| /external/webrtc/talk/session/media/ |
| D | channel_unittest.cc | 127 enum Flags { RTCP = 0x1, RTCP_MUX = 0x2, SECURE = 0x4, SSRC_MUX = 0x8, enumerator
157 (flags1 & RTCP) != 0)); in CreateChannels()
160 (flags2 & RTCP) != 0)); in CreateChannels()
524 CreateChannels(RTCP, RTCP); in TestSetContentsRtcpMux()
544 CreateChannels(RTCP, RTCP); in TestSetContentsRtcpMuxWithPrAnswer()
568 CreateContent(RTCP | RTCP_MUX | SECURE, in TestSetRemoteContentUpdate()
967 RTCP | RTCP_MUX, RTCP | RTCP_MUX, in TestCallTeardownRtcpMux()
1008 CreateChannels(0, RTCP); in SendNoRtcpToRtcp()
1023 CreateChannels(RTCP, 0); in SendRtcpToNoRtcp()
1038 CreateChannels(RTCP, RTCP); in SendRtcpToRtcp()
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| /external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
| D | test_api.cc | 142 TEST_F(RtpRtcpAPITest, RTCP) { in TEST_F() argument
143 EXPECT_EQ(RtcpMode::kOff, module_->RTCP()); in TEST_F()
145 EXPECT_EQ(RtcpMode::kCompound, module_->RTCP()); in TEST_F()
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| /external/srtp/doc/ |
| D | intro.txt | 249 ports for RTP and RTCP. RTCP, the RTP control protocol, is used to
260 RTP allows multiple sources to send RTP and RTCP traffic during the
283 libSRTP provides functions for protecting RTP and RTCP. RTP packets
288 functions apply security to RTCP packets.
294 needed to protect a particular RTP and RTCP stream. This datatype
300 stream context to protect the RTP and RTCP stream that it is
320 that describe the cryptograhic policies for RTP and RTCP, as well as
330 for RTP and RTCP protection, respectively.
390 RTP nor RTCP, nor confidentiality for RTCP.
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| D | rfc3711.txt | 37 Real-time Transport Control Protocol (RTCP).
54 3.4. Secure RTCP . . . . . . . . . . . . . . . . . . . . . . . 15
130 RTCP (the Real-time Transport Control Protocol) [RFC3350].
133 of RTP and RTCP streams (Section 3). SRTP defines a set of default
145 RTP sequence number for SRTP and an index number for Secure RTCP
188 * the confidentiality of the RTP and RTCP payloads, and
190 * the integrity of the entire RTP and RTCP packets, together with
195 erroneous alteration of RTCP messages could otherwise disrupt the
220 RTP/RTCP in both wired and wireless scenarios.
267 Secure RTCP (SRTCP) provides the same security services to RTCP as
[all …]
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| /external/webrtc/webrtc/call/ |
| D | rtc_event_log.proto | 128 // required - Sender SSRC used for sending RTCP (such as receiver reports).
132 // RTCP mode is described by RFC 5506.
137 // required - RTCP mode to use.
228 // required - Sender SSRC used for sending RTCP (such as receiver reports).
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| /external/srtp/googlepatches/ |
| D | google-10-srtp-ios-fixes.pack | 25 …ied a fix to not crash when receiving a RTCP packet that had a much larger sequence number than th…
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| D | vidyo-4-srtp-rtx.patch | 46 /* DAM - no RTCP key limit at present */
85 policy.rtcp.sec_serv = sec_serv_none; /* we don't do RTCP anyway */
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| D | vidyo-3-srtp-ws.patch | 805 policy.rtcp.sec_serv = sec_serv_none; /* we don't do RTCP anyway */
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_utility.h | 52 bool RTCP() const;
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| D | rtp_sender_unittest.cc | 212 ASSERT_FALSE(rtp_parser.RTCP()); in VerifyCVOPacket()
341 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F()
376 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F()
416 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F()
443 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F()
480 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F()
508 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F()
536 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F()
595 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F()
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| D | rtp_header_parser.cc | 46 return rtp_parser.RTCP(); in IsRtcp()
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| D | rtp_utility.cc | 83 bool RtpHeaderParser::RTCP() const { in RTCP() function in webrtc::RtpUtility::RtpHeaderParser
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| D | rtp_rtcp_impl.h | 131 RtcpMode RTCP() const override;
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| D | rtp_rtcp_impl.cc | 483 RtcpMode ModuleRtpRtcpImpl::RTCP() const { in RTCP() function in webrtc::ModuleRtpRtcpImpl
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| /external/webrtc/webrtc/test/ |
| D | rtp_file_reader_unittest.cc | 88 if (!rtp_header_parser.RTCP() && in CountRtpPacketsPerSsrc()
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| D | rtp_file_reader.cc | 457 if (rtp_parser.RTCP()) { in ReadPacket()
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| /external/webrtc/tools/matlab/ |
| D | rtpAnalyze.m | 20 %% Filter out RTCP packets.
23 fprintf('Removing %i RTCP packets\n', length(SeqNo) - sum(ix));
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| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| D | rtp_rtcp.h | 331 virtual RtcpMode RTCP() const = 0;
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| /external/webrtc/webrtc/modules/rtp_rtcp/mocks/ |
| D | mock_rtp_rtcp.h | 140 MOCK_CONST_METHOD0(RTCP, RtcpMode());
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| /external/webrtc/webrtc/voice_engine/ |
| D | channel.cc | 2961 RtcpMode method = _rtpRtcpModule->RTCP(); in GetRTCPStatus()
3108 RtcpMode status = _rtpRtcpModule->RTCP(); in SendApplicationDefinedRTCPPacket()
3139 if (_rtpRtcpModule->RTCP() == RtcpMode::kOff) { in GetRTPStatistics()
3212 &statistics, _rtpRtcpModule->RTCP() == RtcpMode::kOff)) { in GetRTPStatistics()
4069 RtcpMode method = _rtpRtcpModule->RTCP(); in GetRTT()
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| /external/webrtc/webrtc/video/ |
| D | vie_channel.cc | 514 if (rtp_rtcp_modules_[0]->RTCP() == RtcpMode::kOff) in ProcessNACKRequest()
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| /external/libvorbis/doc/ |
| D | rfc5215.txt | 792 payloads; this will affect the RTCP jitter measurement. The length
1360 Control Protocol Extended Reports (RTCP XR)",
|