return false;
|
|
}
|
|
|
|
voice_channels_.push_back(voice_channel);
|
|
|
|
voice_channel->SignalRtcpMuxFullyActive.connect(
|
|
this, &WebRtcSession::DestroyRtcpTransport_n);
|
|
voice_channel->SignalDtlsSrtpSetupFailure.connect(
|
|
this, &WebRtcSession::OnDtlsSrtpSetupFailure);
|
|
|
|
// TODO(steveanton): This should signal which voice channel was created since
|
|
// we can have multiple.
|
|
SignalVoiceChannelCreated();
|
|
voice_channel->SignalSentPacket.connect(this, &WebRtcSession::OnSentPacket_w);
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content,
|
|
const std::string* bundle_transport) {
|
|
// TODO(steveanton): Check to see if it's safe to create multiple video
|
|
// channels.
|
|
RTC_DCHECK(video_channels_.empty());
|
|
|
|
bool require_rtcp_mux =
|
|
rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
|
|
|
|
std::string transport_name =
|
|
bundle_transport ? *bundle_transport : content->name;
|
|
|
|
cricket::DtlsTransportInternal* rtp_dtls_transport =
|
|
transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
|
|
if (!require_rtcp_mux) {
|
|
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
|
|
cricket::VideoChannel* video_channel = channel_manager_->CreateVideoChannel(
|
|
call_, media_config_, rtp_dtls_transport, rtcp_dtls_transport,
|
|
transport_controller_->signaling_thread(), content->name, SrtpRequired(),
|
|
video_options_);
|
|
|
|
if (!video_channel) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
if (rtcp_dtls_transport) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
return false;
|
|
}
|
|
|
|
video_channels_.push_back(video_channel);
|
|
|
|
video_channel->SignalRtcpMuxFullyActive.connect(
|
|
this, &WebRtcSession::DestroyRtcpTransport_n);
|
|
video_channel->SignalDtlsSrtpSetupFailure.connect(
|
|
this, &WebRtcSession::OnDtlsSrtpSetupFailure);
|
|
|
|
// TODO(steveanton): This should signal which video channel was created since
|
|
// we can have multiple.
|
|
SignalVideoChannelCreated();
|
|
video_channel->SignalSentPacket.connect(this, &WebRtcSession::OnSentPacket_w);
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content,
|
|
const std::string* bundle_transport) {
|
|
const std::string transport_name =
|
|
bundle_transport ? *bundle_transport : content->name;
|
|
#ifdef HAVE_QUIC
|
|
if (data_channel_type_ == cricket::DCT_QUIC) {
|
|
RTC_DCHECK(transport_controller_->quic());
|
|
quic_data_transport_->SetTransports(transport_name);
|
|
return true;
|
|
}
|
|
#endif // HAVE_QUIC
|
|
bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
|
|
if (sctp) {
|
|
if (!sctp_factory_) {
|
|
LOG(LS_ERROR)
|
|
<< "Trying to create SCTP transport, but didn't compile with "
|
|
"SCTP support (HAVE_SCTP)";
|
|
return false;
|
|
}
|
|
if (!network_thread_->Invoke<bool>(
|
|
RTC_FROM_HERE, rtc::Bind(&WebRtcSession::CreateSctpTransport_n,
|
|
this, content->name, transport_name))) {
|
|
return false;
|
|
};
|
|
} else {
|
|
bool require_rtcp_mux =
|
|
rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
|
|
|
|
std::string transport_name =
|
|
bundle_transport ? *bundle_transport : content->name;
|
|
cricket::DtlsTransportInternal* rtp_dtls_transport =
|
|
transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
|
|
if (!require_rtcp_mux) {
|
|
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
|
|
rtp_data_channel_.reset(channel_manager_->CreateRtpDataChannel(
|
|
media_config_, rtp_dtls_transport, rtcp_dtls_transport,
|
|
transport_controller_->signaling_thread(), content->name,
|
|
SrtpRequired()));
|
|
|
|
if (!rtp_data_channel_) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
if (rtcp_dtls_transport) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
return false;
|
|
}
|
|
|
|
rtp_data_channel_->SignalRtcpMuxFullyActive.connect(
|
|
this, &WebRtcSession::DestroyRtcpTransport_n);
|
|
rtp_data_channel_->SignalDtlsSrtpSetupFailure.connect(
|
|
this, &WebRtcSession::OnDtlsSrtpSetupFailure);
|
|
rtp_data_channel_->SignalSentPacket.connect(this,
|
|
&WebRtcSession::OnSentPacket_w);
|
|
}
|
|
|
|
SignalDataChannelCreated();
|
|
|
|
return true;
|
|
}
|
|
|
|
Call::Stats WebRtcSession::GetCallStats() {
|
|
if (!worker_thread()->IsCurrent()) {
|
|
return worker_thread()->Invoke<Call::Stats>(
|
|
RTC_FROM_HERE, rtc::Bind(&WebRtcSession::GetCallStats, this));
|
|
}
|
|
if (!call_)
|
|
return Call::Stats();
|
|
return call_->GetStats();
|
|
}
|
|
|
|
std::unique_ptr<SessionStats> WebRtcSession::GetStats_n(
|
|
const ChannelNamePairs& channel_name_pairs) {
|
|
RTC_DCHECK(network_thread()->IsCurrent());
|
|
std::unique_ptr<SessionStats> session_stats(new SessionStats());
|
|
for (const auto channel_name_pair : { &channel_name_pairs.voice,
|
|
&channel_name_pairs.video,
|
|
&channel_name_pairs.data }) {
|
|
if (*channel_name_pair) {
|
|
cricket::TransportStats transport_stats;
|
|
if (!transport_controller_->GetStats((*channel_name_pair)->transport_name,
|
|
&transport_stats)) {
|
|
return nullptr;
|
|
}
|
|
session_stats->proxy_to_transport[(*channel_name_pair)->content_name] =
|
|
(*channel_name_pair)->transport_name;
|
|
session_stats->transport_stats[(*channel_name_pair)->transport_name] =
|
|
std::move(transport_stats);
|
|
}
|
|
}
|
|
return session_stats;
|
|
}
|
|
|
|
bool WebRtcSession::CreateSctpTransport_n(const std::string& content_name,
|
|
const std::string& transport_name) {
|
|
RTC_DCHECK(network_thread_->IsCurrent());
|
|
RTC_DCHECK(sctp_factory_);
|
|
cricket::DtlsTransportInternal* tc =
|
|
transport_controller_->CreateDtlsTransport_n(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
sctp_transport_ = sctp_factory_->CreateSctpTransport(tc);
|
|
RTC_DCHECK(sctp_transport_);
|
|
sctp_invoker_.reset(new rtc::AsyncInvoker());
|
|
sctp_transport_->SignalReadyToSendData.connect(
|
|
this, &WebRtcSession::OnSctpTransportReadyToSendData_n);
|
|
sctp_transport_->SignalDataReceived.connect(
|
|
this, &WebRtcSession::OnSctpTransportDataReceived_n);
|
|
sctp_transport_->SignalStreamClosedRemotely.connect(
|
|
this, &WebRtcSession::OnSctpStreamClosedRemotely_n);
|
|
sctp_transport_name_ = rtc::Optional<std::string>(transport_name);
|
|
sctp_content_name_ = rtc::Optional<std::string>(content_name);
|
|
return true;
|
|
}
|
|
|
|
void WebRtcSession::ChangeSctpTransport_n(const std::string& transport_name) {
|
|
RTC_DCHECK(network_thread_->IsCurrent());
|
|
RTC_DCHECK(sctp_transport_);
|
|
RTC_DCHECK(sctp_transport_name_);
|
|
std::string old_sctp_transport_name = *sctp_transport_name_;
|
|
sctp_transport_name_ = rtc::Optional<std::string>(transport_name);
|
|
cricket::DtlsTransportInternal* tc =
|
|
transport_controller_->CreateDtlsTransport_n(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
sctp_transport_->SetTransportChannel(tc);
|
|
transport_controller_->DestroyDtlsTransport_n(
|
|
old_sctp_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
}
|
|
|
|
void WebRtcSession::DestroySctpTransport_n() {
|
|
RTC_DCHECK(network_thread_->IsCurrent());
|
|
sctp_transport_.reset(nullptr);
|
|
sctp_content_name_.reset();
|
|
sctp_transport_name_.reset();
|
|
sctp_invoker_.reset(nullptr);
|
|
sctp_ready_to_send_data_ = false;
|
|
}
|
|
|
|
void WebRtcSession::OnSctpTransportReadyToSendData_n() {
|
|
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
|
|
RTC_DCHECK(network_thread_->IsCurrent());
|
|
sctp_invoker_->AsyncInvoke<void>(
|
|
RTC_FROM_HERE, signaling_thread_,
|
|
rtc::Bind(&WebRtcSession::OnSctpTransportReadyToSendData_s, this, true));
|
|
}
|
|
|
|
void WebRtcSession::OnSctpTransportReadyToSendData_s(bool ready) {
|
|
RTC_DCHECK(signaling_thread_->IsCurrent());
|
|
sctp_ready_to_send_data_ = ready;
|
|
SignalSctpReadyToSendData(ready);
|
|
}
|
|
|
|
void WebRtcSession::OnSctpTransportDataReceived_n(
|
|
const cricket::ReceiveDataParams& params,
|
|
const rtc::CopyOnWriteBuffer& payload) {
|
|
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
|
|
RTC_DCHECK(network_thread_->IsCurrent());
|
|
sctp_invoker_->AsyncInvoke<void>(
|
|
RTC_FROM_HERE, signaling_thread_,
|
|
rtc::Bind(&WebRtcSession::OnSctpTransportDataReceived_s, this, params,
|
|
payload));
|
|
}
|
|
|
|
void WebRtcSession::OnSctpTransportDataReceived_s(
|
|
const cricket::ReceiveDataParams& params,
|
|
const rtc::CopyOnWriteBuffer& payload) {
|
|
RTC_DCHECK(signaling_thread_->IsCurrent());
|
|
if (params.type == cricket::DMT_CONTROL && IsOpenMessage(payload)) {
|
|
// Received OPEN message; parse and signal that a new data channel should
|
|
// be created.
|
|
std::string label;
|
|
InternalDataChannelInit config;
|
|
config.id = params.ssrc;
|
|
if (!ParseDataChannelOpenMessage(payload, &label, &config)) {
|
|
LOG(LS_WARNING) << "Failed to parse the OPEN message for sid "
|
|
<< params.ssrc;
|
|
return;
|
|
}
|
|
config.open_handshake_role = InternalDataChannelInit::kAcker;
|
|
SignalDataChannelOpenMessage(label, config);
|
|
} else {
|
|
// Otherwise just forward the signal.
|
|
SignalSctpDataReceived(params, payload);
|
|
}
|
|
}
|
|
|
|
void WebRtcSession::OnSctpStreamClosedRemotely_n(int sid) {
|
|
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
|
|
RTC_DCHECK(network_thread_->IsCurrent());
|
|
sctp_invoker_->AsyncInvoke<void>(
|
|
RTC_FROM_HERE, signaling_thread_,
|
|
rtc::Bind(&sigslot::signal1<int>::operator(),
|
|
&SignalSctpStreamClosedRemotely, sid));
|
|
}
|
|
|
|
// Returns false if bundle is enabled and rtcp_mux is disabled.
|
|
bool WebRtcSession::ValidateBundleSettings(const SessionDescription* desc) {
|
|
bool bundle_enabled = desc->HasGroup(cricket::GROUP_TYPE_BUNDLE);
|
|
if (!bundle_enabled)
|
|
return true;
|
|
|
|
const cricket::ContentGroup* bundle_group =
|
|
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
|
|
RTC_DCHECK(bundle_group != NULL);
|
|
|
|
const cricket::ContentInfos& contents = desc->contents();
|
|
for (cricket::ContentInfos::const_iterator citer = contents.begin();
|
|
citer != contents.end(); ++citer) {
|
|
const cricket::ContentInfo* content = (&*citer);
|
|
RTC_DCHECK(content != NULL);
|
|
if (bundle_group->HasContentName(content->name) &&
|
|
!content->rejected && content->type == cricket::NS_JINGLE_RTP) {
|
|
if (!HasRtcpMuxEnabled(content))
|
|
return false;
|
|
}
|
|
}
|
|
// RTCP-MUX is enabled in all the contents.
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcSession::HasRtcpMuxEnabled(
|
|
const cricket::ContentInfo* content) {
|
|
const cricket::MediaContentDescription* description =
|
|
static_cast<cricket::MediaContentDescription*>(content->description);
|
|
return description->rtcp_mux();
|
|
}
|
|
|
|
bool WebRtcSession::ValidateSessionDescription(
|
|
const SessionDescriptionInterface* sdesc,
|
|
cricket::ContentSource source, std::string* err_desc) {
|
|
std::string type;
|
|
if (error() != ERROR_NONE) {
|
|
return BadSdp(source, type, GetSessionErrorMsg(), err_desc);
|
|
}
|
|
|
|
if (!sdesc || !sdesc->description()) {
|
|
return BadSdp(source, type, kInvalidSdp, err_desc);
|
|
}
|
|
|
|
type = sdesc->type();
|
|
Action action = GetAction(sdesc->type());
|
|
if (source == cricket::CS_LOCAL) {
|
|
if (!ExpectSetLocalDescription(action))
|
|
return BadLocalSdp(type, BadStateErrMsg(state()), err_desc);
|
|
} else {
|
|
if (!ExpectSetRemoteDescription(action))
|
|
return BadRemoteSdp(type, BadStateErrMsg(state()), err_desc);
|
|
}
|
|
|
|
// Verify crypto settings.
|
|
std::string crypto_error;
|
|
if ((webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED ||
|
|
dtls_enabled_) &&
|
|
!VerifyCrypto(sdesc->description(), dtls_enabled_, &crypto_error)) {
|
|
return BadSdp(source, type, crypto_error, err_desc);
|
|
}
|
|
|
|
// Verify ice-ufrag and ice-pwd.
|
|
if (!VerifyIceUfragPwdPresent(sdesc->description())) {
|
|
return BadSdp(source, type, kSdpWithoutIceUfragPwd, err_desc);
|
|
}
|
|
|
|
if (!ValidateBundleSettings(sdesc->description())) {
|
|
return BadSdp(source, type, kBundleWithoutRtcpMux, err_desc);
|
|
}
|
|
|
|
// TODO(skvlad): When the local rtcp-mux policy is Require, reject any
|
|
// m-lines that do not rtcp-mux enabled.
|
|
|
|
// Verify m-lines in Answer when compared against Offer.
|
|
if (action == kAnswer || action == kPrAnswer) {
|
|
const cricket::SessionDescription* offer_desc =
|
|
(source == cricket::CS_LOCAL) ? remote_description()->description()
|
|
: local_description()->description();
|
|
if (!MediaSectionsHaveSameCount(sdesc->description(), offer_desc) ||
|
|
!MediaSectionsInSameOrder(sdesc->description(), offer_desc)) {
|
|
return BadAnswerSdp(source, kMlineMismatchInAnswer, err_desc);
|
|
}
|
|
} else {
|
|
// The re-offers should respect the order of m= sections in current local
|
|
// description. See RFC3264 Section 8 paragraph 4 for more details.
|
|
if (local_description() &&
|
|
!MediaSectionsInSameOrder(sdesc->description(),
|
|
local_description()->description())) {
|
|
return BadOfferSdp(source, kMlineMismatchInSubsequentOffer, err_desc);
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool WebRtcSession::ExpectSetLocalDescription(Action action) {
|
|
return ((action == kOffer && state() == STATE_INIT) ||
|
|
// update local offer
|
|
(action == kOffer && state() == STATE_SENTOFFER) ||
|
|
// update the current ongoing session.
|
|
(action == kOffer && state() == STATE_INPROGRESS) ||
|
|
// accept remote offer
|
|
(action == kAnswer && state() == STATE_RECEIVEDOFFER) ||
|
|
(action == kAnswer && state() == STATE_SENTPRANSWER) ||
|
|
(action == kPrAnswer && state() == STATE_RECEIVEDOFFER) ||
|
|
(action == kPrAnswer && state() == STATE_SENTPRANSWER));
|
|
}
|
|
|
|
bool WebRtcSession::ExpectSetRemoteDescription(Action action) {
|
|
return ((action == kOffer && state() == STATE_INIT) ||
|
|
// update remote offer
|
|
(action == kOffer && state() == STATE_RECEIVEDOFFER) ||
|
|
// update the current ongoing session
|
|
(action == kOffer && state() == STATE_INPROGRESS) ||
|
|
// accept local offer
|
|
(action == kAnswer && state() == STATE_SENTOFFER) ||
|
|
(action == kAnswer && state() == STATE_RECEIVEDPRANSWER) ||
|
|
(action == kPrAnswer && state() == STATE_SENTOFFER) ||
|
|
(action == kPrAnswer && state() == STATE_RECEIVEDPRANSWER));
|
|
}
|
|
|
|
std::string WebRtcSession::GetSessionErrorMsg() {
|
|
std::ostringstream desc;
|
|
desc << kSessionError << GetErrorCodeString(error()) << ". ";
|
|
desc << kSessionErrorDesc << error_desc() << ".";
|
|
return desc.str();
|
|
}
|
|
|
|
// We need to check the local/remote description for the Transport instead of
|
|
// the session, because a new Transport added during renegotiation may have
|
|
// them unset while the session has them set from the previous negotiation.
|
|
// Not doing so may trigger the auto generation of transport description and
|
|
// mess up DTLS identity information, ICE credential, etc.
|
|
bool WebRtcSession::ReadyToUseRemoteCandidate(
|
|
const IceCandidateInterface* candidate,
|
|
const SessionDescriptionInterface* remote_desc,
|
|
bool* valid) {
|
|
*valid = true;
|
|
|
|
const SessionDescriptionInterface* current_remote_desc =
|
|
remote_desc ? remote_desc : remote_description();
|
|
|
|
if (!current_remote_desc) {
|
|
return false;
|
|
}
|
|
|
|
size_t mediacontent_index =
|
|
static_cast<size_t>(candidate->sdp_mline_index());
|
|
size_t remote_content_size =
|
|
current_remote_desc->description()->contents().size();
|
|
if (mediacontent_index >= remote_content_size) {
|
|
LOG(LS_ERROR) << "ReadyToUseRemoteCandidate: Invalid candidate media index "
|
|
<< mediacontent_index;
|
|
|
|
*valid = false;
|
|
return false;
|
|
}
|
|
|
|
cricket::ContentInfo content =
|
|
current_remote_desc->description()->contents()[mediacontent_index];
|
|
|
|
const std::string transport_name = GetTransportName(content.name);
|
|
if (transport_name.empty()) {
|
|
return false;
|
|
}
|
|
return transport_controller_->ReadyForRemoteCandidates(transport_name);
|
|
}
|
|
|
|
bool WebRtcSession::SrtpRequired() const {
|
|
return dtls_enabled_ ||
|
|
webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED;
|
|
}
|
|
|
|
void WebRtcSession::OnTransportControllerGatheringState(
|
|
cricket::IceGatheringState state) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (state == cricket::kIceGatheringGathering) {
|
|
if (ice_observer_) {
|
|
ice_observer_->OnIceGatheringChange(
|
|
PeerConnectionInterface::kIceGatheringGathering);
|
|
}
|
|
} else if (state == cricket::kIceGatheringComplete) {
|
|
if (ice_observer_) {
|
|
ice_observer_->OnIceGatheringChange(
|
|
PeerConnectionInterface::kIceGatheringComplete);
|
|
}
|
|
}
|
|
}
|
|
|
|
void WebRtcSession::ReportTransportStats() {
|
|
// Use a set so we don't report the same stats twice if two channels share
|
|
// a transport.
|
|
std::set<std::string> transport_names;
|
|
if (voice_channel()) {
|
|
transport_names.insert(voice_channel()->transport_name());
|
|
}
|
|
if (video_channel()) {
|
|
transport_names.insert(video_channel()->transport_name());
|
|
}
|
|
if (rtp_data_channel()) {
|
|
transport_names.insert(rtp_data_channel()->transport_name());
|
|
}
|
|
if (sctp_transport_name_) {
|
|
transport_names.insert(*sctp_transport_name_);
|
|
}
|
|
for (const auto& name : transport_names) {
|
|
cricket::TransportStats stats;
|
|
if (transport_controller_->GetStats(name, &stats)) {
|
|
ReportBestConnectionState(stats);
|
|
ReportNegotiatedCiphers(stats);
|
|
}
|
|
}
|
|
}
|
|
// Walk through the ConnectionInfos to gather best connection usage
|
|
// for IPv4 and IPv6.
|
|
void WebRtcSession::ReportBestConnectionState(
|
|
const cricket::TransportStats& stats) {
|
|
RTC_DCHECK(metrics_observer_ != NULL);
|
|
for (cricket::TransportChannelStatsList::const_iterator it =
|
|
stats.channel_stats.begin();
|
|
it != stats.channel_stats.end(); ++it) {
|
|
for (cricket::ConnectionInfos::const_iterator it_info =
|
|
it->connection_infos.begin();
|
|
it_info != it->connection_infos.end(); ++it_info) {
|
|
if (!it_info->best_connection) {
|
|
continue;
|
|
}
|
|
|
|
PeerConnectionEnumCounterType type = kPeerConnectionEnumCounterMax;
|
|
const cricket::Candidate& local = it_info->local_candidate;
|
|
const cricket::Candidate& remote = it_info->remote_candidate;
|
|
|
|
// Increment the counter for IceCandidatePairType.
|
|
if (local.protocol() == cricket::TCP_PROTOCOL_NAME ||
|
|
(local.type() == RELAY_PORT_TYPE &&
|
|
local.relay_protocol() == cricket::TCP_PROTOCOL_NAME)) {
|
|
type = kEnumCounterIceCandidatePairTypeTcp;
|
|
} else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
|
|
type = kEnumCounterIceCandidatePairTypeUdp;
|
|
} else {
|
|
RTC_CHECK(0);
|
|
}
|
|
metrics_observer_->IncrementEnumCounter(
|
|
type, GetIceCandidatePairCounter(local, remote),
|
|
kIceCandidatePairMax);
|
|
|
|
// Increment the counter for IP type.
|
|
if (local.address().family() == AF_INET) {
|
|
metrics_observer_->IncrementEnumCounter(
|
|
kEnumCounterAddressFamily, kBestConnections_IPv4,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
|
|
} else if (local.address().family() == AF_INET6) {
|
|
metrics_observer_->IncrementEnumCounter(
|
|
kEnumCounterAddressFamily, kBestConnections_IPv6,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
} else {
|
|
RTC_CHECK(0);
|
|
}
|
|
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
void WebRtcSession::ReportNegotiatedCiphers(
|
|
const cricket::TransportStats& stats) {
|
|
RTC_DCHECK(metrics_observer_ != NULL);
|
|
if (!dtls_enabled_ || stats.channel_stats.empty()) {
|
|
return;
|
|
}
|
|
|
|
int srtp_crypto_suite = stats.channel_stats[0].srtp_crypto_suite;
|
|
int ssl_cipher_suite = stats.channel_stats[0].ssl_cipher_suite;
|
|
if (srtp_crypto_suite == rtc::SRTP_INVALID_CRYPTO_SUITE &&
|
|
ssl_cipher_suite == rtc::TLS_NULL_WITH_NULL_NULL) {
|
|
return;
|
|
}
|
|
|
|
PeerConnectionEnumCounterType srtp_counter_type;
|
|
PeerConnectionEnumCounterType ssl_counter_type;
|
|
if (stats.transport_name == cricket::CN_AUDIO) {
|
|
srtp_counter_type = kEnumCounterAudioSrtpCipher;
|
|
ssl_counter_type = kEnumCounterAudioSslCipher;
|
|
} else if (stats.transport_name == cricket::CN_VIDEO) {
|
|
srtp_counter_type = kEnumCounterVideoSrtpCipher;
|
|
ssl_counter_type = kEnumCounterVideoSslCipher;
|
|
} else if (stats.transport_name == cricket::CN_DATA) {
|
|
srtp_counter_type = kEnumCounterDataSrtpCipher;
|
|
ssl_counter_type = kEnumCounterDataSslCipher;
|
|
} else {
|
|
RTC_NOTREACHED();
|
|
return;
|
|
}
|
|
|
|
if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
|
|
metrics_observer_->IncrementSparseEnumCounter(srtp_counter_type,
|
|
srtp_crypto_suite);
|
|
}
|
|
if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
|
|
metrics_observer_->IncrementSparseEnumCounter(ssl_counter_type,
|
|
ssl_cipher_suite);
|
|
}
|
|
}
|
|
|
|
void WebRtcSession::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
|
|
RTC_DCHECK(worker_thread()->IsCurrent());
|
|
RTC_DCHECK(call_);
|
|
call_->OnSentPacket(sent_packet);
|
|
}
|
|
|
|
const std::string WebRtcSession::GetTransportName(
|
|
const std::string& content_name) {
|
|
cricket::BaseChannel* channel = GetChannel(content_name);
|
|
if (!channel) {
|
|
#ifdef HAVE_QUIC
|
|
if (data_channel_type_ == cricket::DCT_QUIC && quic_data_transport_ &&
|
|
content_name == quic_data_transport_->transport_name()) {
|
|
return quic_data_transport_->transport_name();
|
|
}
|
|
#endif
|
|
if (sctp_transport_) {
|
|
RTC_DCHECK(sctp_content_name_);
|
|
RTC_DCHECK(sctp_transport_name_);
|
|
if (content_name == *sctp_content_name_) {
|
|
return *sctp_transport_name_;
|
|
}
|
|
}
|
|
// Return an empty string if failed to retrieve the transport name.
|
|
return "";
|
|
}
|
|
return channel->transport_name();
|
|
}
|
|
|
|
void WebRtcSession::DestroyRtcpTransport_n(const std::string& transport_name) {
|
|
RTC_DCHECK(network_thread()->IsCurrent());
|
|
transport_controller_->DestroyDtlsTransport_n(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
|
|
void WebRtcSession::RemoveAndDestroyVideoChannel(
|
|
cricket::VideoChannel* video_channel) {
|
|
auto it =
|
|
std::find(video_channels_.begin(), video_channels_.end(), video_channel);
|
|
RTC_DCHECK(it != video_channels_.end());
|
|
if (it == video_channels_.end()) {
|
|
return;
|
|
}
|
|
video_channels_.erase(it);
|
|
DestroyVideoChannel(video_channel);
|
|
}
|
|
|
|
void WebRtcSession::DestroyVideoChannel(cricket::VideoChannel* video_channel) {
|
|
// TODO(steveanton): This should take an identifier for the video channel
|
|
// since we now support more than one.
|
|
SignalVideoChannelDestroyed();
|
|
RTC_DCHECK(video_channel->rtp_dtls_transport());
|
|
const std::string transport_name =
|
|
video_channel->rtp_dtls_transport()->transport_name();
|
|
const bool need_to_delete_rtcp =
|
|
(video_channel->rtcp_dtls_transport() != nullptr);
|
|
// The above need to be cached before destroying the video channel so that we
|
|
// do not access uninitialized memory.
|
|
channel_manager_->DestroyVideoChannel(video_channel);
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
if (need_to_delete_rtcp) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
}
|
|
|
|
void WebRtcSession::RemoveAndDestroyVoiceChannel(
|
|
cricket::VoiceChannel* voice_channel) {
|
|
auto it =
|
|
std::find(voice_channels_.begin(), voice_channels_.end(), voice_channel);
|
|
RTC_DCHECK(it != voice_channels_.end());
|
|
if (it == voice_channels_.end()) {
|
|
return;
|
|
}
|
|
voice_channels_.erase(it);
|
|
DestroyVoiceChannel(voice_channel);
|
|
}
|
|
|
|
void WebRtcSession::DestroyVoiceChannel(cricket::VoiceChannel* voice_channel) {
|
|
// TODO(steveanton): This should take an identifier for the voice channel
|
|
// since we now support more than one.
|
|
SignalVoiceChannelDestroyed();
|
|
RTC_DCHECK(voice_channel->rtp_dtls_transport());
|
|
const std::string transport_name =
|
|
voice_channel->rtp_dtls_transport()->transport_name();
|
|
const bool need_to_delete_rtcp =
|
|
(voice_channel->rtcp_dtls_transport() != nullptr);
|
|
// The above need to be cached before destroying the video channel so that we
|
|
// do not access uninitialized memory.
|
|
channel_manager_->DestroyVoiceChannel(voice_channel);
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
if (need_to_delete_rtcp) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
}
|
|
|
|
void WebRtcSession::DestroyDataChannel() {
|
|
SignalDataChannelDestroyed();
|
|
RTC_DCHECK(rtp_data_channel_->rtp_dtls_transport());
|
|
std::string transport_name;
|
|
transport_name = rtp_data_channel_->rtp_dtls_transport()->transport_name();
|
|
bool need_to_delete_rtcp =
|
|
(rtp_data_channel_->rtcp_dtls_transport() != nullptr);
|
|
channel_manager_->DestroyRtpDataChannel(rtp_data_channel_.release());
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
if (need_to_delete_rtcp) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
}
|
|
} // namespace webrtc
|