2174 lines
75 KiB
C++
2174 lines
75 KiB
C++
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/*
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* Copyright 2004 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 "webrtc/pc/mediasession.h"
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#include <algorithm> // For std::find_if, std::sort.
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#include <functional>
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#include <map>
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#include <memory>
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#include <set>
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#include <unordered_map>
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#include <utility>
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#include "webrtc/base/helpers.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/base/stringutils.h"
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#include "webrtc/media/base/cryptoparams.h"
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#include "webrtc/media/base/mediaconstants.h"
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#include "webrtc/p2p/base/p2pconstants.h"
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#include "webrtc/pc/channelmanager.h"
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#include "webrtc/pc/srtpfilter.h"
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#ifdef HAVE_SCTP
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#include "webrtc/media/sctp/sctpdataengine.h"
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#else
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static const uint32_t kMaxSctpSid = 1023;
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#endif
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namespace {
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const char kInline[] = "inline:";
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void GetSupportedCryptoSuiteNames(void (*func)(std::vector<int>*),
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std::vector<std::string>* names) {
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#ifdef HAVE_SRTP
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std::vector<int> crypto_suites;
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func(&crypto_suites);
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for (const auto crypto : crypto_suites) {
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names->push_back(rtc::SrtpCryptoSuiteToName(crypto));
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}
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#endif
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}
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} // namespace
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namespace cricket {
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// RTP Profile names
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// http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xml
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// RFC4585
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const char kMediaProtocolAvpf[] = "RTP/AVPF";
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// RFC5124
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const char kMediaProtocolDtlsSavpf[] = "UDP/TLS/RTP/SAVPF";
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// We always generate offers with "UDP/TLS/RTP/SAVPF" when using DTLS-SRTP,
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// but we tolerate "RTP/SAVPF" in offers we receive, for compatibility.
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const char kMediaProtocolSavpf[] = "RTP/SAVPF";
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const char kMediaProtocolRtpPrefix[] = "RTP/";
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const char kMediaProtocolSctp[] = "SCTP";
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const char kMediaProtocolDtlsSctp[] = "DTLS/SCTP";
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const char kMediaProtocolUdpDtlsSctp[] = "UDP/DTLS/SCTP";
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const char kMediaProtocolTcpDtlsSctp[] = "TCP/DTLS/SCTP";
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RtpTransceiverDirection RtpTransceiverDirection::FromMediaContentDirection(
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MediaContentDirection md) {
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const bool send = (md == MD_SENDRECV || md == MD_SENDONLY);
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const bool recv = (md == MD_SENDRECV || md == MD_RECVONLY);
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return RtpTransceiverDirection(send, recv);
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}
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MediaContentDirection RtpTransceiverDirection::ToMediaContentDirection() const {
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if (send && recv) {
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return MD_SENDRECV;
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} else if (send) {
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return MD_SENDONLY;
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} else if (recv) {
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return MD_RECVONLY;
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}
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return MD_INACTIVE;
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}
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RtpTransceiverDirection
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NegotiateRtpTransceiverDirection(RtpTransceiverDirection offer,
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RtpTransceiverDirection wants) {
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return RtpTransceiverDirection(offer.recv && wants.send,
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offer.send && wants.recv);
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}
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static bool IsMediaContentOfType(const ContentInfo* content,
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MediaType media_type) {
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if (!IsMediaContent(content)) {
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return false;
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}
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const MediaContentDescription* mdesc =
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static_cast<const MediaContentDescription*>(content->description);
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return mdesc && mdesc->type() == media_type;
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}
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static bool CreateCryptoParams(int tag, const std::string& cipher,
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CryptoParams *out) {
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std::string key;
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key.reserve(SRTP_MASTER_KEY_BASE64_LEN);
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if (!rtc::CreateRandomString(SRTP_MASTER_KEY_BASE64_LEN, &key)) {
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return false;
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}
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out->tag = tag;
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out->cipher_suite = cipher;
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out->key_params = kInline;
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out->key_params += key;
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return true;
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}
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#ifdef HAVE_SRTP
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static bool AddCryptoParams(const std::string& cipher_suite,
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CryptoParamsVec *out) {
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int size = static_cast<int>(out->size());
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out->resize(size + 1);
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return CreateCryptoParams(size, cipher_suite, &out->at(size));
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}
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void AddMediaCryptos(const CryptoParamsVec& cryptos,
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MediaContentDescription* media) {
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for (CryptoParamsVec::const_iterator crypto = cryptos.begin();
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crypto != cryptos.end(); ++crypto) {
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media->AddCrypto(*crypto);
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}
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}
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bool CreateMediaCryptos(const std::vector<std::string>& crypto_suites,
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MediaContentDescription* media) {
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CryptoParamsVec cryptos;
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for (std::vector<std::string>::const_iterator it = crypto_suites.begin();
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it != crypto_suites.end(); ++it) {
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if (!AddCryptoParams(*it, &cryptos)) {
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return false;
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}
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}
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AddMediaCryptos(cryptos, media);
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return true;
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}
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#endif
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const CryptoParamsVec* GetCryptos(const MediaContentDescription* media) {
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if (!media) {
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return NULL;
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}
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return &media->cryptos();
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}
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bool FindMatchingCrypto(const CryptoParamsVec& cryptos,
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const CryptoParams& crypto,
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CryptoParams* out) {
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for (CryptoParamsVec::const_iterator it = cryptos.begin();
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it != cryptos.end(); ++it) {
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if (crypto.Matches(*it)) {
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*out = *it;
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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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// For audio, HMAC 32 is prefered because of the low overhead.
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void GetSupportedAudioCryptoSuites(std::vector<int>* crypto_suites) {
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#ifdef HAVE_SRTP
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crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_32);
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crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
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#endif
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}
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void GetSupportedAudioCryptoSuiteNames(
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std::vector<std::string>* crypto_suite_names) {
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GetSupportedCryptoSuiteNames(GetSupportedAudioCryptoSuites,
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crypto_suite_names);
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}
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void GetSupportedVideoCryptoSuites(std::vector<int>* crypto_suites) {
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GetDefaultSrtpCryptoSuites(crypto_suites);
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}
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void GetSupportedVideoCryptoSuiteNames(
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std::vector<std::string>* crypto_suite_names) {
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GetSupportedCryptoSuiteNames(GetSupportedVideoCryptoSuites,
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crypto_suite_names);
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}
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void GetSupportedDataCryptoSuites(std::vector<int>* crypto_suites) {
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GetDefaultSrtpCryptoSuites(crypto_suites);
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}
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void GetSupportedDataCryptoSuiteNames(
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std::vector<std::string>* crypto_suite_names) {
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GetSupportedCryptoSuiteNames(GetSupportedDataCryptoSuites,
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crypto_suite_names);
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}
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void GetDefaultSrtpCryptoSuites(std::vector<int>* crypto_suites) {
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#ifdef HAVE_SRTP
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crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
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#endif
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}
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void GetDefaultSrtpCryptoSuiteNames(
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std::vector<std::string>* crypto_suite_names) {
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GetSupportedCryptoSuiteNames(GetDefaultSrtpCryptoSuites, crypto_suite_names);
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}
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// For video support only 80-bit SHA1 HMAC. For audio 32-bit HMAC is
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// tolerated unless bundle is enabled because it is low overhead. Pick the
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// crypto in the list that is supported.
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static bool SelectCrypto(const MediaContentDescription* offer,
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bool bundle,
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CryptoParams *crypto) {
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bool audio = offer->type() == MEDIA_TYPE_AUDIO;
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const CryptoParamsVec& cryptos = offer->cryptos();
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for (CryptoParamsVec::const_iterator i = cryptos.begin();
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i != cryptos.end(); ++i) {
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if (rtc::CS_AES_CM_128_HMAC_SHA1_80 == i->cipher_suite ||
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(rtc::CS_AES_CM_128_HMAC_SHA1_32 == i->cipher_suite && audio &&
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!bundle)) {
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return CreateCryptoParams(i->tag, i->cipher_suite, crypto);
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}
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}
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return false;
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}
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// Generate random SSRC values that are not already present in |params_vec|.
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// The generated values are added to |ssrcs|.
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// |num_ssrcs| is the number of the SSRC will be generated.
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static void GenerateSsrcs(const StreamParamsVec& params_vec,
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int num_ssrcs,
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std::vector<uint32_t>* ssrcs) {
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for (int i = 0; i < num_ssrcs; i++) {
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uint32_t candidate;
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do {
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candidate = rtc::CreateRandomNonZeroId();
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} while (GetStreamBySsrc(params_vec, candidate) ||
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std::count(ssrcs->begin(), ssrcs->end(), candidate) > 0);
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ssrcs->push_back(candidate);
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}
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}
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// Returns false if we exhaust the range of SIDs.
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static bool GenerateSctpSid(const StreamParamsVec& params_vec, uint32_t* sid) {
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if (params_vec.size() > kMaxSctpSid) {
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LOG(LS_WARNING) <<
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"Could not generate an SCTP SID: too many SCTP streams.";
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return false;
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}
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while (true) {
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uint32_t candidate = rtc::CreateRandomNonZeroId() % kMaxSctpSid;
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if (!GetStreamBySsrc(params_vec, candidate)) {
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*sid = candidate;
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return true;
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}
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}
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}
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static bool GenerateSctpSids(const StreamParamsVec& params_vec,
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std::vector<uint32_t>* sids) {
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uint32_t sid;
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if (!GenerateSctpSid(params_vec, &sid)) {
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LOG(LS_WARNING) << "Could not generated an SCTP SID.";
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return false;
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}
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sids->push_back(sid);
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return true;
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}
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// Finds all StreamParams of all media types and attach them to stream_params.
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static void GetCurrentStreamParams(const SessionDescription* sdesc,
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StreamParamsVec* stream_params) {
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if (!sdesc)
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return;
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const ContentInfos& contents = sdesc->contents();
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for (ContentInfos::const_iterator content = contents.begin();
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content != contents.end(); ++content) {
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if (!IsMediaContent(&*content)) {
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continue;
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}
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const MediaContentDescription* media =
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static_cast<const MediaContentDescription*>(
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content->description);
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const StreamParamsVec& streams = media->streams();
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for (StreamParamsVec::const_iterator it = streams.begin();
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it != streams.end(); ++it) {
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stream_params->push_back(*it);
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}
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}
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}
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// Filters the data codecs for the data channel type.
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void FilterDataCodecs(std::vector<DataCodec>* codecs, bool sctp) {
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// Filter RTP codec for SCTP and vice versa.
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int codec_id = sctp ? kGoogleRtpDataCodecId : kGoogleSctpDataCodecId;
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for (std::vector<DataCodec>::iterator iter = codecs->begin();
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iter != codecs->end();) {
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if (iter->id == codec_id) {
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iter = codecs->erase(iter);
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} else {
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++iter;
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}
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}
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}
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template <typename IdStruct>
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class UsedIds {
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public:
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UsedIds(int min_allowed_id, int max_allowed_id)
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: min_allowed_id_(min_allowed_id),
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max_allowed_id_(max_allowed_id),
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next_id_(max_allowed_id) {
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}
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// Loops through all Id in |ids| and changes its id if it is
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// already in use by another IdStruct. Call this methods with all Id
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// in a session description to make sure no duplicate ids exists.
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// Note that typename Id must be a type of IdStruct.
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template <typename Id>
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void FindAndSetIdUsed(std::vector<Id>* ids) {
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for (typename std::vector<Id>::iterator it = ids->begin();
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it != ids->end(); ++it) {
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FindAndSetIdUsed(&*it);
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}
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}
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// Finds and sets an unused id if the |idstruct| id is already in use.
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void FindAndSetIdUsed(IdStruct* idstruct) {
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const int original_id = idstruct->id;
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int new_id = idstruct->id;
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if (original_id > max_allowed_id_ || original_id < min_allowed_id_) {
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// If the original id is not in range - this is an id that can't be
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// dynamically changed.
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return;
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}
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if (IsIdUsed(original_id)) {
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new_id = FindUnusedId();
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LOG(LS_WARNING) << "Duplicate id found. Reassigning from " << original_id
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<< " to " << new_id;
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idstruct->id = new_id;
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}
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SetIdUsed(new_id);
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}
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private:
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// Returns the first unused id in reverse order.
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// This hopefully reduce the risk of more collisions. We want to change the
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// default ids as little as possible.
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int FindUnusedId() {
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while (IsIdUsed(next_id_) && next_id_ >= min_allowed_id_) {
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--next_id_;
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}
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ASSERT(next_id_ >= min_allowed_id_);
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return next_id_;
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}
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bool IsIdUsed(int new_id) {
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return id_set_.find(new_id) != id_set_.end();
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}
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void SetIdUsed(int new_id) {
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id_set_.insert(new_id);
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}
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const int min_allowed_id_;
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const int max_allowed_id_;
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int next_id_;
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std::set<int> id_set_;
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};
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||
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// Helper class used for finding duplicate RTP payload types among audio, video
|
||
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// and data codecs. When bundle is used the payload types may not collide.
|
||
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class UsedPayloadTypes : public UsedIds<Codec> {
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public:
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UsedPayloadTypes()
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: UsedIds<Codec>(kDynamicPayloadTypeMin, kDynamicPayloadTypeMax) {
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}
|
||
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|
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private:
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static const int kDynamicPayloadTypeMin = 96;
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static const int kDynamicPayloadTypeMax = 127;
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};
|
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|
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// Helper class used for finding duplicate RTP Header extension ids among
|
||
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// audio and video extensions.
|
||
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class UsedRtpHeaderExtensionIds : public UsedIds<webrtc::RtpExtension> {
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public:
|
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UsedRtpHeaderExtensionIds()
|
||
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: UsedIds<webrtc::RtpExtension>(kLocalIdMin, kLocalIdMax) {}
|
||
|
|
||
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private:
|
||
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// Min and Max local identifier for one-byte header extensions, per RFC5285.
|
||
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static const int kLocalIdMin = 1;
|
||
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static const int kLocalIdMax = 14;
|
||
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};
|
||
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|
||
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static bool IsSctp(const MediaContentDescription* desc) {
|
||
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return ((desc->protocol() == kMediaProtocolSctp) ||
|
||
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(desc->protocol() == kMediaProtocolDtlsSctp));
|
||
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}
|
||
|
|
||
|
// Adds a StreamParams for each Stream in Streams with media type
|
||
|
// media_type to content_description.
|
||
|
// |current_params| - All currently known StreamParams of any media type.
|
||
|
template <class C>
|
||
|
static bool AddStreamParams(MediaType media_type,
|
||
|
const MediaSessionOptions& options,
|
||
|
StreamParamsVec* current_streams,
|
||
|
MediaContentDescriptionImpl<C>* content_description,
|
||
|
const bool add_legacy_stream) {
|
||
|
const bool include_rtx_streams =
|
||
|
ContainsRtxCodec(content_description->codecs());
|
||
|
|
||
|
const MediaSessionOptions::Streams& streams = options.streams;
|
||
|
if (streams.empty() && add_legacy_stream) {
|
||
|
// TODO(perkj): Remove this legacy stream when all apps use StreamParams.
|
||
|
std::vector<uint32_t> ssrcs;
|
||
|
if (IsSctp(content_description)) {
|
||
|
GenerateSctpSids(*current_streams, &ssrcs);
|
||
|
} else {
|
||
|
int num_ssrcs = include_rtx_streams ? 2 : 1;
|
||
|
GenerateSsrcs(*current_streams, num_ssrcs, &ssrcs);
|
||
|
}
|
||
|
if (include_rtx_streams) {
|
||
|
content_description->AddLegacyStream(ssrcs[0], ssrcs[1]);
|
||
|
content_description->set_multistream(true);
|
||
|
} else {
|
||
|
content_description->AddLegacyStream(ssrcs[0]);
|
||
|
}
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
MediaSessionOptions::Streams::const_iterator stream_it;
|
||
|
for (stream_it = streams.begin();
|
||
|
stream_it != streams.end(); ++stream_it) {
|
||
|
if (stream_it->type != media_type)
|
||
|
continue; // Wrong media type.
|
||
|
|
||
|
const StreamParams* param =
|
||
|
GetStreamByIds(*current_streams, "", stream_it->id);
|
||
|
// groupid is empty for StreamParams generated using
|
||
|
// MediaSessionDescriptionFactory.
|
||
|
if (!param) {
|
||
|
// This is a new stream.
|
||
|
std::vector<uint32_t> ssrcs;
|
||
|
if (IsSctp(content_description)) {
|
||
|
GenerateSctpSids(*current_streams, &ssrcs);
|
||
|
} else {
|
||
|
GenerateSsrcs(*current_streams, stream_it->num_sim_layers, &ssrcs);
|
||
|
}
|
||
|
StreamParams stream_param;
|
||
|
stream_param.id = stream_it->id;
|
||
|
// Add the generated ssrc.
|
||
|
for (size_t i = 0; i < ssrcs.size(); ++i) {
|
||
|
stream_param.ssrcs.push_back(ssrcs[i]);
|
||
|
}
|
||
|
if (stream_it->num_sim_layers > 1) {
|
||
|
SsrcGroup group(kSimSsrcGroupSemantics, stream_param.ssrcs);
|
||
|
stream_param.ssrc_groups.push_back(group);
|
||
|
}
|
||
|
// Generate extra ssrcs for include_rtx_streams case.
|
||
|
if (include_rtx_streams) {
|
||
|
// Generate an RTX ssrc for every ssrc in the group.
|
||
|
std::vector<uint32_t> rtx_ssrcs;
|
||
|
GenerateSsrcs(*current_streams, static_cast<int>(ssrcs.size()),
|
||
|
&rtx_ssrcs);
|
||
|
for (size_t i = 0; i < ssrcs.size(); ++i) {
|
||
|
stream_param.AddFidSsrc(ssrcs[i], rtx_ssrcs[i]);
|
||
|
}
|
||
|
content_description->set_multistream(true);
|
||
|
}
|
||
|
stream_param.cname = options.rtcp_cname;
|
||
|
stream_param.sync_label = stream_it->sync_label;
|
||
|
content_description->AddStream(stream_param);
|
||
|
|
||
|
// Store the new StreamParams in current_streams.
|
||
|
// This is necessary so that we can use the CNAME for other media types.
|
||
|
current_streams->push_back(stream_param);
|
||
|
} else {
|
||
|
content_description->AddStream(*param);
|
||
|
}
|
||
|
}
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
// Updates the transport infos of the |sdesc| according to the given
|
||
|
// |bundle_group|. The transport infos of the content names within the
|
||
|
// |bundle_group| should be updated to use the ufrag, pwd and DTLS role of the
|
||
|
// first content within the |bundle_group|.
|
||
|
static bool UpdateTransportInfoForBundle(const ContentGroup& bundle_group,
|
||
|
SessionDescription* sdesc) {
|
||
|
// The bundle should not be empty.
|
||
|
if (!sdesc || !bundle_group.FirstContentName()) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
// We should definitely have a transport for the first content.
|
||
|
const std::string& selected_content_name = *bundle_group.FirstContentName();
|
||
|
const TransportInfo* selected_transport_info =
|
||
|
sdesc->GetTransportInfoByName(selected_content_name);
|
||
|
if (!selected_transport_info) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
// Set the other contents to use the same ICE credentials.
|
||
|
const std::string& selected_ufrag =
|
||
|
selected_transport_info->description.ice_ufrag;
|
||
|
const std::string& selected_pwd =
|
||
|
selected_transport_info->description.ice_pwd;
|
||
|
ConnectionRole selected_connection_role =
|
||
|
selected_transport_info->description.connection_role;
|
||
|
for (TransportInfos::iterator it =
|
||
|
sdesc->transport_infos().begin();
|
||
|
it != sdesc->transport_infos().end(); ++it) {
|
||
|
if (bundle_group.HasContentName(it->content_name) &&
|
||
|
it->content_name != selected_content_name) {
|
||
|
it->description.ice_ufrag = selected_ufrag;
|
||
|
it->description.ice_pwd = selected_pwd;
|
||
|
it->description.connection_role = selected_connection_role;
|
||
|
}
|
||
|
}
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
// Gets the CryptoParamsVec of the given |content_name| from |sdesc|, and
|
||
|
// sets it to |cryptos|.
|
||
|
static bool GetCryptosByName(const SessionDescription* sdesc,
|
||
|
const std::string& content_name,
|
||
|
CryptoParamsVec* cryptos) {
|
||
|
if (!sdesc || !cryptos) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
const ContentInfo* content = sdesc->GetContentByName(content_name);
|
||
|
if (!IsMediaContent(content) || !content->description) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
const MediaContentDescription* media_desc =
|
||
|
static_cast<const MediaContentDescription*>(content->description);
|
||
|
*cryptos = media_desc->cryptos();
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
// Predicate function used by the remove_if.
|
||
|
// Returns true if the |crypto|'s cipher_suite is not found in |filter|.
|
||
|
static bool CryptoNotFound(const CryptoParams crypto,
|
||
|
const CryptoParamsVec* filter) {
|
||
|
if (filter == NULL) {
|
||
|
return true;
|
||
|
}
|
||
|
for (CryptoParamsVec::const_iterator it = filter->begin();
|
||
|
it != filter->end(); ++it) {
|
||
|
if (it->cipher_suite == crypto.cipher_suite) {
|
||
|
return false;
|
||
|
}
|
||
|
}
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
// Prunes the |target_cryptos| by removing the crypto params (cipher_suite)
|
||
|
// which are not available in |filter|.
|
||
|
static void PruneCryptos(const CryptoParamsVec& filter,
|
||
|
CryptoParamsVec* target_cryptos) {
|
||
|
if (!target_cryptos) {
|
||
|
return;
|
||
|
}
|
||
|
target_cryptos->erase(std::remove_if(target_cryptos->begin(),
|
||
|
target_cryptos->end(),
|
||
|
bind2nd(ptr_fun(CryptoNotFound),
|
||
|
&filter)),
|
||
|
target_cryptos->end());
|
||
|
}
|
||
|
|
||
|
static bool IsRtpProtocol(const std::string& protocol) {
|
||
|
return protocol.empty() ||
|
||
|
(protocol.find(cricket::kMediaProtocolRtpPrefix) != std::string::npos);
|
||
|
}
|
||
|
|
||
|
static bool IsRtpContent(SessionDescription* sdesc,
|
||
|
const std::string& content_name) {
|
||
|
bool is_rtp = false;
|
||
|
ContentInfo* content = sdesc->GetContentByName(content_name);
|
||
|
if (IsMediaContent(content)) {
|
||
|
MediaContentDescription* media_desc =
|
||
|
static_cast<MediaContentDescription*>(content->description);
|
||
|
if (!media_desc) {
|
||
|
return false;
|
||
|
}
|
||
|
is_rtp = IsRtpProtocol(media_desc->protocol());
|
||
|
}
|
||
|
return is_rtp;
|
||
|
}
|
||
|
|
||
|
// Updates the crypto parameters of the |sdesc| according to the given
|
||
|
// |bundle_group|. The crypto parameters of all the contents within the
|
||
|
// |bundle_group| should be updated to use the common subset of the
|
||
|
// available cryptos.
|
||
|
static bool UpdateCryptoParamsForBundle(const ContentGroup& bundle_group,
|
||
|
SessionDescription* sdesc) {
|
||
|
// The bundle should not be empty.
|
||
|
if (!sdesc || !bundle_group.FirstContentName()) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
bool common_cryptos_needed = false;
|
||
|
// Get the common cryptos.
|
||
|
const ContentNames& content_names = bundle_group.content_names();
|
||
|
CryptoParamsVec common_cryptos;
|
||
|
for (ContentNames::const_iterator it = content_names.begin();
|
||
|
it != content_names.end(); ++it) {
|
||
|
if (!IsRtpContent(sdesc, *it)) {
|
||
|
continue;
|
||
|
}
|
||
|
// The common cryptos are needed if any of the content does not have DTLS
|
||
|
// enabled.
|
||
|
if (!sdesc->GetTransportInfoByName(*it)->description.secure()) {
|
||
|
common_cryptos_needed = true;
|
||
|
}
|
||
|
if (it == content_names.begin()) {
|
||
|
// Initial the common_cryptos with the first content in the bundle group.
|
||
|
if (!GetCryptosByName(sdesc, *it, &common_cryptos)) {
|
||
|
return false;
|
||
|
}
|
||
|
if (common_cryptos.empty()) {
|
||
|
// If there's no crypto params, we should just return.
|
||
|
return true;
|
||
|
}
|
||
|
} else {
|
||
|
CryptoParamsVec cryptos;
|
||
|
if (!GetCryptosByName(sdesc, *it, &cryptos)) {
|
||
|
return false;
|
||
|
}
|
||
|
PruneCryptos(cryptos, &common_cryptos);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
if (common_cryptos.empty() && common_cryptos_needed) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
// Update to use the common cryptos.
|
||
|
for (ContentNames::const_iterator it = content_names.begin();
|
||
|
it != content_names.end(); ++it) {
|
||
|
if (!IsRtpContent(sdesc, *it)) {
|
||
|
continue;
|
||
|
}
|
||
|
ContentInfo* content = sdesc->GetContentByName(*it);
|
||
|
if (IsMediaContent(content)) {
|
||
|
MediaContentDescription* media_desc =
|
||
|
static_cast<MediaContentDescription*>(content->description);
|
||
|
if (!media_desc) {
|
||
|
return false;
|
||
|
}
|
||
|
media_desc->set_cryptos(common_cryptos);
|
||
|
}
|
||
|
}
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
template <class C>
|
||
|
static bool ContainsRtxCodec(const std::vector<C>& codecs) {
|
||
|
typename std::vector<C>::const_iterator it;
|
||
|
for (it = codecs.begin(); it != codecs.end(); ++it) {
|
||
|
if (IsRtxCodec(*it)) {
|
||
|
return true;
|
||
|
}
|
||
|
}
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
template <class C>
|
||
|
static bool IsRtxCodec(const C& codec) {
|
||
|
return stricmp(codec.name.c_str(), kRtxCodecName) == 0;
|
||
|
}
|
||
|
|
||
|
static TransportOptions GetTransportOptions(const MediaSessionOptions& options,
|
||
|
const std::string& content_name) {
|
||
|
auto it = options.transport_options.find(content_name);
|
||
|
if (it == options.transport_options.end()) {
|
||
|
return TransportOptions();
|
||
|
}
|
||
|
return it->second;
|
||
|
}
|
||
|
|
||
|
// Create a media content to be offered in a session-initiate,
|
||
|
// according to the given options.rtcp_mux, options.is_muc,
|
||
|
// options.streams, codecs, secure_transport, crypto, and streams. If we don't
|
||
|
// currently have crypto (in current_cryptos) and it is enabled (in
|
||
|
// secure_policy), crypto is created (according to crypto_suites). If
|
||
|
// add_legacy_stream is true, and current_streams is empty, a legacy
|
||
|
// stream is created. The created content is added to the offer.
|
||
|
template <class C>
|
||
|
static bool CreateMediaContentOffer(
|
||
|
const MediaSessionOptions& options,
|
||
|
const std::vector<C>& codecs,
|
||
|
const SecurePolicy& secure_policy,
|
||
|
const CryptoParamsVec* current_cryptos,
|
||
|
const std::vector<std::string>& crypto_suites,
|
||
|
const RtpHeaderExtensions& rtp_extensions,
|
||
|
bool add_legacy_stream,
|
||
|
StreamParamsVec* current_streams,
|
||
|
MediaContentDescriptionImpl<C>* offer) {
|
||
|
offer->AddCodecs(codecs);
|
||
|
|
||
|
if (secure_policy == SEC_REQUIRED) {
|
||
|
offer->set_crypto_required(CT_SDES);
|
||
|
}
|
||
|
offer->set_rtcp_mux(options.rtcp_mux_enabled);
|
||
|
if (offer->type() == cricket::MEDIA_TYPE_VIDEO) {
|
||
|
offer->set_rtcp_reduced_size(true);
|
||
|
}
|
||
|
offer->set_multistream(options.is_muc);
|
||
|
offer->set_rtp_header_extensions(rtp_extensions);
|
||
|
|
||
|
if (!AddStreamParams(offer->type(), options, current_streams, offer,
|
||
|
add_legacy_stream)) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
#ifdef HAVE_SRTP
|
||
|
if (secure_policy != SEC_DISABLED) {
|
||
|
if (current_cryptos) {
|
||
|
AddMediaCryptos(*current_cryptos, offer);
|
||
|
}
|
||
|
if (offer->cryptos().empty()) {
|
||
|
if (!CreateMediaCryptos(crypto_suites, offer)) {
|
||
|
return false;
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
#endif
|
||
|
|
||
|
if (offer->crypto_required() == CT_SDES && offer->cryptos().empty()) {
|
||
|
return false;
|
||
|
}
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
template <class C>
|
||
|
static bool ReferencedCodecsMatch(const std::vector<C>& codecs1,
|
||
|
const std::string& codec1_id_str,
|
||
|
const std::vector<C>& codecs2,
|
||
|
const std::string& codec2_id_str) {
|
||
|
int codec1_id;
|
||
|
int codec2_id;
|
||
|
C codec1;
|
||
|
C codec2;
|
||
|
if (!rtc::FromString(codec1_id_str, &codec1_id) ||
|
||
|
!rtc::FromString(codec2_id_str, &codec2_id) ||
|
||
|
!FindCodecById(codecs1, codec1_id, &codec1) ||
|
||
|
!FindCodecById(codecs2, codec2_id, &codec2)) {
|
||
|
return false;
|
||
|
}
|
||
|
return codec1.Matches(codec2);
|
||
|
}
|
||
|
|
||
|
template <class C>
|
||
|
static void NegotiateCodecs(const std::vector<C>& local_codecs,
|
||
|
const std::vector<C>& offered_codecs,
|
||
|
std::vector<C>* negotiated_codecs) {
|
||
|
for (const C& ours : local_codecs) {
|
||
|
C theirs;
|
||
|
// Note that we intentionally only find one matching codec for each of our
|
||
|
// local codecs, in case the remote offer contains duplicate codecs.
|
||
|
if (FindMatchingCodec(local_codecs, offered_codecs, ours, &theirs)) {
|
||
|
C negotiated = ours;
|
||
|
negotiated.IntersectFeedbackParams(theirs);
|
||
|
if (IsRtxCodec(negotiated)) {
|
||
|
std::string offered_apt_value;
|
||
|
theirs.GetParam(kCodecParamAssociatedPayloadType, &offered_apt_value);
|
||
|
// FindMatchingCodec shouldn't return something with no apt value.
|
||
|
RTC_DCHECK(!offered_apt_value.empty());
|
||
|
negotiated.SetParam(kCodecParamAssociatedPayloadType,
|
||
|
offered_apt_value);
|
||
|
}
|
||
|
negotiated.id = theirs.id;
|
||
|
negotiated.name = theirs.name;
|
||
|
negotiated_codecs->push_back(negotiated);
|
||
|
}
|
||
|
}
|
||
|
// RFC3264: Although the answerer MAY list the formats in their desired
|
||
|
// order of preference, it is RECOMMENDED that unless there is a
|
||
|
// specific reason, the answerer list formats in the same relative order
|
||
|
// they were present in the offer.
|
||
|
std::unordered_map<int, int> payload_type_preferences;
|
||
|
int preference = static_cast<int>(offered_codecs.size() + 1);
|
||
|
for (const C& codec : offered_codecs) {
|
||
|
payload_type_preferences[codec.id] = preference--;
|
||
|
}
|
||
|
std::sort(negotiated_codecs->begin(), negotiated_codecs->end(),
|
||
|
[&payload_type_preferences](const C& a, const C& b) {
|
||
|
return payload_type_preferences[a.id] >
|
||
|
payload_type_preferences[b.id];
|
||
|
});
|
||
|
}
|
||
|
|
||
|
// Finds a codec in |codecs2| that matches |codec_to_match|, which is
|
||
|
// a member of |codecs1|. If |codec_to_match| is an RTX codec, both
|
||
|
// the codecs themselves and their associated codecs must match.
|
||
|
template <class C>
|
||
|
static bool FindMatchingCodec(const std::vector<C>& codecs1,
|
||
|
const std::vector<C>& codecs2,
|
||
|
const C& codec_to_match,
|
||
|
C* found_codec) {
|
||
|
for (const C& potential_match : codecs2) {
|
||
|
if (potential_match.Matches(codec_to_match)) {
|
||
|
if (IsRtxCodec(codec_to_match)) {
|
||
|
std::string apt_value_1;
|
||
|
std::string apt_value_2;
|
||
|
if (!codec_to_match.GetParam(kCodecParamAssociatedPayloadType,
|
||
|
&apt_value_1) ||
|
||
|
!potential_match.GetParam(kCodecParamAssociatedPayloadType,
|
||
|
&apt_value_2)) {
|
||
|
LOG(LS_WARNING) << "RTX missing associated payload type.";
|
||
|
continue;
|
||
|
}
|
||
|
if (!ReferencedCodecsMatch(codecs1, apt_value_1, codecs2,
|
||
|
apt_value_2)) {
|
||
|
continue;
|
||
|
}
|
||
|
}
|
||
|
if (found_codec) {
|
||
|
*found_codec = potential_match;
|
||
|
}
|
||
|
return true;
|
||
|
}
|
||
|
}
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
// Adds all codecs from |reference_codecs| to |offered_codecs| that dont'
|
||
|
// already exist in |offered_codecs| and ensure the payload types don't
|
||
|
// collide.
|
||
|
template <class C>
|
||
|
static void FindCodecsToOffer(
|
||
|
const std::vector<C>& reference_codecs,
|
||
|
std::vector<C>* offered_codecs,
|
||
|
UsedPayloadTypes* used_pltypes) {
|
||
|
|
||
|
// Add all new codecs that are not RTX codecs.
|
||
|
for (const C& reference_codec : reference_codecs) {
|
||
|
if (!IsRtxCodec(reference_codec) &&
|
||
|
!FindMatchingCodec<C>(reference_codecs, *offered_codecs,
|
||
|
reference_codec, nullptr)) {
|
||
|
C codec = reference_codec;
|
||
|
used_pltypes->FindAndSetIdUsed(&codec);
|
||
|
offered_codecs->push_back(codec);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
// Add all new RTX codecs.
|
||
|
for (const C& reference_codec : reference_codecs) {
|
||
|
if (IsRtxCodec(reference_codec) &&
|
||
|
!FindMatchingCodec<C>(reference_codecs, *offered_codecs,
|
||
|
reference_codec, nullptr)) {
|
||
|
C rtx_codec = reference_codec;
|
||
|
|
||
|
std::string associated_pt_str;
|
||
|
if (!rtx_codec.GetParam(kCodecParamAssociatedPayloadType,
|
||
|
&associated_pt_str)) {
|
||
|
LOG(LS_WARNING) << "RTX codec " << rtx_codec.name
|
||
|
<< " is missing an associated payload type.";
|
||
|
continue;
|
||
|
}
|
||
|
|
||
|
int associated_pt;
|
||
|
if (!rtc::FromString(associated_pt_str, &associated_pt)) {
|
||
|
LOG(LS_WARNING) << "Couldn't convert payload type " << associated_pt_str
|
||
|
<< " of RTX codec " << rtx_codec.name
|
||
|
<< " to an integer.";
|
||
|
continue;
|
||
|
}
|
||
|
|
||
|
// Find the associated reference codec for the reference RTX codec.
|
||
|
C associated_codec;
|
||
|
if (!FindCodecById(reference_codecs, associated_pt, &associated_codec)) {
|
||
|
LOG(LS_WARNING) << "Couldn't find associated codec with payload type "
|
||
|
<< associated_pt << " for RTX codec " << rtx_codec.name
|
||
|
<< ".";
|
||
|
continue;
|
||
|
}
|
||
|
|
||
|
// Find a codec in the offered list that matches the reference codec.
|
||
|
// Its payload type may be different than the reference codec.
|
||
|
C matching_codec;
|
||
|
if (!FindMatchingCodec<C>(reference_codecs, *offered_codecs,
|
||
|
associated_codec, &matching_codec)) {
|
||
|
LOG(LS_WARNING) << "Couldn't find matching " << associated_codec.name
|
||
|
<< " codec.";
|
||
|
continue;
|
||
|
}
|
||
|
|
||
|
rtx_codec.params[kCodecParamAssociatedPayloadType] =
|
||
|
rtc::ToString(matching_codec.id);
|
||
|
used_pltypes->FindAndSetIdUsed(&rtx_codec);
|
||
|
offered_codecs->push_back(rtx_codec);
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
|
||
|
static bool FindByUri(const RtpHeaderExtensions& extensions,
|
||
|
const webrtc::RtpExtension& ext_to_match,
|
||
|
webrtc::RtpExtension* found_extension) {
|
||
|
for (RtpHeaderExtensions::const_iterator it = extensions.begin();
|
||
|
it != extensions.end(); ++it) {
|
||
|
// We assume that all URIs are given in a canonical format.
|
||
|
if (it->uri == ext_to_match.uri) {
|
||
|
if (found_extension != NULL) {
|
||
|
*found_extension = *it;
|
||
|
}
|
||
|
return true;
|
||
|
}
|
||
|
}
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
// Iterates through |offered_extensions|, adding each one to |all_extensions|
|
||
|
// and |used_ids|, and resolving ID conflicts. If an offered extension has the
|
||
|
// same URI as one in |all_extensions|, it will re-use the same ID and won't be
|
||
|
// treated as a conflict.
|
||
|
static void FindAndSetRtpHdrExtUsed(RtpHeaderExtensions* offered_extensions,
|
||
|
RtpHeaderExtensions* all_extensions,
|
||
|
UsedRtpHeaderExtensionIds* used_ids) {
|
||
|
for (auto& extension : *offered_extensions) {
|
||
|
webrtc::RtpExtension existing;
|
||
|
if (FindByUri(*all_extensions, extension, &existing)) {
|
||
|
extension.id = existing.id;
|
||
|
} else {
|
||
|
used_ids->FindAndSetIdUsed(&extension);
|
||
|
all_extensions->push_back(extension);
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
|
||
|
// Adds |reference_extensions| to |offered_extensions|, while updating
|
||
|
// |all_extensions| and |used_ids|.
|
||
|
static void FindRtpHdrExtsToOffer(
|
||
|
const RtpHeaderExtensions& reference_extensions,
|
||
|
RtpHeaderExtensions* offered_extensions,
|
||
|
RtpHeaderExtensions* all_extensions,
|
||
|
UsedRtpHeaderExtensionIds* used_ids) {
|
||
|
for (auto reference_extension : reference_extensions) {
|
||
|
if (!FindByUri(*offered_extensions, reference_extension, NULL)) {
|
||
|
webrtc::RtpExtension existing;
|
||
|
if (FindByUri(*all_extensions, reference_extension, &existing)) {
|
||
|
offered_extensions->push_back(existing);
|
||
|
} else {
|
||
|
used_ids->FindAndSetIdUsed(&reference_extension);
|
||
|
all_extensions->push_back(reference_extension);
|
||
|
offered_extensions->push_back(reference_extension);
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
|
||
|
static void NegotiateRtpHeaderExtensions(
|
||
|
const RtpHeaderExtensions& local_extensions,
|
||
|
const RtpHeaderExtensions& offered_extensions,
|
||
|
RtpHeaderExtensions* negotiated_extenstions) {
|
||
|
RtpHeaderExtensions::const_iterator ours;
|
||
|
for (ours = local_extensions.begin();
|
||
|
ours != local_extensions.end(); ++ours) {
|
||
|
webrtc::RtpExtension theirs;
|
||
|
if (FindByUri(offered_extensions, *ours, &theirs)) {
|
||
|
// We respond with their RTP header extension id.
|
||
|
negotiated_extenstions->push_back(theirs);
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
|
||
|
static void StripCNCodecs(AudioCodecs* audio_codecs) {
|
||
|
AudioCodecs::iterator iter = audio_codecs->begin();
|
||
|
while (iter != audio_codecs->end()) {
|
||
|
if (stricmp(iter->name.c_str(), kComfortNoiseCodecName) == 0) {
|
||
|
iter = audio_codecs->erase(iter);
|
||
|
} else {
|
||
|
++iter;
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
|
||
|
// Create a media content to be answered in a session-accept,
|
||
|
// according to the given options.rtcp_mux, options.streams, codecs,
|
||
|
// crypto, and streams. If we don't currently have crypto (in
|
||
|
// current_cryptos) and it is enabled (in secure_policy), crypto is
|
||
|
// created (according to crypto_suites). If add_legacy_stream is
|
||
|
// true, and current_streams is empty, a legacy stream is created.
|
||
|
// The codecs, rtcp_mux, and crypto are all negotiated with the offer
|
||
|
// from the incoming session-initiate. If the negotiation fails, this
|
||
|
// method returns false. The created content is added to the offer.
|
||
|
template <class C>
|
||
|
static bool CreateMediaContentAnswer(
|
||
|
const MediaContentDescriptionImpl<C>* offer,
|
||
|
const MediaSessionOptions& options,
|
||
|
const std::vector<C>& local_codecs,
|
||
|
const SecurePolicy& sdes_policy,
|
||
|
const CryptoParamsVec* current_cryptos,
|
||
|
const RtpHeaderExtensions& local_rtp_extenstions,
|
||
|
StreamParamsVec* current_streams,
|
||
|
bool add_legacy_stream,
|
||
|
bool bundle_enabled,
|
||
|
MediaContentDescriptionImpl<C>* answer) {
|
||
|
std::vector<C> negotiated_codecs;
|
||
|
NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs);
|
||
|
answer->AddCodecs(negotiated_codecs);
|
||
|
answer->set_protocol(offer->protocol());
|
||
|
RtpHeaderExtensions negotiated_rtp_extensions;
|
||
|
NegotiateRtpHeaderExtensions(local_rtp_extenstions,
|
||
|
offer->rtp_header_extensions(),
|
||
|
&negotiated_rtp_extensions);
|
||
|
answer->set_rtp_header_extensions(negotiated_rtp_extensions);
|
||
|
|
||
|
answer->set_rtcp_mux(options.rtcp_mux_enabled && offer->rtcp_mux());
|
||
|
if (answer->type() == cricket::MEDIA_TYPE_VIDEO) {
|
||
|
answer->set_rtcp_reduced_size(offer->rtcp_reduced_size());
|
||
|
}
|
||
|
|
||
|
if (sdes_policy != SEC_DISABLED) {
|
||
|
CryptoParams crypto;
|
||
|
if (SelectCrypto(offer, bundle_enabled, &crypto)) {
|
||
|
if (current_cryptos) {
|
||
|
FindMatchingCrypto(*current_cryptos, crypto, &crypto);
|
||
|
}
|
||
|
answer->AddCrypto(crypto);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
if (answer->cryptos().empty() &&
|
||
|
(offer->crypto_required() == CT_SDES || sdes_policy == SEC_REQUIRED)) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
if (!AddStreamParams(answer->type(), options, current_streams, answer,
|
||
|
add_legacy_stream)) {
|
||
|
return false; // Something went seriously wrong.
|
||
|
}
|
||
|
|
||
|
// Make sure the answer media content direction is per default set as
|
||
|
// described in RFC3264 section 6.1.
|
||
|
const bool is_data = !IsRtpProtocol(answer->protocol());
|
||
|
const bool has_send_streams = !answer->streams().empty();
|
||
|
const bool wants_send = has_send_streams || is_data;
|
||
|
const bool recv_audio =
|
||
|
answer->type() == cricket::MEDIA_TYPE_AUDIO && options.recv_audio;
|
||
|
const bool recv_video =
|
||
|
answer->type() == cricket::MEDIA_TYPE_VIDEO && options.recv_video;
|
||
|
const bool recv_data =
|
||
|
answer->type() == cricket::MEDIA_TYPE_DATA;
|
||
|
const bool wants_receive = recv_audio || recv_video || recv_data;
|
||
|
|
||
|
auto offer_rtd =
|
||
|
RtpTransceiverDirection::FromMediaContentDirection(offer->direction());
|
||
|
auto wants_rtd = RtpTransceiverDirection(wants_send, wants_receive);
|
||
|
answer->set_direction(NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd)
|
||
|
.ToMediaContentDirection());
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
static bool IsDtlsRtp(const std::string& protocol) {
|
||
|
// Most-likely values first.
|
||
|
return protocol == "UDP/TLS/RTP/SAVPF" || protocol == "TCP/TLS/RTP/SAVPF" ||
|
||
|
protocol == "UDP/TLS/RTP/SAVP" || protocol == "TCP/TLS/RTP/SAVP";
|
||
|
}
|
||
|
|
||
|
static bool IsPlainRtp(const std::string& protocol) {
|
||
|
// Most-likely values first.
|
||
|
return protocol == "RTP/SAVPF" || protocol == "RTP/AVPF" ||
|
||
|
protocol == "RTP/SAVP" || protocol == "RTP/AVP";
|
||
|
}
|
||
|
|
||
|
static bool IsDtlsSctp(const std::string& protocol) {
|
||
|
return protocol == "DTLS/SCTP";
|
||
|
}
|
||
|
|
||
|
static bool IsPlainSctp(const std::string& protocol) {
|
||
|
return protocol == "SCTP";
|
||
|
}
|
||
|
|
||
|
static bool IsMediaProtocolSupported(MediaType type,
|
||
|
const std::string& protocol,
|
||
|
bool secure_transport) {
|
||
|
// Since not all applications serialize and deserialize the media protocol,
|
||
|
// we will have to accept |protocol| to be empty.
|
||
|
if (protocol.empty()) {
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
if (type == MEDIA_TYPE_DATA) {
|
||
|
// Check for SCTP, but also for RTP for RTP-based data channels.
|
||
|
// TODO(pthatcher): Remove RTP once RTP-based data channels are gone.
|
||
|
if (secure_transport) {
|
||
|
// Most likely scenarios first.
|
||
|
return IsDtlsSctp(protocol) || IsDtlsRtp(protocol) ||
|
||
|
IsPlainRtp(protocol);
|
||
|
} else {
|
||
|
return IsPlainSctp(protocol) || IsPlainRtp(protocol);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
// Allow for non-DTLS RTP protocol even when using DTLS because that's what
|
||
|
// JSEP specifies.
|
||
|
if (secure_transport) {
|
||
|
// Most likely scenarios first.
|
||
|
return IsDtlsRtp(protocol) || IsPlainRtp(protocol);
|
||
|
} else {
|
||
|
return IsPlainRtp(protocol);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
static void SetMediaProtocol(bool secure_transport,
|
||
|
MediaContentDescription* desc) {
|
||
|
if (!desc->cryptos().empty())
|
||
|
desc->set_protocol(kMediaProtocolSavpf);
|
||
|
else if (secure_transport)
|
||
|
desc->set_protocol(kMediaProtocolDtlsSavpf);
|
||
|
else
|
||
|
desc->set_protocol(kMediaProtocolAvpf);
|
||
|
}
|
||
|
|
||
|
// Gets the TransportInfo of the given |content_name| from the
|
||
|
// |current_description|. If doesn't exist, returns a new one.
|
||
|
static const TransportDescription* GetTransportDescription(
|
||
|
const std::string& content_name,
|
||
|
const SessionDescription* current_description) {
|
||
|
const TransportDescription* desc = NULL;
|
||
|
if (current_description) {
|
||
|
const TransportInfo* info =
|
||
|
current_description->GetTransportInfoByName(content_name);
|
||
|
if (info) {
|
||
|
desc = &info->description;
|
||
|
}
|
||
|
}
|
||
|
return desc;
|
||
|
}
|
||
|
|
||
|
// Gets the current DTLS state from the transport description.
|
||
|
static bool IsDtlsActive(
|
||
|
const std::string& content_name,
|
||
|
const SessionDescription* current_description) {
|
||
|
if (!current_description)
|
||
|
return false;
|
||
|
|
||
|
const ContentInfo* content =
|
||
|
current_description->GetContentByName(content_name);
|
||
|
if (!content)
|
||
|
return false;
|
||
|
|
||
|
const TransportDescription* current_tdesc =
|
||
|
GetTransportDescription(content_name, current_description);
|
||
|
if (!current_tdesc)
|
||
|
return false;
|
||
|
|
||
|
return current_tdesc->secure();
|
||
|
}
|
||
|
|
||
|
std::string MediaTypeToString(MediaType type) {
|
||
|
std::string type_str;
|
||
|
switch (type) {
|
||
|
case MEDIA_TYPE_AUDIO:
|
||
|
type_str = "audio";
|
||
|
break;
|
||
|
case MEDIA_TYPE_VIDEO:
|
||
|
type_str = "video";
|
||
|
break;
|
||
|
case MEDIA_TYPE_DATA:
|
||
|
type_str = "data";
|
||
|
break;
|
||
|
default:
|
||
|
ASSERT(false);
|
||
|
break;
|
||
|
}
|
||
|
return type_str;
|
||
|
}
|
||
|
|
||
|
std::string MediaContentDirectionToString(MediaContentDirection direction) {
|
||
|
std::string dir_str;
|
||
|
switch (direction) {
|
||
|
case MD_INACTIVE:
|
||
|
dir_str = "inactive";
|
||
|
break;
|
||
|
case MD_SENDONLY:
|
||
|
dir_str = "sendonly";
|
||
|
break;
|
||
|
case MD_RECVONLY:
|
||
|
dir_str = "recvonly";
|
||
|
break;
|
||
|
case MD_SENDRECV:
|
||
|
dir_str = "sendrecv";
|
||
|
break;
|
||
|
default:
|
||
|
ASSERT(false);
|
||
|
break;
|
||
|
}
|
||
|
|
||
|
return dir_str;
|
||
|
}
|
||
|
|
||
|
void MediaSessionOptions::AddSendStream(MediaType type,
|
||
|
const std::string& id,
|
||
|
const std::string& sync_label) {
|
||
|
AddSendStreamInternal(type, id, sync_label, 1);
|
||
|
}
|
||
|
|
||
|
void MediaSessionOptions::AddSendVideoStream(
|
||
|
const std::string& id,
|
||
|
const std::string& sync_label,
|
||
|
int num_sim_layers) {
|
||
|
AddSendStreamInternal(MEDIA_TYPE_VIDEO, id, sync_label, num_sim_layers);
|
||
|
}
|
||
|
|
||
|
void MediaSessionOptions::AddSendStreamInternal(
|
||
|
MediaType type,
|
||
|
const std::string& id,
|
||
|
const std::string& sync_label,
|
||
|
int num_sim_layers) {
|
||
|
streams.push_back(Stream(type, id, sync_label, num_sim_layers));
|
||
|
|
||
|
// If we haven't already set the data_channel_type, and we add a
|
||
|
// stream, we assume it's an RTP data stream.
|
||
|
if (type == MEDIA_TYPE_DATA && data_channel_type == DCT_NONE)
|
||
|
data_channel_type = DCT_RTP;
|
||
|
}
|
||
|
|
||
|
void MediaSessionOptions::RemoveSendStream(MediaType type,
|
||
|
const std::string& id) {
|
||
|
Streams::iterator stream_it = streams.begin();
|
||
|
for (; stream_it != streams.end(); ++stream_it) {
|
||
|
if (stream_it->type == type && stream_it->id == id) {
|
||
|
streams.erase(stream_it);
|
||
|
return;
|
||
|
}
|
||
|
}
|
||
|
ASSERT(false);
|
||
|
}
|
||
|
|
||
|
bool MediaSessionOptions::HasSendMediaStream(MediaType type) const {
|
||
|
Streams::const_iterator stream_it = streams.begin();
|
||
|
for (; stream_it != streams.end(); ++stream_it) {
|
||
|
if (stream_it->type == type) {
|
||
|
return true;
|
||
|
}
|
||
|
}
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
|
||
|
const TransportDescriptionFactory* transport_desc_factory)
|
||
|
: secure_(SEC_DISABLED),
|
||
|
add_legacy_(true),
|
||
|
transport_desc_factory_(transport_desc_factory) {
|
||
|
}
|
||
|
|
||
|
MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
|
||
|
ChannelManager* channel_manager,
|
||
|
const TransportDescriptionFactory* transport_desc_factory)
|
||
|
: secure_(SEC_DISABLED),
|
||
|
add_legacy_(true),
|
||
|
transport_desc_factory_(transport_desc_factory) {
|
||
|
channel_manager->GetSupportedAudioSendCodecs(&audio_send_codecs_);
|
||
|
channel_manager->GetSupportedAudioReceiveCodecs(&audio_recv_codecs_);
|
||
|
channel_manager->GetSupportedAudioSendCodecs(&audio_send_codecs_);
|
||
|
channel_manager->GetSupportedAudioRtpHeaderExtensions(&audio_rtp_extensions_);
|
||
|
channel_manager->GetSupportedVideoCodecs(&video_codecs_);
|
||
|
channel_manager->GetSupportedVideoRtpHeaderExtensions(&video_rtp_extensions_);
|
||
|
channel_manager->GetSupportedDataCodecs(&data_codecs_);
|
||
|
NegotiateCodecs(audio_recv_codecs_, audio_send_codecs_,
|
||
|
&audio_sendrecv_codecs_);
|
||
|
}
|
||
|
|
||
|
const AudioCodecs& MediaSessionDescriptionFactory::audio_sendrecv_codecs()
|
||
|
const {
|
||
|
return audio_sendrecv_codecs_;
|
||
|
}
|
||
|
|
||
|
const AudioCodecs& MediaSessionDescriptionFactory::audio_send_codecs() const {
|
||
|
return audio_send_codecs_;
|
||
|
}
|
||
|
|
||
|
const AudioCodecs& MediaSessionDescriptionFactory::audio_recv_codecs() const {
|
||
|
return audio_recv_codecs_;
|
||
|
}
|
||
|
|
||
|
void MediaSessionDescriptionFactory::set_audio_codecs(
|
||
|
const AudioCodecs& send_codecs, const AudioCodecs& recv_codecs) {
|
||
|
audio_send_codecs_ = send_codecs;
|
||
|
audio_recv_codecs_ = recv_codecs;
|
||
|
audio_sendrecv_codecs_.clear();
|
||
|
// Use NegotiateCodecs to merge our codec lists, since the operation is
|
||
|
// essentially the same. Put send_codecs as the offered_codecs, which is the
|
||
|
// order we'd like to follow. The reasoning is that encoding is usually more
|
||
|
// expensive than decoding, and prioritizing a codec in the send list probably
|
||
|
// means it's a codec we can handle efficiently.
|
||
|
NegotiateCodecs(recv_codecs, send_codecs, &audio_sendrecv_codecs_);
|
||
|
}
|
||
|
|
||
|
SessionDescription* MediaSessionDescriptionFactory::CreateOffer(
|
||
|
const MediaSessionOptions& options,
|
||
|
const SessionDescription* current_description) const {
|
||
|
std::unique_ptr<SessionDescription> offer(new SessionDescription());
|
||
|
|
||
|
StreamParamsVec current_streams;
|
||
|
GetCurrentStreamParams(current_description, ¤t_streams);
|
||
|
|
||
|
const bool wants_send =
|
||
|
options.HasSendMediaStream(MEDIA_TYPE_AUDIO) || add_legacy_;
|
||
|
const AudioCodecs& supported_audio_codecs =
|
||
|
GetAudioCodecsForOffer({wants_send, options.recv_audio});
|
||
|
|
||
|
AudioCodecs audio_codecs;
|
||
|
VideoCodecs video_codecs;
|
||
|
DataCodecs data_codecs;
|
||
|
GetCodecsToOffer(current_description, supported_audio_codecs,
|
||
|
video_codecs_, data_codecs_,
|
||
|
&audio_codecs, &video_codecs, &data_codecs);
|
||
|
|
||
|
if (!options.vad_enabled) {
|
||
|
// If application doesn't want CN codecs in offer.
|
||
|
StripCNCodecs(&audio_codecs);
|
||
|
}
|
||
|
|
||
|
RtpHeaderExtensions audio_rtp_extensions;
|
||
|
RtpHeaderExtensions video_rtp_extensions;
|
||
|
GetRtpHdrExtsToOffer(current_description, &audio_rtp_extensions,
|
||
|
&video_rtp_extensions);
|
||
|
|
||
|
bool audio_added = false;
|
||
|
bool video_added = false;
|
||
|
bool data_added = false;
|
||
|
|
||
|
// Iterate through the contents of |current_description| to maintain the order
|
||
|
// of the m-lines in the new offer.
|
||
|
if (current_description) {
|
||
|
ContentInfos::const_iterator it = current_description->contents().begin();
|
||
|
for (; it != current_description->contents().end(); ++it) {
|
||
|
if (IsMediaContentOfType(&*it, MEDIA_TYPE_AUDIO)) {
|
||
|
if (!AddAudioContentForOffer(options, current_description,
|
||
|
audio_rtp_extensions, audio_codecs,
|
||
|
¤t_streams, offer.get())) {
|
||
|
return NULL;
|
||
|
}
|
||
|
audio_added = true;
|
||
|
} else if (IsMediaContentOfType(&*it, MEDIA_TYPE_VIDEO)) {
|
||
|
if (!AddVideoContentForOffer(options, current_description,
|
||
|
video_rtp_extensions, video_codecs,
|
||
|
¤t_streams, offer.get())) {
|
||
|
return NULL;
|
||
|
}
|
||
|
video_added = true;
|
||
|
} else if (IsMediaContentOfType(&*it, MEDIA_TYPE_DATA)) {
|
||
|
MediaSessionOptions options_copy(options);
|
||
|
if (IsSctp(static_cast<const MediaContentDescription*>(
|
||
|
it->description))) {
|
||
|
options_copy.data_channel_type = DCT_SCTP;
|
||
|
}
|
||
|
if (!AddDataContentForOffer(options_copy, current_description,
|
||
|
&data_codecs, ¤t_streams,
|
||
|
offer.get())) {
|
||
|
return NULL;
|
||
|
}
|
||
|
data_added = true;
|
||
|
} else {
|
||
|
ASSERT(false);
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
|
||
|
// Append contents that are not in |current_description|.
|
||
|
if (!audio_added && options.has_audio() &&
|
||
|
!AddAudioContentForOffer(options, current_description,
|
||
|
audio_rtp_extensions, audio_codecs,
|
||
|
¤t_streams, offer.get())) {
|
||
|
return NULL;
|
||
|
}
|
||
|
if (!video_added && options.has_video() &&
|
||
|
!AddVideoContentForOffer(options, current_description,
|
||
|
video_rtp_extensions, video_codecs,
|
||
|
¤t_streams, offer.get())) {
|
||
|
return NULL;
|
||
|
}
|
||
|
if (!data_added && options.has_data() &&
|
||
|
!AddDataContentForOffer(options, current_description, &data_codecs,
|
||
|
¤t_streams, offer.get())) {
|
||
|
return NULL;
|
||
|
}
|
||
|
|
||
|
// Bundle the contents together, if we've been asked to do so, and update any
|
||
|
// parameters that need to be tweaked for BUNDLE.
|
||
|
if (options.bundle_enabled) {
|
||
|
ContentGroup offer_bundle(GROUP_TYPE_BUNDLE);
|
||
|
for (ContentInfos::const_iterator content = offer->contents().begin();
|
||
|
content != offer->contents().end(); ++content) {
|
||
|
offer_bundle.AddContentName(content->name);
|
||
|
}
|
||
|
offer->AddGroup(offer_bundle);
|
||
|
if (!UpdateTransportInfoForBundle(offer_bundle, offer.get())) {
|
||
|
LOG(LS_ERROR) << "CreateOffer failed to UpdateTransportInfoForBundle.";
|
||
|
return NULL;
|
||
|
}
|
||
|
if (!UpdateCryptoParamsForBundle(offer_bundle, offer.get())) {
|
||
|
LOG(LS_ERROR) << "CreateOffer failed to UpdateCryptoParamsForBundle.";
|
||
|
return NULL;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
return offer.release();
|
||
|
}
|
||
|
|
||
|
SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
|
||
|
const SessionDescription* offer, const MediaSessionOptions& options,
|
||
|
const SessionDescription* current_description) const {
|
||
|
// The answer contains the intersection of the codecs in the offer with the
|
||
|
// codecs we support. As indicated by XEP-0167, we retain the same payload ids
|
||
|
// from the offer in the answer.
|
||
|
std::unique_ptr<SessionDescription> answer(new SessionDescription());
|
||
|
|
||
|
StreamParamsVec current_streams;
|
||
|
GetCurrentStreamParams(current_description, ¤t_streams);
|
||
|
|
||
|
if (offer) {
|
||
|
ContentInfos::const_iterator it = offer->contents().begin();
|
||
|
for (; it != offer->contents().end(); ++it) {
|
||
|
if (IsMediaContentOfType(&*it, MEDIA_TYPE_AUDIO)) {
|
||
|
if (!AddAudioContentForAnswer(offer, options, current_description,
|
||
|
¤t_streams, answer.get())) {
|
||
|
return NULL;
|
||
|
}
|
||
|
} else if (IsMediaContentOfType(&*it, MEDIA_TYPE_VIDEO)) {
|
||
|
if (!AddVideoContentForAnswer(offer, options, current_description,
|
||
|
¤t_streams, answer.get())) {
|
||
|
return NULL;
|
||
|
}
|
||
|
} else {
|
||
|
ASSERT(IsMediaContentOfType(&*it, MEDIA_TYPE_DATA));
|
||
|
if (!AddDataContentForAnswer(offer, options, current_description,
|
||
|
¤t_streams, answer.get())) {
|
||
|
return NULL;
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
|
||
|
// If the offer supports BUNDLE, and we want to use it too, create a BUNDLE
|
||
|
// group in the answer with the appropriate content names.
|
||
|
if (offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled) {
|
||
|
const ContentGroup* offer_bundle = offer->GetGroupByName(GROUP_TYPE_BUNDLE);
|
||
|
ContentGroup answer_bundle(GROUP_TYPE_BUNDLE);
|
||
|
for (ContentInfos::const_iterator content = answer->contents().begin();
|
||
|
content != answer->contents().end(); ++content) {
|
||
|
if (!content->rejected && offer_bundle->HasContentName(content->name)) {
|
||
|
answer_bundle.AddContentName(content->name);
|
||
|
}
|
||
|
}
|
||
|
if (answer_bundle.FirstContentName()) {
|
||
|
answer->AddGroup(answer_bundle);
|
||
|
|
||
|
// Share the same ICE credentials and crypto params across all contents,
|
||
|
// as BUNDLE requires.
|
||
|
if (!UpdateTransportInfoForBundle(answer_bundle, answer.get())) {
|
||
|
LOG(LS_ERROR) << "CreateAnswer failed to UpdateTransportInfoForBundle.";
|
||
|
return NULL;
|
||
|
}
|
||
|
|
||
|
if (!UpdateCryptoParamsForBundle(answer_bundle, answer.get())) {
|
||
|
LOG(LS_ERROR) << "CreateAnswer failed to UpdateCryptoParamsForBundle.";
|
||
|
return NULL;
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
|
||
|
return answer.release();
|
||
|
}
|
||
|
|
||
|
const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForOffer(
|
||
|
const RtpTransceiverDirection& direction) const {
|
||
|
// If stream is inactive - generate list as if sendrecv.
|
||
|
if (direction.send == direction.recv) {
|
||
|
return audio_sendrecv_codecs_;
|
||
|
} else if (direction.send) {
|
||
|
return audio_send_codecs_;
|
||
|
} else {
|
||
|
return audio_recv_codecs_;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForAnswer(
|
||
|
const RtpTransceiverDirection& offer,
|
||
|
const RtpTransceiverDirection& answer) const {
|
||
|
// For inactive and sendrecv answers, generate lists as if we were to accept
|
||
|
// the offer's direction. See RFC 3264 Section 6.1.
|
||
|
if (answer.send == answer.recv) {
|
||
|
if (offer.send == offer.recv) {
|
||
|
return audio_sendrecv_codecs_;
|
||
|
} else if (offer.send) {
|
||
|
return audio_recv_codecs_;
|
||
|
} else {
|
||
|
return audio_send_codecs_;
|
||
|
}
|
||
|
} else if (answer.send) {
|
||
|
return audio_send_codecs_;
|
||
|
} else {
|
||
|
return audio_recv_codecs_;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
void MediaSessionDescriptionFactory::GetCodecsToOffer(
|
||
|
const SessionDescription* current_description,
|
||
|
const AudioCodecs& supported_audio_codecs,
|
||
|
const VideoCodecs& supported_video_codecs,
|
||
|
const DataCodecs& supported_data_codecs,
|
||
|
AudioCodecs* audio_codecs,
|
||
|
VideoCodecs* video_codecs,
|
||
|
DataCodecs* data_codecs) const {
|
||
|
UsedPayloadTypes used_pltypes;
|
||
|
audio_codecs->clear();
|
||
|
video_codecs->clear();
|
||
|
data_codecs->clear();
|
||
|
|
||
|
|
||
|
// First - get all codecs from the current description if the media type
|
||
|
// is used.
|
||
|
// Add them to |used_pltypes| so the payloadtype is not reused if a new media
|
||
|
// type is added.
|
||
|
if (current_description) {
|
||
|
const AudioContentDescription* audio =
|
||
|
GetFirstAudioContentDescription(current_description);
|
||
|
if (audio) {
|
||
|
*audio_codecs = audio->codecs();
|
||
|
used_pltypes.FindAndSetIdUsed<AudioCodec>(audio_codecs);
|
||
|
}
|
||
|
const VideoContentDescription* video =
|
||
|
GetFirstVideoContentDescription(current_description);
|
||
|
if (video) {
|
||
|
*video_codecs = video->codecs();
|
||
|
used_pltypes.FindAndSetIdUsed<VideoCodec>(video_codecs);
|
||
|
}
|
||
|
const DataContentDescription* data =
|
||
|
GetFirstDataContentDescription(current_description);
|
||
|
if (data) {
|
||
|
*data_codecs = data->codecs();
|
||
|
used_pltypes.FindAndSetIdUsed<DataCodec>(data_codecs);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
// Add our codecs that are not in |current_description|.
|
||
|
FindCodecsToOffer<AudioCodec>(supported_audio_codecs, audio_codecs,
|
||
|
&used_pltypes);
|
||
|
FindCodecsToOffer<VideoCodec>(supported_video_codecs, video_codecs,
|
||
|
&used_pltypes);
|
||
|
FindCodecsToOffer<DataCodec>(supported_data_codecs, data_codecs,
|
||
|
&used_pltypes);
|
||
|
}
|
||
|
|
||
|
void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer(
|
||
|
const SessionDescription* current_description,
|
||
|
RtpHeaderExtensions* audio_extensions,
|
||
|
RtpHeaderExtensions* video_extensions) const {
|
||
|
// All header extensions allocated from the same range to avoid potential
|
||
|
// issues when using BUNDLE.
|
||
|
UsedRtpHeaderExtensionIds used_ids;
|
||
|
RtpHeaderExtensions all_extensions;
|
||
|
audio_extensions->clear();
|
||
|
video_extensions->clear();
|
||
|
|
||
|
// First - get all extensions from the current description if the media type
|
||
|
// is used.
|
||
|
// Add them to |used_ids| so the local ids are not reused if a new media
|
||
|
// type is added.
|
||
|
if (current_description) {
|
||
|
const AudioContentDescription* audio =
|
||
|
GetFirstAudioContentDescription(current_description);
|
||
|
if (audio) {
|
||
|
*audio_extensions = audio->rtp_header_extensions();
|
||
|
FindAndSetRtpHdrExtUsed(audio_extensions, &all_extensions, &used_ids);
|
||
|
}
|
||
|
const VideoContentDescription* video =
|
||
|
GetFirstVideoContentDescription(current_description);
|
||
|
if (video) {
|
||
|
*video_extensions = video->rtp_header_extensions();
|
||
|
FindAndSetRtpHdrExtUsed(video_extensions, &all_extensions, &used_ids);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
// Add our default RTP header extensions that are not in
|
||
|
// |current_description|.
|
||
|
FindRtpHdrExtsToOffer(audio_rtp_header_extensions(), audio_extensions,
|
||
|
&all_extensions, &used_ids);
|
||
|
FindRtpHdrExtsToOffer(video_rtp_header_extensions(), video_extensions,
|
||
|
&all_extensions, &used_ids);
|
||
|
}
|
||
|
|
||
|
bool MediaSessionDescriptionFactory::AddTransportOffer(
|
||
|
const std::string& content_name,
|
||
|
const TransportOptions& transport_options,
|
||
|
const SessionDescription* current_desc,
|
||
|
SessionDescription* offer_desc) const {
|
||
|
if (!transport_desc_factory_)
|
||
|
return false;
|
||
|
const TransportDescription* current_tdesc =
|
||
|
GetTransportDescription(content_name, current_desc);
|
||
|
std::unique_ptr<TransportDescription> new_tdesc(
|
||
|
transport_desc_factory_->CreateOffer(transport_options, current_tdesc));
|
||
|
bool ret = (new_tdesc.get() != NULL &&
|
||
|
offer_desc->AddTransportInfo(TransportInfo(content_name, *new_tdesc)));
|
||
|
if (!ret) {
|
||
|
LOG(LS_ERROR)
|
||
|
<< "Failed to AddTransportOffer, content name=" << content_name;
|
||
|
}
|
||
|
return ret;
|
||
|
}
|
||
|
|
||
|
TransportDescription* MediaSessionDescriptionFactory::CreateTransportAnswer(
|
||
|
const std::string& content_name,
|
||
|
const SessionDescription* offer_desc,
|
||
|
const TransportOptions& transport_options,
|
||
|
const SessionDescription* current_desc) const {
|
||
|
if (!transport_desc_factory_)
|
||
|
return NULL;
|
||
|
const TransportDescription* offer_tdesc =
|
||
|
GetTransportDescription(content_name, offer_desc);
|
||
|
const TransportDescription* current_tdesc =
|
||
|
GetTransportDescription(content_name, current_desc);
|
||
|
return
|
||
|
transport_desc_factory_->CreateAnswer(offer_tdesc, transport_options,
|
||
|
current_tdesc);
|
||
|
}
|
||
|
|
||
|
bool MediaSessionDescriptionFactory::AddTransportAnswer(
|
||
|
const std::string& content_name,
|
||
|
const TransportDescription& transport_desc,
|
||
|
SessionDescription* answer_desc) const {
|
||
|
if (!answer_desc->AddTransportInfo(TransportInfo(content_name,
|
||
|
transport_desc))) {
|
||
|
LOG(LS_ERROR)
|
||
|
<< "Failed to AddTransportAnswer, content name=" << content_name;
|
||
|
return false;
|
||
|
}
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
bool MediaSessionDescriptionFactory::AddAudioContentForOffer(
|
||
|
const MediaSessionOptions& options,
|
||
|
const SessionDescription* current_description,
|
||
|
const RtpHeaderExtensions& audio_rtp_extensions,
|
||
|
const AudioCodecs& audio_codecs,
|
||
|
StreamParamsVec* current_streams,
|
||
|
SessionDescription* desc) const {
|
||
|
const ContentInfo* current_audio_content =
|
||
|
GetFirstAudioContent(current_description);
|
||
|
std::string content_name =
|
||
|
current_audio_content ? current_audio_content->name : CN_AUDIO;
|
||
|
|
||
|
cricket::SecurePolicy sdes_policy =
|
||
|
IsDtlsActive(content_name, current_description) ? cricket::SEC_DISABLED
|
||
|
: secure();
|
||
|
|
||
|
std::unique_ptr<AudioContentDescription> audio(new AudioContentDescription());
|
||
|
std::vector<std::string> crypto_suites;
|
||
|
GetSupportedAudioCryptoSuiteNames(&crypto_suites);
|
||
|
if (!CreateMediaContentOffer(
|
||
|
options,
|
||
|
audio_codecs,
|
||
|
sdes_policy,
|
||
|
GetCryptos(GetFirstAudioContentDescription(current_description)),
|
||
|
crypto_suites,
|
||
|
audio_rtp_extensions,
|
||
|
add_legacy_,
|
||
|
current_streams,
|
||
|
audio.get())) {
|
||
|
return false;
|
||
|
}
|
||
|
audio->set_lang(lang_);
|
||
|
|
||
|
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
|
||
|
SetMediaProtocol(secure_transport, audio.get());
|
||
|
|
||
|
auto offer_rtd =
|
||
|
RtpTransceiverDirection(!audio->streams().empty(), options.recv_audio);
|
||
|
audio->set_direction(offer_rtd.ToMediaContentDirection());
|
||
|
|
||
|
desc->AddContent(content_name, NS_JINGLE_RTP, audio.release());
|
||
|
if (!AddTransportOffer(content_name,
|
||
|
GetTransportOptions(options, content_name),
|
||
|
current_description, desc)) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
bool MediaSessionDescriptionFactory::AddVideoContentForOffer(
|
||
|
const MediaSessionOptions& options,
|
||
|
const SessionDescription* current_description,
|
||
|
const RtpHeaderExtensions& video_rtp_extensions,
|
||
|
const VideoCodecs& video_codecs,
|
||
|
StreamParamsVec* current_streams,
|
||
|
SessionDescription* desc) const {
|
||
|
const ContentInfo* current_video_content =
|
||
|
GetFirstVideoContent(current_description);
|
||
|
std::string content_name =
|
||
|
current_video_content ? current_video_content->name : CN_VIDEO;
|
||
|
|
||
|
cricket::SecurePolicy sdes_policy =
|
||
|
IsDtlsActive(content_name, current_description) ? cricket::SEC_DISABLED
|
||
|
: secure();
|
||
|
|
||
|
std::unique_ptr<VideoContentDescription> video(new VideoContentDescription());
|
||
|
std::vector<std::string> crypto_suites;
|
||
|
GetSupportedVideoCryptoSuiteNames(&crypto_suites);
|
||
|
if (!CreateMediaContentOffer(
|
||
|
options,
|
||
|
video_codecs,
|
||
|
sdes_policy,
|
||
|
GetCryptos(GetFirstVideoContentDescription(current_description)),
|
||
|
crypto_suites,
|
||
|
video_rtp_extensions,
|
||
|
add_legacy_,
|
||
|
current_streams,
|
||
|
video.get())) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
video->set_bandwidth(options.video_bandwidth);
|
||
|
|
||
|
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
|
||
|
SetMediaProtocol(secure_transport, video.get());
|
||
|
|
||
|
if (!video->streams().empty()) {
|
||
|
if (options.recv_video) {
|
||
|
video->set_direction(MD_SENDRECV);
|
||
|
} else {
|
||
|
video->set_direction(MD_SENDONLY);
|
||
|
}
|
||
|
} else {
|
||
|
if (options.recv_video) {
|
||
|
video->set_direction(MD_RECVONLY);
|
||
|
} else {
|
||
|
video->set_direction(MD_INACTIVE);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
desc->AddContent(content_name, NS_JINGLE_RTP, video.release());
|
||
|
if (!AddTransportOffer(content_name,
|
||
|
GetTransportOptions(options, content_name),
|
||
|
current_description, desc)) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
bool MediaSessionDescriptionFactory::AddDataContentForOffer(
|
||
|
const MediaSessionOptions& options,
|
||
|
const SessionDescription* current_description,
|
||
|
DataCodecs* data_codecs,
|
||
|
StreamParamsVec* current_streams,
|
||
|
SessionDescription* desc) const {
|
||
|
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
|
||
|
|
||
|
std::unique_ptr<DataContentDescription> data(new DataContentDescription());
|
||
|
bool is_sctp = (options.data_channel_type == DCT_SCTP);
|
||
|
|
||
|
FilterDataCodecs(data_codecs, is_sctp);
|
||
|
|
||
|
const ContentInfo* current_data_content =
|
||
|
GetFirstDataContent(current_description);
|
||
|
std::string content_name =
|
||
|
current_data_content ? current_data_content->name : CN_DATA;
|
||
|
|
||
|
cricket::SecurePolicy sdes_policy =
|
||
|
IsDtlsActive(content_name, current_description) ? cricket::SEC_DISABLED
|
||
|
: secure();
|
||
|
std::vector<std::string> crypto_suites;
|
||
|
if (is_sctp) {
|
||
|
// SDES doesn't make sense for SCTP, so we disable it, and we only
|
||
|
// get SDES crypto suites for RTP-based data channels.
|
||
|
sdes_policy = cricket::SEC_DISABLED;
|
||
|
// Unlike SetMediaProtocol below, we need to set the protocol
|
||
|
// before we call CreateMediaContentOffer. Otherwise,
|
||
|
// CreateMediaContentOffer won't know this is SCTP and will
|
||
|
// generate SSRCs rather than SIDs.
|
||
|
data->set_protocol(
|
||
|
secure_transport ? kMediaProtocolDtlsSctp : kMediaProtocolSctp);
|
||
|
} else {
|
||
|
GetSupportedDataCryptoSuiteNames(&crypto_suites);
|
||
|
}
|
||
|
|
||
|
if (!CreateMediaContentOffer(
|
||
|
options,
|
||
|
*data_codecs,
|
||
|
sdes_policy,
|
||
|
GetCryptos(GetFirstDataContentDescription(current_description)),
|
||
|
crypto_suites,
|
||
|
RtpHeaderExtensions(),
|
||
|
add_legacy_,
|
||
|
current_streams,
|
||
|
data.get())) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
if (is_sctp) {
|
||
|
desc->AddContent(content_name, NS_JINGLE_DRAFT_SCTP, data.release());
|
||
|
} else {
|
||
|
data->set_bandwidth(options.data_bandwidth);
|
||
|
SetMediaProtocol(secure_transport, data.get());
|
||
|
desc->AddContent(content_name, NS_JINGLE_RTP, data.release());
|
||
|
}
|
||
|
if (!AddTransportOffer(content_name,
|
||
|
GetTransportOptions(options, content_name),
|
||
|
current_description, desc)) {
|
||
|
return false;
|
||
|
}
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
bool MediaSessionDescriptionFactory::AddAudioContentForAnswer(
|
||
|
const SessionDescription* offer,
|
||
|
const MediaSessionOptions& options,
|
||
|
const SessionDescription* current_description,
|
||
|
StreamParamsVec* current_streams,
|
||
|
SessionDescription* answer) const {
|
||
|
const ContentInfo* audio_content = GetFirstAudioContent(offer);
|
||
|
const AudioContentDescription* offer_audio =
|
||
|
static_cast<const AudioContentDescription*>(audio_content->description);
|
||
|
|
||
|
std::unique_ptr<TransportDescription> audio_transport(CreateTransportAnswer(
|
||
|
audio_content->name, offer,
|
||
|
GetTransportOptions(options, audio_content->name), current_description));
|
||
|
if (!audio_transport) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
// Pick codecs based on the requested communications direction in the offer.
|
||
|
const bool wants_send =
|
||
|
options.HasSendMediaStream(MEDIA_TYPE_AUDIO) || add_legacy_;
|
||
|
auto wants_rtd = RtpTransceiverDirection(wants_send, options.recv_audio);
|
||
|
auto offer_rtd =
|
||
|
RtpTransceiverDirection::FromMediaContentDirection(
|
||
|
offer_audio->direction());
|
||
|
auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
|
||
|
AudioCodecs audio_codecs = GetAudioCodecsForAnswer(offer_rtd, answer_rtd);
|
||
|
if (!options.vad_enabled) {
|
||
|
StripCNCodecs(&audio_codecs);
|
||
|
}
|
||
|
|
||
|
bool bundle_enabled =
|
||
|
offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled;
|
||
|
std::unique_ptr<AudioContentDescription> audio_answer(
|
||
|
new AudioContentDescription());
|
||
|
// Do not require or create SDES cryptos if DTLS is used.
|
||
|
cricket::SecurePolicy sdes_policy =
|
||
|
audio_transport->secure() ? cricket::SEC_DISABLED : secure();
|
||
|
if (!CreateMediaContentAnswer(
|
||
|
offer_audio,
|
||
|
options,
|
||
|
audio_codecs,
|
||
|
sdes_policy,
|
||
|
GetCryptos(GetFirstAudioContentDescription(current_description)),
|
||
|
audio_rtp_extensions_,
|
||
|
current_streams,
|
||
|
add_legacy_,
|
||
|
bundle_enabled,
|
||
|
audio_answer.get())) {
|
||
|
return false; // Fails the session setup.
|
||
|
}
|
||
|
|
||
|
bool rejected = !options.has_audio() || audio_content->rejected ||
|
||
|
!IsMediaProtocolSupported(MEDIA_TYPE_AUDIO,
|
||
|
audio_answer->protocol(),
|
||
|
audio_transport->secure());
|
||
|
if (!rejected) {
|
||
|
AddTransportAnswer(audio_content->name, *(audio_transport.get()), answer);
|
||
|
} else {
|
||
|
// RFC 3264
|
||
|
// The answer MUST contain the same number of m-lines as the offer.
|
||
|
LOG(LS_INFO) << "Audio is not supported in the answer.";
|
||
|
}
|
||
|
|
||
|
answer->AddContent(audio_content->name, audio_content->type, rejected,
|
||
|
audio_answer.release());
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
bool MediaSessionDescriptionFactory::AddVideoContentForAnswer(
|
||
|
const SessionDescription* offer,
|
||
|
const MediaSessionOptions& options,
|
||
|
const SessionDescription* current_description,
|
||
|
StreamParamsVec* current_streams,
|
||
|
SessionDescription* answer) const {
|
||
|
const ContentInfo* video_content = GetFirstVideoContent(offer);
|
||
|
std::unique_ptr<TransportDescription> video_transport(CreateTransportAnswer(
|
||
|
video_content->name, offer,
|
||
|
GetTransportOptions(options, video_content->name), current_description));
|
||
|
if (!video_transport) {
|
||
|
return false;
|
||
|
}
|
||
|
|
||
|
std::unique_ptr<VideoContentDescription> video_answer(
|
||
|
new VideoContentDescription());
|
||
|
// Do not require or create SDES cryptos if DTLS is used.
|
||
|
cricket::SecurePolicy sdes_policy =
|
||
|
video_transport->secure() ? cricket::SEC_DISABLED : secure();
|
||
|
bool bundle_enabled =
|
||
|
offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled;
|
||
|
if (!CreateMediaContentAnswer(
|
||
|
static_cast<const VideoContentDescription*>(
|
||
|
video_content->description),
|
||
|
options,
|
||
|
video_codecs_,
|
||
|
sdes_policy,
|
||
|
GetCryptos(GetFirstVideoContentDescription(current_description)),
|
||
|
video_rtp_extensions_,
|
||
|
current_streams,
|
||
|
add_legacy_,
|
||
|
bundle_enabled,
|
||
|
video_answer.get())) {
|
||
|
return false;
|
||
|
}
|
||
|
bool rejected = !options.has_video() || video_content->rejected ||
|
||
|
!IsMediaProtocolSupported(MEDIA_TYPE_VIDEO,
|
||
|
video_answer->protocol(),
|
||
|
video_transport->secure());
|
||
|
if (!rejected) {
|
||
|
if (!AddTransportAnswer(video_content->name, *(video_transport.get()),
|
||
|
answer)) {
|
||
|
return false;
|
||
|
}
|
||
|
video_answer->set_bandwidth(options.video_bandwidth);
|
||
|
} else {
|
||
|
// RFC 3264
|
||
|
// The answer MUST contain the same number of m-lines as the offer.
|
||
|
LOG(LS_INFO) << "Video is not supported in the answer.";
|
||
|
}
|
||
|
answer->AddContent(video_content->name, video_content->type, rejected,
|
||
|
video_answer.release());
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
bool MediaSessionDescriptionFactory::AddDataContentForAnswer(
|
||
|
const SessionDescription* offer,
|
||
|
const MediaSessionOptions& options,
|
||
|
const SessionDescription* current_description,
|
||
|
StreamParamsVec* current_streams,
|
||
|
SessionDescription* answer) const {
|
||
|
const ContentInfo* data_content = GetFirstDataContent(offer);
|
||
|
std::unique_ptr<TransportDescription> data_transport(CreateTransportAnswer(
|
||
|
data_content->name, offer,
|
||
|
GetTransportOptions(options, data_content->name), current_description));
|
||
|
if (!data_transport) {
|
||
|
return false;
|
||
|
}
|
||
|
bool is_sctp = (options.data_channel_type == DCT_SCTP);
|
||
|
std::vector<DataCodec> data_codecs(data_codecs_);
|
||
|
FilterDataCodecs(&data_codecs, is_sctp);
|
||
|
|
||
|
std::unique_ptr<DataContentDescription> data_answer(
|
||
|
new DataContentDescription());
|
||
|
// Do not require or create SDES cryptos if DTLS is used.
|
||
|
cricket::SecurePolicy sdes_policy =
|
||
|
data_transport->secure() ? cricket::SEC_DISABLED : secure();
|
||
|
bool bundle_enabled =
|
||
|
offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled;
|
||
|
if (!CreateMediaContentAnswer(
|
||
|
static_cast<const DataContentDescription*>(
|
||
|
data_content->description),
|
||
|
options,
|
||
|
data_codecs_,
|
||
|
sdes_policy,
|
||
|
GetCryptos(GetFirstDataContentDescription(current_description)),
|
||
|
RtpHeaderExtensions(),
|
||
|
current_streams,
|
||
|
add_legacy_,
|
||
|
bundle_enabled,
|
||
|
data_answer.get())) {
|
||
|
return false; // Fails the session setup.
|
||
|
}
|
||
|
|
||
|
bool rejected = !options.has_data() || data_content->rejected ||
|
||
|
!IsMediaProtocolSupported(MEDIA_TYPE_DATA,
|
||
|
data_answer->protocol(),
|
||
|
data_transport->secure());
|
||
|
if (!rejected) {
|
||
|
data_answer->set_bandwidth(options.data_bandwidth);
|
||
|
if (!AddTransportAnswer(data_content->name, *(data_transport.get()),
|
||
|
answer)) {
|
||
|
return false;
|
||
|
}
|
||
|
} else {
|
||
|
// RFC 3264
|
||
|
// The answer MUST contain the same number of m-lines as the offer.
|
||
|
LOG(LS_INFO) << "Data is not supported in the answer.";
|
||
|
}
|
||
|
answer->AddContent(data_content->name, data_content->type, rejected,
|
||
|
data_answer.release());
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
bool IsMediaContent(const ContentInfo* content) {
|
||
|
return (content &&
|
||
|
(content->type == NS_JINGLE_RTP ||
|
||
|
content->type == NS_JINGLE_DRAFT_SCTP));
|
||
|
}
|
||
|
|
||
|
bool IsAudioContent(const ContentInfo* content) {
|
||
|
return IsMediaContentOfType(content, MEDIA_TYPE_AUDIO);
|
||
|
}
|
||
|
|
||
|
bool IsVideoContent(const ContentInfo* content) {
|
||
|
return IsMediaContentOfType(content, MEDIA_TYPE_VIDEO);
|
||
|
}
|
||
|
|
||
|
bool IsDataContent(const ContentInfo* content) {
|
||
|
return IsMediaContentOfType(content, MEDIA_TYPE_DATA);
|
||
|
}
|
||
|
|
||
|
const ContentInfo* GetFirstMediaContent(const ContentInfos& contents,
|
||
|
MediaType media_type) {
|
||
|
for (const ContentInfo& content : contents) {
|
||
|
if (IsMediaContentOfType(&content, media_type)) {
|
||
|
return &content;
|
||
|
}
|
||
|
}
|
||
|
return nullptr;
|
||
|
}
|
||
|
|
||
|
const ContentInfo* GetFirstAudioContent(const ContentInfos& contents) {
|
||
|
return GetFirstMediaContent(contents, MEDIA_TYPE_AUDIO);
|
||
|
}
|
||
|
|
||
|
const ContentInfo* GetFirstVideoContent(const ContentInfos& contents) {
|
||
|
return GetFirstMediaContent(contents, MEDIA_TYPE_VIDEO);
|
||
|
}
|
||
|
|
||
|
const ContentInfo* GetFirstDataContent(const ContentInfos& contents) {
|
||
|
return GetFirstMediaContent(contents, MEDIA_TYPE_DATA);
|
||
|
}
|
||
|
|
||
|
static const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc,
|
||
|
MediaType media_type) {
|
||
|
if (sdesc == nullptr) {
|
||
|
return nullptr;
|
||
|
}
|
||
|
|
||
|
return GetFirstMediaContent(sdesc->contents(), media_type);
|
||
|
}
|
||
|
|
||
|
const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc) {
|
||
|
return GetFirstMediaContent(sdesc, MEDIA_TYPE_AUDIO);
|
||
|
}
|
||
|
|
||
|
const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc) {
|
||
|
return GetFirstMediaContent(sdesc, MEDIA_TYPE_VIDEO);
|
||
|
}
|
||
|
|
||
|
const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc) {
|
||
|
return GetFirstMediaContent(sdesc, MEDIA_TYPE_DATA);
|
||
|
}
|
||
|
|
||
|
const MediaContentDescription* GetFirstMediaContentDescription(
|
||
|
const SessionDescription* sdesc, MediaType media_type) {
|
||
|
const ContentInfo* content = GetFirstMediaContent(sdesc, media_type);
|
||
|
const ContentDescription* description = content ? content->description : NULL;
|
||
|
return static_cast<const MediaContentDescription*>(description);
|
||
|
}
|
||
|
|
||
|
const AudioContentDescription* GetFirstAudioContentDescription(
|
||
|
const SessionDescription* sdesc) {
|
||
|
return static_cast<const AudioContentDescription*>(
|
||
|
GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_AUDIO));
|
||
|
}
|
||
|
|
||
|
const VideoContentDescription* GetFirstVideoContentDescription(
|
||
|
const SessionDescription* sdesc) {
|
||
|
return static_cast<const VideoContentDescription*>(
|
||
|
GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_VIDEO));
|
||
|
}
|
||
|
|
||
|
const DataContentDescription* GetFirstDataContentDescription(
|
||
|
const SessionDescription* sdesc) {
|
||
|
return static_cast<const DataContentDescription*>(
|
||
|
GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA));
|
||
|
}
|
||
|
|
||
|
//
|
||
|
// Non-const versions of the above functions.
|
||
|
//
|
||
|
|
||
|
ContentInfo* GetFirstMediaContent(ContentInfos& contents,
|
||
|
MediaType media_type) {
|
||
|
for (ContentInfo& content : contents) {
|
||
|
if (IsMediaContentOfType(&content, media_type)) {
|
||
|
return &content;
|
||
|
}
|
||
|
}
|
||
|
return nullptr;
|
||
|
}
|
||
|
|
||
|
ContentInfo* GetFirstAudioContent(ContentInfos& contents) {
|
||
|
return GetFirstMediaContent(contents, MEDIA_TYPE_AUDIO);
|
||
|
}
|
||
|
|
||
|
ContentInfo* GetFirstVideoContent(ContentInfos& contents) {
|
||
|
return GetFirstMediaContent(contents, MEDIA_TYPE_VIDEO);
|
||
|
}
|
||
|
|
||
|
ContentInfo* GetFirstDataContent(ContentInfos& contents) {
|
||
|
return GetFirstMediaContent(contents, MEDIA_TYPE_DATA);
|
||
|
}
|
||
|
|
||
|
static ContentInfo* GetFirstMediaContent(SessionDescription* sdesc,
|
||
|
MediaType media_type) {
|
||
|
if (sdesc == nullptr) {
|
||
|
return nullptr;
|
||
|
}
|
||
|
|
||
|
return GetFirstMediaContent(sdesc->contents(), media_type);
|
||
|
}
|
||
|
|
||
|
ContentInfo* GetFirstAudioContent(SessionDescription* sdesc) {
|
||
|
return GetFirstMediaContent(sdesc, MEDIA_TYPE_AUDIO);
|
||
|
}
|
||
|
|
||
|
ContentInfo* GetFirstVideoContent(SessionDescription* sdesc) {
|
||
|
return GetFirstMediaContent(sdesc, MEDIA_TYPE_VIDEO);
|
||
|
}
|
||
|
|
||
|
ContentInfo* GetFirstDataContent(SessionDescription* sdesc) {
|
||
|
return GetFirstMediaContent(sdesc, MEDIA_TYPE_DATA);
|
||
|
}
|
||
|
|
||
|
MediaContentDescription* GetFirstMediaContentDescription(
|
||
|
SessionDescription* sdesc,
|
||
|
MediaType media_type) {
|
||
|
ContentInfo* content = GetFirstMediaContent(sdesc, media_type);
|
||
|
ContentDescription* description = content ? content->description : NULL;
|
||
|
return static_cast<MediaContentDescription*>(description);
|
||
|
}
|
||
|
|
||
|
AudioContentDescription* GetFirstAudioContentDescription(
|
||
|
SessionDescription* sdesc) {
|
||
|
return static_cast<AudioContentDescription*>(
|
||
|
GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_AUDIO));
|
||
|
}
|
||
|
|
||
|
VideoContentDescription* GetFirstVideoContentDescription(
|
||
|
SessionDescription* sdesc) {
|
||
|
return static_cast<VideoContentDescription*>(
|
||
|
GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_VIDEO));
|
||
|
}
|
||
|
|
||
|
DataContentDescription* GetFirstDataContentDescription(
|
||
|
SessionDescription* sdesc) {
|
||
|
return static_cast<DataContentDescription*>(
|
||
|
GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA));
|
||
|
}
|
||
|
|
||
|
} // namespace cricket
|