/* * 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. */ #ifndef PC_SESSION_DESCRIPTION_H_ #define PC_SESSION_DESCRIPTION_H_ #include #include #include #include #include #include #include #include "absl/memory/memory.h" #include "api/crypto_params.h" #include "api/media_types.h" #include "api/rtp_parameters.h" #include "api/rtp_transceiver_interface.h" #include "media/base/media_channel.h" #include "media/base/stream_params.h" #include "p2p/base/transport_description.h" #include "p2p/base/transport_info.h" #include "pc/media_protocol_names.h" #include "pc/simulcast_description.h" #include "rtc_base/deprecation.h" #include "rtc_base/socket_address.h" #include "rtc_base/system/rtc_export.h" namespace cricket { typedef std::vector AudioCodecs; typedef std::vector VideoCodecs; typedef std::vector RtpDataCodecs; typedef std::vector CryptoParamsVec; typedef std::vector RtpHeaderExtensions; // RTC4585 RTP/AVPF extern const char kMediaProtocolAvpf[]; // RFC5124 RTP/SAVPF extern const char kMediaProtocolSavpf[]; extern const char kMediaProtocolDtlsSavpf[]; // Options to control how session descriptions are generated. const int kAutoBandwidth = -1; class AudioContentDescription; class VideoContentDescription; class RtpDataContentDescription; class SctpDataContentDescription; // Describes a session description media section. There are subclasses for each // media type (audio, video, data) that will have additional information. class MediaContentDescription { public: MediaContentDescription() = default; virtual ~MediaContentDescription() = default; virtual MediaType type() const = 0; // Try to cast this media description to an AudioContentDescription. Returns // nullptr if the cast fails. virtual AudioContentDescription* as_audio() { return nullptr; } virtual const AudioContentDescription* as_audio() const { return nullptr; } // Try to cast this media description to a VideoContentDescription. Returns // nullptr if the cast fails. virtual VideoContentDescription* as_video() { return nullptr; } virtual const VideoContentDescription* as_video() const { return nullptr; } virtual RtpDataContentDescription* as_rtp_data() { return nullptr; } virtual const RtpDataContentDescription* as_rtp_data() const { return nullptr; } virtual SctpDataContentDescription* as_sctp() { return nullptr; } virtual const SctpDataContentDescription* as_sctp() const { return nullptr; } virtual bool has_codecs() const = 0; // Copy operator that returns an unique_ptr. // Not a virtual function. // If a type-specific variant of Clone() is desired, override it, or // simply use std::make_unique(*this) instead of Clone(). std::unique_ptr Clone() const { return absl::WrapUnique(CloneInternal()); } // |protocol| is the expected media transport protocol, such as RTP/AVPF, // RTP/SAVPF or SCTP/DTLS. virtual std::string protocol() const { return protocol_; } virtual void set_protocol(const std::string& protocol) { protocol_ = protocol; } virtual webrtc::RtpTransceiverDirection direction() const { return direction_; } virtual void set_direction(webrtc::RtpTransceiverDirection direction) { direction_ = direction; } virtual bool rtcp_mux() const { return rtcp_mux_; } virtual void set_rtcp_mux(bool mux) { rtcp_mux_ = mux; } virtual bool rtcp_reduced_size() const { return rtcp_reduced_size_; } virtual void set_rtcp_reduced_size(bool reduced_size) { rtcp_reduced_size_ = reduced_size; } // Indicates support for the remote network estimate packet type. This // functionality is experimental and subject to change without notice. virtual bool remote_estimate() const { return remote_estimate_; } virtual void set_remote_estimate(bool remote_estimate) { remote_estimate_ = remote_estimate; } virtual int bandwidth() const { return bandwidth_; } virtual void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; } virtual const std::vector& cryptos() const { return cryptos_; } virtual void AddCrypto(const CryptoParams& params) { cryptos_.push_back(params); } virtual void set_cryptos(const std::vector& cryptos) { cryptos_ = cryptos; } virtual const RtpHeaderExtensions& rtp_header_extensions() const { return rtp_header_extensions_; } virtual void set_rtp_header_extensions( const RtpHeaderExtensions& extensions) { rtp_header_extensions_ = extensions; rtp_header_extensions_set_ = true; } virtual void AddRtpHeaderExtension(const webrtc::RtpExtension& ext) { rtp_header_extensions_.push_back(ext); rtp_header_extensions_set_ = true; } virtual void ClearRtpHeaderExtensions() { rtp_header_extensions_.clear(); rtp_header_extensions_set_ = true; } // We can't always tell if an empty list of header extensions is // because the other side doesn't support them, or just isn't hooked up to // signal them. For now we assume an empty list means no signaling, but // provide the ClearRtpHeaderExtensions method to allow "no support" to be // clearly indicated (i.e. when derived from other information). virtual bool rtp_header_extensions_set() const { return rtp_header_extensions_set_; } virtual const StreamParamsVec& streams() const { return send_streams_; } // TODO(pthatcher): Remove this by giving mediamessage.cc access // to MediaContentDescription virtual StreamParamsVec& mutable_streams() { return send_streams_; } virtual void AddStream(const StreamParams& stream) { send_streams_.push_back(stream); } // Legacy streams have an ssrc, but nothing else. void AddLegacyStream(uint32_t ssrc) { AddStream(StreamParams::CreateLegacy(ssrc)); } void AddLegacyStream(uint32_t ssrc, uint32_t fid_ssrc) { StreamParams sp = StreamParams::CreateLegacy(ssrc); sp.AddFidSsrc(ssrc, fid_ssrc); AddStream(sp); } // Sets the CNAME of all StreamParams if it have not been set. virtual void SetCnameIfEmpty(const std::string& cname) { for (cricket::StreamParamsVec::iterator it = send_streams_.begin(); it != send_streams_.end(); ++it) { if (it->cname.empty()) it->cname = cname; } } virtual uint32_t first_ssrc() const { if (send_streams_.empty()) { return 0; } return send_streams_[0].first_ssrc(); } virtual bool has_ssrcs() const { if (send_streams_.empty()) { return false; } return send_streams_[0].has_ssrcs(); } virtual void set_conference_mode(bool enable) { conference_mode_ = enable; } virtual bool conference_mode() const { return conference_mode_; } // https://tools.ietf.org/html/rfc4566#section-5.7 // May be present at the media or session level of SDP. If present at both // levels, the media-level attribute overwrites the session-level one. virtual void set_connection_address(const rtc::SocketAddress& address) { connection_address_ = address; } virtual const rtc::SocketAddress& connection_address() const { return connection_address_; } // Determines if it's allowed to mix one- and two-byte rtp header extensions // within the same rtp stream. enum ExtmapAllowMixed { kNo, kSession, kMedia }; virtual void set_extmap_allow_mixed_enum( ExtmapAllowMixed new_extmap_allow_mixed) { if (new_extmap_allow_mixed == kMedia && extmap_allow_mixed_enum_ == kSession) { // Do not downgrade from session level to media level. return; } extmap_allow_mixed_enum_ = new_extmap_allow_mixed; } virtual ExtmapAllowMixed extmap_allow_mixed_enum() const { return extmap_allow_mixed_enum_; } virtual bool extmap_allow_mixed() const { return extmap_allow_mixed_enum_ != kNo; } // Simulcast functionality. virtual bool HasSimulcast() const { return !simulcast_.empty(); } virtual SimulcastDescription& simulcast_description() { return simulcast_; } virtual const SimulcastDescription& simulcast_description() const { return simulcast_; } virtual void set_simulcast_description( const SimulcastDescription& simulcast) { simulcast_ = simulcast; } virtual const std::vector& receive_rids() const { return receive_rids_; } virtual void set_receive_rids(const std::vector& rids) { receive_rids_ = rids; } protected: bool rtcp_mux_ = false; bool rtcp_reduced_size_ = false; bool remote_estimate_ = false; int bandwidth_ = kAutoBandwidth; std::string protocol_; std::vector cryptos_; std::vector rtp_header_extensions_; bool rtp_header_extensions_set_ = false; StreamParamsVec send_streams_; bool conference_mode_ = false; webrtc::RtpTransceiverDirection direction_ = webrtc::RtpTransceiverDirection::kSendRecv; rtc::SocketAddress connection_address_; // Mixed one- and two-byte header not included in offer on media level or // session level, but we will respond that we support it. The plan is to add // it to our offer on session level. See todo in SessionDescription. ExtmapAllowMixed extmap_allow_mixed_enum_ = kNo; SimulcastDescription simulcast_; std::vector receive_rids_; private: // Copy function that returns a raw pointer. Caller will assert ownership. // Should only be called by the Clone() function. Must be implemented // by each final subclass. virtual MediaContentDescription* CloneInternal() const = 0; }; template class MediaContentDescriptionImpl : public MediaContentDescription { public: void set_protocol(const std::string& protocol) override { RTC_DCHECK(IsRtpProtocol(protocol)); protocol_ = protocol; } typedef C CodecType; // Codecs should be in preference order (most preferred codec first). virtual const std::vector& codecs() const { return codecs_; } virtual void set_codecs(const std::vector& codecs) { codecs_ = codecs; } bool has_codecs() const override { return !codecs_.empty(); } virtual bool HasCodec(int id) { bool found = false; for (typename std::vector::iterator iter = codecs_.begin(); iter != codecs_.end(); ++iter) { if (iter->id == id) { found = true; break; } } return found; } virtual void AddCodec(const C& codec) { codecs_.push_back(codec); } virtual void AddOrReplaceCodec(const C& codec) { for (typename std::vector::iterator iter = codecs_.begin(); iter != codecs_.end(); ++iter) { if (iter->id == codec.id) { *iter = codec; return; } } AddCodec(codec); } virtual void AddCodecs(const std::vector& codecs) { typename std::vector::const_iterator codec; for (codec = codecs.begin(); codec != codecs.end(); ++codec) { AddCodec(*codec); } } private: std::vector codecs_; }; class AudioContentDescription : public MediaContentDescriptionImpl { public: AudioContentDescription() {} virtual MediaType type() const { return MEDIA_TYPE_AUDIO; } virtual AudioContentDescription* as_audio() { return this; } virtual const AudioContentDescription* as_audio() const { return this; } private: virtual AudioContentDescription* CloneInternal() const { return new AudioContentDescription(*this); } }; class VideoContentDescription : public MediaContentDescriptionImpl { public: virtual MediaType type() const { return MEDIA_TYPE_VIDEO; } virtual VideoContentDescription* as_video() { return this; } virtual const VideoContentDescription* as_video() const { return this; } private: virtual VideoContentDescription* CloneInternal() const { return new VideoContentDescription(*this); } }; class RtpDataContentDescription : public MediaContentDescriptionImpl { public: RtpDataContentDescription() {} MediaType type() const override { return MEDIA_TYPE_DATA; } RtpDataContentDescription* as_rtp_data() override { return this; } const RtpDataContentDescription* as_rtp_data() const override { return this; } private: RtpDataContentDescription* CloneInternal() const override { return new RtpDataContentDescription(*this); } }; class SctpDataContentDescription : public MediaContentDescription { public: SctpDataContentDescription() {} SctpDataContentDescription(const SctpDataContentDescription& o) : MediaContentDescription(o), use_sctpmap_(o.use_sctpmap_), port_(o.port_), max_message_size_(o.max_message_size_) {} MediaType type() const override { return MEDIA_TYPE_DATA; } SctpDataContentDescription* as_sctp() override { return this; } const SctpDataContentDescription* as_sctp() const override { return this; } bool has_codecs() const override { return false; } void set_protocol(const std::string& protocol) override { RTC_DCHECK(IsSctpProtocol(protocol)); protocol_ = protocol; } bool use_sctpmap() const { return use_sctpmap_; } void set_use_sctpmap(bool enable) { use_sctpmap_ = enable; } int port() const { return port_; } void set_port(int port) { port_ = port; } int max_message_size() const { return max_message_size_; } void set_max_message_size(int max_message_size) { max_message_size_ = max_message_size; } private: SctpDataContentDescription* CloneInternal() const override { return new SctpDataContentDescription(*this); } bool use_sctpmap_ = true; // Note: "true" is no longer conformant. // Defaults should be constants imported from SCTP. Quick hack. int port_ = 5000; // draft-ietf-mmusic-sdp-sctp-23: Max message size default is 64K int max_message_size_ = 64 * 1024; }; // Protocol used for encoding media. This is the "top level" protocol that may // be wrapped by zero or many transport protocols (UDP, ICE, etc.). enum class MediaProtocolType { kRtp, // Section will use the RTP protocol (e.g., for audio or video). // https://tools.ietf.org/html/rfc3550 kSctp // Section will use the SCTP protocol (e.g., for a data channel). // https://tools.ietf.org/html/rfc4960 }; // Represents a session description section. Most information about the section // is stored in the description, which is a subclass of MediaContentDescription. // Owns the description. class RTC_EXPORT ContentInfo { public: explicit ContentInfo(MediaProtocolType type) : type(type) {} ~ContentInfo(); // Copy ContentInfo(const ContentInfo& o); ContentInfo& operator=(const ContentInfo& o); ContentInfo(ContentInfo&& o) = default; ContentInfo& operator=(ContentInfo&& o) = default; // Alias for |name|. std::string mid() const { return name; } void set_mid(const std::string& mid) { this->name = mid; } // Alias for |description|. MediaContentDescription* media_description(); const MediaContentDescription* media_description() const; void set_media_description(std::unique_ptr desc) { description_ = std::move(desc); } // TODO(bugs.webrtc.org/8620): Rename this to mid. std::string name; MediaProtocolType type; bool rejected = false; bool bundle_only = false; private: friend class SessionDescription; std::unique_ptr description_; }; typedef std::vector ContentNames; // This class provides a mechanism to aggregate different media contents into a // group. This group can also be shared with the peers in a pre-defined format. // GroupInfo should be populated only with the |content_name| of the // MediaDescription. class ContentGroup { public: explicit ContentGroup(const std::string& semantics); ContentGroup(const ContentGroup&); ContentGroup(ContentGroup&&); ContentGroup& operator=(const ContentGroup&); ContentGroup& operator=(ContentGroup&&); ~ContentGroup(); const std::string& semantics() const { return semantics_; } const ContentNames& content_names() const { return content_names_; } const std::string* FirstContentName() const; bool HasContentName(const std::string& content_name) const; void AddContentName(const std::string& content_name); bool RemoveContentName(const std::string& content_name); private: std::string semantics_; ContentNames content_names_; }; typedef std::vector ContentInfos; typedef std::vector ContentGroups; const ContentInfo* FindContentInfoByName(const ContentInfos& contents, const std::string& name); const ContentInfo* FindContentInfoByType(const ContentInfos& contents, const std::string& type); // Determines how the MSID will be signaled in the SDP. These can be used as // flags to indicate both or none. enum MsidSignaling { // Signal MSID with one a=msid line in the media section. kMsidSignalingMediaSection = 0x1, // Signal MSID with a=ssrc: msid lines in the media section. kMsidSignalingSsrcAttribute = 0x2 }; // Describes a collection of contents, each with its own name and // type. Analogous to a or stanza. Assumes that // contents are unique be name, but doesn't enforce that. class SessionDescription { public: SessionDescription(); ~SessionDescription(); std::unique_ptr Clone() const; // Content accessors. const ContentInfos& contents() const { return contents_; } ContentInfos& contents() { return contents_; } const ContentInfo* GetContentByName(const std::string& name) const; ContentInfo* GetContentByName(const std::string& name); const MediaContentDescription* GetContentDescriptionByName( const std::string& name) const; MediaContentDescription* GetContentDescriptionByName(const std::string& name); const ContentInfo* FirstContentByType(MediaProtocolType type) const; const ContentInfo* FirstContent() const; // Content mutators. // Adds a content to this description. Takes ownership of ContentDescription*. void AddContent(const std::string& name, MediaProtocolType type, std::unique_ptr description); void AddContent(const std::string& name, MediaProtocolType type, bool rejected, std::unique_ptr description); void AddContent(const std::string& name, MediaProtocolType type, bool rejected, bool bundle_only, std::unique_ptr description); void AddContent(ContentInfo&& content); bool RemoveContentByName(const std::string& name); // Transport accessors. const TransportInfos& transport_infos() const { return transport_infos_; } TransportInfos& transport_infos() { return transport_infos_; } const TransportInfo* GetTransportInfoByName(const std::string& name) const; TransportInfo* GetTransportInfoByName(const std::string& name); const TransportDescription* GetTransportDescriptionByName( const std::string& name) const { const TransportInfo* tinfo = GetTransportInfoByName(name); return tinfo ? &tinfo->description : NULL; } // Transport mutators. void set_transport_infos(const TransportInfos& transport_infos) { transport_infos_ = transport_infos; } // Adds a TransportInfo to this description. void AddTransportInfo(const TransportInfo& transport_info); bool RemoveTransportInfoByName(const std::string& name); // Group accessors. const ContentGroups& groups() const { return content_groups_; } const ContentGroup* GetGroupByName(const std::string& name) const; bool HasGroup(const std::string& name) const; // Group mutators. void AddGroup(const ContentGroup& group) { content_groups_.push_back(group); } // Remove the first group with the same semantics specified by |name|. void RemoveGroupByName(const std::string& name); // Global attributes. void set_msid_supported(bool supported) { msid_supported_ = supported; } bool msid_supported() const { return msid_supported_; } // Determines how the MSIDs were/will be signaled. Flag value composed of // MsidSignaling bits (see enum above). void set_msid_signaling(int msid_signaling) { msid_signaling_ = msid_signaling; } int msid_signaling() const { return msid_signaling_; } // Determines if it's allowed to mix one- and two-byte rtp header extensions // within the same rtp stream. void set_extmap_allow_mixed(bool supported) { extmap_allow_mixed_ = supported; MediaContentDescription::ExtmapAllowMixed media_level_setting = supported ? MediaContentDescription::kSession : MediaContentDescription::kNo; for (auto& content : contents_) { // Do not set to kNo if the current setting is kMedia. if (supported || content.media_description()->extmap_allow_mixed_enum() != MediaContentDescription::kMedia) { content.media_description()->set_extmap_allow_mixed_enum( media_level_setting); } } } bool extmap_allow_mixed() const { return extmap_allow_mixed_; } private: SessionDescription(const SessionDescription&); ContentInfos contents_; TransportInfos transport_infos_; ContentGroups content_groups_; bool msid_supported_ = true; // Default to what Plan B would do. // TODO(bugs.webrtc.org/8530): Change default to kMsidSignalingMediaSection. int msid_signaling_ = kMsidSignalingSsrcAttribute; // TODO(webrtc:9985): Activate mixed one- and two-byte header extension in // offer at session level. It's currently not included in offer by default // because clients prior to https://bugs.webrtc.org/9712 cannot parse this // correctly. If it's included in offer to us we will respond that we support // it. bool extmap_allow_mixed_ = false; }; // Indicates whether a session description was sent by the local client or // received from the remote client. enum ContentSource { CS_LOCAL, CS_REMOTE }; } // namespace cricket #endif // PC_SESSION_DESCRIPTION_H_