| /* |
| * Copyright 2004 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "pc/media_session.h" |
| |
| #include <stddef.h> |
| |
| #include <algorithm> |
| #include <map> |
| #include <memory> |
| #include <set> |
| #include <unordered_map> |
| #include <utility> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/strings/match.h" |
| #include "absl/strings/string_view.h" |
| #include "absl/types/optional.h" |
| #include "api/crypto_params.h" |
| #include "api/video_codecs/h264_profile_level_id.h" |
| #include "media/base/codec.h" |
| #include "media/base/media_constants.h" |
| #include "media/base/sdp_video_format_utils.h" |
| #include "media/sctp/sctp_transport_internal.h" |
| #include "p2p/base/p2p_constants.h" |
| #include "pc/channel_manager.h" |
| #include "pc/media_protocol_names.h" |
| #include "pc/rtp_media_utils.h" |
| #include "pc/used_ids.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/helpers.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/ssl_stream_adapter.h" |
| #include "rtc_base/string_encode.h" |
| #include "rtc_base/third_party/base64/base64.h" |
| #include "rtc_base/unique_id_generator.h" |
| #include "system_wrappers/include/field_trial.h" |
| |
| namespace { |
| |
| using rtc::UniqueRandomIdGenerator; |
| using webrtc::RtpTransceiverDirection; |
| |
| const char kInline[] = "inline:"; |
| |
| void GetSupportedSdesCryptoSuiteNames( |
| void (*func)(const webrtc::CryptoOptions&, std::vector<int>*), |
| const webrtc::CryptoOptions& crypto_options, |
| std::vector<std::string>* names) { |
| std::vector<int> crypto_suites; |
| func(crypto_options, &crypto_suites); |
| for (const auto crypto : crypto_suites) { |
| names->push_back(rtc::SrtpCryptoSuiteToName(crypto)); |
| } |
| } |
| |
| webrtc::RtpExtension RtpExtensionFromCapability( |
| const webrtc::RtpHeaderExtensionCapability& capability) { |
| return webrtc::RtpExtension(capability.uri, |
| capability.preferred_id.value_or(1)); |
| } |
| |
| cricket::RtpHeaderExtensions RtpHeaderExtensionsFromCapabilities( |
| const std::vector<webrtc::RtpHeaderExtensionCapability>& capabilities) { |
| cricket::RtpHeaderExtensions exts; |
| for (const auto& capability : capabilities) { |
| exts.push_back(RtpExtensionFromCapability(capability)); |
| } |
| return exts; |
| } |
| |
| std::vector<webrtc::RtpHeaderExtensionCapability> |
| UnstoppedRtpHeaderExtensionCapabilities( |
| std::vector<webrtc::RtpHeaderExtensionCapability> capabilities) { |
| capabilities.erase( |
| std::remove_if( |
| capabilities.begin(), capabilities.end(), |
| [](const webrtc::RtpHeaderExtensionCapability& capability) { |
| return capability.direction == RtpTransceiverDirection::kStopped; |
| }), |
| capabilities.end()); |
| return capabilities; |
| } |
| |
| bool IsCapabilityPresent(const webrtc::RtpHeaderExtensionCapability& capability, |
| const cricket::RtpHeaderExtensions& extensions) { |
| return std::find_if(extensions.begin(), extensions.end(), |
| [&capability](const webrtc::RtpExtension& extension) { |
| return capability.uri == extension.uri; |
| }) != extensions.end(); |
| } |
| |
| cricket::RtpHeaderExtensions UnstoppedOrPresentRtpHeaderExtensions( |
| const std::vector<webrtc::RtpHeaderExtensionCapability>& capabilities, |
| const cricket::RtpHeaderExtensions& unencrypted, |
| const cricket::RtpHeaderExtensions& encrypted) { |
| cricket::RtpHeaderExtensions extensions; |
| for (const auto& capability : capabilities) { |
| if (capability.direction != RtpTransceiverDirection::kStopped || |
| IsCapabilityPresent(capability, unencrypted) || |
| IsCapabilityPresent(capability, encrypted)) { |
| extensions.push_back(RtpExtensionFromCapability(capability)); |
| } |
| } |
| return extensions; |
| } |
| |
| } // namespace |
| |
| namespace cricket { |
| |
| // RTP Profile names |
| // http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xml |
| // RFC4585 |
| const char kMediaProtocolAvpf[] = "RTP/AVPF"; |
| // RFC5124 |
| const char kMediaProtocolDtlsSavpf[] = "UDP/TLS/RTP/SAVPF"; |
| |
| // We always generate offers with "UDP/TLS/RTP/SAVPF" when using DTLS-SRTP, |
| // but we tolerate "RTP/SAVPF" in offers we receive, for compatibility. |
| const char kMediaProtocolSavpf[] = "RTP/SAVPF"; |
| |
| // Note that the below functions support some protocol strings purely for |
| // legacy compatibility, as required by JSEP in Section 5.1.2, Profile Names |
| // and Interoperability. |
| |
| 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 RtpTransceiverDirection NegotiateRtpTransceiverDirection( |
| RtpTransceiverDirection offer, |
| RtpTransceiverDirection wants) { |
| bool offer_send = webrtc::RtpTransceiverDirectionHasSend(offer); |
| bool offer_recv = webrtc::RtpTransceiverDirectionHasRecv(offer); |
| bool wants_send = webrtc::RtpTransceiverDirectionHasSend(wants); |
| bool wants_recv = webrtc::RtpTransceiverDirectionHasRecv(wants); |
| return webrtc::RtpTransceiverDirectionFromSendRecv(offer_recv && wants_send, |
| offer_send && wants_recv); |
| } |
| |
| static bool IsMediaContentOfType(const ContentInfo* content, |
| MediaType media_type) { |
| if (!content || !content->media_description()) { |
| return false; |
| } |
| return content->media_description()->type() == media_type; |
| } |
| |
| static bool CreateCryptoParams(int tag, |
| const std::string& cipher, |
| CryptoParams* crypto_out) { |
| int key_len; |
| int salt_len; |
| if (!rtc::GetSrtpKeyAndSaltLengths(rtc::SrtpCryptoSuiteFromName(cipher), |
| &key_len, &salt_len)) { |
| return false; |
| } |
| |
| int master_key_len = key_len + salt_len; |
| std::string master_key; |
| if (!rtc::CreateRandomData(master_key_len, &master_key)) { |
| return false; |
| } |
| |
| RTC_CHECK_EQ(master_key_len, master_key.size()); |
| std::string key = rtc::Base64::Encode(master_key); |
| |
| crypto_out->tag = tag; |
| crypto_out->cipher_suite = cipher; |
| crypto_out->key_params = kInline; |
| crypto_out->key_params += key; |
| return true; |
| } |
| |
| static bool AddCryptoParams(const std::string& cipher_suite, |
| CryptoParamsVec* cryptos_out) { |
| int size = static_cast<int>(cryptos_out->size()); |
| |
| cryptos_out->resize(size + 1); |
| return CreateCryptoParams(size, cipher_suite, &cryptos_out->at(size)); |
| } |
| |
| void AddMediaCryptos(const CryptoParamsVec& cryptos, |
| MediaContentDescription* media) { |
| for (const CryptoParams& crypto : cryptos) { |
| media->AddCrypto(crypto); |
| } |
| } |
| |
| bool CreateMediaCryptos(const std::vector<std::string>& crypto_suites, |
| MediaContentDescription* media) { |
| CryptoParamsVec cryptos; |
| for (const std::string& crypto_suite : crypto_suites) { |
| if (!AddCryptoParams(crypto_suite, &cryptos)) { |
| return false; |
| } |
| } |
| AddMediaCryptos(cryptos, media); |
| return true; |
| } |
| |
| const CryptoParamsVec* GetCryptos(const ContentInfo* content) { |
| if (!content || !content->media_description()) { |
| return nullptr; |
| } |
| return &content->media_description()->cryptos(); |
| } |
| |
| bool FindMatchingCrypto(const CryptoParamsVec& cryptos, |
| const CryptoParams& crypto, |
| CryptoParams* crypto_out) { |
| auto it = absl::c_find_if( |
| cryptos, [&crypto](const CryptoParams& c) { return crypto.Matches(c); }); |
| if (it == cryptos.end()) { |
| return false; |
| } |
| *crypto_out = *it; |
| return true; |
| } |
| |
| // For audio, HMAC 32 (if enabled) is prefered over HMAC 80 because of the |
| // low overhead. |
| void GetSupportedAudioSdesCryptoSuites( |
| const webrtc::CryptoOptions& crypto_options, |
| std::vector<int>* crypto_suites) { |
| if (crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher) { |
| crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_32); |
| } |
| crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80); |
| if (crypto_options.srtp.enable_gcm_crypto_suites) { |
| crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM); |
| crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM); |
| } |
| } |
| |
| void GetSupportedAudioSdesCryptoSuiteNames( |
| const webrtc::CryptoOptions& crypto_options, |
| std::vector<std::string>* crypto_suite_names) { |
| GetSupportedSdesCryptoSuiteNames(GetSupportedAudioSdesCryptoSuites, |
| crypto_options, crypto_suite_names); |
| } |
| |
| void GetSupportedVideoSdesCryptoSuites( |
| const webrtc::CryptoOptions& crypto_options, |
| std::vector<int>* crypto_suites) { |
| crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80); |
| if (crypto_options.srtp.enable_gcm_crypto_suites) { |
| crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM); |
| crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM); |
| } |
| } |
| |
| void GetSupportedVideoSdesCryptoSuiteNames( |
| const webrtc::CryptoOptions& crypto_options, |
| std::vector<std::string>* crypto_suite_names) { |
| GetSupportedSdesCryptoSuiteNames(GetSupportedVideoSdesCryptoSuites, |
| crypto_options, crypto_suite_names); |
| } |
| |
| void GetSupportedDataSdesCryptoSuites( |
| const webrtc::CryptoOptions& crypto_options, |
| std::vector<int>* crypto_suites) { |
| crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80); |
| if (crypto_options.srtp.enable_gcm_crypto_suites) { |
| crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM); |
| crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM); |
| } |
| } |
| |
| void GetSupportedDataSdesCryptoSuiteNames( |
| const webrtc::CryptoOptions& crypto_options, |
| std::vector<std::string>* crypto_suite_names) { |
| GetSupportedSdesCryptoSuiteNames(GetSupportedDataSdesCryptoSuites, |
| crypto_options, crypto_suite_names); |
| } |
| |
| // Support any GCM cipher (if enabled through options). For video support only |
| // 80-bit SHA1 HMAC. For audio 32-bit HMAC is tolerated (if enabled) unless |
| // bundle is enabled because it is low overhead. |
| // Pick the crypto in the list that is supported. |
| static bool SelectCrypto(const MediaContentDescription* offer, |
| bool bundle, |
| const webrtc::CryptoOptions& crypto_options, |
| CryptoParams* crypto_out) { |
| bool audio = offer->type() == MEDIA_TYPE_AUDIO; |
| const CryptoParamsVec& cryptos = offer->cryptos(); |
| |
| for (const CryptoParams& crypto : cryptos) { |
| if ((crypto_options.srtp.enable_gcm_crypto_suites && |
| rtc::IsGcmCryptoSuiteName(crypto.cipher_suite)) || |
| rtc::CS_AES_CM_128_HMAC_SHA1_80 == crypto.cipher_suite || |
| (rtc::CS_AES_CM_128_HMAC_SHA1_32 == crypto.cipher_suite && audio && |
| !bundle && crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher)) { |
| return CreateCryptoParams(crypto.tag, crypto.cipher_suite, crypto_out); |
| } |
| } |
| return false; |
| } |
| |
| // Finds all StreamParams of all media types and attach them to stream_params. |
| static StreamParamsVec GetCurrentStreamParams( |
| const std::vector<const ContentInfo*>& active_local_contents) { |
| StreamParamsVec stream_params; |
| for (const ContentInfo* content : active_local_contents) { |
| for (const StreamParams& params : content->media_description()->streams()) { |
| stream_params.push_back(params); |
| } |
| } |
| return stream_params; |
| } |
| |
| static StreamParams CreateStreamParamsForNewSenderWithSsrcs( |
| const SenderOptions& sender, |
| const std::string& rtcp_cname, |
| bool include_rtx_streams, |
| bool include_flexfec_stream, |
| UniqueRandomIdGenerator* ssrc_generator) { |
| StreamParams result; |
| result.id = sender.track_id; |
| |
| // TODO(brandtr): Update when we support multistream protection. |
| if (include_flexfec_stream && sender.num_sim_layers > 1) { |
| include_flexfec_stream = false; |
| RTC_LOG(LS_WARNING) |
| << "Our FlexFEC implementation only supports protecting " |
| "a single media streams. This session has multiple " |
| "media streams however, so no FlexFEC SSRC will be generated."; |
| } |
| if (include_flexfec_stream && |
| !webrtc::field_trial::IsEnabled("WebRTC-FlexFEC-03")) { |
| include_flexfec_stream = false; |
| RTC_LOG(LS_WARNING) |
| << "WebRTC-FlexFEC trial is not enabled, not sending FlexFEC"; |
| } |
| |
| result.GenerateSsrcs(sender.num_sim_layers, include_rtx_streams, |
| include_flexfec_stream, ssrc_generator); |
| |
| result.cname = rtcp_cname; |
| result.set_stream_ids(sender.stream_ids); |
| |
| return result; |
| } |
| |
| static bool ValidateSimulcastLayers( |
| const std::vector<RidDescription>& rids, |
| const SimulcastLayerList& simulcast_layers) { |
| return absl::c_all_of( |
| simulcast_layers.GetAllLayers(), [&rids](const SimulcastLayer& layer) { |
| return absl::c_any_of(rids, [&layer](const RidDescription& rid) { |
| return rid.rid == layer.rid; |
| }); |
| }); |
| } |
| |
| static StreamParams CreateStreamParamsForNewSenderWithRids( |
| const SenderOptions& sender, |
| const std::string& rtcp_cname) { |
| RTC_DCHECK(!sender.rids.empty()); |
| RTC_DCHECK_EQ(sender.num_sim_layers, 0) |
| << "RIDs are the compliant way to indicate simulcast."; |
| RTC_DCHECK(ValidateSimulcastLayers(sender.rids, sender.simulcast_layers)); |
| StreamParams result; |
| result.id = sender.track_id; |
| result.cname = rtcp_cname; |
| result.set_stream_ids(sender.stream_ids); |
| |
| // More than one rid should be signaled. |
| if (sender.rids.size() > 1) { |
| result.set_rids(sender.rids); |
| } |
| |
| return result; |
| } |
| |
| // Adds SimulcastDescription if indicated by the media description options. |
| // MediaContentDescription should already be set up with the send rids. |
| static void AddSimulcastToMediaDescription( |
| const MediaDescriptionOptions& media_description_options, |
| MediaContentDescription* description) { |
| RTC_DCHECK(description); |
| |
| // Check if we are using RIDs in this scenario. |
| if (absl::c_all_of(description->streams(), [](const StreamParams& params) { |
| return !params.has_rids(); |
| })) { |
| return; |
| } |
| |
| RTC_DCHECK_EQ(1, description->streams().size()) |
| << "RIDs are only supported in Unified Plan semantics."; |
| RTC_DCHECK_EQ(1, media_description_options.sender_options.size()); |
| RTC_DCHECK(description->type() == MediaType::MEDIA_TYPE_AUDIO || |
| description->type() == MediaType::MEDIA_TYPE_VIDEO); |
| |
| // One RID or less indicates that simulcast is not needed. |
| if (description->streams()[0].rids().size() <= 1) { |
| return; |
| } |
| |
| // Only negotiate the send layers. |
| SimulcastDescription simulcast; |
| simulcast.send_layers() = |
| media_description_options.sender_options[0].simulcast_layers; |
| description->set_simulcast_description(simulcast); |
| } |
| |
| // Adds a StreamParams for each SenderOptions in |sender_options| to |
| // content_description. |
| // |current_params| - All currently known StreamParams of any media type. |
| template <class C> |
| static bool AddStreamParams( |
| const std::vector<SenderOptions>& sender_options, |
| const std::string& rtcp_cname, |
| UniqueRandomIdGenerator* ssrc_generator, |
| StreamParamsVec* current_streams, |
| MediaContentDescriptionImpl<C>* content_description) { |
| // SCTP streams are not negotiated using SDP/ContentDescriptions. |
| if (IsSctpProtocol(content_description->protocol())) { |
| return true; |
| } |
| |
| const bool include_rtx_streams = |
| ContainsRtxCodec(content_description->codecs()); |
| |
| const bool include_flexfec_stream = |
| ContainsFlexfecCodec(content_description->codecs()); |
| |
| for (const SenderOptions& sender : sender_options) { |
| // groupid is empty for StreamParams generated using |
| // MediaSessionDescriptionFactory. |
| StreamParams* param = |
| GetStreamByIds(*current_streams, "" /*group_id*/, sender.track_id); |
| if (!param) { |
| // This is a new sender. |
| StreamParams stream_param = |
| sender.rids.empty() |
| ? |
| // Signal SSRCs and legacy simulcast (if requested). |
| CreateStreamParamsForNewSenderWithSsrcs( |
| sender, rtcp_cname, include_rtx_streams, |
| include_flexfec_stream, ssrc_generator) |
| : |
| // Signal RIDs and spec-compliant simulcast (if requested). |
| CreateStreamParamsForNewSenderWithRids(sender, rtcp_cname); |
| |
| 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 { |
| // Use existing generated SSRCs/groups, but update the sync_label if |
| // necessary. This may be needed if a MediaStreamTrack was moved from one |
| // MediaStream to another. |
| param->set_stream_ids(sender.stream_ids); |
| 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 (TransportInfo& transport_info : sdesc->transport_infos()) { |
| if (bundle_group.HasContentName(transport_info.content_name) && |
| transport_info.content_name != selected_content_name) { |
| transport_info.description.ice_ufrag = selected_ufrag; |
| transport_info.description.ice_pwd = selected_pwd; |
| transport_info.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 (!content || !content->media_description()) { |
| return false; |
| } |
| *cryptos = content->media_description()->cryptos(); |
| 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(), |
| // Returns true if the |crypto|'s cipher_suite is not |
| // found in |filter|. |
| [&filter](const CryptoParams& crypto) { |
| for (const CryptoParams& entry : filter) { |
| if (entry.cipher_suite == crypto.cipher_suite) |
| return false; |
| } |
| return true; |
| }), |
| target_cryptos->end()); |
| } |
| |
| static bool IsRtpContent(SessionDescription* sdesc, |
| const std::string& content_name) { |
| bool is_rtp = false; |
| ContentInfo* content = sdesc->GetContentByName(content_name); |
| if (content && content->media_description()) { |
| is_rtp = IsRtpProtocol(content->media_description()->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; |
| bool first = true; |
| for (const std::string& content_name : content_names) { |
| if (!IsRtpContent(sdesc, content_name)) { |
| continue; |
| } |
| // The common cryptos are needed if any of the content does not have DTLS |
| // enabled. |
| if (!sdesc->GetTransportInfoByName(content_name)->description.secure()) { |
| common_cryptos_needed = true; |
| } |
| if (first) { |
| first = false; |
| // Initial the common_cryptos with the first content in the bundle group. |
| if (!GetCryptosByName(sdesc, content_name, &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, content_name, &cryptos)) { |
| return false; |
| } |
| PruneCryptos(cryptos, &common_cryptos); |
| } |
| } |
| |
| if (common_cryptos.empty() && common_cryptos_needed) { |
| return false; |
| } |
| |
| // Update to use the common cryptos. |
| for (const std::string& content_name : content_names) { |
| if (!IsRtpContent(sdesc, content_name)) { |
| continue; |
| } |
| ContentInfo* content = sdesc->GetContentByName(content_name); |
| if (IsMediaContent(content)) { |
| MediaContentDescription* media_desc = content->media_description(); |
| if (!media_desc) { |
| return false; |
| } |
| media_desc->set_cryptos(common_cryptos); |
| } |
| } |
| return true; |
| } |
| |
| static std::vector<const ContentInfo*> GetActiveContents( |
| const SessionDescription& description, |
| const MediaSessionOptions& session_options) { |
| std::vector<const ContentInfo*> active_contents; |
| for (size_t i = 0; i < description.contents().size(); ++i) { |
| RTC_DCHECK_LT(i, session_options.media_description_options.size()); |
| const ContentInfo& content = description.contents()[i]; |
| const MediaDescriptionOptions& media_options = |
| session_options.media_description_options[i]; |
| if (!content.rejected && !media_options.stopped && |
| content.name == media_options.mid) { |
| active_contents.push_back(&content); |
| } |
| } |
| return active_contents; |
| } |
| |
| template <class C> |
| static bool ContainsRtxCodec(const std::vector<C>& codecs) { |
| for (const auto& codec : codecs) { |
| if (IsRtxCodec(codec)) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| template <class C> |
| static bool IsRtxCodec(const C& codec) { |
| return absl::EqualsIgnoreCase(codec.name, kRtxCodecName); |
| } |
| |
| template <class C> |
| static bool ContainsFlexfecCodec(const std::vector<C>& codecs) { |
| for (const auto& codec : codecs) { |
| if (IsFlexfecCodec(codec)) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| template <class C> |
| static bool IsFlexfecCodec(const C& codec) { |
| return absl::EqualsIgnoreCase(codec.name, kFlexfecCodecName); |
| } |
| |
| // Create a media content to be offered for the given |sender_options|, |
| // according to the given options.rtcp_mux, session_options.is_muc, codecs, |
| // secure_transport, crypto, and current_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). The created content is added to the |
| // offer. |
| static bool CreateContentOffer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const SecurePolicy& secure_policy, |
| const CryptoParamsVec* current_cryptos, |
| const std::vector<std::string>& crypto_suites, |
| const RtpHeaderExtensions& rtp_extensions, |
| UniqueRandomIdGenerator* ssrc_generator, |
| StreamParamsVec* current_streams, |
| MediaContentDescription* offer) { |
| offer->set_rtcp_mux(session_options.rtcp_mux_enabled); |
| if (offer->type() == cricket::MEDIA_TYPE_VIDEO) { |
| offer->set_rtcp_reduced_size(true); |
| } |
| |
| // Build the vector of header extensions with directions for this |
| // media_description's options. |
| RtpHeaderExtensions extensions; |
| for (auto extension_with_id : rtp_extensions) { |
| for (const auto& extension : media_description_options.header_extensions) { |
| if (extension_with_id.uri == extension.uri) { |
| // TODO(crbug.com/1051821): Configure the extension direction from |
| // the information in the media_description_options extension |
| // capability. |
| extensions.push_back(extension_with_id); |
| } |
| } |
| } |
| offer->set_rtp_header_extensions(extensions); |
| |
| AddSimulcastToMediaDescription(media_description_options, offer); |
| |
| if (secure_policy != SEC_DISABLED) { |
| if (current_cryptos) { |
| AddMediaCryptos(*current_cryptos, offer); |
| } |
| if (offer->cryptos().empty()) { |
| if (!CreateMediaCryptos(crypto_suites, offer)) { |
| return false; |
| } |
| } |
| } |
| |
| if (secure_policy == SEC_REQUIRED && offer->cryptos().empty()) { |
| return false; |
| } |
| return true; |
| } |
| template <class C> |
| static bool CreateMediaContentOffer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_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, |
| UniqueRandomIdGenerator* ssrc_generator, |
| StreamParamsVec* current_streams, |
| MediaContentDescriptionImpl<C>* offer) { |
| offer->AddCodecs(codecs); |
| if (!AddStreamParams(media_description_options.sender_options, |
| session_options.rtcp_cname, ssrc_generator, |
| current_streams, offer)) { |
| return false; |
| } |
| |
| return CreateContentOffer(media_description_options, session_options, |
| secure_policy, current_cryptos, crypto_suites, |
| rtp_extensions, ssrc_generator, current_streams, |
| offer); |
| } |
| |
| template <class C> |
| static bool ReferencedCodecsMatch(const std::vector<C>& codecs1, |
| const int codec1_id, |
| const std::vector<C>& codecs2, |
| const int codec2_id) { |
| const C* codec1 = FindCodecById(codecs1, codec1_id); |
| const C* codec2 = FindCodecById(codecs2, codec2_id); |
| return codec1 != nullptr && codec2 != nullptr && codec1->Matches(*codec2); |
| } |
| |
| template <class C> |
| static void NegotiatePacketization(const C& local_codec, |
| const C& remote_codec, |
| C* negotiated_codec) {} |
| |
| template <> |
| void NegotiatePacketization(const VideoCodec& local_codec, |
| const VideoCodec& remote_codec, |
| VideoCodec* negotiated_codec) { |
| negotiated_codec->packetization = |
| VideoCodec::IntersectPacketization(local_codec, remote_codec); |
| } |
| |
| template <class C> |
| static void NegotiateCodecs(const std::vector<C>& local_codecs, |
| const std::vector<C>& offered_codecs, |
| std::vector<C>* negotiated_codecs, |
| bool keep_offer_order) { |
| 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; |
| NegotiatePacketization(ours, theirs, &negotiated); |
| negotiated.IntersectFeedbackParams(theirs); |
| if (IsRtxCodec(negotiated)) { |
| const auto apt_it = |
| theirs.params.find(kCodecParamAssociatedPayloadType); |
| // FindMatchingCodec shouldn't return something with no apt value. |
| RTC_DCHECK(apt_it != theirs.params.end()); |
| negotiated.SetParam(kCodecParamAssociatedPayloadType, apt_it->second); |
| |
| // We support parsing the declarative rtx-time parameter. |
| const auto rtx_time_it = theirs.params.find(kCodecParamRtxTime); |
| if (rtx_time_it != theirs.params.end()) { |
| negotiated.SetParam(kCodecParamRtxTime, rtx_time_it->second); |
| } |
| } |
| if (absl::EqualsIgnoreCase(ours.name, kH264CodecName)) { |
| webrtc::H264GenerateProfileLevelIdForAnswer(ours.params, theirs.params, |
| &negotiated.params); |
| } |
| negotiated.id = theirs.id; |
| negotiated.name = theirs.name; |
| negotiated_codecs->push_back(std::move(negotiated)); |
| } |
| } |
| if (keep_offer_order) { |
| // 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. |
| // This can be skipped when the transceiver has any codec preferences. |
| 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--; |
| } |
| absl::c_sort(*negotiated_codecs, [&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) { |
| // |codec_to_match| should be a member of |codecs1|, in order to look up RTX |
| // codecs' associated codecs correctly. If not, that's a programming error. |
| RTC_DCHECK(absl::c_any_of(codecs1, [&codec_to_match](const C& codec) { |
| return &codec == &codec_to_match; |
| })); |
| for (const C& potential_match : codecs2) { |
| if (potential_match.Matches(codec_to_match)) { |
| if (IsRtxCodec(codec_to_match)) { |
| int apt_value_1 = 0; |
| int apt_value_2 = 0; |
| if (!codec_to_match.GetParam(kCodecParamAssociatedPayloadType, |
| &apt_value_1) || |
| !potential_match.GetParam(kCodecParamAssociatedPayloadType, |
| &apt_value_2)) { |
| RTC_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; |
| } |
| |
| // Find the codec in |codec_list| that |rtx_codec| is associated with. |
| template <class C> |
| static const C* GetAssociatedCodec(const std::vector<C>& codec_list, |
| const C& rtx_codec) { |
| std::string associated_pt_str; |
| if (!rtx_codec.GetParam(kCodecParamAssociatedPayloadType, |
| &associated_pt_str)) { |
| RTC_LOG(LS_WARNING) << "RTX codec " << rtx_codec.name |
| << " is missing an associated payload type."; |
| return nullptr; |
| } |
| |
| int associated_pt; |
| if (!rtc::FromString(associated_pt_str, &associated_pt)) { |
| RTC_LOG(LS_WARNING) << "Couldn't convert payload type " << associated_pt_str |
| << " of RTX codec " << rtx_codec.name |
| << " to an integer."; |
| return nullptr; |
| } |
| |
| // Find the associated reference codec for the reference RTX codec. |
| const C* associated_codec = FindCodecById(codec_list, associated_pt); |
| if (!associated_codec) { |
| RTC_LOG(LS_WARNING) << "Couldn't find associated codec with payload type " |
| << associated_pt << " for RTX codec " << rtx_codec.name |
| << "."; |
| } |
| return associated_codec; |
| } |
| |
| // Adds all codecs from |reference_codecs| to |offered_codecs| that don't |
| // already exist in |offered_codecs| and ensure the payload types don't |
| // collide. |
| template <class C> |
| static void MergeCodecs(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; |
| const C* associated_codec = |
| GetAssociatedCodec(reference_codecs, rtx_codec); |
| if (!associated_codec) { |
| 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)) { |
| RTC_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); |
| } |
| } |
| } |
| |
| template <typename Codecs> |
| static Codecs MatchCodecPreference( |
| const std::vector<webrtc::RtpCodecCapability>& codec_preferences, |
| const Codecs& codecs) { |
| Codecs filtered_codecs; |
| std::set<std::string> kept_codecs_ids; |
| bool want_rtx = false; |
| |
| for (const auto& codec_preference : codec_preferences) { |
| auto found_codec = absl::c_find_if( |
| codecs, [&codec_preference](const typename Codecs::value_type& codec) { |
| webrtc::RtpCodecParameters codec_parameters = |
| codec.ToCodecParameters(); |
| return codec_parameters.name == codec_preference.name && |
| codec_parameters.kind == codec_preference.kind && |
| codec_parameters.num_channels == |
| codec_preference.num_channels && |
| codec_parameters.clock_rate == codec_preference.clock_rate && |
| codec_parameters.parameters == codec_preference.parameters; |
| }); |
| |
| if (found_codec != codecs.end()) { |
| filtered_codecs.push_back(*found_codec); |
| kept_codecs_ids.insert(std::to_string(found_codec->id)); |
| } else if (IsRtxCodec(codec_preference)) { |
| want_rtx = true; |
| } |
| } |
| |
| if (want_rtx) { |
| for (const auto& codec : codecs) { |
| if (IsRtxCodec(codec)) { |
| const auto apt = |
| codec.params.find(cricket::kCodecParamAssociatedPayloadType); |
| if (apt != codec.params.end() && |
| kept_codecs_ids.count(apt->second) > 0) { |
| filtered_codecs.push_back(codec); |
| } |
| } |
| } |
| } |
| |
| return filtered_codecs; |
| } |
| |
| // Adds all extensions from |reference_extensions| to |offered_extensions| that |
| // don't already exist in |offered_extensions| and ensure the IDs don't |
| // collide. If an extension is added, it's also added to |regular_extensions| or |
| // |encrypted_extensions|, and if the extension is in |regular_extensions| or |
| // |encrypted_extensions|, its ID is marked as used in |used_ids|. |
| // |offered_extensions| is for either audio or video while |regular_extensions| |
| // and |encrypted_extensions| are used for both audio and video. There could be |
| // overlap between audio extensions and video extensions. |
| static void MergeRtpHdrExts(const RtpHeaderExtensions& reference_extensions, |
| RtpHeaderExtensions* offered_extensions, |
| RtpHeaderExtensions* regular_extensions, |
| RtpHeaderExtensions* encrypted_extensions, |
| UsedRtpHeaderExtensionIds* used_ids) { |
| for (auto reference_extension : reference_extensions) { |
| if (!webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption( |
| *offered_extensions, reference_extension.uri, |
| reference_extension.encrypt)) { |
| if (reference_extension.encrypt) { |
| const webrtc::RtpExtension* existing = |
| webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption( |
| *encrypted_extensions, reference_extension.uri, |
| reference_extension.encrypt); |
| if (existing) { |
| offered_extensions->push_back(*existing); |
| } else { |
| used_ids->FindAndSetIdUsed(&reference_extension); |
| encrypted_extensions->push_back(reference_extension); |
| offered_extensions->push_back(reference_extension); |
| } |
| } else { |
| const webrtc::RtpExtension* existing = |
| webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption( |
| *regular_extensions, reference_extension.uri, |
| reference_extension.encrypt); |
| if (existing) { |
| offered_extensions->push_back(*existing); |
| } else { |
| used_ids->FindAndSetIdUsed(&reference_extension); |
| regular_extensions->push_back(reference_extension); |
| offered_extensions->push_back(reference_extension); |
| } |
| } |
| } |
| } |
| } |
| |
| static void AddEncryptedVersionsOfHdrExts( |
| RtpHeaderExtensions* offered_extensions, |
| RtpHeaderExtensions* encrypted_extensions, |
| UsedRtpHeaderExtensionIds* used_ids) { |
| RtpHeaderExtensions encrypted_extensions_to_add; |
| for (const auto& extension : *offered_extensions) { |
| // Skip existing encrypted offered extension |
| if (extension.encrypt) { |
| continue; |
| } |
| |
| // Skip if we cannot encrypt the extension |
| if (!webrtc::RtpExtension::IsEncryptionSupported(extension.uri)) { |
| continue; |
| } |
| |
| // Skip if an encrypted extension with that URI already exists in the |
| // offered extensions. |
| const bool have_encrypted_extension = |
| webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption( |
| *offered_extensions, extension.uri, true); |
| if (have_encrypted_extension) { |
| continue; |
| } |
| |
| // Determine if a shared encrypted extension with that URI already exists. |
| const webrtc::RtpExtension* shared_encrypted_extension = |
| webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption( |
| *encrypted_extensions, extension.uri, true); |
| if (shared_encrypted_extension) { |
| // Re-use the shared encrypted extension |
| encrypted_extensions_to_add.push_back(*shared_encrypted_extension); |
| continue; |
| } |
| |
| // None exists. Create a new shared encrypted extension from the |
| // non-encrypted one. |
| webrtc::RtpExtension new_encrypted_extension(extension); |
| new_encrypted_extension.encrypt = true; |
| used_ids->FindAndSetIdUsed(&new_encrypted_extension); |
| encrypted_extensions->push_back(new_encrypted_extension); |
| encrypted_extensions_to_add.push_back(new_encrypted_extension); |
| } |
| |
| // Append the additional encrypted extensions to be offered |
| offered_extensions->insert(offered_extensions->end(), |
| encrypted_extensions_to_add.begin(), |
| encrypted_extensions_to_add.end()); |
| } |
| |
| // Mostly identical to RtpExtension::FindHeaderExtensionByUri but discards any |
| // encrypted extensions that this implementation cannot encrypt. |
| static const webrtc::RtpExtension* FindHeaderExtensionByUriDiscardUnsupported( |
| const std::vector<webrtc::RtpExtension>& extensions, |
| absl::string_view uri, |
| webrtc::RtpExtension::Filter filter) { |
| // Note: While it's technically possible to decrypt extensions that we don't |
| // encrypt, the symmetric API of libsrtp does not allow us to supply |
| // different IDs for encryption/decryption of header extensions depending on |
| // whether the packet is inbound or outbound. Thereby, we are limited to |
| // what we can send in encrypted form. |
| if (!webrtc::RtpExtension::IsEncryptionSupported(uri)) { |
| // If there's no encryption support and we only want encrypted extensions, |
| // there's no point in continuing the search here. |
| if (filter == webrtc::RtpExtension::kRequireEncryptedExtension) { |
| return nullptr; |
| } |
| |
| // Instruct to only return non-encrypted extensions |
| filter = webrtc::RtpExtension::Filter::kDiscardEncryptedExtension; |
| } |
| |
| return webrtc::RtpExtension::FindHeaderExtensionByUri(extensions, uri, |
| filter); |
| } |
| |
| static void NegotiateRtpHeaderExtensions( |
| const RtpHeaderExtensions& local_extensions, |
| const RtpHeaderExtensions& offered_extensions, |
| webrtc::RtpExtension::Filter filter, |
| RtpHeaderExtensions* negotiated_extensions) { |
| // TransportSequenceNumberV2 is not offered by default. The special logic for |
| // the TransportSequenceNumber extensions works as follows: |
| // Offer Answer |
| // V1 V1 if in local_extensions. |
| // V1 and V2 V2 regardless of local_extensions. |
| // V2 V2 regardless of local_extensions. |
| const webrtc::RtpExtension* transport_sequence_number_v2_offer = |
| FindHeaderExtensionByUriDiscardUnsupported( |
| offered_extensions, |
| webrtc::RtpExtension::kTransportSequenceNumberV2Uri, filter); |
| |
| bool frame_descriptor_in_local = false; |
| bool dependency_descriptor_in_local = false; |
| bool abs_capture_time_in_local = false; |
| |
| for (const webrtc::RtpExtension& ours : local_extensions) { |
| if (ours.uri == webrtc::RtpExtension::kGenericFrameDescriptorUri00) |
| frame_descriptor_in_local = true; |
| else if (ours.uri == webrtc::RtpExtension::kDependencyDescriptorUri) |
| dependency_descriptor_in_local = true; |
| else if (ours.uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri) |
| abs_capture_time_in_local = true; |
| const webrtc::RtpExtension* theirs = |
| FindHeaderExtensionByUriDiscardUnsupported(offered_extensions, ours.uri, |
| filter); |
| if (theirs) { |
| if (transport_sequence_number_v2_offer && |
| ours.uri == webrtc::RtpExtension::kTransportSequenceNumberUri) { |
| // Don't respond to |
| // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 |
| // if we get an offer including |
| // http://www.webrtc.org/experiments/rtp-hdrext/transport-wide-cc-02 |
| continue; |
| } else { |
| // We respond with their RTP header extension id. |
| negotiated_extensions->push_back(*theirs); |
| } |
| } |
| } |
| |
| if (transport_sequence_number_v2_offer) { |
| // Respond that we support kTransportSequenceNumberV2Uri. |
| negotiated_extensions->push_back(*transport_sequence_number_v2_offer); |
| } |
| |
| // Frame descriptors support. If the extension is not present locally, but is |
| // in the offer, we add it to the list. |
| if (!dependency_descriptor_in_local) { |
| const webrtc::RtpExtension* theirs = |
| FindHeaderExtensionByUriDiscardUnsupported( |
| offered_extensions, webrtc::RtpExtension::kDependencyDescriptorUri, |
| filter); |
| if (theirs) { |
| negotiated_extensions->push_back(*theirs); |
| } |
| } |
| if (!frame_descriptor_in_local) { |
| const webrtc::RtpExtension* theirs = |
| FindHeaderExtensionByUriDiscardUnsupported( |
| offered_extensions, |
| webrtc::RtpExtension::kGenericFrameDescriptorUri00, filter); |
| if (theirs) { |
| negotiated_extensions->push_back(*theirs); |
| } |
| } |
| |
| // Absolute capture time support. If the extension is not present locally, but |
| // is in the offer, we add it to the list. |
| if (!abs_capture_time_in_local) { |
| const webrtc::RtpExtension* theirs = |
| FindHeaderExtensionByUriDiscardUnsupported( |
| offered_extensions, webrtc::RtpExtension::kAbsoluteCaptureTimeUri, |
| filter); |
| if (theirs) { |
| negotiated_extensions->push_back(*theirs); |
| } |
| } |
| } |
| |
| static void StripCNCodecs(AudioCodecs* audio_codecs) { |
| audio_codecs->erase(std::remove_if(audio_codecs->begin(), audio_codecs->end(), |
| [](const AudioCodec& codec) { |
| return absl::EqualsIgnoreCase( |
| codec.name, kComfortNoiseCodecName); |
| }), |
| audio_codecs->end()); |
| } |
| |
| template <class C> |
| static bool SetCodecsInAnswer( |
| const MediaContentDescriptionImpl<C>* offer, |
| const std::vector<C>& local_codecs, |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| UniqueRandomIdGenerator* ssrc_generator, |
| StreamParamsVec* current_streams, |
| MediaContentDescriptionImpl<C>* answer) { |
| std::vector<C> negotiated_codecs; |
| NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs, |
| media_description_options.codec_preferences.empty()); |
| answer->AddCodecs(negotiated_codecs); |
| answer->set_protocol(offer->protocol()); |
| if (!AddStreamParams(media_description_options.sender_options, |
| session_options.rtcp_cname, ssrc_generator, |
| current_streams, answer)) { |
| return false; // Something went seriously wrong. |
| } |
| return true; |
| } |
| |
| // Create a media content to be answered for the given |sender_options| |
| // according to the given session_options.rtcp_mux, session_options.streams, |
| // codecs, crypto, and current_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). The codecs, rtcp_mux, and crypto are all |
| // negotiated with the offer. If the negotiation fails, this method returns |
| // false. The created content is added to the offer. |
| static bool CreateMediaContentAnswer( |
| const MediaContentDescription* offer, |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const SecurePolicy& sdes_policy, |
| const CryptoParamsVec* current_cryptos, |
| const RtpHeaderExtensions& local_rtp_extensions, |
| UniqueRandomIdGenerator* ssrc_generator, |
| bool enable_encrypted_rtp_header_extensions, |
| StreamParamsVec* current_streams, |
| bool bundle_enabled, |
| MediaContentDescription* answer) { |
| answer->set_extmap_allow_mixed_enum(offer->extmap_allow_mixed_enum()); |
| const webrtc::RtpExtension::Filter extensions_filter = |
| enable_encrypted_rtp_header_extensions |
| ? webrtc::RtpExtension::Filter::kPreferEncryptedExtension |
| : webrtc::RtpExtension::Filter::kDiscardEncryptedExtension; |
| RtpHeaderExtensions negotiated_rtp_extensions; |
| NegotiateRtpHeaderExtensions(local_rtp_extensions, |
| offer->rtp_header_extensions(), |
| extensions_filter, &negotiated_rtp_extensions); |
| answer->set_rtp_header_extensions(negotiated_rtp_extensions); |
| |
| answer->set_rtcp_mux(session_options.rtcp_mux_enabled && offer->rtcp_mux()); |
| if (answer->type() == cricket::MEDIA_TYPE_VIDEO) { |
| answer->set_rtcp_reduced_size(offer->rtcp_reduced_size()); |
| } |
| |
| answer->set_remote_estimate(offer->remote_estimate()); |
| |
| if (sdes_policy != SEC_DISABLED) { |
| CryptoParams crypto; |
| if (SelectCrypto(offer, bundle_enabled, session_options.crypto_options, |
| &crypto)) { |
| if (current_cryptos) { |
| FindMatchingCrypto(*current_cryptos, crypto, &crypto); |
| } |
| answer->AddCrypto(crypto); |
| } |
| } |
| |
| if (answer->cryptos().empty() && sdes_policy == SEC_REQUIRED) { |
| return false; |
| } |
| |
| AddSimulcastToMediaDescription(media_description_options, answer); |
| |
| answer->set_direction(NegotiateRtpTransceiverDirection( |
| offer->direction(), media_description_options.direction)); |
| |
| return true; |
| } |
| |
| 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 ContentInfo* content, |
| const SessionDescription* current_description) { |
| if (!content) { |
| return false; |
| } |
| |
| size_t msection_index = content - ¤t_description->contents()[0]; |
| |
| if (current_description->transport_infos().size() <= msection_index) { |
| return false; |
| } |
| |
| return current_description->transport_infos()[msection_index] |
| .description.secure(); |
| } |
| |
| void MediaDescriptionOptions::AddAudioSender( |
| const std::string& track_id, |
| const std::vector<std::string>& stream_ids) { |
| RTC_DCHECK(type == MEDIA_TYPE_AUDIO); |
| AddSenderInternal(track_id, stream_ids, {}, SimulcastLayerList(), 1); |
| } |
| |
| void MediaDescriptionOptions::AddVideoSender( |
| const std::string& track_id, |
| const std::vector<std::string>& stream_ids, |
| const std::vector<RidDescription>& rids, |
| const SimulcastLayerList& simulcast_layers, |
| int num_sim_layers) { |
| RTC_DCHECK(type == MEDIA_TYPE_VIDEO); |
| RTC_DCHECK(rids.empty() || num_sim_layers == 0) |
| << "RIDs are the compliant way to indicate simulcast."; |
| RTC_DCHECK(ValidateSimulcastLayers(rids, simulcast_layers)); |
| AddSenderInternal(track_id, stream_ids, rids, simulcast_layers, |
| num_sim_layers); |
| } |
| |
| void MediaDescriptionOptions::AddSenderInternal( |
| const std::string& track_id, |
| const std::vector<std::string>& stream_ids, |
| const std::vector<RidDescription>& rids, |
| const SimulcastLayerList& simulcast_layers, |
| int num_sim_layers) { |
| // TODO(steveanton): Support any number of stream ids. |
| RTC_CHECK(stream_ids.size() == 1U); |
| SenderOptions options; |
| options.track_id = track_id; |
| options.stream_ids = stream_ids; |
| options.simulcast_layers = simulcast_layers; |
| options.rids = rids; |
| options.num_sim_layers = num_sim_layers; |
| sender_options.push_back(options); |
| } |
| |
| bool MediaSessionOptions::HasMediaDescription(MediaType type) const { |
| return absl::c_any_of( |
| media_description_options, |
| [type](const MediaDescriptionOptions& t) { return t.type == type; }); |
| } |
| |
| MediaSessionDescriptionFactory::MediaSessionDescriptionFactory( |
| const TransportDescriptionFactory* transport_desc_factory, |
| rtc::UniqueRandomIdGenerator* ssrc_generator) |
| : ssrc_generator_(ssrc_generator), |
| transport_desc_factory_(transport_desc_factory) { |
| RTC_DCHECK(ssrc_generator_); |
| } |
| |
| MediaSessionDescriptionFactory::MediaSessionDescriptionFactory( |
| ChannelManager* channel_manager, |
| const TransportDescriptionFactory* transport_desc_factory, |
| rtc::UniqueRandomIdGenerator* ssrc_generator) |
| : MediaSessionDescriptionFactory(transport_desc_factory, ssrc_generator) { |
| channel_manager->GetSupportedAudioSendCodecs(&audio_send_codecs_); |
| channel_manager->GetSupportedAudioReceiveCodecs(&audio_recv_codecs_); |
| channel_manager->GetSupportedVideoSendCodecs(&video_send_codecs_); |
| channel_manager->GetSupportedVideoReceiveCodecs(&video_recv_codecs_); |
| ComputeAudioCodecsIntersectionAndUnion(); |
| ComputeVideoCodecsIntersectionAndUnion(); |
| } |
| |
| 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; |
| ComputeAudioCodecsIntersectionAndUnion(); |
| } |
| |
| const VideoCodecs& MediaSessionDescriptionFactory::video_sendrecv_codecs() |
| const { |
| return video_sendrecv_codecs_; |
| } |
| |
| const VideoCodecs& MediaSessionDescriptionFactory::video_send_codecs() const { |
| return video_send_codecs_; |
| } |
| |
| const VideoCodecs& MediaSessionDescriptionFactory::video_recv_codecs() const { |
| return video_recv_codecs_; |
| } |
| |
| void MediaSessionDescriptionFactory::set_video_codecs( |
| const VideoCodecs& send_codecs, |
| const VideoCodecs& recv_codecs) { |
| video_send_codecs_ = send_codecs; |
| video_recv_codecs_ = recv_codecs; |
| ComputeVideoCodecsIntersectionAndUnion(); |
| } |
| |
| static void RemoveUnifiedPlanExtensions(RtpHeaderExtensions* extensions) { |
| RTC_DCHECK(extensions); |
| |
| extensions->erase( |
| std::remove_if(extensions->begin(), extensions->end(), |
| [](auto extension) { |
| return extension.uri == webrtc::RtpExtension::kMidUri || |
| extension.uri == webrtc::RtpExtension::kRidUri || |
| extension.uri == |
| webrtc::RtpExtension::kRepairedRidUri; |
| }), |
| extensions->end()); |
| } |
| |
| RtpHeaderExtensions |
| MediaSessionDescriptionFactory::filtered_rtp_header_extensions( |
| RtpHeaderExtensions extensions) const { |
| if (!is_unified_plan_) { |
| RemoveUnifiedPlanExtensions(&extensions); |
| } |
| return extensions; |
| } |
| |
| std::unique_ptr<SessionDescription> MediaSessionDescriptionFactory::CreateOffer( |
| const MediaSessionOptions& session_options, |
| const SessionDescription* current_description) const { |
| // Must have options for each existing section. |
| if (current_description) { |
| RTC_DCHECK_LE(current_description->contents().size(), |
| session_options.media_description_options.size()); |
| } |
| |
| IceCredentialsIterator ice_credentials( |
| session_options.pooled_ice_credentials); |
| |
| std::vector<const ContentInfo*> current_active_contents; |
| if (current_description) { |
| current_active_contents = |
| GetActiveContents(*current_description, session_options); |
| } |
| |
| StreamParamsVec current_streams = |
| GetCurrentStreamParams(current_active_contents); |
| |
| AudioCodecs offer_audio_codecs; |
| VideoCodecs offer_video_codecs; |
| GetCodecsForOffer(current_active_contents, &offer_audio_codecs, |
| &offer_video_codecs); |
| AudioVideoRtpHeaderExtensions extensions_with_ids = |
| GetOfferedRtpHeaderExtensionsWithIds( |
| current_active_contents, session_options.offer_extmap_allow_mixed, |
| session_options.media_description_options); |
| |
| auto offer = std::make_unique<SessionDescription>(); |
| |
| // Iterate through the media description options, matching with existing media |
| // descriptions in |current_description|. |
| size_t msection_index = 0; |
| for (const MediaDescriptionOptions& media_description_options : |
| session_options.media_description_options) { |
| const ContentInfo* current_content = nullptr; |
| if (current_description && |
| msection_index < current_description->contents().size()) { |
| current_content = ¤t_description->contents()[msection_index]; |
| // Media type must match unless this media section is being recycled. |
| RTC_DCHECK(current_content->name != media_description_options.mid || |
| IsMediaContentOfType(current_content, |
| media_description_options.type)); |
| } |
| switch (media_description_options.type) { |
| case MEDIA_TYPE_AUDIO: |
| if (!AddAudioContentForOffer(media_description_options, session_options, |
| current_content, current_description, |
| extensions_with_ids.audio, |
| offer_audio_codecs, ¤t_streams, |
| offer.get(), &ice_credentials)) { |
| return nullptr; |
| } |
| break; |
| case MEDIA_TYPE_VIDEO: |
| if (!AddVideoContentForOffer(media_description_options, session_options, |
| current_content, current_description, |
| extensions_with_ids.video, |
| offer_video_codecs, ¤t_streams, |
| offer.get(), &ice_credentials)) { |
| return nullptr; |
| } |
| break; |
| case MEDIA_TYPE_DATA: |
| if (!AddDataContentForOffer(media_description_options, session_options, |
| current_content, current_description, |
| ¤t_streams, offer.get(), |
| &ice_credentials)) { |
| return nullptr; |
| } |
| break; |
| case MEDIA_TYPE_UNSUPPORTED: |
| if (!AddUnsupportedContentForOffer( |
| media_description_options, session_options, current_content, |
| current_description, offer.get(), &ice_credentials)) { |
| return nullptr; |
| } |
| break; |
| default: |
| RTC_NOTREACHED(); |
| } |
| ++msection_index; |
| } |
| |
| // Bundle the contents together, if we've been asked to do so, and update any |
| // parameters that need to be tweaked for BUNDLE. |
| if (session_options.bundle_enabled) { |
| ContentGroup offer_bundle(GROUP_TYPE_BUNDLE); |
| for (const ContentInfo& content : offer->contents()) { |
| if (content.rejected) { |
| continue; |
| } |
| // TODO(deadbeef): There are conditions that make bundling two media |
| // descriptions together illegal. For example, they use the same payload |
| // type to represent different codecs, or same IDs for different header |
| // extensions. We need to detect this and not try to bundle those media |
| // descriptions together. |
| offer_bundle.AddContentName(content.name); |
| } |
| if (!offer_bundle.content_names().empty()) { |
| offer->AddGroup(offer_bundle); |
| if (!UpdateTransportInfoForBundle(offer_bundle, offer.get())) { |
| RTC_LOG(LS_ERROR) |
| << "CreateOffer failed to UpdateTransportInfoForBundle."; |
| return nullptr; |
| } |
| if (!UpdateCryptoParamsForBundle(offer_bundle, offer.get())) { |
| RTC_LOG(LS_ERROR) |
| << "CreateOffer failed to UpdateCryptoParamsForBundle."; |
| return nullptr; |
| } |
| } |
| } |
| |
| // The following determines how to signal MSIDs to ensure compatibility with |
| // older endpoints (in particular, older Plan B endpoints). |
| if (is_unified_plan_) { |
| // Be conservative and signal using both a=msid and a=ssrc lines. Unified |
| // Plan answerers will look at a=msid and Plan B answerers will look at the |
| // a=ssrc MSID line. |
| offer->set_msid_signaling(cricket::kMsidSignalingMediaSection | |
| cricket::kMsidSignalingSsrcAttribute); |
| } else { |
| // Plan B always signals MSID using a=ssrc lines. |
| offer->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute); |
| } |
| |
| offer->set_extmap_allow_mixed(session_options.offer_extmap_allow_mixed); |
| |
| return offer; |
| } |
| |
| std::unique_ptr<SessionDescription> |
| MediaSessionDescriptionFactory::CreateAnswer( |
| const SessionDescription* offer, |
| const MediaSessionOptions& session_options, |
| const SessionDescription* current_description) const { |
| if (!offer) { |
| return nullptr; |
| } |
| |
| // Must have options for exactly as many sections as in the offer. |
| RTC_DCHECK_EQ(offer->contents().size(), |
| session_options.media_description_options.size()); |
| |
| IceCredentialsIterator ice_credentials( |
| session_options.pooled_ice_credentials); |
| |
| std::vector<const ContentInfo*> current_active_contents; |
| if (current_description) { |
| current_active_contents = |
| GetActiveContents(*current_description, session_options); |
| } |
| |
| StreamParamsVec current_streams = |
| GetCurrentStreamParams(current_active_contents); |
| |
| // Get list of all possible codecs that respects existing payload type |
| // mappings and uses a single payload type space. |
| // |
| // Note that these lists may be further filtered for each m= section; this |
| // step is done just to establish the payload type mappings shared by all |
| // sections. |
| AudioCodecs answer_audio_codecs; |
| VideoCodecs answer_video_codecs; |
| GetCodecsForAnswer(current_active_contents, *offer, &answer_audio_codecs, |
| &answer_video_codecs); |
| |
| auto answer = std::make_unique<SessionDescription>(); |
| |
| // If the offer supports BUNDLE, and we want to use it too, create a BUNDLE |
| // group in the answer with the appropriate content names. |
| std::vector<const ContentGroup*> offer_bundles = |
| offer->GetGroupsByName(GROUP_TYPE_BUNDLE); |
| // There are as many answer BUNDLE groups as offer BUNDLE groups (even if |
| // rejected, we respond with an empty group). |offer_bundles|, |
| // |answer_bundles| and |bundle_transports| share the same size and indices. |
| std::vector<ContentGroup> answer_bundles; |
| std::vector<std::unique_ptr<TransportInfo>> bundle_transports; |
| answer_bundles.reserve(offer_bundles.size()); |
| bundle_transports.reserve(offer_bundles.size()); |
| for (size_t i = 0; i < offer_bundles.size(); ++i) { |
| answer_bundles.emplace_back(GROUP_TYPE_BUNDLE); |
| bundle_transports.emplace_back(nullptr); |
| } |
| |
| answer->set_extmap_allow_mixed(offer->extmap_allow_mixed()); |
| |
| // Iterate through the media description options, matching with existing |
| // media descriptions in |current_description|. |
| size_t msection_index = 0; |
| for (const MediaDescriptionOptions& media_description_options : |
| session_options.media_description_options) { |
| const ContentInfo* offer_content = &offer->contents()[msection_index]; |
| // Media types and MIDs must match between the remote offer and the |
| // MediaDescriptionOptions. |
| RTC_DCHECK( |
| IsMediaContentOfType(offer_content, media_description_options.type)); |
| RTC_DCHECK(media_description_options.mid == offer_content->name); |
| // Get the index of the BUNDLE group that this MID belongs to, if any. |
| absl::optional<size_t> bundle_index; |
| for (size_t i = 0; i < offer_bundles.size(); ++i) { |
| if (offer_bundles[i]->HasContentName(media_description_options.mid)) { |
| bundle_index = i; |
| break; |
| } |
| } |
| TransportInfo* bundle_transport = |
| bundle_index.has_value() ? bundle_transports[bundle_index.value()].get() |
| : nullptr; |
| |
| const ContentInfo* current_content = nullptr; |
| if (current_description && |
| msection_index < current_description->contents().size()) { |
| current_content = ¤t_description->contents()[msection_index]; |
| } |
| RtpHeaderExtensions header_extensions = RtpHeaderExtensionsFromCapabilities( |
| UnstoppedRtpHeaderExtensionCapabilities( |
| media_description_options.header_extensions)); |
| switch (media_description_options.type) { |
| case MEDIA_TYPE_AUDIO: |
| if (!AddAudioContentForAnswer( |
| media_description_options, session_options, offer_content, |
| offer, current_content, current_description, bundle_transport, |
| answer_audio_codecs, header_extensions, ¤t_streams, |
| answer.get(), &ice_credentials)) { |
| return nullptr; |
| } |
| break; |
| case MEDIA_TYPE_VIDEO: |
| if (!AddVideoContentForAnswer( |
| media_description_options, session_options, offer_content, |
| offer, current_content, current_description, bundle_transport, |
| answer_video_codecs, header_extensions, ¤t_streams, |
| answer.get(), &ice_credentials)) { |
| return nullptr; |
| } |
| break; |
| case MEDIA_TYPE_DATA: |
| if (!AddDataContentForAnswer( |
| media_description_options, session_options, offer_content, |
| offer, current_content, current_description, bundle_transport, |
| ¤t_streams, answer.get(), &ice_credentials)) { |
| return nullptr; |
| } |
| break; |
| case MEDIA_TYPE_UNSUPPORTED: |
| if (!AddUnsupportedContentForAnswer( |
| media_description_options, session_options, offer_content, |
| offer, current_content, current_description, bundle_transport, |
| answer.get(), &ice_credentials)) { |
| return nullptr; |
| } |
| break; |
| default: |
| RTC_NOTREACHED(); |
| } |
| ++msection_index; |
| // See if we can add the newly generated m= section to the BUNDLE group in |
| // the answer. |
| ContentInfo& added = answer->contents().back(); |
| if (!added.rejected && session_options.bundle_enabled && |
| bundle_index.has_value()) { |
| // The |bundle_index| is for |media_description_options.mid|. |
| RTC_DCHECK_EQ(media_description_options.mid, added.name); |
| answer_bundles[bundle_index.value()].AddContentName(added.name); |
| bundle_transports[bundle_index.value()].reset( |
| new TransportInfo(*answer->GetTransportInfoByName(added.name))); |
| } |
| } |
| |
| // If BUNDLE group(s) were offered, put the same number of BUNDLE groups in |
| // the answer even if they're empty. RFC5888 says: |
| // |
| // A SIP entity that receives an offer that contains an "a=group" line |
| // with semantics that are understood MUST return an answer that |
| // contains an "a=group" line with the same semantics. |
| if (!offer_bundles.empty()) { |
| for (const ContentGroup& answer_bundle : answer_bundles) { |
| answer->AddGroup(answer_bundle); |
| |
| if (answer_bundle.FirstContentName()) { |
| // Share the same ICE credentials and crypto params across all contents, |
| // as BUNDLE requires. |
| if (!UpdateTransportInfoForBundle(answer_bundle, answer.get())) { |
| RTC_LOG(LS_ERROR) |
| << "CreateAnswer failed to UpdateTransportInfoForBundle."; |
| return NULL; |
| } |
| |
| if (!UpdateCryptoParamsForBundle(answer_bundle, answer.get())) { |
| RTC_LOG(LS_ERROR) |
| << "CreateAnswer failed to UpdateCryptoParamsForBundle."; |
| return NULL; |
| } |
| } |
| } |
| } |
| |
| // The following determines how to signal MSIDs to ensure compatibility with |
| // older endpoints (in particular, older Plan B endpoints). |
| if (is_unified_plan_) { |
| // Unified Plan needs to look at what the offer included to find the most |
| // compatible answer. |
| if (offer->msid_signaling() == 0) { |
| // We end up here in one of three cases: |
| // 1. An empty offer. We'll reply with an empty answer so it doesn't |
| // matter what we pick here. |
| // 2. A data channel only offer. We won't add any MSIDs to the answer so |
| // it also doesn't matter what we pick here. |
| // 3. Media that's either sendonly or inactive from the remote endpoint. |
| // We don't have any information to say whether the endpoint is Plan B |
| // or Unified Plan, so be conservative and send both. |
| answer->set_msid_signaling(cricket::kMsidSignalingMediaSection | |
| cricket::kMsidSignalingSsrcAttribute); |
| } else if (offer->msid_signaling() == |
| (cricket::kMsidSignalingMediaSection | |
| cricket::kMsidSignalingSsrcAttribute)) { |
| // If both a=msid and a=ssrc MSID signaling methods were used, we're |
| // probably talking to a Unified Plan endpoint so respond with just |
| // a=msid. |
| answer->set_msid_signaling(cricket::kMsidSignalingMediaSection); |
| } else { |
| // Otherwise, it's clear which method the offerer is using so repeat that |
| // back to them. |
| answer->set_msid_signaling(offer->msid_signaling()); |
| } |
| } else { |
| // Plan B always signals MSID using a=ssrc lines. |
| answer->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute); |
| } |
| |
| return answer; |
| } |
| |
| const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForOffer( |
| const RtpTransceiverDirection& direction) const { |
| switch (direction) { |
| // If stream is inactive - generate list as if sendrecv. |
| case RtpTransceiverDirection::kSendRecv: |
| case RtpTransceiverDirection::kStopped: |
| case RtpTransceiverDirection::kInactive: |
| return audio_sendrecv_codecs_; |
| case RtpTransceiverDirection::kSendOnly: |
| return audio_send_codecs_; |
| case RtpTransceiverDirection::kRecvOnly: |
| return audio_recv_codecs_; |
| } |
| RTC_CHECK_NOTREACHED(); |
| } |
| |
| const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForAnswer( |
| const RtpTransceiverDirection& offer, |
| const RtpTransceiverDirection& answer) const { |
| switch (answer) { |
| // For inactive and sendrecv answers, generate lists as if we were to accept |
| // the offer's direction. See RFC 3264 Section 6.1. |
| case RtpTransceiverDirection::kSendRecv: |
| case RtpTransceiverDirection::kStopped: |
| case RtpTransceiverDirection::kInactive: |
| return GetAudioCodecsForOffer( |
| webrtc::RtpTransceiverDirectionReversed(offer)); |
| case RtpTransceiverDirection::kSendOnly: |
| return audio_send_codecs_; |
| case RtpTransceiverDirection::kRecvOnly: |
| return audio_recv_codecs_; |
| } |
| RTC_CHECK_NOTREACHED(); |
| } |
| |
| const VideoCodecs& MediaSessionDescriptionFactory::GetVideoCodecsForOffer( |
| const RtpTransceiverDirection& direction) const { |
| switch (direction) { |
| // If stream is inactive - generate list as if sendrecv. |
| case RtpTransceiverDirection::kSendRecv: |
| case RtpTransceiverDirection::kStopped: |
| case RtpTransceiverDirection::kInactive: |
| return video_sendrecv_codecs_; |
| case RtpTransceiverDirection::kSendOnly: |
| return video_send_codecs_; |
| case RtpTransceiverDirection::kRecvOnly: |
| return video_recv_codecs_; |
| } |
| RTC_CHECK_NOTREACHED(); |
| } |
| |
| const VideoCodecs& MediaSessionDescriptionFactory::GetVideoCodecsForAnswer( |
| const RtpTransceiverDirection& offer, |
| const RtpTransceiverDirection& answer) const { |
| switch (answer) { |
| // For inactive and sendrecv answers, generate lists as if we were to accept |
| // the offer's direction. See RFC 3264 Section 6.1. |
| case RtpTransceiverDirection::kSendRecv: |
| case RtpTransceiverDirection::kStopped: |
| case RtpTransceiverDirection::kInactive: |
| return GetVideoCodecsForOffer( |
| webrtc::RtpTransceiverDirectionReversed(offer)); |
| case RtpTransceiverDirection::kSendOnly: |
| return video_send_codecs_; |
| case RtpTransceiverDirection::kRecvOnly: |
| return video_recv_codecs_; |
| } |
| RTC_CHECK_NOTREACHED(); |
| } |
| |
| void MergeCodecsFromDescription( |
| const std::vector<const ContentInfo*>& current_active_contents, |
| AudioCodecs* audio_codecs, |
| VideoCodecs* video_codecs, |
| UsedPayloadTypes* used_pltypes) { |
| for (const ContentInfo* content : current_active_contents) { |
| if (IsMediaContentOfType(content, MEDIA_TYPE_AUDIO)) { |
| const AudioContentDescription* audio = |
| content->media_description()->as_audio(); |
| MergeCodecs<AudioCodec>(audio->codecs(), audio_codecs, used_pltypes); |
| } else if (IsMediaContentOfType(content, MEDIA_TYPE_VIDEO)) { |
| const VideoContentDescription* video = |
| content->media_description()->as_video(); |
| MergeCodecs<VideoCodec>(video->codecs(), video_codecs, used_pltypes); |
| } |
| } |
| } |
| |
| // Getting codecs for an offer involves these steps: |
| // |
| // 1. Construct payload type -> codec mappings for current description. |
| // 2. Add any reference codecs that weren't already present |
| // 3. For each individual media description (m= section), filter codecs based |
| // on the directional attribute (happens in another method). |
| void MediaSessionDescriptionFactory::GetCodecsForOffer( |
| const std::vector<const ContentInfo*>& current_active_contents, |
| AudioCodecs* audio_codecs, |
| VideoCodecs* video_codecs) const { |
| // First - get all codecs from the current description if the media type |
| // is used. Add them to |used_pltypes| so the payload type is not reused if a |
| // new media type is added. |
| UsedPayloadTypes used_pltypes; |
| MergeCodecsFromDescription(current_active_contents, audio_codecs, |
| video_codecs, &used_pltypes); |
| |
| // Add our codecs that are not in the current description. |
| MergeCodecs<AudioCodec>(all_audio_codecs_, audio_codecs, &used_pltypes); |
| MergeCodecs<VideoCodec>(all_video_codecs_, video_codecs, &used_pltypes); |
| } |
| |
| // Getting codecs for an answer involves these steps: |
| // |
| // 1. Construct payload type -> codec mappings for current description. |
| // 2. Add any codecs from the offer that weren't already present. |
| // 3. Add any remaining codecs that weren't already present. |
| // 4. For each individual media description (m= section), filter codecs based |
| // on the directional attribute (happens in another method). |
| void MediaSessionDescriptionFactory::GetCodecsForAnswer( |
| const std::vector<const ContentInfo*>& current_active_contents, |
| const SessionDescription& remote_offer, |
| AudioCodecs* audio_codecs, |
| VideoCodecs* video_codecs) const { |
| // First - get all codecs from the current description if the media type |
| // is used. Add them to |used_pltypes| so the payload type is not reused if a |
| // new media type is added. |
| UsedPayloadTypes used_pltypes; |
| MergeCodecsFromDescription(current_active_contents, audio_codecs, |
| video_codecs, &used_pltypes); |
| |
| // Second - filter out codecs that we don't support at all and should ignore. |
| AudioCodecs filtered_offered_audio_codecs; |
| VideoCodecs filtered_offered_video_codecs; |
| for (const ContentInfo& content : remote_offer.contents()) { |
| if (IsMediaContentOfType(&content, MEDIA_TYPE_AUDIO)) { |
| const AudioContentDescription* audio = |
| content.media_description()->as_audio(); |
| for (const AudioCodec& offered_audio_codec : audio->codecs()) { |
| if (!FindMatchingCodec<AudioCodec>(audio->codecs(), |
| filtered_offered_audio_codecs, |
| offered_audio_codec, nullptr) && |
| FindMatchingCodec<AudioCodec>(audio->codecs(), all_audio_codecs_, |
| offered_audio_codec, nullptr)) { |
| filtered_offered_audio_codecs.push_back(offered_audio_codec); |
| } |
| } |
| } else if (IsMediaContentOfType(&content, MEDIA_TYPE_VIDEO)) { |
| const VideoContentDescription* video = |
| content.media_description()->as_video(); |
| for (const VideoCodec& offered_video_codec : video->codecs()) { |
| if (!FindMatchingCodec<VideoCodec>(video->codecs(), |
| filtered_offered_video_codecs, |
| offered_video_codec, nullptr) && |
| FindMatchingCodec<VideoCodec>(video->codecs(), all_video_codecs_, |
| offered_video_codec, nullptr)) { |
| filtered_offered_video_codecs.push_back(offered_video_codec); |
| } |
| } |
| } |
| } |
| |
| // Add codecs that are not in the current description but were in |
| // |remote_offer|. |
| MergeCodecs<AudioCodec>(filtered_offered_audio_codecs, audio_codecs, |
| &used_pltypes); |
| MergeCodecs<VideoCodec>(filtered_offered_video_codecs, video_codecs, |
| &used_pltypes); |
| } |
| |
| MediaSessionDescriptionFactory::AudioVideoRtpHeaderExtensions |
| MediaSessionDescriptionFactory::GetOfferedRtpHeaderExtensionsWithIds( |
| const std::vector<const ContentInfo*>& current_active_contents, |
| bool extmap_allow_mixed, |
| const std::vector<MediaDescriptionOptions>& media_description_options) |
| const { |
| // All header extensions allocated from the same range to avoid potential |
| // issues when using BUNDLE. |
| |
| // Strictly speaking the SDP attribute extmap_allow_mixed signals that the |
| // receiver supports an RTP stream where one- and two-byte RTP header |
| // extensions are mixed. For backwards compatibility reasons it's used in |
| // WebRTC to signal that two-byte RTP header extensions are supported. |
| UsedRtpHeaderExtensionIds used_ids( |
| extmap_allow_mixed ? UsedRtpHeaderExtensionIds::IdDomain::kTwoByteAllowed |
| : UsedRtpHeaderExtensionIds::IdDomain::kOneByteOnly); |
| RtpHeaderExtensions all_regular_extensions; |
| RtpHeaderExtensions all_encrypted_extensions; |
| |
| AudioVideoRtpHeaderExtensions offered_extensions; |
| // 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. |
| for (const ContentInfo* content : current_active_contents) { |
| if (IsMediaContentOfType(content, MEDIA_TYPE_AUDIO)) { |
| const AudioContentDescription* audio = |
| content->media_description()->as_audio(); |
| MergeRtpHdrExts(audio->rtp_header_extensions(), &offered_extensions.audio, |
| &all_regular_extensions, &all_encrypted_extensions, |
| &used_ids); |
| } else if (IsMediaContentOfType(content, MEDIA_TYPE_VIDEO)) { |
| const VideoContentDescription* video = |
| content->media_description()->as_video(); |
| MergeRtpHdrExts(video->rtp_header_extensions(), &offered_extensions.video, |
| &all_regular_extensions, &all_encrypted_extensions, |
| &used_ids); |
| } |
| } |
| |
| // Add all encountered header extensions in the media description options that |
| // are not in the current description. |
| |
| for (const auto& entry : media_description_options) { |
| RtpHeaderExtensions filtered_extensions = |
| filtered_rtp_header_extensions(UnstoppedOrPresentRtpHeaderExtensions( |
| entry.header_extensions, all_regular_extensions, |
| all_encrypted_extensions)); |
| if (entry.type == MEDIA_TYPE_AUDIO) |
| MergeRtpHdrExts(filtered_extensions, &offered_extensions.audio, |
| &all_regular_extensions, &all_encrypted_extensions, |
| &used_ids); |
| else if (entry.type == MEDIA_TYPE_VIDEO) |
| MergeRtpHdrExts(filtered_extensions, &offered_extensions.video, |
| &all_regular_extensions, &all_encrypted_extensions, |
| &used_ids); |
| } |
| // TODO(jbauch): Support adding encrypted header extensions to existing |
| // sessions. |
| if (enable_encrypted_rtp_header_extensions_ && |
| current_active_contents.empty()) { |
| AddEncryptedVersionsOfHdrExts(&offered_extensions.audio, |
| &all_encrypted_extensions, &used_ids); |
| AddEncryptedVersionsOfHdrExts(&offered_extensions.video, |
| &all_encrypted_extensions, &used_ids); |
| } |
| return offered_extensions; |
| } |
| |
| bool MediaSessionDescriptionFactory::AddTransportOffer( |
| const std::string& content_name, |
| const TransportOptions& transport_options, |
| const SessionDescription* current_desc, |
| SessionDescription* offer_desc, |
| IceCredentialsIterator* ice_credentials) 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, |
| ice_credentials)); |
| if (!new_tdesc) { |
| RTC_LOG(LS_ERROR) << "Failed to AddTransportOffer, content name=" |
| << content_name; |
| } |
| offer_desc->AddTransportInfo(TransportInfo(content_name, *new_tdesc)); |
| return true; |
| } |
| |
| std::unique_ptr<TransportDescription> |
| MediaSessionDescriptionFactory::CreateTransportAnswer( |
| const std::string& content_name, |
| const SessionDescription* offer_desc, |
| const TransportOptions& transport_options, |
| const SessionDescription* current_desc, |
| bool require_transport_attributes, |
| IceCredentialsIterator* ice_credentials) 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, |
| require_transport_attributes, |
| current_tdesc, ice_credentials); |
| } |
| |
| bool MediaSessionDescriptionFactory::AddTransportAnswer( |
| const std::string& content_name, |
| const TransportDescription& transport_desc, |
| SessionDescription* answer_desc) const { |
| answer_desc->AddTransportInfo(TransportInfo(content_name, transport_desc)); |
| return true; |
| } |
| |
| // |audio_codecs| = set of all possible codecs that can be used, with correct |
| // payload type mappings |
| // |
| // |supported_audio_codecs| = set of codecs that are supported for the direction |
| // of this m= section |
| // |
| // acd->codecs() = set of previously negotiated codecs for this m= section |
| // |
| // The payload types should come from audio_codecs, but the order should come |
| // from acd->codecs() and then supported_codecs, to ensure that re-offers don't |
| // change existing codec priority, and that new codecs are added with the right |
| // priority. |
| bool MediaSessionDescriptionFactory::AddAudioContentForOffer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const RtpHeaderExtensions& audio_rtp_extensions, |
| const AudioCodecs& audio_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* desc, |
| IceCredentialsIterator* ice_credentials) const { |
| // Filter audio_codecs (which includes all codecs, with correctly remapped |
| // payload types) based on transceiver direction. |
| const AudioCodecs& supported_audio_codecs = |
| GetAudioCodecsForOffer(media_description_options.direction); |
| |
| AudioCodecs filtered_codecs; |
| |
| if (!media_description_options.codec_preferences.empty()) { |
| // Add the codecs from the current transceiver's codec preferences. |
| // They override any existing codecs from previous negotiations. |
| filtered_codecs = MatchCodecPreference( |
| media_description_options.codec_preferences, supported_audio_codecs); |
| } else { |
| // Add the codecs from current content if it exists and is not rejected nor |
| // recycled. |
| if (current_content && !current_content->rejected && |
| current_content->name == media_description_options.mid) { |
| RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_AUDIO)); |
| const AudioContentDescription* acd = |
| current_content->media_description()->as_audio(); |
| for (const AudioCodec& codec : acd->codecs()) { |
| if (FindMatchingCodec<AudioCodec>(acd->codecs(), audio_codecs, codec, |
| nullptr)) { |
| filtered_codecs.push_back(codec); |
| } |
| } |
| } |
| // Add other supported audio codecs. |
| AudioCodec found_codec; |
| for (const AudioCodec& codec : supported_audio_codecs) { |
| if (FindMatchingCodec<AudioCodec>(supported_audio_codecs, audio_codecs, |
| codec, &found_codec) && |
| !FindMatchingCodec<AudioCodec>(supported_audio_codecs, |
| filtered_codecs, codec, nullptr)) { |
| // Use the |found_codec| from |audio_codecs| because it has the |
| // correctly mapped payload type. |
| filtered_codecs.push_back(found_codec); |
| } |
| } |
| } |
| if (!session_options.vad_enabled) { |
| // If application doesn't want CN codecs in offer. |
| StripCNCodecs(&filtered_codecs); |
| } |
| |
| cricket::SecurePolicy sdes_policy = |
| IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED |
| : secure(); |
| |
| auto audio = std::make_unique<AudioContentDescription>(); |
| std::vector<std::string> crypto_suites; |
| GetSupportedAudioSdesCryptoSuiteNames(session_options.crypto_options, |
| &crypto_suites); |
| if (!CreateMediaContentOffer(media_description_options, session_options, |
| filtered_codecs, sdes_policy, |
| GetCryptos(current_content), crypto_suites, |
| audio_rtp_extensions, ssrc_generator_, |
| current_streams, audio.get())) { |
| return false; |
| } |
| |
| bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED); |
| SetMediaProtocol(secure_transport, audio.get()); |
| |
| audio->set_direction(media_description_options.direction); |
| |
| desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp, |
| media_description_options.stopped, std::move(audio)); |
| if (!AddTransportOffer(media_description_options.mid, |
| media_description_options.transport_options, |
| current_description, desc, ice_credentials)) { |
| return false; |
| } |
| |
| return true; |
| } |
| |
| // TODO(kron): This function is very similar to AddAudioContentForOffer. |
| // Refactor to reuse shared code. |
| bool MediaSessionDescriptionFactory::AddVideoContentForOffer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const RtpHeaderExtensions& video_rtp_extensions, |
| const VideoCodecs& video_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* desc, |
| IceCredentialsIterator* ice_credentials) const { |
| // Filter video_codecs (which includes all codecs, with correctly remapped |
| // payload types) based on transceiver direction. |
| const VideoCodecs& supported_video_codecs = |
| GetVideoCodecsForOffer(media_description_options.direction); |
| |
| VideoCodecs filtered_codecs; |
| |
| if (!media_description_options.codec_preferences.empty()) { |
| // Add the codecs from the current transceiver's codec preferences. |
| // They override any existing codecs from previous negotiations. |
| filtered_codecs = MatchCodecPreference( |
| media_description_options.codec_preferences, supported_video_codecs); |
| } else { |
| // Add the codecs from current content if it exists and is not rejected nor |
| // recycled. |
| if (current_content && !current_content->rejected && |
| current_content->name == media_description_options.mid) { |
| RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_VIDEO)); |
| const VideoContentDescription* vcd = |
| current_content->media_description()->as_video(); |
| for (const VideoCodec& codec : vcd->codecs()) { |
| if (FindMatchingCodec<VideoCodec>(vcd->codecs(), video_codecs, codec, |
| nullptr)) { |
| filtered_codecs.push_back(codec); |
| } |
| } |
| } |
| // Add other supported video codecs. |
| VideoCodec found_codec; |
| for (const VideoCodec& codec : supported_video_codecs) { |
| if (FindMatchingCodec<VideoCodec>(supported_video_codecs, video_codecs, |
| codec, &found_codec) && |
| !FindMatchingCodec<VideoCodec>(supported_video_codecs, |
| filtered_codecs, codec, nullptr)) { |
| // Use the |found_codec| from |video_codecs| because it has the |
| // correctly mapped payload type. |
| filtered_codecs.push_back(found_codec); |
| } |
| } |
| } |
| |
| if (session_options.raw_packetization_for_video) { |
| for (VideoCodec& codec : filtered_codecs) { |
| if (codec.GetCodecType() == VideoCodec::CODEC_VIDEO) { |
| codec.packetization = kPacketizationParamRaw; |
| } |
| } |
| } |
| |
| cricket::SecurePolicy sdes_policy = |
| IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED |
| : secure(); |
| auto video = std::make_unique<VideoContentDescription>(); |
| std::vector<std::string> crypto_suites; |
| GetSupportedVideoSdesCryptoSuiteNames(session_options.crypto_options, |
| &crypto_suites); |
| if (!CreateMediaContentOffer(media_description_options, session_options, |
| filtered_codecs, sdes_policy, |
| GetCryptos(current_content), crypto_suites, |
| video_rtp_extensions, ssrc_generator_, |
| current_streams, video.get())) { |
| return false; |
| } |
| |
| video->set_bandwidth(kAutoBandwidth); |
| |
| bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED); |
| SetMediaProtocol(secure_transport, video.get()); |
| |
| video->set_direction(media_description_options.direction); |
| |
| desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp, |
| media_description_options.stopped, std::move(video)); |
| if (!AddTransportOffer(media_description_options.mid, |
| media_description_options.transport_options, |
| current_description, desc, ice_credentials)) { |
| return false; |
| } |
| |
| return true; |
| } |
| |
| bool MediaSessionDescriptionFactory::AddDataContentForOffer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| StreamParamsVec* current_streams, |
| SessionDescription* desc, |
| IceCredentialsIterator* ice_credentials) const { |
| auto data = std::make_unique<SctpDataContentDescription>(); |
| |
| bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED); |
| |
| cricket::SecurePolicy sdes_policy = |
| IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED |
| : secure(); |
| std::vector<std::string> crypto_suites; |
| // 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 ? kMediaProtocolUdpDtlsSctp |
| : kMediaProtocolSctp); |
| data->set_use_sctpmap(session_options.use_obsolete_sctp_sdp); |
| data->set_max_message_size(kSctpSendBufferSize); |
| |
| if (!CreateContentOffer(media_description_options, session_options, |
| sdes_policy, GetCryptos(current_content), |
| crypto_suites, RtpHeaderExtensions(), ssrc_generator_, |
| current_streams, data.get())) { |
| return false; |
| } |
| |
| desc->AddContent(media_description_options.mid, MediaProtocolType::kSctp, |
| media_description_options.stopped, std::move(data)); |
| if (!AddTransportOffer(media_description_options.mid, |
| media_description_options.transport_options, |
| current_description, desc, ice_credentials)) { |
| return false; |
| } |
| return true; |
| } |
| |
| bool MediaSessionDescriptionFactory::AddUnsupportedContentForOffer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| SessionDescription* desc, |
| IceCredentialsIterator* ice_credentials) const { |
| RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_UNSUPPORTED)); |
| |
| const UnsupportedContentDescription* current_unsupported_description = |
| current_content->media_description()->as_unsupported(); |
| auto unsupported = std::make_unique<UnsupportedContentDescription>( |
| current_unsupported_description->media_type()); |
| unsupported->set_protocol(current_content->media_description()->protocol()); |
| desc->AddContent(media_description_options.mid, MediaProtocolType::kOther, |
| /*rejected=*/true, std::move(unsupported)); |
| |
| if (!AddTransportOffer(media_description_options.mid, |
| media_description_options.transport_options, |
| current_description, desc, ice_credentials)) { |
| return false; |
| } |
| return true; |
| } |
| |
| // |audio_codecs| = set of all possible codecs that can be used, with correct |
| // payload type mappings |
| // |
| // |supported_audio_codecs| = set of codecs that are supported for the direction |
| // of this m= section |
| // |
| // acd->codecs() = set of previously negotiated codecs for this m= section |
| // |
| // The payload types should come from audio_codecs, but the order should come |
| // from acd->codecs() and then supported_codecs, to ensure that re-offers don't |
| // change existing codec priority, and that new codecs are added with the right |
| // priority. |
| bool MediaSessionDescriptionFactory::AddAudioContentForAnswer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* offer_content, |
| const SessionDescription* offer_description, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const TransportInfo* bundle_transport, |
| const AudioCodecs& audio_codecs, |
| const RtpHeaderExtensions& default_audio_rtp_header_extensions, |
| StreamParamsVec* current_streams, |
| SessionDescription* answer, |
| IceCredentialsIterator* ice_credentials) const { |
| RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_AUDIO)); |
| const AudioContentDescription* offer_audio_description = |
| offer_content->media_description()->as_audio(); |
| |
| std::unique_ptr<TransportDescription> audio_transport = CreateTransportAnswer( |
| media_description_options.mid, offer_description, |
| media_description_options.transport_options, current_description, |
| bundle_transport != nullptr, ice_credentials); |
| if (!audio_transport) { |
| return false; |
| } |
| |
| // Pick codecs based on the requested communications direction in the offer |
| // and the selected direction in the answer. |
| // Note these will be filtered one final time in CreateMediaContentAnswer. |
| auto wants_rtd = media_description_options.direction; |
| auto offer_rtd = offer_audio_description->direction(); |
| auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd); |
| AudioCodecs supported_audio_codecs = |
| GetAudioCodecsForAnswer(offer_rtd, answer_rtd); |
| |
| AudioCodecs filtered_codecs; |
| |
| if (!media_description_options.codec_preferences.empty()) { |
| filtered_codecs = MatchCodecPreference( |
| media_description_options.codec_preferences, supported_audio_codecs); |
| } else { |
| // Add the codecs from current content if it exists and is not rejected nor |
| // recycled. |
| if (current_content && !current_content->rejected && |
| current_content->name == media_description_options.mid) { |
| RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_AUDIO)); |
| const AudioContentDescription* acd = |
| current_content->media_description()->as_audio(); |
| for (const AudioCodec& codec : acd->codecs()) { |
| if (FindMatchingCodec<AudioCodec>(acd->codecs(), audio_codecs, codec, |
| nullptr)) { |
| filtered_codecs.push_back(codec); |
| } |
| } |
| } |
| // Add other supported audio codecs. |
| for (const AudioCodec& codec : supported_audio_codecs) { |
| if (FindMatchingCodec<AudioCodec>(supported_audio_codecs, audio_codecs, |
| codec, nullptr) && |
| !FindMatchingCodec<AudioCodec>(supported_audio_codecs, |
| filtered_codecs, codec, nullptr)) { |
| // We should use the local codec with local parameters and the codec id |
| // would be correctly mapped in |NegotiateCodecs|. |
| filtered_codecs.push_back(codec); |
| } |
| } |
| } |
| if (!session_options.vad_enabled) { |
| // If application doesn't want CN codecs in answer. |
| StripCNCodecs(&filtered_codecs); |
| } |
| |
| bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) && |
| session_options.bundle_enabled; |
| auto audio_answer = std::make_unique<AudioContentDescription>(); |
| // Do not require or create SDES cryptos if DTLS is used. |
| cricket::SecurePolicy sdes_policy = |
| audio_transport->secure() ? cricket::SEC_DISABLED : secure(); |
| if (!SetCodecsInAnswer(offer_audio_description, filtered_codecs, |
| media_description_options, session_options, |
| ssrc_generator_, current_streams, |
| audio_answer.get())) { |
| return false; |
| } |
| if (!CreateMediaContentAnswer( |
| offer_audio_description, media_description_options, session_options, |
| sdes_policy, GetCryptos(current_content), |
| filtered_rtp_header_extensions(default_audio_rtp_header_extensions), |
| ssrc_generator_, enable_encrypted_rtp_header_extensions_, |
| current_streams, bundle_enabled, audio_answer.get())) { |
| return false; // Fails the session setup. |
| } |
| |
| bool secure = bundle_transport ? bundle_transport->description.secure() |
| : audio_transport->secure(); |
| bool rejected = media_description_options.stopped || |
| offer_content->rejected || |
| !IsMediaProtocolSupported(MEDIA_TYPE_AUDIO, |
| audio_answer->protocol(), secure); |
| if (!AddTransportAnswer(media_description_options.mid, |
| *(audio_transport.get()), answer)) { |
| return false; |
| } |
| |
| if (rejected) { |
| RTC_LOG(LS_INFO) << "Audio m= section '" << media_description_options.mid |
| << "' being rejected in answer."; |
| } |
| |
| answer->AddContent(media_description_options.mid, offer_content->type, |
| rejected, std::move(audio_answer)); |
| return true; |
| } |
| |
| // TODO(kron): This function is very similar to AddAudioContentForAnswer. |
| // Refactor to reuse shared code. |
| bool MediaSessionDescriptionFactory::AddVideoContentForAnswer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* offer_content, |
| const SessionDescription* offer_description, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const TransportInfo* bundle_transport, |
| const VideoCodecs& video_codecs, |
| const RtpHeaderExtensions& default_video_rtp_header_extensions, |
| StreamParamsVec* current_streams, |
| SessionDescription* answer, |
| IceCredentialsIterator* ice_credentials) const { |
| RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_VIDEO)); |
| const VideoContentDescription* offer_video_description = |
| offer_content->media_description()->as_video(); |
| |
| std::unique_ptr<TransportDescription> video_transport = CreateTransportAnswer( |
| media_description_options.mid, offer_description, |
| media_description_options.transport_options, current_description, |
| bundle_transport != nullptr, ice_credentials); |
| if (!video_transport) { |
| return false; |
| } |
| |
| // Pick codecs based on the requested communications direction in the offer |
| // and the selected direction in the answer. |
| // Note these will be filtered one final time in CreateMediaContentAnswer. |
| auto wants_rtd = media_description_options.direction; |
| auto offer_rtd = offer_video_description->direction(); |
| auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd); |
| VideoCodecs supported_video_codecs = |
| GetVideoCodecsForAnswer(offer_rtd, answer_rtd); |
| |
| VideoCodecs filtered_codecs; |
| |
| if (!media_description_options.codec_preferences.empty()) { |
| filtered_codecs = MatchCodecPreference( |
| media_description_options.codec_preferences, supported_video_codecs); |
| } else { |
| // Add the codecs from current content if it exists and is not rejected nor |
| // recycled. |
| if (current_content && !current_content->rejected && |
| current_content->name == media_description_options.mid) { |
| RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_VIDEO)); |
| const VideoContentDescription* vcd = |
| current_content->media_description()->as_video(); |
| for (const VideoCodec& codec : vcd->codecs()) { |
| if (FindMatchingCodec<VideoCodec>(vcd->codecs(), video_codecs, codec, |
| nullptr)) { |
| filtered_codecs.push_back(codec); |
| } |
| } |
| } |
| // Add other supported video codecs. |
| for (const VideoCodec& codec : supported_video_codecs) { |
| if (FindMatchingCodec<VideoCodec>(supported_video_codecs, video_codecs, |
| codec, nullptr) && |
| !FindMatchingCodec<VideoCodec>(supported_video_codecs, |
| filtered_codecs, codec, nullptr)) { |
| // We should use the local codec with local parameters and the codec id |
| // would be correctly mapped in |NegotiateCodecs|. |
| filtered_codecs.push_back(codec); |
| } |
| } |
| } |
| |
| if (session_options.raw_packetization_for_video) { |
| for (VideoCodec& codec : filtered_codecs) { |
| if (codec.GetCodecType() == VideoCodec::CODEC_VIDEO) { |
| codec.packetization = kPacketizationParamRaw; |
| } |
| } |
| } |
| |
| bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) && |
| session_options.bundle_enabled; |
| auto video_answer = std::make_unique<VideoContentDescription>(); |
| // Do not require or create SDES cryptos if DTLS is used. |
| cricket::SecurePolicy sdes_policy = |
| video_transport->secure() ? cricket::SEC_DISABLED : secure(); |
| if (!SetCodecsInAnswer(offer_video_description, filtered_codecs, |
| media_description_options, session_options, |
| ssrc_generator_, current_streams, |
| video_answer.get())) { |
| return false; |
| } |
| if (!CreateMediaContentAnswer( |
| offer_video_description, media_description_options, session_options, |
| sdes_policy, GetCryptos(current_content), |
| filtered_rtp_header_extensions(default_video_rtp_header_extensions), |
| ssrc_generator_, enable_encrypted_rtp_header_extensions_, |
| current_streams, bundle_enabled, video_answer.get())) { |
| return false; // Failed the sessin setup. |
| } |
| bool secure = bundle_transport ? bundle_transport->description.secure() |
| : video_transport->secure(); |
| bool rejected = media_description_options.stopped || |
| offer_content->rejected || |
| !IsMediaProtocolSupported(MEDIA_TYPE_VIDEO, |
| video_answer->protocol(), secure); |
| if (!AddTransportAnswer(media_description_options.mid, |
| *(video_transport.get()), answer)) { |
| return false; |
| } |
| |
| if (!rejected) { |
| video_answer->set_bandwidth(kAutoBandwidth); |
| } else { |
| RTC_LOG(LS_INFO) << "Video m= section '" << media_description_options.mid |
| << "' being rejected in answer."; |
| } |
| answer->AddContent(media_description_options.mid, offer_content->type, |
| rejected, std::move(video_answer)); |
| return true; |
| } |
| |
| bool MediaSessionDescriptionFactory::AddDataContentForAnswer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* offer_content, |
| const SessionDescription* offer_description, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const TransportInfo* bundle_transport, |
| StreamParamsVec* current_streams, |
| SessionDescription* answer, |
| IceCredentialsIterator* ice_credentials) const { |
| std::unique_ptr<TransportDescription> data_transport = CreateTransportAnswer( |
| media_description_options.mid, offer_description, |
| media_description_options.transport_options, current_description, |
| bundle_transport != nullptr, ice_credentials); |
| if (!data_transport) { |
| return false; |
| } |
| |
| // 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_description->HasGroup(GROUP_TYPE_BUNDLE) && |
| session_options.bundle_enabled; |
| RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_DATA)); |
| std::unique_ptr<MediaContentDescription> data_answer; |
| if (offer_content->media_description()->as_sctp()) { |
| // SCTP data content |
| data_answer = std::make_unique<SctpDataContentDescription>(); |
| const SctpDataContentDescription* offer_data_description = |
| offer_content->media_description()->as_sctp(); |
| // Respond with the offerer's proto, whatever it is. |
| data_answer->as_sctp()->set_protocol(offer_data_description->protocol()); |
| // Respond with our max message size or the remote max messsage size, |
| // whichever is smaller. |
| // 0 is treated specially - it means "I can accept any size". Since |
| // we do not implement infinite size messages, reply with |
| // kSctpSendBufferSize. |
| if (offer_data_description->max_message_size() == 0) { |
| data_answer->as_sctp()->set_max_message_size(kSctpSendBufferSize); |
| } else { |
| data_answer->as_sctp()->set_max_message_size(std::min( |
| offer_data_description->max_message_size(), kSctpSendBufferSize)); |
| } |
| if (!CreateMediaContentAnswer( |
| offer_data_description, media_description_options, session_options, |
| sdes_policy, GetCryptos(current_content), RtpHeaderExtensions(), |
| ssrc_generator_, enable_encrypted_rtp_header_extensions_, |
| current_streams, bundle_enabled, data_answer.get())) { |
| return false; // Fails the session setup. |
| } |
| // Respond with sctpmap if the offer uses sctpmap. |
| bool offer_uses_sctpmap = offer_data_description->use_sctpmap(); |
| data_answer->as_sctp()->set_use_sctpmap(offer_uses_sctpmap); |
| } else { |
| RTC_NOTREACHED() << "Non-SCTP data content found"; |
| } |
| |
| bool secure = bundle_transport ? bundle_transport->description.secure() |
| : data_transport->secure(); |
| |
| bool rejected = media_description_options.stopped || |
| offer_content->rejected || |
| !IsMediaProtocolSupported(MEDIA_TYPE_DATA, |
| data_answer->protocol(), secure); |
| if (!AddTransportAnswer(media_description_options.mid, |
| *(data_transport.get()), answer)) { |
| return false; |
| } |
| |
| answer->AddContent(media_description_options.mid, offer_content->type, |
| rejected, std::move(data_answer)); |
| return true; |
| } |
| |
| bool MediaSessionDescriptionFactory::AddUnsupportedContentForAnswer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* offer_content, |
| const SessionDescription* offer_description, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const TransportInfo* bundle_transport, |
| SessionDescription* answer, |
| IceCredentialsIterator* ice_credentials) const { |
| std::unique_ptr<TransportDescription> unsupported_transport = |
| CreateTransportAnswer(media_description_options.mid, offer_description, |
| media_description_options.transport_options, |
| current_description, bundle_transport != nullptr, |
| ice_credentials); |
| if (!unsupported_transport) { |
| return false; |
| } |
| RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_UNSUPPORTED)); |
| |
| const UnsupportedContentDescription* offer_unsupported_description = |
| offer_content->media_description()->as_unsupported(); |
| std::unique_ptr<MediaContentDescription> unsupported_answer = |
| std::make_unique<UnsupportedContentDescription>( |
| offer_unsupported_description->media_type()); |
| unsupported_answer->set_protocol(offer_unsupported_description->protocol()); |
| |
| if (!AddTransportAnswer(media_description_options.mid, |
| *(unsupported_transport.get()), answer)) { |
| return false; |
| } |
| answer->AddContent(media_description_options.mid, offer_content->type, |
| /*rejected=*/true, std::move(unsupported_answer)); |
| return true; |
| } |
| |
| void MediaSessionDescriptionFactory::ComputeAudioCodecsIntersectionAndUnion() { |
| audio_sendrecv_codecs_.clear(); |
| all_audio_codecs_.clear(); |
| // Compute the audio codecs union. |
| for (const AudioCodec& send : audio_send_codecs_) { |
| all_audio_codecs_.push_back(send); |
| if (!FindMatchingCodec<AudioCodec>(audio_send_codecs_, audio_recv_codecs_, |
| send, nullptr)) { |
| // It doesn't make sense to have an RTX codec we support sending but not |
| // receiving. |
| RTC_DCHECK(!IsRtxCodec(send)); |
| } |
| } |
| for (const AudioCodec& recv : audio_recv_codecs_) { |
| if (!FindMatchingCodec<AudioCodec>(audio_recv_codecs_, audio_send_codecs_, |
| recv, nullptr)) { |
| all_audio_codecs_.push_back(recv); |
| } |
| } |
| // 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(audio_recv_codecs_, audio_send_codecs_, |
| &audio_sendrecv_codecs_, true); |
| } |
| |
| void MediaSessionDescriptionFactory::ComputeVideoCodecsIntersectionAndUnion() { |
| video_sendrecv_codecs_.clear(); |
| all_video_codecs_.clear(); |
| // Compute the video codecs union. |
| for (const VideoCodec& send : video_send_codecs_) { |
| all_video_codecs_.push_back(send); |
| if (!FindMatchingCodec<VideoCodec>(video_send_codecs_, video_recv_codecs_, |
| send, nullptr)) { |
| // TODO(kron): This check is violated by the unit test: |
| // MediaSessionDescriptionFactoryTest.RtxWithoutApt |
| // Remove either the test or the check. |
| |
| // It doesn't make sense to have an RTX codec we support sending but not |
| // receiving. |
| // RTC_DCHECK(!IsRtxCodec(send)); |
| } |
| } |
| for (const VideoCodec& recv : video_recv_codecs_) { |
| if (!FindMatchingCodec<VideoCodec>(video_recv_codecs_, video_send_codecs_, |
| recv, nullptr)) { |
| all_video_codecs_.push_back(recv); |
| } |
| } |
| // 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(video_recv_codecs_, video_send_codecs_, |
| &video_sendrecv_codecs_, true); |
| } |
| |
| bool IsMediaContent(const ContentInfo* content) { |
| return (content && (content->type == MediaProtocolType::kRtp || |
| content->type == MediaProtocolType::kSctp)); |
| } |
| |
| 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); |
| } |
| |
| bool IsUnsupportedContent(const ContentInfo* content) { |
| return IsMediaContentOfType(content, MEDIA_TYPE_UNSUPPORTED); |
| } |
| |
| 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); |
| } |
| |
| 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); |
| return (content ? content->media_description() : nullptr); |
| } |
| |
| const AudioContentDescription* GetFirstAudioContentDescription( |
| const
|