Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(273)

Side by Side Diff: talk/media/base/rtputils.cc

Issue 1578323002: Add rtppacketuil.h/cc (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2011 Google Inc. 3 * Copyright 2011 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation 11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution. 12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products 13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission. 14 * derived from this software without specific prior written permission.
15 * 15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED 16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF 17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO 18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, 19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, 20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; 21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */ 26 */
27 27
28 #include "talk/media/base/rtputils.h" 28 #include "talk/media/base/rtputils.h"
29 29
30 #include "talk/media/base/turnutils.h"
31 #include "webrtc/base/asyncpacketsocket.h"
pthatcher1 2016/01/14 19:34:02 Dependency on a socket is weird for an rtputils cl
Sergey Ulanov 2016/01/14 20:13:10 Done.
32 #include "webrtc/base/checks.h"
33 #include "webrtc/base/messagedigest.h"
34
30 namespace cricket { 35 namespace cricket {
31 36
32 static const uint8_t kRtpVersion = 2; 37 static const uint8_t kRtpVersion = 2;
33 static const size_t kRtpFlagsOffset = 0; 38 static const size_t kRtpFlagsOffset = 0;
34 static const size_t kRtpPayloadTypeOffset = 1; 39 static const size_t kRtpPayloadTypeOffset = 1;
35 static const size_t kRtpSeqNumOffset = 2; 40 static const size_t kRtpSeqNumOffset = 2;
36 static const size_t kRtpTimestampOffset = 4; 41 static const size_t kRtpTimestampOffset = 4;
37 static const size_t kRtpSsrcOffset = 8; 42 static const size_t kRtpSsrcOffset = 8;
38 static const size_t kRtcpPayloadTypeOffset = 1; 43 static const size_t kRtcpPayloadTypeOffset = 1;
44 static const size_t kRtpExtensionHeaderLen = 4;
45 static const size_t kAbsSendTimeExtensionLen = 3;
46 static const size_t kOneByteExtensionHeaderLen = 1;
47
48 namespace {
49
50 // Fake auth tag written by the sender when external authentication is enabled.
51 // HMAC in packet will be compared against this value before updating packet
52 // with actual HMAC value.
53 static const uint8_t kFakeAuthTag[10] = {
54 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd
55 };
56
57 void UpdateAbsSendTimeExtensionValue(uint8_t* extension_data,
58 size_t length,
59 uint64_t time_us) {
60 // Absolute send time in RTP streams.
61 //
62 // The absolute send time is signaled to the receiver in-band using the
63 // general mechanism for RTP header extensions [RFC5285]. The payload
64 // of this extension (the transmitted value) is a 24-bit unsigned integer
65 // containing the sender's current time in seconds as a fixed point number
66 // with 18 bits fractional part.
67 //
68 // The form of the absolute send time extension block:
69 //
70 // 0 1 2 3
71 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
72 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
73 // | ID | len=2 | absolute send time |
74 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
75 if (length != kAbsSendTimeExtensionLen) {
76 RTC_NOTREACHED();
77 return;
78 }
79
80 // Convert microseconds to a 6.18 fixed point value in seconds.
81 uint32_t send_time = ((time_us << 18) / 1000000) & 0x00FFFFFF;
82 extension_data[0] = static_cast<uint8_t>(send_time >> 16);
83 extension_data[1] = static_cast<uint8_t>(send_time >> 8);
84 extension_data[2] = static_cast<uint8_t>(send_time);
85 }
86
87 // Assumes |length| is actual packet length + tag length. Updates HMAC at end of
88 // the RTP packet.
89 void UpdateRtpAuthTag(uint8_t* rtp,
90 size_t length,
91 const rtc::PacketTimeUpdateParams& packet_time_params) {
92 // If there is no key, return.
93 if (packet_time_params.srtp_auth_key.empty()) {
94 return;
95 }
96
97 size_t tag_length = packet_time_params.srtp_auth_tag_len;
98
99 // ROC (rollover counter) is at the beginning of the auth tag.
100 const size_t kRocLength = 4;
101 if (tag_length < kRocLength || tag_length > length) {
102 RTC_NOTREACHED();
103 return;
104 }
105
106 uint8_t* auth_tag = rtp + (length - tag_length);
107
108 // We should have a fake HMAC value @ auth_tag.
109 RTC_DCHECK_EQ(0, memcmp(auth_tag, kFakeAuthTag, tag_length));
110
111 // Copy ROC after end of rtp packet.
112 memcpy(auth_tag, &packet_time_params.srtp_packet_index, kRocLength);
113 // Authentication of a RTP packet will have RTP packet + ROC size.
114 int auth_required_length = length - tag_length + kRocLength;
115
116 uint8_t output[64];
117 size_t result = rtc::ComputeHmac(
118 rtc::DIGEST_SHA_1, &packet_time_params.srtp_auth_key[0],
119 packet_time_params.srtp_auth_key.size(), rtp,
120 auth_required_length, output, sizeof(output));
121
122 if (result < tag_length) {
123 RTC_NOTREACHED();
124 return;
125 }
126
127 // Copy HMAC from output to packet. This is required as auth tag length
128 // may not be equal to the actual HMAC length.
129 memcpy(auth_tag, output, tag_length);
130 }
131
132 }
39 133
40 bool GetUint8(const void* data, size_t offset, int* value) { 134 bool GetUint8(const void* data, size_t offset, int* value) {
41 if (!data || !value) { 135 if (!data || !value) {
42 return false; 136 return false;
43 } 137 }
44 *value = *(static_cast<const uint8_t*>(data) + offset); 138 *value = *(static_cast<const uint8_t*>(data) + offset);
45 return true; 139 return true;
46 } 140 }
47 141
48 bool GetUint16(const void* data, size_t offset, int* value) { 142 bool GetUint16(const void* data, size_t offset, int* value) {
(...skipping 144 matching lines...) Expand 10 before | Expand all | Expand 10 after
193 if (len < kMinRtpPacketLen) 287 if (len < kMinRtpPacketLen)
194 return false; 288 return false;
195 289
196 return (static_cast<const uint8_t*>(data)[0] >> 6) == kRtpVersion; 290 return (static_cast<const uint8_t*>(data)[0] >> 6) == kRtpVersion;
197 } 291 }
198 292
199 bool IsValidRtpPayloadType(int payload_type) { 293 bool IsValidRtpPayloadType(int payload_type) {
200 return payload_type >= 0 && payload_type <= 127; 294 return payload_type >= 0 && payload_type <= 127;
201 } 295 }
202 296
297 bool ValidateRtpHeader(const uint8_t* rtp,
298 size_t length,
299 size_t* header_length) {
300 if (header_length) {
301 *header_length = 0;
302 }
303
304 if (length < kMinRtpPacketLen) {
305 return false;
306 }
307
308 size_t cc_count = rtp[0] & 0x0F;
309 size_t header_length_without_extension = kMinRtpPacketLen + 4 * cc_count;
310 if (header_length_without_extension > length) {
311 return false;
312 }
313
314 // If extension bit is not set, we are done with header processing, as input
315 // length is verified above.
316 if (!(rtp[0] & 0x10)) {
317 if (header_length)
318 *header_length = header_length_without_extension;
319
320 return true;
321 }
322
323 rtp += header_length_without_extension;
324
325 if (header_length_without_extension + kRtpExtensionHeaderLen > length) {
326 return false;
327 }
328
329 // Getting extension profile length.
330 // Length is in 32 bit words.
331 uint16_t extension_length_in_32bits = rtc::GetBE16(rtp + 2);
332 size_t extension_length = extension_length_in_32bits * 4;
333
334 size_t rtp_header_length = extension_length +
335 header_length_without_extension +
336 kRtpExtensionHeaderLen;
337
338 // Verify input length against total header size.
339 if (rtp_header_length > length) {
340 return false;
341 }
342
343 if (header_length) {
344 *header_length = rtp_header_length;
345 }
346 return true;
347 }
348
349 // ValidateRtpHeader() must be called before this method to make sure, we have
350 // a sane rtp packet.
351 bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp,
352 size_t length,
353 int extension_id,
354 uint64_t time_us) {
355 // 0 1 2 3
356 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
357 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
358 // |V=2|P|X| CC |M| PT | sequence number |
359 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
360 // | timestamp |
361 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
362 // | synchronization source (SSRC) identifier |
363 // +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
364 // | contributing source (CSRC) identifiers |
365 // | .... |
366 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
367
368 // Return if extension bit is not set.
369 if (!(rtp[0] & 0x10)) {
370 return true;
371 }
372
373 size_t cc_count = rtp[0] & 0x0F;
374 size_t header_length_without_extension = kMinRtpPacketLen + 4 * cc_count;
375
376 rtp += header_length_without_extension;
377
378 // Getting extension profile ID and length.
379 uint16_t profile_id = rtc::GetBE16(rtp);
380 // Length is in 32 bit words.
381 uint16_t extension_length_in_32bits = rtc::GetBE16(rtp + 2);
382 size_t extension_length = extension_length_in_32bits * 4;
383
384 rtp += kRtpExtensionHeaderLen; // Moving past extension header.
385
386 bool found = false;
387 // WebRTC is using one byte header extension.
388 // TODO(mallinath) - Handle two byte header extension.
389 if (profile_id == 0xBEDE) { // OneByte extension header
390 // 0
391 // 0 1 2 3 4 5 6 7
392 // +-+-+-+-+-+-+-+-+
393 // | ID |length |
394 // +-+-+-+-+-+-+-+-+
395
396 // 0 1 2 3
397 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
398 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
399 // | 0xBE | 0xDE | length=3 |
400 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
401 // | ID | L=0 | data | ID | L=1 | data...
402 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
403 // ...data | 0 (pad) | 0 (pad) | ID | L=3 |
404 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
405 // | data |
406 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
407 const uint8_t* extension_start = rtp;
408 const uint8_t* extension_end = extension_start + extension_length;
409
410 while (rtp < extension_end) {
411 const int id = (*rtp & 0xF0) >> 4;
412 const size_t length = (*rtp & 0x0F) + 1;
413 if (rtp + kOneByteExtensionHeaderLen + length > extension_end) {
414 return false;
415 }
416 // The 4-bit length is the number minus one of data bytes of this header
417 // extension element following the one-byte header.
418 if (id == extension_id) {
419 UpdateAbsSendTimeExtensionValue(rtp + kOneByteExtensionHeaderLen,
420 length, time_us);
421 found = true;
422 break;
423 }
424 rtp += kOneByteExtensionHeaderLen + length;
425 // Counting padding bytes.
426 while ((rtp < extension_end) && (*rtp == 0)) {
427 ++rtp;
428 }
429 }
430 }
431 return found;
432 }
433
434 bool ApplyPacketOptions(uint8_t* data,
435 size_t length,
436 const rtc::PacketTimeUpdateParams& packet_time_params,
437 uint64_t time_us) {
438 RTC_DCHECK(data);
439 RTC_DCHECK(length);
440
441 // if there is no valid |rtp_sendtime_extension_id| and |srtp_auth_key| in
442 // PacketOptions, nothing to be updated in this packet.
443 if (packet_time_params.rtp_sendtime_extension_id == -1 &&
444 packet_time_params.srtp_auth_key.empty()) {
445 return true;
446 }
447
448 // If there is a srtp auth key present then the packet must be an RTP packet.
449 // RTP packet may have been wrapped in a TURN Channel Data or TURN send
450 // indication.
451 size_t rtp_start_pos;
452 size_t rtp_length;
453 if (!UnwrapTurnPacket(data, length, &rtp_start_pos, &rtp_length)) {
454 RTC_NOTREACHED();
455 return false;
456 }
457
458 // Making sure we have a valid RTP packet at the end.
459 if (!IsRtpPacket(data + rtp_start_pos, rtp_length) ||
460 !ValidateRtpHeader(data + rtp_start_pos, rtp_length, nullptr)) {
461 RTC_NOTREACHED();
462 return false;
463 }
464
465 uint8_t* start = data + rtp_start_pos;
466 // If packet option has non default value (-1) for sendtime extension id,
467 // then we should parse the rtp packet to update the timestamp. Otherwise
468 // just calculate HMAC and update packet with it.
469 if (packet_time_params.rtp_sendtime_extension_id != -1) {
470 UpdateRtpAbsSendTimeExtension(start, rtp_length,
471 packet_time_params.rtp_sendtime_extension_id,
472 time_us);
473 }
474
475 UpdateRtpAuthTag(start, rtp_length, packet_time_params);
476 return true;
477 }
478
203 } // namespace cricket 479 } // namespace cricket
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698