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/*
* Copyright 2012 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.
*/
// This file contains the PeerConnection interface as defined in
//
// The PeerConnectionFactory class provides factory methods to create
// PeerConnection, MediaStream and MediaStreamTrack objects.
//
// The following steps are needed to setup a typical call using WebRTC:
//
// 1. Create a PeerConnectionFactoryInterface. Check constructors for more
// information about input parameters.
//
// 2. Create a PeerConnection object. Provide a configuration struct which
// points to STUN and/or TURN servers used to generate ICE candidates, and
// provide an object that implements the PeerConnectionObserver interface,
// which is used to receive callbacks from the PeerConnection.
//
// 3. Create local MediaStreamTracks using the PeerConnectionFactory and add
// them to PeerConnection by calling AddTrack (or legacy method, AddStream).
//
// 4. Create an offer, call SetLocalDescription with it, serialize it, and send
// it to the remote peer
//
// 5. Once an ICE candidate has been gathered, the PeerConnection will call the
// observer function OnIceCandidate. The candidates must also be serialized and
// sent to the remote peer.
//
// 6. Once an answer is received from the remote peer, call
// SetRemoteDescription with the remote answer.
//
// 7. Once a remote candidate is received from the remote peer, provide it to
// the PeerConnection by calling AddIceCandidate.
//
// The receiver of a call (assuming the application is "call"-based) can decide
// to accept or reject the call; this decision will be taken by the application,
// not the PeerConnection.
//
// If the application decides to accept the call, it should:
//
// 1. Create PeerConnectionFactoryInterface if it doesn't exist.
//
// 2. Create a new PeerConnection.
//
// 3. Provide the remote offer to the new PeerConnection object by calling
// SetRemoteDescription.
//
// 4. Generate an answer to the remote offer by calling CreateAnswer and send it
// back to the remote peer.
//
// 5. Provide the local answer to the new PeerConnection by calling
// SetLocalDescription with the answer.
//
// 6. Provide the remote ICE candidates by calling AddIceCandidate.
//
// 7. Once a candidate has been gathered, the PeerConnection will call the
// observer function OnIceCandidate. Send these candidates to the remote peer.
#ifndef API_PEER_CONNECTION_INTERFACE_H_
#define API_PEER_CONNECTION_INTERFACE_H_
// IWYU pragma: no_include "pc/media_factory.h"
#include <stdint.h>
#include <stdio.h>
#include <functional>
#include <memory>
#include <string>
#include <vector>
#include "absl/base/attributes.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/async_dns_resolver.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_options.h"
#include "api/candidate.h"
#include "api/crypto/crypto_options.h"
#include "api/data_channel_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/fec_controller.h"
#include "api/field_trials_view.h"
#include "api/ice_transport_interface.h"
#include "api/jsep.h"
#include "api/legacy_stats_types.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/metronome/metronome.h"
#include "api/neteq/neteq_factory.h"
#include "api/network_state_predictor.h"
#include "api/packet_socket_factory.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
#include "api/rtc_event_log_output.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/sctp_transport_interface.h"
#include "api/set_local_description_observer_interface.h"
#include "api/set_remote_description_observer_interface.h"
#include "api/stats/rtc_stats_collector_callback.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/bandwidth_estimation_settings.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/enums.h"
#include "api/transport/network_control.h"
#include "api/transport/sctp_transport_factory_interface.h"
#include "api/turn_customizer.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "call/rtp_transport_controller_send_factory_interface.h"
#include "media/base/media_config.h"
// TODO(bugs.webrtc.org/7447): We plan to provide a way to let applications
// inject a PacketSocketFactory and/or NetworkManager, and not expose
// PortAllocator in the PeerConnection api.
#include "api/audio/audio_frame_processor.h"
#include "api/ref_count.h"
#include "api/units/time_delta.h"
#include "p2p/base/port.h"
#include "p2p/base/port_allocator.h"
#include "rtc_base/checks.h"
#include "rtc_base/network.h"
#include "rtc_base/network_constants.h"
#include "rtc_base/network_monitor_factory.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/socket_factory.h"
#include "rtc_base/ssl_certificate.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/system/rtc_export.h"
#include "rtc_base/thread.h"
namespace rtc {
class Thread; // IWYU pragma: keep
} // namespace rtc
namespace webrtc {
// IWYU pragma: begin_keep
// MediaFactory class definition is not part of the api.
class MediaFactory;
// IWYU pragma: end_keep
// MediaStream container interface.
class StreamCollectionInterface : public webrtc::RefCountInterface {
public:
// TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
virtual size_t count() = 0;
virtual MediaStreamInterface* at(size_t index) = 0;
virtual MediaStreamInterface* find(const std::string& label) = 0;
virtual MediaStreamTrackInterface* FindAudioTrack(const std::string& id) = 0;
virtual MediaStreamTrackInterface* FindVideoTrack(const std::string& id) = 0;
protected:
// Dtor protected as objects shouldn't be deleted via this interface.
~StreamCollectionInterface() override = default;
};
class StatsObserver : public webrtc::RefCountInterface {
public:
virtual void OnComplete(const StatsReports& reports) = 0;
protected:
~StatsObserver() override = default;
};
enum class SdpSemantics {
kPlanB_DEPRECATED,
kPlanB [[deprecated]] = kPlanB_DEPRECATED,
kUnifiedPlan,
};
class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
public:
enum SignalingState {
kStable,
kHaveLocalOffer,
kHaveLocalPrAnswer,
kHaveRemoteOffer,
kHaveRemotePrAnswer,
kClosed,
};
static constexpr absl::string_view AsString(SignalingState);
enum IceGatheringState {
kIceGatheringNew,
kIceGatheringGathering,
kIceGatheringComplete
};
static constexpr absl::string_view AsString(IceGatheringState state);
enum class PeerConnectionState {
kNew,
kConnecting,
kConnected,
kDisconnected,
kFailed,
kClosed,
};
static constexpr absl::string_view AsString(PeerConnectionState state);
enum IceConnectionState {
kIceConnectionNew,
kIceConnectionChecking,
kIceConnectionConnected,
kIceConnectionCompleted,
kIceConnectionFailed,
kIceConnectionDisconnected,
kIceConnectionClosed,
kIceConnectionMax,
};
static constexpr absl::string_view AsString(IceConnectionState state);
// TLS certificate policy.
enum TlsCertPolicy {
// For TLS based protocols, ensure the connection is secure by not
// circumventing certificate validation.
kTlsCertPolicySecure,
// For TLS based protocols, disregard security completely by skipping
// certificate validation. This is insecure and should never be used unless
// security is irrelevant in that particular context.
kTlsCertPolicyInsecureNoCheck,
};
struct RTC_EXPORT IceServer {
IceServer();
IceServer(const IceServer&);
~IceServer();
// TODO(jbauch): Remove uri when all code using it has switched to urls.
// List of URIs associated with this server. Valid formats are described
// in RFC7064 and RFC7065, and more may be added in the future. The "host"
// part of the URI may contain either an IP address or a hostname.
std::string uri;
std::vector<std::string> urls;
std::string username;
std::string password;
TlsCertPolicy tls_cert_policy = kTlsCertPolicySecure;
// If the URIs in `urls` only contain IP addresses, this field can be used
// to indicate the hostname, which may be necessary for TLS (using the SNI
// extension). If `urls` itself contains the hostname, this isn't
// necessary.
std::string hostname;
// List of protocols to be used in the TLS ALPN extension.
std::vector<std::string> tls_alpn_protocols;
// List of elliptic curves to be used in the TLS elliptic curves extension.
std::vector<std::string> tls_elliptic_curves;
bool operator==(const IceServer& o) const {
return uri == o.uri && urls == o.urls && username == o.username &&
password == o.password && tls_cert_policy == o.tls_cert_policy &&
hostname == o.hostname &&
tls_alpn_protocols == o.tls_alpn_protocols &&
tls_elliptic_curves == o.tls_elliptic_curves;
}
bool operator!=(const IceServer& o) const { return !(*this == o); }
};
typedef std::vector<IceServer> IceServers;
enum IceTransportsType {
// TODO(pthatcher): Rename these kTransporTypeXXX, but update
// Chromium at the same time.
kNone,
kRelay,
kNoHost,
kAll
};
enum BundlePolicy {
kBundlePolicyBalanced,
kBundlePolicyMaxBundle,
kBundlePolicyMaxCompat
};
enum RtcpMuxPolicy {
kRtcpMuxPolicyNegotiate,
kRtcpMuxPolicyRequire,
};
enum TcpCandidatePolicy {
kTcpCandidatePolicyEnabled,
kTcpCandidatePolicyDisabled
};
enum CandidateNetworkPolicy {
kCandidateNetworkPolicyAll,
kCandidateNetworkPolicyLowCost
};
enum ContinualGatheringPolicy { GATHER_ONCE, GATHER_CONTINUALLY };
struct PortAllocatorConfig {
// For min_port and max_port, 0 means not specified.
int min_port = 0;
int max_port = 0;
uint32_t flags = 0; // Same as kDefaultPortAllocatorFlags.
};
enum class RTCConfigurationType {
// A configuration that is safer to use, despite not having the best
// performance. Currently this is the default configuration.
kSafe,
// An aggressive configuration that has better performance, although it
// may be riskier and may need extra support in the application.
kAggressive
};
// TODO(hbos): Change into class with private data and public getters.
// TODO(nisse): In particular, accessing fields directly from an
// application is brittle, since the organization mirrors the
// organization of the implementation, which isn't stable. So we
// need getters and setters at least for fields which applications
// are interested in.
struct RTC_EXPORT RTCConfiguration {
// This struct is subject to reorganization, both for naming
// consistency, and to group settings to match where they are used
// in the implementation. To do that, we need getter and setter
// methods for all settings which are of interest to applications,
// Chrome in particular.
RTCConfiguration();
RTCConfiguration(const RTCConfiguration&);
explicit RTCConfiguration(RTCConfigurationType type);
~RTCConfiguration();
bool operator==(const RTCConfiguration& o) const;
bool operator!=(const RTCConfiguration& o) const;
bool dscp() const { return media_config.enable_dscp; }
void set_dscp(bool enable) { media_config.enable_dscp = enable; }
bool cpu_adaptation() const {
return media_config.video.enable_cpu_adaptation;
}
void set_cpu_adaptation(bool enable) {
media_config.video.enable_cpu_adaptation = enable;
}
bool suspend_below_min_bitrate() const {
return media_config.video.suspend_below_min_bitrate;
}
void set_suspend_below_min_bitrate(bool enable) {
media_config.video.suspend_below_min_bitrate = enable;
}
bool prerenderer_smoothing() const {
return media_config.video.enable_prerenderer_smoothing;
}
void set_prerenderer_smoothing(bool enable) {
media_config.video.enable_prerenderer_smoothing = enable;
}
bool experiment_cpu_load_estimator() const {
return media_config.video.experiment_cpu_load_estimator;
}
void set_experiment_cpu_load_estimator(bool enable) {
media_config.video.experiment_cpu_load_estimator = enable;
}
int audio_rtcp_report_interval_ms() const {
return media_config.audio.rtcp_report_interval_ms;
}
void set_audio_rtcp_report_interval_ms(int audio_rtcp_report_interval_ms) {
media_config.audio.rtcp_report_interval_ms =
audio_rtcp_report_interval_ms;
}
int video_rtcp_report_interval_ms() const {
return media_config.video.rtcp_report_interval_ms;
}
void set_video_rtcp_report_interval_ms(int video_rtcp_report_interval_ms) {
media_config.video.rtcp_report_interval_ms =
video_rtcp_report_interval_ms;
}
// Settings for the port allcoator. Applied only if the port allocator is
// created by PeerConnectionFactory, not if it is injected with
// PeerConnectionDependencies
int min_port() const { return port_allocator_config.min_port; }
void set_min_port(int port) { port_allocator_config.min_port = port; }
int max_port() const { return port_allocator_config.max_port; }
void set_max_port(int port) { port_allocator_config.max_port = port; }
uint32_t port_allocator_flags() { return port_allocator_config.flags; }
void set_port_allocator_flags(uint32_t flags) {
port_allocator_config.flags = flags;
}
static const int kUndefined = -1;
// Default maximum number of packets in the audio jitter buffer.
static const int kAudioJitterBufferMaxPackets = 200;
// ICE connection receiving timeout for aggressive configuration.
static const int kAggressiveIceConnectionReceivingTimeout = 1000;
////////////////////////////////////////////////////////////////////////
// The below few fields mirror the standard RTCConfiguration dictionary:
////////////////////////////////////////////////////////////////////////
// TODO(pthatcher): Rename this ice_servers, but update Chromium
// at the same time.
IceServers servers;
// TODO(pthatcher): Rename this ice_transport_type, but update
// Chromium at the same time.
IceTransportsType type = kAll;
BundlePolicy bundle_policy = kBundlePolicyBalanced;
RtcpMuxPolicy rtcp_mux_policy = kRtcpMuxPolicyRequire;
std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
int ice_candidate_pool_size = 0;
//////////////////////////////////////////////////////////////////////////
// The below fields correspond to constraints from the deprecated
// constraints interface for constructing a PeerConnection.
//
// absl::optional fields can be "missing", in which case the implementation
// default will be used.
//////////////////////////////////////////////////////////////////////////
// If set to true, don't gather IPv6 ICE candidates on Wi-Fi.
// Only intended to be used on specific devices. Certain phones disable IPv6
// when the screen is turned off and it would be better to just disable the
// IPv6 ICE candidates on Wi-Fi in those cases.
bool disable_ipv6_on_wifi = false;
// By default, the PeerConnection will use a limited number of IPv6 network
// interfaces, in order to avoid too many ICE candidate pairs being created
// and delaying ICE completion.
//
// Can be set to INT_MAX to effectively disable the limit.
int max_ipv6_networks = cricket::kDefaultMaxIPv6Networks;
// Exclude link-local network interfaces
// from consideration for gathering ICE candidates.
bool disable_link_local_networks = false;
// Minimum bitrate at which screencast video tracks will be encoded at.
// This means adding padding bits up to this bitrate, which can help
// when switching from a static scene to one with motion.
absl::optional<int> screencast_min_bitrate;
/////////////////////////////////////////////////
// The below fields are not part of the standard.
/////////////////////////////////////////////////
// Can be used to disable TCP candidate generation.
TcpCandidatePolicy tcp_candidate_policy = kTcpCandidatePolicyEnabled;
// Can be used to avoid gathering candidates for a "higher cost" network,
// if a lower cost one exists. For example, if both Wi-Fi and cellular
// interfaces are available, this could be used to avoid using the cellular
// interface.
CandidateNetworkPolicy candidate_network_policy =
kCandidateNetworkPolicyAll;
// The maximum number of packets that can be stored in the NetEq audio
// jitter buffer. Can be reduced to lower tolerated audio latency.
int audio_jitter_buffer_max_packets = kAudioJitterBufferMaxPackets;
// Whether to use the NetEq "fast mode" which will accelerate audio quicker
// if it falls behind.
bool audio_jitter_buffer_fast_accelerate = false;
// The minimum delay in milliseconds for the audio jitter buffer.
int audio_jitter_buffer_min_delay_ms = 0;
// Timeout in milliseconds before an ICE candidate pair is considered to be
// "not receiving", after which a lower priority candidate pair may be
// selected.
int ice_connection_receiving_timeout = kUndefined;
// Interval in milliseconds at which an ICE "backup" candidate pair will be
// pinged. This is a candidate pair which is not actively in use, but may
// be switched to if the active candidate pair becomes unusable.
//
// This is relevant mainly to Wi-Fi/cell handoff; the application may not
// want this backup cellular candidate pair pinged frequently, since it
// consumes data/battery.
int ice_backup_candidate_pair_ping_interval = kUndefined;
// Can be used to enable continual gathering, which means new candidates
// will be gathered as network interfaces change. Note that if continual
// gathering is used, the candidate removal API should also be used, to
// avoid an ever-growing list of candidates.
ContinualGatheringPolicy continual_gathering_policy = GATHER_ONCE;
// If set to true, candidate pairs will be pinged in order of most likely
// to work (which means using a TURN server, generally), rather than in
// standard priority order.
bool prioritize_most_likely_ice_candidate_pairs = false;
// Implementation defined settings. A public member only for the benefit of
// the implementation. Applications must not access it directly, and should
// instead use provided accessor methods, e.g., set_cpu_adaptation.
struct cricket::MediaConfig media_config;
// If set to true, only one preferred TURN allocation will be used per
// network interface. UDP is preferred over TCP and IPv6 over IPv4. This
// can be used to cut down on the number of candidate pairings.
// Deprecated. TODO(webrtc:11026) Remove this flag once the downstream
// dependency is removed.
bool prune_turn_ports = false;
// The policy used to prune turn port.
PortPrunePolicy turn_port_prune_policy = NO_PRUNE;
PortPrunePolicy GetTurnPortPrunePolicy() const {
return prune_turn_ports ? PRUNE_BASED_ON_PRIORITY
: turn_port_prune_policy;
}
// If set to true, this means the ICE transport should presume TURN-to-TURN
// candidate pairs will succeed, even before a binding response is received.
// This can be used to optimize the initial connection time, since the DTLS
// handshake can begin immediately.
bool presume_writable_when_fully_relayed = false;
// If true, "renomination" will be added to the ice options in the transport
// description.
bool enable_ice_renomination = false;
// If true, the ICE role is re-determined when the PeerConnection sets a
// local transport description that indicates an ICE restart.
//
// This is standard RFC5245 ICE behavior, but causes unnecessary role
// thrashing, so an application may wish to avoid it. This role
// re-determining was removed in ICEbis (ICE v2).
bool redetermine_role_on_ice_restart = true;
// This flag is only effective when `continual_gathering_policy` is
// GATHER_CONTINUALLY.
//
// If true, after the ICE transport type is changed such that new types of
// ICE candidates are allowed by the new transport type, e.g. from
// IceTransportsType::kRelay to IceTransportsType::kAll, candidates that
// have been gathered by the ICE transport but not matching the previous
// transport type and as a result not observed by PeerConnectionObserver,
// will be surfaced to the observer.
bool surface_ice_candidates_on_ice_transport_type_changed = false;
// The following fields define intervals in milliseconds at which ICE
// connectivity checks are sent.
//
// We consider ICE is "strongly connected" for an agent when there is at
// least one candidate pair that currently succeeds in connectivity check
// from its direction i.e. sending a STUN ping and receives a STUN ping
// response, AND all candidate pairs have sent a minimum number of pings for
// connectivity (this number is implementation-specific). Otherwise, ICE is
// considered in "weak connectivity".
//
// Note that the above notion of strong and weak connectivity is not defined
// in RFC 5245, and they apply to our current ICE implementation only.
//
// 1) ice_check_interval_strong_connectivity defines the interval applied to
// ALL candidate pairs when ICE is strongly connected, and it overrides the
// default value of this interval in the ICE implementation;
// 2) ice_check_interval_weak_connectivity defines the counterpart for ALL
// pairs when ICE is weakly connected, and it overrides the default value of
// this interval in the ICE implementation;
// 3) ice_check_min_interval defines the minimal interval (equivalently the
// maximum rate) that overrides the above two intervals when either of them
// is less.
absl::optional<int> ice_check_interval_strong_connectivity;
absl::optional<int> ice_check_interval_weak_connectivity;
absl::optional<int> ice_check_min_interval;
// The min time period for which a candidate pair must wait for response to
// connectivity checks before it becomes unwritable. This parameter
// overrides the default value in the ICE implementation if set.
absl::optional<int> ice_unwritable_timeout;
// The min number of connectivity checks that a candidate pair must sent
// without receiving response before it becomes unwritable. This parameter
// overrides the default value in the ICE implementation if set.
absl::optional<int> ice_unwritable_min_checks;
// The min time period for which a candidate pair must wait for response to
// connectivity checks it becomes inactive. This parameter overrides the
// default value in the ICE implementation if set.
absl::optional<int> ice_inactive_timeout;
// The interval in milliseconds at which STUN candidates will resend STUN
// binding requests to keep NAT bindings open.
absl::optional<int> stun_candidate_keepalive_interval;
// Optional TurnCustomizer.
// With this class one can modify outgoing TURN messages.
// The object passed in must remain valid until PeerConnection::Close() is
// called.
webrtc::TurnCustomizer* turn_customizer = nullptr;
// Preferred network interface.
// A candidate pair on a preferred network has a higher precedence in ICE
// than one on an un-preferred network, regardless of priority or network
// cost.
absl::optional<rtc::AdapterType> network_preference;
// Configure the SDP semantics used by this PeerConnection. By default, this
// is Unified Plan which is compliant to the WebRTC 1.0 specification. It is
// possible to overrwite this to the deprecated Plan B SDP format, but note
// that kPlanB will be deleted at some future date, see
//
// kUnifiedPlan will cause the PeerConnection to create offers and answers
// with multiple m= sections where each m= section maps to one RtpSender and
// one RtpReceiver (an RtpTransceiver), either both audio or both video.
// This will also cause the PeerConnection to ignore all but the first
// a=ssrc lines that form a Plan B streams (if the PeerConnection is given
// Plan B SDP to process).
//
// kPlanB will cause the PeerConnection to create offers and answers with at
// most one audio and one video m= section with multiple RtpSenders and
// RtpReceivers specified as multiple a=ssrc lines within the section. This
// will also cause PeerConnection to ignore all but the first m= section of
// the same media type (if the PeerConnection is given Unified Plan SDP to
// process).
SdpSemantics sdp_semantics = SdpSemantics::kUnifiedPlan;
// TODO(bugs.webrtc.org/9891) - Move to crypto_options or remove.
// Actively reset the SRTP parameters whenever the DTLS transports
// underneath are reset for every offer/answer negotiation.
// This is only intended to be a workaround for crbug.com/835958
// WARNING: This would cause RTP/RTCP packets decryption failure if not used
// correctly. This flag will be deprecated soon. Do not rely on it.
bool active_reset_srtp_params = false;
// Defines advanced optional cryptographic settings related to SRTP and
// frame encryption for native WebRTC. Setting this will overwrite any
// settings set in PeerConnectionFactory (which is deprecated).
absl::optional<CryptoOptions> crypto_options;
// Configure if we should include the SDP attribute extmap-allow-mixed in
// our offer on session level.
bool offer_extmap_allow_mixed = true;
// TURN logging identifier.
// This identifier is added to a TURN allocation
// and it intended to be used to be able to match client side
// logs with TURN server logs. It will not be added if it's an empty string.
std::string turn_logging_id;
// Added to be able to control rollout of this feature.
bool enable_implicit_rollback = false;
// The delay before doing a usage histogram report for long-lived
// PeerConnections. Used for testing only.
absl::optional<int> report_usage_pattern_delay_ms;
// The ping interval (ms) when the connection is stable and writable. This
// parameter overrides the default value in the ICE implementation if set.
absl::optional<int> stable_writable_connection_ping_interval_ms;
// Whether this PeerConnection will avoid VPNs (kAvoidVpn), prefer VPNs
// (kPreferVpn), only work over VPN (kOnlyUseVpn) or only work over non-VPN
// (kNeverUseVpn) interfaces. This controls which local interfaces the
// PeerConnection will prefer to connect over. Since VPN detection is not
// perfect, adherence to this preference cannot be guaranteed.
VpnPreference vpn_preference = VpnPreference::kDefault;
// List of address/length subnets that should be treated like
// VPN (in case webrtc fails to auto detect them).
std::vector<rtc::NetworkMask> vpn_list;
PortAllocatorConfig port_allocator_config;
// The burst interval of the pacer, see TaskQueuePacedSender constructor.
absl::optional<TimeDelta> pacer_burst_interval;
//
// Don't forget to update operator== if adding something.
//
};
struct RTCOfferAnswerOptions {
static const int kUndefined = -1;
static const int kMaxOfferToReceiveMedia = 1;
// The default value for constraint offerToReceiveX:true.
static const int kOfferToReceiveMediaTrue = 1;
// These options are left as backwards compatibility for clients who need
// "Plan B" semantics. Clients who have switched to "Unified Plan" semantics
// should use the RtpTransceiver API (AddTransceiver) instead.
//
// offer_to_receive_X set to 1 will cause a media description to be
// generated in the offer, even if no tracks of that type have been added.
// Values greater than 1 are treated the same.
//
// If set to 0, the generated directional attribute will not include the
// "recv" direction (meaning it will be "sendonly" or "inactive".
int offer_to_receive_video = kUndefined;
int offer_to_receive_audio = kUndefined;
bool voice_activity_detection = true;
bool ice_restart = false;
// If true, will offer to BUNDLE audio/video/data together. Not to be
// confused with RTCP mux (multiplexing RTP and RTCP together).
bool use_rtp_mux = true;
// If true, "a=packetization:<payload_type> raw" attribute will be offered
// in the SDP for all video payload and accepted in the answer if offered.
bool raw_packetization_for_video = false;
// This will apply to all video tracks with a Plan B SDP offer/answer.
int num_simulcast_layers = 1;
// If true: Use SDP format from draft-ietf-mmusic-scdp-sdp-03
// If false: Use SDP format from draft-ietf-mmusic-sdp-sdp-26 or later
bool use_obsolete_sctp_sdp = false;
RTCOfferAnswerOptions() = default;
RTCOfferAnswerOptions(int offer_to_receive_video,
int offer_to_receive_audio,
bool voice_activity_detection,
bool ice_restart,
bool use_rtp_mux)
: offer_to_receive_video(offer_to_receive_video),
offer_to_receive_audio(offer_to_receive_audio),
voice_activity_detection(voice_activity_detection),
ice_restart(ice_restart),
use_rtp_mux(use_rtp_mux) {}
};
// Used by GetStats to decide which stats to include in the stats reports.
// `kStatsOutputLevelStandard` includes the standard stats for Javascript API;
// `kStatsOutputLevelDebug` includes both the standard stats and additional
// stats for debugging purposes.
enum StatsOutputLevel {
kStatsOutputLevelStandard,
kStatsOutputLevelDebug,
};
// Accessor methods to active local streams.
// This method is not supported with kUnifiedPlan semantics. Please use
// GetSenders() instead.
virtual rtc::scoped_refptr<StreamCollectionInterface> local_streams() = 0;
// Accessor methods to remote streams.
// This method is not supported with kUnifiedPlan semantics. Please use
// GetReceivers() instead.
virtual rtc::scoped_refptr<StreamCollectionInterface> remote_streams() = 0;
// Add a new MediaStream to be sent on this PeerConnection.
// Note that a SessionDescription negotiation is needed before the
// remote peer can receive the stream.
//
// This has been removed from the standard in favor of a track-based API. So,
// this is equivalent to simply calling AddTrack for each track within the
// stream, with the one difference that if "stream->AddTrack(...)" is called
// later, the PeerConnection will automatically pick up the new track. Though
// this functionality will be deprecated in the future.
//
// This method is not supported with kUnifiedPlan semantics. Please use
// AddTrack instead.
virtual bool AddStream(MediaStreamInterface* stream) = 0;
// Remove a MediaStream from this PeerConnection.
// Note that a SessionDescription negotiation is needed before the
// remote peer is notified.
//
// This method is not supported with kUnifiedPlan semantics. Please use
// RemoveTrack instead.
virtual void RemoveStream(MediaStreamInterface* stream) = 0;
// Add a new MediaStreamTrack to be sent on this PeerConnection, and return
// the newly created RtpSender. The RtpSender will be associated with the
// streams specified in the `stream_ids` list.
//
// Errors:
// - INVALID_PARAMETER: `track` is null, has a kind other than audio or video,
// or a sender already exists for the track.
// - INVALID_STATE: The PeerConnection is closed.
virtual RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) = 0;
// Add a new MediaStreamTrack as above, but with an additional parameter,
// `init_send_encodings` : initial RtpEncodingParameters for RtpSender,
// similar to init_send_encodings in RtpTransceiverInit.
// Note that a new transceiver will always be created.
//
virtual RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids,
const std::vector<RtpEncodingParameters>& init_send_encodings) = 0;
// Removes the connection between a MediaStreamTrack and the PeerConnection.
// Stops sending on the RtpSender and marks the
// corresponding RtpTransceiver direction as no longer sending.
//
// Errors:
// - INVALID_PARAMETER: `sender` is null or (Plan B only) the sender is not
// associated with this PeerConnection.
// - INVALID_STATE: PeerConnection is closed.
//
// Plan B semantics: Removes the RtpSender from this PeerConnection.
//
// TODO(bugs.webrtc.org/9534): Rename to RemoveTrack once the other signature
// is removed; remove default implementation once upstream is updated.
virtual RTCError RemoveTrackOrError(
rtc::scoped_refptr<RtpSenderInterface> sender) {
RTC_CHECK_NOTREACHED();
return RTCError();
}
// AddTransceiver creates a new RtpTransceiver and adds it to the set of
// transceivers. Adding a transceiver will cause future calls to CreateOffer
// to add a media description for the corresponding transceiver.
//
// The initial value of `mid` in the returned transceiver is null. Setting a
// new session description may change it to a non-null value.
//
//
// Optionally, an RtpTransceiverInit structure can be specified to configure
// the transceiver from construction. If not specified, the transceiver will
// default to having a direction of kSendRecv and not be part of any streams.
//
// These methods are only available when Unified Plan is enabled (see
// RTCConfiguration).
//
// Common errors:
// - INTERNAL_ERROR: The configuration does not have Unified Plan enabled.
// Adds a transceiver with a sender set to transmit the given track. The kind
// of the transceiver (and sender/receiver) will be derived from the kind of
// the track.
// Errors:
// - INVALID_PARAMETER: `track` is null.
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track) = 0;
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init) = 0;
// Adds a transceiver with the given kind. Can either be MEDIA_TYPE_AUDIO or
// MEDIA_TYPE_VIDEO.
// Errors:
// - INVALID_PARAMETER: `media_type` is not MEDIA_TYPE_AUDIO or
// MEDIA_TYPE_VIDEO.
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(cricket::MediaType media_type) = 0;
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(cricket::MediaType media_type,
const RtpTransceiverInit& init) = 0;
// Creates a sender without a track. Can be used for "early media"/"warmup"
// use cases, where the application may want to negotiate video attributes
// before a track is available to send.
//
// The standard way to do this would be through "addTransceiver", but we
// don't support that API yet.
//
// `kind` must be "audio" or "video".
//
// `stream_id` is used to populate the msid attribute; if empty, one will
// be generated automatically.
//
// This method is not supported with kUnifiedPlan semantics. Please use
// AddTransceiver instead.
virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
const std::string& kind,
const std::string& stream_id) = 0;
// If Plan B semantics are specified, gets all RtpSenders, created either
// through AddStream, AddTrack, or CreateSender. All senders of a specific
// media type share the same media description.
//
// If Unified Plan semantics are specified, gets the RtpSender for each
// RtpTransceiver.
virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const = 0;
// If Plan B semantics are specified, gets all RtpReceivers created when a
// remote description is applied. All receivers of a specific media type share
// the same media description. It is also possible to have a media description
// with no associated RtpReceivers, if the directional attribute does not
// indicate that the remote peer is sending any media.
//
// If Unified Plan semantics are specified, gets the RtpReceiver for each
// RtpTransceiver.
virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
const = 0;
// Get all RtpTransceivers, created either through AddTransceiver, AddTrack or
// by a remote description applied with SetRemoteDescription.
//
// Note: This method is only available when Unified Plan is enabled (see
// RTCConfiguration).
virtual std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
GetTransceivers() const = 0;
// The legacy non-compliant GetStats() API. This correspond to the
// callback-based version of getStats() in JavaScript. The returned metrics
// are UNDOCUMENTED and many of them rely on implementation-specific details.
// The goal is to DELETE THIS VERSION but we can't today because it is heavily
//
// This version is wired up into Chrome. Any stats implemented are
// automatically exposed to the Web Platform. This has BYPASSED the Chrome
// release processes for years and lead to cross-browser incompatibility
// issues and web application reliance on Chrome-only behavior.
//
// This API is in "maintenance mode", serious regressions should be fixed but
// adding new stats is highly discouraged.
//
// TODO(hbos): Deprecate and remove this when third parties have migrated to
virtual bool GetStats(StatsObserver* observer,
MediaStreamTrackInterface* track, // Optional
StatsOutputLevel level) = 0;
// The spec-compliant GetStats() API. This correspond to the promise-based
// version of getStats() in JavaScript. Implementation status is described in
// api/stats/rtcstats_objects.h. For more details on stats, see spec:
// TODO(hbos): Takes shared ownership, use rtc::scoped_refptr<> instead. This
// requires stop overriding the current version in third party or making third
// party calls explicit to avoid ambiguity during switch. Make the future
// version abstract as soon as third party projects implement it.
virtual void GetStats(RTCStatsCollectorCallback* callback) = 0;
// Spec-compliant getStats() performing the stats selection algorithm with the
virtual void GetStats(
rtc::scoped_refptr<RtpSenderInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) = 0;
// Spec-compliant getStats() performing the stats selection algorithm with the
virtual void GetStats(
rtc::scoped_refptr<RtpReceiverInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) = 0;
// Clear cached stats in the RTCStatsCollector.
virtual void ClearStatsCache() {}
// Create a data channel with the provided config, or default config if none
// is provided. Note that an offer/answer negotiation is still necessary
// before the data channel can be used.
//
// Also, calling CreateDataChannel is the only way to get a data "m=" section
// in SDP, so it should be done before CreateOffer is called, if the
// application plans to use data channels.
virtual RTCErrorOr<rtc::scoped_refptr<DataChannelInterface>>
CreateDataChannelOrError(const std::string& label,
const DataChannelInit* config) {
return RTCError(RTCErrorType::INTERNAL_ERROR, "dummy function called");
}
// TODO(crbug.com/788659): Remove "virtual" below and default implementation
// above once mock in Chrome is fixed.
ABSL_DEPRECATED("Use CreateDataChannelOrError")
virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
const std::string& label,
const DataChannelInit* config) {
auto result = CreateDataChannelOrError(label, config);
if (!result.ok()) {
return nullptr;
} else {
return result.MoveValue();
}
}
// NOTE: For the following 6 methods, it's only safe to dereference the
// SessionDescriptionInterface on signaling_thread() (for example, calling
// ToString).
// Returns the more recently applied description; "pending" if it exists, and
// otherwise "current". See below.
virtual const SessionDescriptionInterface* local_description() const = 0;
virtual const SessionDescriptionInterface* remote_description() const = 0;
// A "current" description the one currently negotiated from a complete
// offer/answer exchange.
virtual const SessionDescriptionInterface* current_local_description()
const = 0;
virtual const SessionDescriptionInterface* current_remote_description()
const = 0;
// A "pending" description is one that's part of an incomplete offer/answer
// exchange (thus, either an offer or a pranswer). Once the offer/answer
// exchange is finished, the "pending" description will become "current".
virtual const SessionDescriptionInterface* pending_local_description()
const = 0;
virtual const SessionDescriptionInterface* pending_remote_description()
const = 0;
// Tells the PeerConnection that ICE should be restarted. This triggers a need
// for negotiation and subsequent CreateOffer() calls will act as if
// RTCOfferAnswerOptions::ice_restart is true.
virtual void RestartIce() = 0;
// Create a new offer.
// The CreateSessionDescriptionObserver callback will be called when done.
virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) = 0;
// Create an answer to an offer.
// The CreateSessionDescriptionObserver callback will be called when done.
virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) = 0;
// Sets the local session description.
//
// According to spec, the local session description MUST be the same as was
// returned by CreateOffer() or CreateAnswer() or else the operation should
// fail. Our implementation however allows some amount of "SDP munging", but
// please note that this is HIGHLY DISCOURAGED. If you do not intent to munge
// SDP, the method below that doesn't take `desc` as an argument will create
// the offer or answer for you.
//
// The observer is invoked as soon as the operation completes, which could be
// before or after the SetLocalDescription() method has exited.
virtual void SetLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {}
// Creates an offer or answer (depending on current signaling state) and sets
// it as the local session description.
//
// The observer is invoked as soon as the operation completes, which could be
// before or after the SetLocalDescription() method has exited.
virtual void SetLocalDescription(
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {}
// Like SetLocalDescription() above, but the observer is invoked with a delay
// after the operation completes. This helps avoid recursive calls by the
// observer but also makes it possible for states to change in-between the
// operation completing and the observer getting called. This makes them racy
// for synchronizing peer connection states to the application.
// ones taking SetLocalDescriptionObserverInterface as argument.
virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) = 0;
virtual void SetLocalDescription(SetSessionDescriptionObserver* observer) {}
// Sets the remote session description.
//
// (Unlike "SDP munging" before SetLocalDescription(), modifying a remote
// offer or answer is allowed by the spec.)
//
// The observer is invoked as soon as the operation completes, which could be
// before or after the SetRemoteDescription() method has exited.
virtual void SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) = 0;
// Like SetRemoteDescription() above, but the observer is invoked with a delay
// after the operation completes. This helps avoid recursive calls by the
// observer but also makes it possible for states to change in-between the
// operation completing and the observer getting called. This makes them racy
// for synchronizing peer connection states to the application.
// ones taking SetRemoteDescriptionObserverInterface as argument.
virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {}
// According to spec, we must only fire "negotiationneeded" if the Operations
// Chain is empty. This method takes care of validating an event previously
// generated with PeerConnectionObserver::OnNegotiationNeededEvent() to make
// sure that even if there was a delay (e.g. due to a PostTask) between the
// event being generated and the time of firing, the Operations Chain is empty
// and the event is still valid to be fired.
virtual bool ShouldFireNegotiationNeededEvent(uint32_t event_id) = 0;
virtual PeerConnectionInterface::RTCConfiguration GetConfiguration() = 0;
// Sets the PeerConnection's global configuration to `config`.
//
// The members of `config` that may be changed are `type`, `servers`,
// `ice_candidate_pool_size` and `prune_turn_ports` (though the candidate
// pool size can't be changed after the first call to SetLocalDescription).
// Note that this means the BUNDLE and RTCP-multiplexing policies cannot be
// changed with this method.
//
// Any changes to STUN/TURN servers or ICE candidate policy will affect the
// next gathering phase, and cause the next call to createOffer to generate
// new ICE credentials, as described in JSEP. This also occurs when
// `prune_turn_ports` changes, for the same reasoning.
//
// If an error occurs, returns false and populates `error` if non-null:
// - INVALID_MODIFICATION if `config` contains a modified parameter other
// than one of the parameters listed above.
// - INVALID_RANGE if `ice_candidate_pool_size` is out of range.
// - SYNTAX_ERROR if parsing an ICE server URL failed.
// - INVALID_PARAMETER if a TURN server is missing `username` or `password`.
// - INTERNAL_ERROR if an unexpected error occurred.
virtual RTCError SetConfiguration(
const PeerConnectionInterface::RTCConfiguration& config) = 0;
// Provides a remote candidate to the ICE Agent.
// A copy of the `candidate` will be created and added to the remote
// description. So the caller of this method still has the ownership of the
// `candidate`.
// TODO(hbos): The spec mandates chaining this operation onto the operations
// chain; deprecate and remove this version in favor of the callback-based
// signature.
virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
// TODO(hbos): Remove default implementation once implemented by downstream
// projects.
virtual void AddIceCandidate(std::unique_ptr<IceCandidateInterface> candidate,
std::function<void(RTCError)> callback) {}
// Removes a group of remote candidates from the ICE agent. Needed mainly for
// continual gathering, to avoid an ever-growing list of candidates as
// networks come and go. Note that the candidates' transport_name must be set
// to the MID of the m= section that generated the candidate.
// TODO(bugs.webrtc.org/8395): Use IceCandidateInterface instead of
// cricket::Candidate, which would avoid the transport_name oddity.
virtual bool RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) = 0;
// SetBitrate limits the bandwidth allocated for all RTP streams sent by
// this PeerConnection. Other limitations might affect these limits and
// are respected (for example "b=AS" in SDP).
//
// Setting `current_bitrate_bps` will reset the current bitrate estimate
// to the provided value.
virtual RTCError SetBitrate(const BitrateSettings& bitrate) = 0;
// Allows an application to reconfigure bandwidth estimation.
// The method can be called both before and after estimation has started.
// Estimation starts when the first RTP packet is sent.
// Estimation will be restarted if already started.
virtual void ReconfigureBandwidthEstimation(
const BandwidthEstimationSettings& settings) = 0;
// Enable/disable playout of received audio streams. Enabled by default. Note
// that even if playout is enabled, streams will only be played out if the
// appropriate SDP is also applied. Setting `playout` to false will stop
// playout of the underlying audio device but starts a task which will poll
// for audio data every 10ms to ensure that audio processing happens and the
// audio statistics are updated.
virtual void SetAudioPlayout(bool playout) = 0;
// Enable/disable recording of transmitted audio streams. Enabled by default.
// Note that even if recording is enabled, streams will only be recorded if
// the appropriate SDP is also applied.
virtual void SetAudioRecording(bool recording) = 0;
// Looks up the DtlsTransport associated with a MID value.
// In the Javascript API, DtlsTransport is a property of a sender, but
// because the PeerConnection owns the DtlsTransport in this implementation,
// it is better to look them up on the PeerConnection.
virtual rtc::scoped_refptr<DtlsTransportInterface> LookupDtlsTransportByMid(
const std::string& mid) = 0;
// Returns the SCTP transport, if any.
virtual rtc::scoped_refptr<SctpTransportInterface> GetSctpTransport()
const = 0;
// Returns the current SignalingState.
virtual SignalingState signaling_state() = 0;
// Returns an aggregate state of all ICE *and* DTLS transports.
// This is left in place to avoid breaking native clients who expect our old,
// nonstandard behavior.
// TODO(jonasolsson): deprecate and remove this.
virtual IceConnectionState ice_connection_state() = 0;
// Returns an aggregated state of all ICE transports.
virtual IceConnectionState standardized_ice_connection_state() = 0;
// Returns an aggregated state of all ICE and DTLS transports.
virtual PeerConnectionState peer_connection_state() = 0;
virtual IceGatheringState ice_gathering_state() = 0;
// Returns the current state of canTrickleIceCandidates per
virtual absl::optional<bool> can_trickle_ice_candidates() = 0;
// When a resource is overused, the PeerConnection will try to reduce the load
// on the sysem, for example by reducing the resolution or frame rate of
// encoded streams. The Resource API allows injecting platform-specific usage
// measurements. The conditions to trigger kOveruse or kUnderuse are up to the
// implementation.
virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) = 0;
// Start RtcEventLog using an existing output-sink. Takes ownership of
// `output` and passes it on to Call, which will take the ownership. If
// the operation fails the output will be closed and deallocated. The
// event log will send serialized events to the output object every
// `output_period_ms`. Applications using the event log should generally
// make their own trade-off regarding the output period. A long period is
// generally more efficient, with potential drawbacks being more bursty
// thread usage, and more events lost in case the application crashes. If
// the `output_period_ms` argument is omitted, webrtc selects a default
// deemed to be workable in most cases.
virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) = 0;
virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output) = 0;
// Stops logging the RtcEventLog.
virtual void StopRtcEventLog() = 0;
// Terminates all media, closes the transports, and in general releases any
// resources used by the PeerConnection. This is an irreversible operation.
//
// Note that after this method completes, the PeerConnection will no longer
// use the PeerConnectionObserver interface passed in on construction, and
// thus the observer object can be safely destroyed.
virtual void Close() = 0;
// The thread on which all PeerConnectionObserver callbacks will be invoked,
// as well as callbacks for other classes such as DataChannelObserver.
//
// Also the only thread on which it's safe to use SessionDescriptionInterface
// pointers.
virtual rtc::Thread* signaling_thread() const = 0;
// NetworkController instance being used by this PeerConnection, to be used
// to identify instances when using a custom NetworkControllerFactory.
virtual NetworkControllerInterface* GetNetworkController() = 0;
protected:
// Dtor protected as objects shouldn't be deleted via this interface.
~PeerConnectionInterface() override = default;
};
// PeerConnection callback interface, used for RTCPeerConnection events.
// Application should implement these methods.
class PeerConnectionObserver {
public:
virtual ~PeerConnectionObserver() = default;
// Triggered when the SignalingState changed.
virtual void OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) = 0;
// Triggered when media is received on a new stream from remote peer.
virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) {}
// Triggered when a remote peer closes a stream.
virtual void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) {
}
// Triggered when a remote peer opens a data channel.
virtual void OnDataChannel(
rtc::scoped_refptr<DataChannelInterface> data_channel) = 0;
// Triggered when renegotiation is needed. For example, an ICE restart
// has begun.
// TODO(hbos): Delete in favor of OnNegotiationNeededEvent() when downstream
// projects have migrated.
virtual void OnRenegotiationNeeded() {}
// Used to fire spec-compliant onnegotiationneeded events, which should only
// fire when the Operations Chain is empty. The observer is responsible for
// queuing a task (e.g. Chromium: jump to main thread) to maybe fire the
// event. The event identified using `event_id` must only fire if
// PeerConnection::ShouldFireNegotiationNeededEvent() returns true since it is
// possible for the event to become invalidated by operations subsequently
// chained.
virtual void OnNegotiationNeededEvent(uint32_t event_id) {}
// Called any time the legacy IceConnectionState changes.
//
// Note that our ICE states lag behind the standard slightly. The most
// notable differences include the fact that "failed" occurs after 15
// seconds, not 30, and this actually represents a combination ICE + DTLS
// state, so it may be "failed" if DTLS fails while ICE succeeds.
//
// TODO(jonasolsson): deprecate and remove this.
virtual void OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) {}
// Called any time the standards-compliant IceConnectionState changes.
virtual void OnStandardizedIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) {}
// Called any time the PeerConnectionState changes.
virtual void OnConnectionChange(
PeerConnectionInterface::PeerConnectionState new_state) {}
// Called any time the IceGatheringState changes.
virtual void OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) = 0;
// A new ICE candidate has been gathered.
virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
// Gathering of an ICE candidate failed.
virtual void OnIceCandidateError(const std::string& address,
int port,
const std::string& url,
int error_code,
const std::string& error_text) {}
// Ice candidates have been removed.
// TODO(honghaiz): Make this a pure virtual method when all its subclasses
// implement it.
virtual void OnIceCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {}
// Called when the ICE connection receiving status changes.
virtual void OnIceConnectionReceivingChange(bool receiving) {}
// Called when the selected candidate pair for the ICE connection changes.
virtual void OnIceSelectedCandidatePairChanged(
const cricket::CandidatePairChangeEvent& event) {}
// This is called when a receiver and its track are created.
// TODO(zhihuang): Make this pure virtual when all subclasses implement it.
// Note: This is called with both Plan B and Unified Plan semantics. Unified
// Plan users should prefer OnTrack, OnAddTrack is only called as backwards
// compatibility (and is called in the exact same situations as OnTrack).
virtual void OnAddTrack(
rtc::scoped_refptr<RtpReceiverInterface> receiver,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {}
// This is called when signaling indicates a transceiver will be receiving
// media from the remote endpoint. This is fired during a call to
// SetRemoteDescription. The receiving track can be accessed by:
// `transceiver->receiver()->track()` and its associated streams by
// `transceiver->receiver()->streams()`.
// Note: This will only be called if Unified Plan semantics are specified.
// This behavior is specified in section 2.2.8.2.5 of the "Set the
// RTCSessionDescription" algorithm:
virtual void OnTrack(
rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {}
// Called when signaling indicates that media will no longer be received on a
// track.
// With Plan B semantics, the given receiver will have been removed from the
// PeerConnection and the track muted.
// With Unified Plan semantics, the receiver will remain but the transceiver
// will have changed direction to either sendonly or inactive.
// TODO(hbos,deadbeef): Make pure virtual when all subclasses implement it.
virtual void OnRemoveTrack(
rtc::scoped_refptr<RtpReceiverInterface> receiver) {}
// Called when an interesting usage is detected by WebRTC.
// An appropriate action is to add information about the context of the
// PeerConnection and write the event to some kind of "interesting events"
// log function.
// The heuristics for defining what constitutes "interesting" are
// implementation-defined.
virtual void OnInterestingUsage(int usage_pattern) {}
};
// PeerConnectionDependencies holds all of PeerConnections dependencies.
// A dependency is distinct from a configuration as it defines significant
// executable code that can be provided by a user of the API.
//
// All new dependencies should be added as a unique_ptr to allow the
// PeerConnection object to be the definitive owner of the dependencies
// lifetime making injection safer.
struct RTC_EXPORT PeerConnectionDependencies final {
explicit PeerConnectionDependencies(PeerConnectionObserver* observer_in);
// This object is not copyable or assignable.
PeerConnectionDependencies(const PeerConnectionDependencies&) = delete;
PeerConnectionDependencies& operator=(const PeerConnectionDependencies&) =
delete;
// This object is only moveable.
PeerConnectionDependencies(PeerConnectionDependencies&&);
PeerConnectionDependencies& operator=(PeerConnectionDependencies&&) = default;
~PeerConnectionDependencies();
// Mandatory dependencies
PeerConnectionObserver* observer = nullptr;
// Optional dependencies
// TODO(bugs.webrtc.org/7447): remove port allocator once downstream is
// updated. The recommended way to inject networking components is to pass a
// PacketSocketFactory when creating the PeerConnectionFactory.
std::unique_ptr<cricket::PortAllocator> allocator;
// Factory for creating resolvers that look up hostnames in DNS
std::unique_ptr<webrtc::AsyncDnsResolverFactoryInterface>
async_dns_resolver_factory;
std::unique_ptr<webrtc::IceTransportFactory> ice_transport_factory;
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier;
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory;
// Optional network controller factory to use.
// Overrides that set in PeerConnectionFactoryDependencies.
std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
// Optional field trials to use.
// Overrides those from PeerConnectionFactoryDependencies.
std::unique_ptr<FieldTrialsView> trials;
};
// PeerConnectionFactoryDependencies holds all of the PeerConnectionFactory
// dependencies. All new dependencies should be added here instead of
// overloading the function. This simplifies dependency injection and makes it
// clear which are mandatory and optional. If possible please allow the peer
// connection factory to take ownership of the dependency by adding a unique_ptr
// to this structure.
struct RTC_EXPORT PeerConnectionFactoryDependencies final {
PeerConnectionFactoryDependencies();
// This object is not copyable or assignable.
PeerConnectionFactoryDependencies(const PeerConnectionFactoryDependencies&) =
delete;
PeerConnectionFactoryDependencies& operator=(
const PeerConnectionFactoryDependencies&) = delete;
// This object is only moveable.
PeerConnectionFactoryDependencies(PeerConnectionFactoryDependencies&&);
PeerConnectionFactoryDependencies& operator=(
PeerConnectionFactoryDependencies&&) = default;
~PeerConnectionFactoryDependencies();
// Optional dependencies
rtc::Thread* network_thread = nullptr;
rtc::Thread* worker_thread = nullptr;
rtc::Thread* signaling_thread = nullptr;
rtc::SocketFactory* socket_factory = nullptr;
// The `packet_socket_factory` will only be used if CreatePeerConnection is
// called without a `port_allocator`.
std::unique_ptr<rtc::PacketSocketFactory> packet_socket_factory;
std::unique_ptr<TaskQueueFactory> task_queue_factory;
std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory;
std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory;
std::unique_ptr<NetworkStatePredictorFactoryInterface>
network_state_predictor_factory;
std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
// The `network_manager` will only be used if CreatePeerConnection is called
// without a `port_allocator`, causing the default allocator and network
// manager to be used.
std::unique_ptr<rtc::NetworkManager> network_manager;
// The `network_monitor_factory` will only be used if CreatePeerConnection is
// called without a `port_allocator`, and the above `network_manager' is null.
std::unique_ptr<rtc::NetworkMonitorFactory> network_monitor_factory;
std::unique_ptr<NetEqFactory> neteq_factory;
std::unique_ptr<SctpTransportFactoryInterface> sctp_factory;
std::unique_ptr<FieldTrialsView> trials;
std::unique_ptr<RtpTransportControllerSendFactoryInterface>
transport_controller_send_factory;
// Metronome used for decoding, must be called on the worker thread.
std::unique_ptr<Metronome> decode_metronome;
// Metronome used for encoding, must be called on the worker thread.
// TODO(b/304158952): Consider merging into a single metronome for all codec
// usage.
std::unique_ptr<Metronome> encode_metronome;
// Media specific dependencies. Unused when `media_factory == nullptr`.
rtc::scoped_refptr<AudioDeviceModule> adm;
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory;
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory;
rtc::scoped_refptr<AudioMixer> audio_mixer;
rtc::scoped_refptr<AudioProcessing> audio_processing;
std::unique_ptr<AudioFrameProcessor> audio_frame_processor;
std::unique_ptr<VideoEncoderFactory> video_encoder_factory;
std::unique_ptr<VideoDecoderFactory> video_decoder_factory;
// The `media_factory` members allows webrtc to be optionally built without
// media support (i.e., if only being used for data channels).
// By default media is disabled. To enable media call
// `EnableMedia(PeerConnectionFactoryDependencies&)`. Definition of the
// `MediaFactory` interface is a webrtc implementation detail.
std::unique_ptr<MediaFactory> media_factory;
};
// PeerConnectionFactoryInterface is the factory interface used for creating
// PeerConnection, MediaStream and MediaStreamTrack objects.
//
// The simplest method for obtaiing one, CreatePeerConnectionFactory will
// create the required libjingle threads, socket and network manager factory
// classes for networking if none are provided, though it requires that the
// application runs a message loop on the thread that called the method (see
// explanation below)
//
// If an application decides to provide its own threads and/or implementation
// of networking classes, it should use the alternate
// CreatePeerConnectionFactory method which accepts threads as input, and use
// the CreatePeerConnection version that takes a PortAllocator as an argument.
class RTC_EXPORT PeerConnectionFactoryInterface
: public webrtc::RefCountInterface {
public:
class Options {
public:
Options() {}
// If set to true, created PeerConnections won't enforce any SRTP
// requirement, allowing unsecured media. Should only be used for
// testing/debugging.
bool disable_encryption = false;
// If set to true, any platform-supported network monitoring capability
// won't be used, and instead networks will only be updated via polling.
//
// This only has an effect if a PeerConnection is created with the default
// PortAllocator implementation.
bool disable_network_monitor = false;
// Sets the network types to ignore. For instance, calling this with
// ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK will ignore Ethernet and
// loopback interfaces.
int network_ignore_mask = rtc::kDefaultNetworkIgnoreMask;
// Sets the maximum supported protocol version. The highest version
// supported by both ends will be used for the connection, i.e. if one
// party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
// Sets crypto related options, e.g. enabled cipher suites.
CryptoOptions crypto_options = {};
};
// Set the options to be used for subsequently created PeerConnections.
virtual void SetOptions(const Options& options) = 0;
// The preferred way to create a new peer connection. Simply provide the
// configuration and a PeerConnectionDependencies structure.
// TODO(benwright): Make pure virtual once downstream mock PC factory classes
// are updated.
virtual RTCErrorOr<rtc::scoped_refptr<PeerConnectionInterface>>
CreatePeerConnectionOrError(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies);
// Deprecated creator - does not return an error code on error.
// TODO(bugs.webrtc.org:12238): Deprecate and remove.
ABSL_DEPRECATED("Use CreatePeerConnectionOrError")
virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies);
// Deprecated; `allocator` and `cert_generator` may be null, in which case
// default implementations will be used.
//
// `observer` must not be null.
//
// Note that this method does not take ownership of `observer`; it's the
// responsibility of the caller to delete it. It can be safely deleted after
// Close has been called on the returned PeerConnection, which ensures no
// more observer callbacks will be invoked.
ABSL_DEPRECATED("Use CreatePeerConnectionOrError")
virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer);
// Returns the capabilities of an RTP sender of type `kind`.
// If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
// TODO(orphis): Make pure virtual when all subclasses implement it.
virtual RtpCapabilities GetRtpSenderCapabilities(
cricket::MediaType kind) const;
// Returns the capabilities of an RTP receiver of type `kind`.
// If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
// TODO(orphis): Make pure virtual when all subclasses implement it.
virtual RtpCapabilities GetRtpReceiverCapabilities(
cricket::MediaType kind) const;
virtual rtc::scoped_refptr<MediaStreamInterface> CreateLocalMediaStream(
const std::string& stream_id) = 0;
// Creates an AudioSourceInterface.
// `options` decides audio processing settings.
virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const cricket::AudioOptions& options) = 0;
// Creates a new local VideoTrack. The same `source` can be used in several
// tracks.
virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
rtc::scoped_refptr<VideoTrackSourceInterface> source,
absl::string_view label) = 0;
ABSL_DEPRECATED("Use version with scoped_refptr")
virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
const std::string& label,
VideoTrackSourceInterface* source) {
return CreateVideoTrack(
rtc::scoped_refptr<VideoTrackSourceInterface>(source), label);
}
// Creates an new AudioTrack. At the moment `source` can be null.
virtual rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
const std::string& label,
AudioSourceInterface* source) = 0;
// Starts AEC dump using existing file. Takes ownership of `file` and passes
// it on to VoiceEngine (via other objects) immediately, which will take
// the ownerhip. If the operation fails, the file will be closed.
// A maximum file size in bytes can be specified. When the file size limit is
// reached, logging is stopped automatically. If max_size_bytes is set to a
// value <= 0, no limit will be used, and logging will continue until the
// StopAecDump function is called.
// TODO(webrtc:6463): Delete default implementation when downstream mocks
// classes are updated.
virtual bool StartAecDump(FILE* file, int64_t max_size_bytes) {
return false;
}
// Stops logging the AEC dump.
virtual void StopAecDump() = 0;
protected:
// Dtor and ctor protected as objects shouldn't be created or deleted via
// this interface.
PeerConnectionFactoryInterface() {}
~PeerConnectionFactoryInterface() override = default;
};
// CreateModularPeerConnectionFactory is implemented in the "peerconnection"
// build target, which doesn't pull in the implementations of every module
// webrtc may use.
//
// If an application knows it will only require certain modules, it can reduce
// webrtc's impact on its binary size by depending only on the "peerconnection"
// target and the modules the application requires, using
// CreateModularPeerConnectionFactory. For example, if an application
// only uses WebRTC for audio, it can pass in null pointers for the
// video-specific interfaces, and omit the corresponding modules from its
// build.
//
// If `network_thread` or `worker_thread` are null, the PeerConnectionFactory
// will create the necessary thread internally. If `signaling_thread` is null,
// the PeerConnectionFactory will use the thread on which this method is called
// as the signaling thread, wrapping it in an rtc::Thread object if needed.
RTC_EXPORT rtc::scoped_refptr<PeerConnectionFactoryInterface>
CreateModularPeerConnectionFactory(
PeerConnectionFactoryDependencies dependencies);
inline constexpr absl::string_view PeerConnectionInterface::AsString(
SignalingState state) {
switch (state) {
case SignalingState::kStable:
return "stable";
case SignalingState::kHaveLocalOffer:
return "have-local-offer";
case SignalingState::kHaveLocalPrAnswer:
return "have-local-pranswer";
case SignalingState::kHaveRemoteOffer:
return "have-remote-offer";
case SignalingState::kHaveRemotePrAnswer:
return "have-remote-pranswer";
case SignalingState::kClosed:
return "closed";
}
// This cannot happen.
// Not using "RTC_CHECK_NOTREACHED()" because AsString() is constexpr.
return "";
}
inline constexpr absl::string_view PeerConnectionInterface::AsString(
IceGatheringState state) {
switch (state) {
case IceGatheringState::kIceGatheringNew:
return "new";
case IceGatheringState::kIceGatheringGathering:
return "gathering";
case IceGatheringState::kIceGatheringComplete:
return "complete";
}
// This cannot happen.
// Not using "RTC_CHECK_NOTREACHED()" because AsString() is constexpr.
return "";
}
inline constexpr absl::string_view PeerConnectionInterface::AsString(
PeerConnectionState state) {
switch (state) {
case PeerConnectionState::kNew:
return "new";
case PeerConnectionState::kConnecting:
return "connecting";
case PeerConnectionState::kConnected:
return "connected";
case PeerConnectionState::kDisconnected:
return "disconnected";
case PeerConnectionState::kFailed:
return "failed";
case PeerConnectionState::kClosed:
return "closed";
}
// This cannot happen.
// Not using "RTC_CHECK_NOTREACHED()" because AsString() is constexpr.
return "";
}
inline constexpr absl::string_view PeerConnectionInterface::AsString(
IceConnectionState state) {
switch (state) {
case kIceConnectionNew:
return "new";
case kIceConnectionChecking:
return "checking";
case kIceConnectionConnected:
return "connected";
case kIceConnectionCompleted:
return "completed";
case kIceConnectionFailed:
return "failed";
case kIceConnectionDisconnected:
return "disconnected";
case kIceConnectionClosed:
return "closed";
case kIceConnectionMax:
// This cannot happen.
// Not using "RTC_CHECK_NOTREACHED()" because AsString() is constexpr.
return "";
}
// This cannot happen.
// Not using "RTC_CHECK_NOTREACHED()" because AsString() is constexpr.
return "";
}
} // namespace webrtc
#endif // API_PEER_CONNECTION_INTERFACE_H_