blob: c49115a116a628fab2a1ab12bd180aaa3204527e [file] [log] [blame]
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +02001/*
2 * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11#ifndef MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_
12#define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_
13
14#include <memory>
15#include <string>
16#include <vector>
17
Ali Tofighd14e8892022-05-13 11:42:16 +020018#include "absl/strings/string_view.h"
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +020019#include "absl/types/optional.h"
Jonas Orelande62c2f22022-03-29 11:04:48 +020020#include "api/field_trials_view.h"
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +020021#include "api/frame_transformer_interface.h"
22#include "api/scoped_refptr.h"
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +020023#include "api/video/video_bitrate_allocation.h"
24#include "modules/rtp_rtcp/include/receive_statistics.h"
25#include "modules/rtp_rtcp/include/report_block_data.h"
26#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
27#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
28#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
29#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
30#include "modules/rtp_rtcp/source/video_fec_generator.h"
Alessio Bazzicabc1c93d2021-03-12 17:45:26 +010031#include "system_wrappers/include/ntp_time.h"
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +020032
33namespace webrtc {
34
35// Forward declarations.
36class FrameEncryptorInterface;
37class RateLimiter;
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +020038class RtcEventLog;
39class RTPSender;
40class Transport;
41class VideoBitrateAllocationObserver;
42
43class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
44 public:
45 struct Configuration {
46 Configuration() = default;
47 Configuration(Configuration&& rhs) = default;
48
Byoungchan Lee604fd2f2022-01-21 09:49:39 +090049 Configuration(const Configuration&) = delete;
50 Configuration& operator=(const Configuration&) = delete;
51
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +020052 // True for a audio version of the RTP/RTCP module object false will create
53 // a video version.
54 bool audio = false;
55 bool receiver_only = false;
56
57 // The clock to use to read time. If nullptr then system clock will be used.
58 Clock* clock = nullptr;
59
60 ReceiveStatisticsProvider* receive_statistics = nullptr;
61
62 // Transport object that will be called when packets are ready to be sent
63 // out on the network.
64 Transport* outgoing_transport = nullptr;
65
66 // Called when the receiver requests an intra frame.
67 RtcpIntraFrameObserver* intra_frame_callback = nullptr;
68
69 // Called when the receiver sends a loss notification.
70 RtcpLossNotificationObserver* rtcp_loss_notification_observer = nullptr;
71
72 // Called when we receive a changed estimate from the receiver of out
73 // stream.
74 RtcpBandwidthObserver* bandwidth_callback = nullptr;
75
76 NetworkStateEstimateObserver* network_state_estimate_observer = nullptr;
77 TransportFeedbackObserver* transport_feedback_callback = nullptr;
78 VideoBitrateAllocationObserver* bitrate_allocation_observer = nullptr;
79 RtcpRttStats* rtt_stats = nullptr;
80 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr;
81 // Called on receipt of RTCP report block from remote side.
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +020082 // TODO(bugs.webrtc.org/10679): Consider whether we want to use
83 // only getters or only callbacks. If we decide on getters, the
84 // ReportBlockDataObserver should also be removed in favor of
85 // GetLatestReportBlockData().
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +020086 RtcpCnameCallback* rtcp_cname_callback = nullptr;
87 ReportBlockDataObserver* report_block_data_observer = nullptr;
88
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +020089 // Spread any bursts of packets into smaller bursts to minimize packet loss.
90 RtpPacketSender* paced_sender = nullptr;
91
92 // Generates FEC packets.
93 // TODO(sprang): Wire up to RtpSenderEgress.
94 VideoFecGenerator* fec_generator = nullptr;
95
96 BitrateStatisticsObserver* send_bitrate_observer = nullptr;
97 SendSideDelayObserver* send_side_delay_observer = nullptr;
98 RtcEventLog* event_log = nullptr;
99 SendPacketObserver* send_packet_observer = nullptr;
100 RateLimiter* retransmission_rate_limiter = nullptr;
101 StreamDataCountersCallback* rtp_stats_callback = nullptr;
102
103 int rtcp_report_interval_ms = 0;
104
105 // Update network2 instead of pacer_exit field of video timing extension.
106 bool populate_network2_timestamp = false;
107
108 rtc::scoped_refptr<FrameTransformerInterface> frame_transformer;
109
110 // E2EE Custom Video Frame Encryption
111 FrameEncryptorInterface* frame_encryptor = nullptr;
112 // Require all outgoing frames to be encrypted with a FrameEncryptor.
113 bool require_frame_encryption = false;
114
115 // Corresponds to extmap-allow-mixed in SDP negotiation.
116 bool extmap_allow_mixed = false;
117
118 // If true, the RTP sender will always annotate outgoing packets with
119 // MID and RID header extensions, if provided and negotiated.
120 // If false, the RTP sender will stop sending MID and RID header extensions,
121 // when it knows that the receiver is ready to demux based on SSRC. This is
122 // done by RTCP RR acking.
123 bool always_send_mid_and_rid = false;
124
Artem Titov913cfa72021-07-28 23:57:33 +0200125 // If set, field trials are read from `field_trials`, otherwise
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200126 // defaults to webrtc::FieldTrialBasedConfig.
Jonas Orelande62c2f22022-03-29 11:04:48 +0200127 const FieldTrialsView* field_trials = nullptr;
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200128
129 // SSRCs for media and retransmission, respectively.
Artem Titov913cfa72021-07-28 23:57:33 +0200130 // FlexFec SSRC is fetched from `flexfec_sender`.
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200131 uint32_t local_media_ssrc = 0;
132 absl::optional<uint32_t> rtx_send_ssrc;
133
134 bool need_rtp_packet_infos = false;
135
136 // If true, the RTP packet history will select RTX packets based on
137 // heuristics such as send time, retransmission count etc, in order to
138 // make padding potentially more useful.
139 // If false, the last packet will always be picked. This may reduce CPU
140 // overhead.
141 bool enable_rtx_padding_prioritization = true;
142
Niels Möllerbe810cb2020-12-02 14:25:03 +0100143 // Estimate RTT as non-sender as described in
144 // https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5
145 bool non_sender_rtt_measurement = false;
Niels Mölleraf785d92022-05-31 10:45:41 +0200146
147 // If non-empty, sets the value for sending in the RID (and Repaired) RTP
148 // header extension. RIDs are used to identify an RTP stream if SSRCs are
149 // not negotiated. If the RID and Repaired RID extensions are not
150 // registered, the RID will not be sent.
151 std::string rid;
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200152 };
153
Alessio Bazzicabc1c93d2021-03-12 17:45:26 +0100154 // Stats for RTCP sender reports (SR) for a specific SSRC.
155 // Refer to https://tools.ietf.org/html/rfc3550#section-6.4.1.
156 struct SenderReportStats {
Ivo Creusen2562cf02021-09-03 14:51:22 +0000157 // Arrival NTP timestamp for the last received RTCP SR.
Alessio Bazzicabc1c93d2021-03-12 17:45:26 +0100158 NtpTime last_arrival_timestamp;
159 // Received (a.k.a., remote) NTP timestamp for the last received RTCP SR.
160 NtpTime last_remote_timestamp;
161 // Total number of RTP data packets transmitted by the sender since starting
162 // transmission up until the time this SR packet was generated. The count
163 // should be reset if the sender changes its SSRC identifier.
164 uint32_t packets_sent;
165 // Total number of payload octets (i.e., not including header or padding)
166 // transmitted in RTP data packets by the sender since starting transmission
167 // up until the time this SR packet was generated. The count should be reset
168 // if the sender changes its SSRC identifier.
169 uint64_t bytes_sent;
170 // Total number of RTCP SR blocks received.
171 // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-reportssent.
172 uint64_t reports_count;
173 };
Ivo Creusen2562cf02021-09-03 14:51:22 +0000174 // Stats about the non-sender SSRC, based on RTCP extended reports (XR).
175 // Refer to https://datatracker.ietf.org/doc/html/rfc3611#section-2.
176 struct NonSenderRttStats {
177 // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
178 absl::optional<TimeDelta> round_trip_time;
179 // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
180 TimeDelta total_round_trip_time = TimeDelta::Zero();
181 // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
182 int round_trip_time_measurements = 0;
183 };
Alessio Bazzicabc1c93d2021-03-12 17:45:26 +0100184
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200185 // **************************************************************************
186 // Receiver functions
187 // **************************************************************************
188
189 virtual void IncomingRtcpPacket(const uint8_t* incoming_packet,
190 size_t incoming_packet_length) = 0;
191
192 virtual void SetRemoteSSRC(uint32_t ssrc) = 0;
193
Tommi08be9ba2021-06-15 23:01:57 +0200194 // Called when the local ssrc changes (post initialization) for receive
195 // streams to match with send. Called on the packet receive thread/tq.
196 virtual void SetLocalSsrc(uint32_t ssrc) = 0;
197
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200198 // **************************************************************************
199 // Sender
200 // **************************************************************************
201
202 // Sets the maximum size of an RTP packet, including RTP headers.
203 virtual void SetMaxRtpPacketSize(size_t size) = 0;
204
205 // Returns max RTP packet size. Takes into account RTP headers and
206 // FEC/ULP/RED overhead (when FEC is enabled).
207 virtual size_t MaxRtpPacketSize() const = 0;
208
209 virtual void RegisterSendPayloadFrequency(int payload_type,
210 int payload_frequency) = 0;
211
212 // Unregisters a send payload.
Artem Titov913cfa72021-07-28 23:57:33 +0200213 // `payload_type` - payload type of codec
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200214 // Returns -1 on failure else 0.
215 virtual int32_t DeRegisterSendPayload(int8_t payload_type) = 0;
216
217 virtual void SetExtmapAllowMixed(bool extmap_allow_mixed) = 0;
218
219 // Register extension by uri, triggers CHECK on falure.
220 virtual void RegisterRtpHeaderExtension(absl::string_view uri, int id) = 0;
221
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200222 virtual void DeregisterSendRtpHeaderExtension(absl::string_view uri) = 0;
223
224 // Returns true if RTP module is send media, and any of the extensions
225 // required for bandwidth estimation is registered.
226 virtual bool SupportsPadding() const = 0;
227 // Same as SupportsPadding(), but additionally requires that
228 // SetRtxSendStatus() has been called with the kRtxRedundantPayloads option
229 // enabled.
230 virtual bool SupportsRtxPayloadPadding() const = 0;
231
232 // Returns start timestamp.
233 virtual uint32_t StartTimestamp() const = 0;
234
235 // Sets start timestamp. Start timestamp is set to a random value if this
236 // function is never called.
237 virtual void SetStartTimestamp(uint32_t timestamp) = 0;
238
239 // Returns SequenceNumber.
240 virtual uint16_t SequenceNumber() const = 0;
241
242 // Sets SequenceNumber, default is a random number.
243 virtual void SetSequenceNumber(uint16_t seq) = 0;
244
245 virtual void SetRtpState(const RtpState& rtp_state) = 0;
246 virtual void SetRtxState(const RtpState& rtp_state) = 0;
247 virtual RtpState GetRtpState() const = 0;
248 virtual RtpState GetRtxState() const = 0;
249
Ivo Creusen8c40d512021-07-13 12:53:22 +0000250 // This can be used to enable/disable receive-side RTT.
251 virtual void SetNonSenderRttMeasurement(bool enabled) = 0;
252
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200253 // Returns SSRC.
254 virtual uint32_t SSRC() const = 0;
255
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200256 // Sets the value for sending in the MID RTP header extension.
257 // The MID RTP header extension should be registered for this to do anything.
258 // Once set, this value can not be changed or removed.
Ali Tofighd14e8892022-05-13 11:42:16 +0200259 virtual void SetMid(absl::string_view mid) = 0;
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200260
261 // Sets CSRC.
Artem Titov913cfa72021-07-28 23:57:33 +0200262 // `csrcs` - vector of CSRCs
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200263 virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0;
264
265 // Turns on/off sending RTX (RFC 4588). The modes can be set as a combination
266 // of values of the enumerator RtxMode.
267 virtual void SetRtxSendStatus(int modes) = 0;
268
269 // Returns status of sending RTX (RFC 4588). The returned value can be
270 // a combination of values of the enumerator RtxMode.
271 virtual int RtxSendStatus() const = 0;
272
273 // Returns the SSRC used for RTX if set, otherwise a nullopt.
274 virtual absl::optional<uint32_t> RtxSsrc() const = 0;
275
276 // Sets the payload type to use when sending RTX packets. Note that this
277 // doesn't enable RTX, only the payload type is set.
278 virtual void SetRtxSendPayloadType(int payload_type,
279 int associated_payload_type) = 0;
280
281 // Returns the FlexFEC SSRC, if there is one.
282 virtual absl::optional<uint32_t> FlexfecSsrc() const = 0;
283
284 // Sets sending status. Sends kRtcpByeCode when going from true to false.
285 // Returns -1 on failure else 0.
286 virtual int32_t SetSendingStatus(bool sending) = 0;
287
288 // Returns current sending status.
289 virtual bool Sending() const = 0;
290
291 // Starts/Stops media packets. On by default.
292 virtual void SetSendingMediaStatus(bool sending) = 0;
293
294 // Returns current media sending status.
295 virtual bool SendingMedia() const = 0;
296
297 // Returns whether audio is configured (i.e. Configuration::audio = true).
298 virtual bool IsAudioConfigured() const = 0;
299
300 // Indicate that the packets sent by this module should be counted towards the
301 // bitrate estimate since the stream participates in the bitrate allocation.
302 virtual void SetAsPartOfAllocation(bool part_of_allocation) = 0;
303
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200304 // Returns bitrate sent (post-pacing) per packet type.
305 virtual RtpSendRates GetSendRates() const = 0;
306
307 virtual RTPSender* RtpSender() = 0;
308 virtual const RTPSender* RtpSender() const = 0;
309
310 // Record that a frame is about to be sent. Returns true on success, and false
311 // if the module isn't ready to send.
312 virtual bool OnSendingRtpFrame(uint32_t timestamp,
313 int64_t capture_time_ms,
314 int payload_type,
315 bool force_sender_report) = 0;
316
317 // Try to send the provided packet. Returns true iff packet matches any of
318 // the SSRCs for this module (media/rtx/fec etc) and was forwarded to the
319 // transport.
320 virtual bool TrySendPacket(RtpPacketToSend* packet,
321 const PacedPacketInfo& pacing_info) = 0;
322
Erik Språng1d50cb62020-07-02 17:41:32 +0200323 // Update the FEC protection parameters to use for delta- and key-frames.
324 // Only used when deferred FEC is active.
325 virtual void SetFecProtectionParams(
326 const FecProtectionParams& delta_params,
327 const FecProtectionParams& key_params) = 0;
328
329 // If deferred FEC generation is enabled, this method should be called after
330 // calling TrySendPacket(). Any generated FEC packets will be removed and
331 // returned from the FEC generator.
332 virtual std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() = 0;
333
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200334 virtual void OnPacketsAcknowledged(
335 rtc::ArrayView<const uint16_t> sequence_numbers) = 0;
336
337 virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
338 size_t target_size_bytes) = 0;
339
340 virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
341 rtc::ArrayView<const uint16_t> sequence_numbers) const = 0;
342
343 // Returns an expected per packet overhead representing the main RTP header,
344 // any CSRCs, and the registered header extensions that are expected on all
345 // packets (i.e. disregarding things like abs capture time which is only
346 // populated on a subset of packets, but counting MID/RID type extensions
347 // when we expect to send them).
348 virtual size_t ExpectedPerPacketOverhead() const = 0;
349
Erik Språngb6bbdeb2021-08-13 16:12:41 +0200350 // Access to packet state (e.g. sequence numbering) must only be access by
351 // one thread at a time. It may be only one thread, or a construction thread
352 // that calls SetRtpState() - handing over to a pacer thread that calls
353 // TrySendPacket() - and at teardown ownership is handed to a destruciton
354 // thread that calls GetRtpState().
355 // This method is used to signal that "ownership" of the rtp state is being
356 // transferred to another thread.
357 virtual void OnPacketSendingThreadSwitched() = 0;
358
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200359 // **************************************************************************
360 // RTCP
361 // **************************************************************************
362
363 // Returns RTCP status.
364 virtual RtcpMode RTCP() const = 0;
365
366 // Sets RTCP status i.e on(compound or non-compound)/off.
Artem Titov913cfa72021-07-28 23:57:33 +0200367 // `method` - RTCP method to use.
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200368 virtual void SetRTCPStatus(RtcpMode method) = 0;
369
370 // Sets RTCP CName (i.e unique identifier).
371 // Returns -1 on failure else 0.
Ali Tofighd14e8892022-05-13 11:42:16 +0200372 virtual int32_t SetCNAME(absl::string_view cname) = 0;
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200373
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200374 // Returns remote NTP.
375 // Returns -1 on failure else 0.
376 virtual int32_t RemoteNTP(uint32_t* received_ntp_secs,
377 uint32_t* received_ntp_frac,
378 uint32_t* rtcp_arrival_time_secs,
379 uint32_t* rtcp_arrival_time_frac,
380 uint32_t* rtcp_timestamp) const = 0;
381
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200382 // Returns current RTT (round-trip time) estimate.
383 // Returns -1 on failure else 0.
384 virtual int32_t RTT(uint32_t remote_ssrc,
385 int64_t* rtt,
386 int64_t* avg_rtt,
387 int64_t* min_rtt,
388 int64_t* max_rtt) const = 0;
389
390 // Returns the estimated RTT, with fallback to a default value.
391 virtual int64_t ExpectedRetransmissionTimeMs() const = 0;
392
393 // Forces a send of a RTCP packet. Periodic SR and RR are triggered via the
394 // process function.
395 // Returns -1 on failure else 0.
396 virtual int32_t SendRTCP(RTCPPacketType rtcp_packet_type) = 0;
397
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200398 // Returns send statistics for the RTP and RTX stream.
399 virtual void GetSendStreamDataCounters(
400 StreamDataCounters* rtp_counters,
401 StreamDataCounters* rtx_counters) const = 0;
402
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200403 // A snapshot of Report Blocks with additional data of interest to statistics.
404 // Within this list, the sender-source SSRC pair is unique and per-pair the
405 // ReportBlockData represents the latest Report Block that was received for
406 // that pair.
407 virtual std::vector<ReportBlockData> GetLatestReportBlockData() const = 0;
Alessio Bazzicabc1c93d2021-03-12 17:45:26 +0100408 // Returns stats based on the received RTCP SRs.
409 virtual absl::optional<SenderReportStats> GetSenderReportStats() const = 0;
Ivo Creusen2562cf02021-09-03 14:51:22 +0000410 // Returns non-sender RTT stats, based on DLRR.
411 virtual absl::optional<NonSenderRttStats> GetNonSenderRttStats() const = 0;
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200412
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200413 // (REMB) Receiver Estimated Max Bitrate.
414 // Schedules sending REMB on next and following sender/receiver reports.
415 void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override = 0;
416 // Stops sending REMB on next and following sender/receiver reports.
417 void UnsetRemb() override = 0;
418
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200419 // (NACK)
420
421 // Sends a Negative acknowledgement packet.
422 // Returns -1 on failure else 0.
423 // TODO(philipel): Deprecate this and start using SendNack instead, mostly
424 // because we want a function that actually send NACK for the specified
425 // packets.
426 virtual int32_t SendNACK(const uint16_t* nack_list, uint16_t size) = 0;
427
428 // Sends NACK for the packets specified.
429 // Note: This assumes the caller keeps track of timing and doesn't rely on
430 // the RTP module to do this.
431 virtual void SendNack(const std::vector<uint16_t>& sequence_numbers) = 0;
432
433 // Store the sent packets, needed to answer to a Negative acknowledgment
434 // requests.
435 virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0;
436
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200437 virtual void SetVideoBitrateAllocation(
438 const VideoBitrateAllocation& bitrate) = 0;
439
440 // **************************************************************************
441 // Video
442 // **************************************************************************
443
444 // Requests new key frame.
445 // using PLI, https://tools.ietf.org/html/rfc4585#section-6.3.1.1
446 void SendPictureLossIndication() { SendRTCP(kRtcpPli); }
447 // using FIR, https://tools.ietf.org/html/rfc5104#section-4.3.1.2
448 void SendFullIntraRequest() { SendRTCP(kRtcpFir); }
449
450 // Sends a LossNotification RTCP message.
451 // Returns -1 on failure else 0.
452 virtual int32_t SendLossNotification(uint16_t last_decoded_seq_num,
453 uint16_t last_received_seq_num,
454 bool decodability_flag,
455 bool buffering_allowed) = 0;
456};
457
458} // namespace webrtc
459
460#endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_