henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1 | /* |
| 2 | * libjingle |
| 3 | * Copyright 2004 Google Inc. |
| 4 | * |
| 5 | * Redistribution and use in source and binary forms, with or without |
| 6 | * modification, are permitted provided that the following conditions are met: |
| 7 | * |
| 8 | * 1. Redistributions of source code must retain the above copyright notice, |
| 9 | * this list of conditions and the following disclaimer. |
| 10 | * 2. Redistributions in binary form must reproduce the above copyright notice, |
| 11 | * this list of conditions and the following disclaimer in the documentation |
| 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 |
| 14 | * derived from this software without specific prior written permission. |
| 15 | * |
| 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 |
| 18 | * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO |
| 19 | * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, |
| 20 | * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, |
| 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, |
| 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 |
| 25 | * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| 26 | */ |
| 27 | |
| 28 | #include "talk/session/media/mediasession.h" |
| 29 | |
| 30 | #include <functional> |
| 31 | #include <map> |
| 32 | #include <set> |
| 33 | #include <utility> |
| 34 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 35 | #include "talk/media/base/constants.h" |
| 36 | #include "talk/media/base/cryptoparams.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 37 | #include "talk/session/media/channelmanager.h" |
| 38 | #include "talk/session/media/srtpfilter.h" |
buildbot@webrtc.org | a09a999 | 2014-08-13 17:26:08 +0000 | [diff] [blame] | 39 | #include "webrtc/base/helpers.h" |
| 40 | #include "webrtc/base/logging.h" |
| 41 | #include "webrtc/base/scoped_ptr.h" |
| 42 | #include "webrtc/base/stringutils.h" |
pthatcher@webrtc.org | 5ad4178 | 2014-12-23 22:14:15 +0000 | [diff] [blame] | 43 | #include "webrtc/p2p/base/constants.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 44 | |
mallinath@webrtc.org | a27be8e | 2013-09-27 23:04:10 +0000 | [diff] [blame] | 45 | #ifdef HAVE_SCTP |
| 46 | #include "talk/media/sctp/sctpdataengine.h" |
| 47 | #else |
wu@webrtc.org | 97077a3 | 2013-10-25 21:18:33 +0000 | [diff] [blame] | 48 | static const uint32 kMaxSctpSid = 1023; |
mallinath@webrtc.org | a27be8e | 2013-09-27 23:04:10 +0000 | [diff] [blame] | 49 | #endif |
| 50 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 51 | namespace { |
| 52 | const char kInline[] = "inline:"; |
| 53 | } |
| 54 | |
| 55 | namespace cricket { |
| 56 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 57 | using rtc::scoped_ptr; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 58 | |
| 59 | // RTP Profile names |
| 60 | // http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xml |
| 61 | // RFC4585 |
| 62 | const char kMediaProtocolAvpf[] = "RTP/AVPF"; |
| 63 | // RFC5124 |
jiayl@webrtc.org | 8dcd43c | 2014-05-29 22:07:59 +0000 | [diff] [blame] | 64 | const char kMediaProtocolDtlsSavpf[] = "UDP/TLS/RTP/SAVPF"; |
| 65 | |
| 66 | // This should be replaced by "UDP/TLS/RTP/SAVPF", but we need to support it for |
| 67 | // now to be compatible with previous Chrome versions. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 68 | const char kMediaProtocolSavpf[] = "RTP/SAVPF"; |
| 69 | |
| 70 | const char kMediaProtocolRtpPrefix[] = "RTP/"; |
| 71 | |
| 72 | const char kMediaProtocolSctp[] = "SCTP"; |
| 73 | const char kMediaProtocolDtlsSctp[] = "DTLS/SCTP"; |
lally@webrtc.org | ec97c65 | 2015-02-24 20:18:48 +0000 | [diff] [blame] | 74 | const char kMediaProtocolUdpDtlsSctp[] = "UDP/DTLS/SCTP"; |
lally@webrtc.org | a747093 | 2015-02-24 20:19:21 +0000 | [diff] [blame] | 75 | const char kMediaProtocolTcpDtlsSctp[] = "TCP/DTLS/SCTP"; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 76 | |
| 77 | static bool IsMediaContentOfType(const ContentInfo* content, |
| 78 | MediaType media_type) { |
| 79 | if (!IsMediaContent(content)) { |
| 80 | return false; |
| 81 | } |
| 82 | |
| 83 | const MediaContentDescription* mdesc = |
| 84 | static_cast<const MediaContentDescription*>(content->description); |
| 85 | return mdesc && mdesc->type() == media_type; |
| 86 | } |
| 87 | |
| 88 | static bool CreateCryptoParams(int tag, const std::string& cipher, |
| 89 | CryptoParams *out) { |
| 90 | std::string key; |
| 91 | key.reserve(SRTP_MASTER_KEY_BASE64_LEN); |
| 92 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 93 | if (!rtc::CreateRandomString(SRTP_MASTER_KEY_BASE64_LEN, &key)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 94 | return false; |
| 95 | } |
| 96 | out->tag = tag; |
| 97 | out->cipher_suite = cipher; |
| 98 | out->key_params = kInline; |
| 99 | out->key_params += key; |
| 100 | return true; |
| 101 | } |
| 102 | |
| 103 | #ifdef HAVE_SRTP |
| 104 | static bool AddCryptoParams(const std::string& cipher_suite, |
| 105 | CryptoParamsVec *out) { |
henrike@webrtc.org | 28654cb | 2013-07-22 21:07:49 +0000 | [diff] [blame] | 106 | int size = static_cast<int>(out->size()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 107 | |
| 108 | out->resize(size + 1); |
| 109 | return CreateCryptoParams(size, cipher_suite, &out->at(size)); |
| 110 | } |
| 111 | |
| 112 | void AddMediaCryptos(const CryptoParamsVec& cryptos, |
| 113 | MediaContentDescription* media) { |
| 114 | for (CryptoParamsVec::const_iterator crypto = cryptos.begin(); |
| 115 | crypto != cryptos.end(); ++crypto) { |
| 116 | media->AddCrypto(*crypto); |
| 117 | } |
| 118 | } |
| 119 | |
| 120 | bool CreateMediaCryptos(const std::vector<std::string>& crypto_suites, |
| 121 | MediaContentDescription* media) { |
| 122 | CryptoParamsVec cryptos; |
| 123 | for (std::vector<std::string>::const_iterator it = crypto_suites.begin(); |
| 124 | it != crypto_suites.end(); ++it) { |
| 125 | if (!AddCryptoParams(*it, &cryptos)) { |
| 126 | return false; |
| 127 | } |
| 128 | } |
| 129 | AddMediaCryptos(cryptos, media); |
| 130 | return true; |
| 131 | } |
| 132 | #endif |
| 133 | |
| 134 | const CryptoParamsVec* GetCryptos(const MediaContentDescription* media) { |
| 135 | if (!media) { |
| 136 | return NULL; |
| 137 | } |
| 138 | return &media->cryptos(); |
| 139 | } |
| 140 | |
| 141 | bool FindMatchingCrypto(const CryptoParamsVec& cryptos, |
| 142 | const CryptoParams& crypto, |
| 143 | CryptoParams* out) { |
| 144 | for (CryptoParamsVec::const_iterator it = cryptos.begin(); |
| 145 | it != cryptos.end(); ++it) { |
| 146 | if (crypto.Matches(*it)) { |
| 147 | *out = *it; |
| 148 | return true; |
| 149 | } |
| 150 | } |
| 151 | return false; |
| 152 | } |
| 153 | |
| 154 | // For audio, HMAC 32 is prefered because of the low overhead. |
| 155 | void GetSupportedAudioCryptoSuites( |
| 156 | std::vector<std::string>* crypto_suites) { |
| 157 | #ifdef HAVE_SRTP |
| 158 | crypto_suites->push_back(CS_AES_CM_128_HMAC_SHA1_32); |
| 159 | crypto_suites->push_back(CS_AES_CM_128_HMAC_SHA1_80); |
| 160 | #endif |
| 161 | } |
| 162 | |
| 163 | void GetSupportedVideoCryptoSuites( |
| 164 | std::vector<std::string>* crypto_suites) { |
| 165 | GetSupportedDefaultCryptoSuites(crypto_suites); |
| 166 | } |
| 167 | |
| 168 | void GetSupportedDataCryptoSuites( |
| 169 | std::vector<std::string>* crypto_suites) { |
| 170 | GetSupportedDefaultCryptoSuites(crypto_suites); |
| 171 | } |
| 172 | |
| 173 | void GetSupportedDefaultCryptoSuites( |
| 174 | std::vector<std::string>* crypto_suites) { |
| 175 | #ifdef HAVE_SRTP |
| 176 | crypto_suites->push_back(CS_AES_CM_128_HMAC_SHA1_80); |
| 177 | #endif |
| 178 | } |
| 179 | |
| 180 | // For video support only 80-bit SHA1 HMAC. For audio 32-bit HMAC is |
| 181 | // tolerated unless bundle is enabled because it is low overhead. Pick the |
| 182 | // crypto in the list that is supported. |
| 183 | static bool SelectCrypto(const MediaContentDescription* offer, |
| 184 | bool bundle, |
| 185 | CryptoParams *crypto) { |
| 186 | bool audio = offer->type() == MEDIA_TYPE_AUDIO; |
| 187 | const CryptoParamsVec& cryptos = offer->cryptos(); |
| 188 | |
| 189 | for (CryptoParamsVec::const_iterator i = cryptos.begin(); |
| 190 | i != cryptos.end(); ++i) { |
| 191 | if (CS_AES_CM_128_HMAC_SHA1_80 == i->cipher_suite || |
| 192 | (CS_AES_CM_128_HMAC_SHA1_32 == i->cipher_suite && audio && !bundle)) { |
| 193 | return CreateCryptoParams(i->tag, i->cipher_suite, crypto); |
| 194 | } |
| 195 | } |
| 196 | return false; |
| 197 | } |
| 198 | |
| 199 | static const StreamParams* FindFirstStreamParamsByCname( |
| 200 | const StreamParamsVec& params_vec, |
| 201 | const std::string& cname) { |
| 202 | for (StreamParamsVec::const_iterator it = params_vec.begin(); |
| 203 | it != params_vec.end(); ++it) { |
| 204 | if (cname == it->cname) |
| 205 | return &*it; |
| 206 | } |
| 207 | return NULL; |
| 208 | } |
| 209 | |
| 210 | // Generates a new CNAME or the CNAME of an already existing StreamParams |
| 211 | // if a StreamParams exist for another Stream in streams with sync_label |
| 212 | // sync_label. |
| 213 | static bool GenerateCname(const StreamParamsVec& params_vec, |
| 214 | const MediaSessionOptions::Streams& streams, |
| 215 | const std::string& synch_label, |
| 216 | std::string* cname) { |
| 217 | ASSERT(cname != NULL); |
| 218 | if (!cname) |
| 219 | return false; |
| 220 | |
| 221 | // Check if a CNAME exist for any of the other synched streams. |
| 222 | for (MediaSessionOptions::Streams::const_iterator stream_it = streams.begin(); |
| 223 | stream_it != streams.end() ; ++stream_it) { |
| 224 | if (synch_label != stream_it->sync_label) |
| 225 | continue; |
| 226 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 227 | // groupid is empty for StreamParams generated using |
| 228 | // MediaSessionDescriptionFactory. |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 229 | const StreamParams* param = GetStreamByIds(params_vec, "", stream_it->id); |
| 230 | if (param) { |
| 231 | *cname = param->cname; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 232 | return true; |
| 233 | } |
| 234 | } |
| 235 | // No other stream seems to exist that we should sync with. |
| 236 | // Generate a random string for the RTCP CNAME, as stated in RFC 6222. |
| 237 | // This string is only used for synchronization, and therefore is opaque. |
| 238 | do { |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 239 | if (!rtc::CreateRandomString(16, cname)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 240 | ASSERT(false); |
| 241 | return false; |
| 242 | } |
| 243 | } while (FindFirstStreamParamsByCname(params_vec, *cname)); |
| 244 | |
| 245 | return true; |
| 246 | } |
| 247 | |
| 248 | // Generate random SSRC values that are not already present in |params_vec|. |
wu@webrtc.org | cecfd18 | 2013-10-30 05:18:12 +0000 | [diff] [blame] | 249 | // The generated values are added to |ssrcs|. |
| 250 | // |num_ssrcs| is the number of the SSRC will be generated. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 251 | static void GenerateSsrcs(const StreamParamsVec& params_vec, |
wu@webrtc.org | cecfd18 | 2013-10-30 05:18:12 +0000 | [diff] [blame] | 252 | int num_ssrcs, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 253 | std::vector<uint32>* ssrcs) { |
wu@webrtc.org | cecfd18 | 2013-10-30 05:18:12 +0000 | [diff] [blame] | 254 | for (int i = 0; i < num_ssrcs; i++) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 255 | uint32 candidate; |
| 256 | do { |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 257 | candidate = rtc::CreateRandomNonZeroId(); |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 258 | } while (GetStreamBySsrc(params_vec, candidate) || |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 259 | std::count(ssrcs->begin(), ssrcs->end(), candidate) > 0); |
| 260 | ssrcs->push_back(candidate); |
| 261 | } |
| 262 | } |
| 263 | |
| 264 | // Returns false if we exhaust the range of SIDs. |
| 265 | static bool GenerateSctpSid(const StreamParamsVec& params_vec, |
| 266 | uint32* sid) { |
| 267 | if (params_vec.size() > kMaxSctpSid) { |
| 268 | LOG(LS_WARNING) << |
| 269 | "Could not generate an SCTP SID: too many SCTP streams."; |
| 270 | return false; |
| 271 | } |
| 272 | while (true) { |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 273 | uint32 candidate = rtc::CreateRandomNonZeroId() % kMaxSctpSid; |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 274 | if (!GetStreamBySsrc(params_vec, candidate)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 275 | *sid = candidate; |
| 276 | return true; |
| 277 | } |
| 278 | } |
| 279 | } |
| 280 | |
| 281 | static bool GenerateSctpSids(const StreamParamsVec& params_vec, |
| 282 | std::vector<uint32>* sids) { |
| 283 | uint32 sid; |
| 284 | if (!GenerateSctpSid(params_vec, &sid)) { |
| 285 | LOG(LS_WARNING) << "Could not generated an SCTP SID."; |
| 286 | return false; |
| 287 | } |
| 288 | sids->push_back(sid); |
| 289 | return true; |
| 290 | } |
| 291 | |
| 292 | // Finds all StreamParams of all media types and attach them to stream_params. |
| 293 | static void GetCurrentStreamParams(const SessionDescription* sdesc, |
| 294 | StreamParamsVec* stream_params) { |
| 295 | if (!sdesc) |
| 296 | return; |
| 297 | |
| 298 | const ContentInfos& contents = sdesc->contents(); |
| 299 | for (ContentInfos::const_iterator content = contents.begin(); |
| 300 | content != contents.end(); ++content) { |
| 301 | if (!IsMediaContent(&*content)) { |
| 302 | continue; |
| 303 | } |
| 304 | const MediaContentDescription* media = |
| 305 | static_cast<const MediaContentDescription*>( |
| 306 | content->description); |
| 307 | const StreamParamsVec& streams = media->streams(); |
| 308 | for (StreamParamsVec::const_iterator it = streams.begin(); |
| 309 | it != streams.end(); ++it) { |
| 310 | stream_params->push_back(*it); |
| 311 | } |
| 312 | } |
| 313 | } |
| 314 | |
jiayl@webrtc.org | 9c16c39 | 2014-05-01 18:30:30 +0000 | [diff] [blame] | 315 | // Filters the data codecs for the data channel type. |
| 316 | void FilterDataCodecs(std::vector<DataCodec>* codecs, bool sctp) { |
| 317 | // Filter RTP codec for SCTP and vice versa. |
| 318 | int codec_id = sctp ? kGoogleRtpDataCodecId : kGoogleSctpDataCodecId; |
| 319 | for (std::vector<DataCodec>::iterator iter = codecs->begin(); |
| 320 | iter != codecs->end();) { |
| 321 | if (iter->id == codec_id) { |
| 322 | iter = codecs->erase(iter); |
| 323 | } else { |
| 324 | ++iter; |
| 325 | } |
| 326 | } |
| 327 | } |
| 328 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 329 | template <typename IdStruct> |
| 330 | class UsedIds { |
| 331 | public: |
| 332 | UsedIds(int min_allowed_id, int max_allowed_id) |
| 333 | : min_allowed_id_(min_allowed_id), |
| 334 | max_allowed_id_(max_allowed_id), |
| 335 | next_id_(max_allowed_id) { |
| 336 | } |
| 337 | |
| 338 | // Loops through all Id in |ids| and changes its id if it is |
| 339 | // already in use by another IdStruct. Call this methods with all Id |
| 340 | // in a session description to make sure no duplicate ids exists. |
| 341 | // Note that typename Id must be a type of IdStruct. |
| 342 | template <typename Id> |
| 343 | void FindAndSetIdUsed(std::vector<Id>* ids) { |
| 344 | for (typename std::vector<Id>::iterator it = ids->begin(); |
| 345 | it != ids->end(); ++it) { |
| 346 | FindAndSetIdUsed(&*it); |
| 347 | } |
| 348 | } |
| 349 | |
| 350 | // Finds and sets an unused id if the |idstruct| id is already in use. |
| 351 | void FindAndSetIdUsed(IdStruct* idstruct) { |
| 352 | const int original_id = idstruct->id; |
| 353 | int new_id = idstruct->id; |
| 354 | |
| 355 | if (original_id > max_allowed_id_ || original_id < min_allowed_id_) { |
| 356 | // If the original id is not in range - this is an id that can't be |
| 357 | // dynamically changed. |
| 358 | return; |
| 359 | } |
| 360 | |
| 361 | if (IsIdUsed(original_id)) { |
| 362 | new_id = FindUnusedId(); |
| 363 | LOG(LS_WARNING) << "Duplicate id found. Reassigning from " << original_id |
| 364 | << " to " << new_id; |
| 365 | idstruct->id = new_id; |
| 366 | } |
| 367 | SetIdUsed(new_id); |
| 368 | } |
| 369 | |
| 370 | private: |
| 371 | // Returns the first unused id in reverse order. |
| 372 | // This hopefully reduce the risk of more collisions. We want to change the |
| 373 | // default ids as little as possible. |
| 374 | int FindUnusedId() { |
| 375 | while (IsIdUsed(next_id_) && next_id_ >= min_allowed_id_) { |
| 376 | --next_id_; |
| 377 | } |
| 378 | ASSERT(next_id_ >= min_allowed_id_); |
| 379 | return next_id_; |
| 380 | } |
| 381 | |
| 382 | bool IsIdUsed(int new_id) { |
| 383 | return id_set_.find(new_id) != id_set_.end(); |
| 384 | } |
| 385 | |
| 386 | void SetIdUsed(int new_id) { |
| 387 | id_set_.insert(new_id); |
| 388 | } |
| 389 | |
| 390 | const int min_allowed_id_; |
| 391 | const int max_allowed_id_; |
| 392 | int next_id_; |
| 393 | std::set<int> id_set_; |
| 394 | }; |
| 395 | |
| 396 | // Helper class used for finding duplicate RTP payload types among audio, video |
| 397 | // and data codecs. When bundle is used the payload types may not collide. |
| 398 | class UsedPayloadTypes : public UsedIds<Codec> { |
| 399 | public: |
| 400 | UsedPayloadTypes() |
| 401 | : UsedIds<Codec>(kDynamicPayloadTypeMin, kDynamicPayloadTypeMax) { |
| 402 | } |
| 403 | |
| 404 | |
| 405 | private: |
| 406 | static const int kDynamicPayloadTypeMin = 96; |
| 407 | static const int kDynamicPayloadTypeMax = 127; |
| 408 | }; |
| 409 | |
| 410 | // Helper class used for finding duplicate RTP Header extension ids among |
| 411 | // audio and video extensions. |
| 412 | class UsedRtpHeaderExtensionIds : public UsedIds<RtpHeaderExtension> { |
| 413 | public: |
| 414 | UsedRtpHeaderExtensionIds() |
| 415 | : UsedIds<RtpHeaderExtension>(kLocalIdMin, kLocalIdMax) { |
| 416 | } |
| 417 | |
| 418 | private: |
henrike@webrtc.org | 79047f9 | 2014-03-06 23:46:59 +0000 | [diff] [blame] | 419 | // Min and Max local identifier for one-byte header extensions, per RFC5285. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 420 | static const int kLocalIdMin = 1; |
henrike@webrtc.org | 79047f9 | 2014-03-06 23:46:59 +0000 | [diff] [blame] | 421 | static const int kLocalIdMax = 14; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 422 | }; |
| 423 | |
| 424 | static bool IsSctp(const MediaContentDescription* desc) { |
| 425 | return ((desc->protocol() == kMediaProtocolSctp) || |
| 426 | (desc->protocol() == kMediaProtocolDtlsSctp)); |
| 427 | } |
| 428 | |
| 429 | // Adds a StreamParams for each Stream in Streams with media type |
| 430 | // media_type to content_description. |
| 431 | // |current_params| - All currently known StreamParams of any media type. |
| 432 | template <class C> |
| 433 | static bool AddStreamParams( |
| 434 | MediaType media_type, |
| 435 | const MediaSessionOptions::Streams& streams, |
| 436 | StreamParamsVec* current_streams, |
| 437 | MediaContentDescriptionImpl<C>* content_description, |
| 438 | const bool add_legacy_stream) { |
Noah Richards | 2e7a098 | 2015-05-18 14:02:54 -0700 | [diff] [blame^] | 439 | const bool include_rtx_streams = |
| 440 | ContainsRtxCodec(content_description->codecs()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 441 | |
| 442 | if (streams.empty() && add_legacy_stream) { |
| 443 | // TODO(perkj): Remove this legacy stream when all apps use StreamParams. |
| 444 | std::vector<uint32> ssrcs; |
| 445 | if (IsSctp(content_description)) { |
| 446 | GenerateSctpSids(*current_streams, &ssrcs); |
| 447 | } else { |
Noah Richards | 2e7a098 | 2015-05-18 14:02:54 -0700 | [diff] [blame^] | 448 | int num_ssrcs = include_rtx_streams ? 2 : 1; |
wu@webrtc.org | cecfd18 | 2013-10-30 05:18:12 +0000 | [diff] [blame] | 449 | GenerateSsrcs(*current_streams, num_ssrcs, &ssrcs); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 450 | } |
Noah Richards | 2e7a098 | 2015-05-18 14:02:54 -0700 | [diff] [blame^] | 451 | if (include_rtx_streams) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 452 | content_description->AddLegacyStream(ssrcs[0], ssrcs[1]); |
| 453 | content_description->set_multistream(true); |
| 454 | } else { |
| 455 | content_description->AddLegacyStream(ssrcs[0]); |
| 456 | } |
| 457 | return true; |
| 458 | } |
| 459 | |
| 460 | MediaSessionOptions::Streams::const_iterator stream_it; |
| 461 | for (stream_it = streams.begin(); |
| 462 | stream_it != streams.end(); ++stream_it) { |
| 463 | if (stream_it->type != media_type) |
| 464 | continue; // Wrong media type. |
| 465 | |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 466 | const StreamParams* param = |
| 467 | GetStreamByIds(*current_streams, "", stream_it->id); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 468 | // groupid is empty for StreamParams generated using |
| 469 | // MediaSessionDescriptionFactory. |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 470 | if (!param) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 471 | // This is a new stream. |
| 472 | // Get a CNAME. Either new or same as one of the other synched streams. |
| 473 | std::string cname; |
| 474 | if (!GenerateCname(*current_streams, streams, stream_it->sync_label, |
| 475 | &cname)) { |
| 476 | return false; |
| 477 | } |
| 478 | |
| 479 | std::vector<uint32> ssrcs; |
| 480 | if (IsSctp(content_description)) { |
| 481 | GenerateSctpSids(*current_streams, &ssrcs); |
| 482 | } else { |
wu@webrtc.org | cecfd18 | 2013-10-30 05:18:12 +0000 | [diff] [blame] | 483 | GenerateSsrcs(*current_streams, stream_it->num_sim_layers, &ssrcs); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 484 | } |
| 485 | StreamParams stream_param; |
| 486 | stream_param.id = stream_it->id; |
wu@webrtc.org | cecfd18 | 2013-10-30 05:18:12 +0000 | [diff] [blame] | 487 | // Add the generated ssrc. |
| 488 | for (size_t i = 0; i < ssrcs.size(); ++i) { |
| 489 | stream_param.ssrcs.push_back(ssrcs[i]); |
| 490 | } |
| 491 | if (stream_it->num_sim_layers > 1) { |
| 492 | SsrcGroup group(kSimSsrcGroupSemantics, stream_param.ssrcs); |
| 493 | stream_param.ssrc_groups.push_back(group); |
| 494 | } |
Noah Richards | 2e7a098 | 2015-05-18 14:02:54 -0700 | [diff] [blame^] | 495 | // Generate extra ssrcs for include_rtx_streams case. |
| 496 | if (include_rtx_streams) { |
| 497 | // Generate an RTX ssrc for every ssrc in the group. |
| 498 | std::vector<uint32> rtx_ssrcs; |
| 499 | GenerateSsrcs(*current_streams, static_cast<int>(ssrcs.size()), |
| 500 | &rtx_ssrcs); |
| 501 | for (size_t i = 0; i < ssrcs.size(); ++i) { |
| 502 | stream_param.AddFidSsrc(ssrcs[i], rtx_ssrcs[i]); |
| 503 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 504 | content_description->set_multistream(true); |
| 505 | } |
| 506 | stream_param.cname = cname; |
| 507 | stream_param.sync_label = stream_it->sync_label; |
| 508 | content_description->AddStream(stream_param); |
| 509 | |
| 510 | // Store the new StreamParams in current_streams. |
| 511 | // This is necessary so that we can use the CNAME for other media types. |
| 512 | current_streams->push_back(stream_param); |
| 513 | } else { |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 514 | content_description->AddStream(*param); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 515 | } |
| 516 | } |
| 517 | return true; |
| 518 | } |
| 519 | |
| 520 | // Updates the transport infos of the |sdesc| according to the given |
| 521 | // |bundle_group|. The transport infos of the content names within the |
| 522 | // |bundle_group| should be updated to use the ufrag and pwd of the first |
| 523 | // content within the |bundle_group|. |
| 524 | static bool UpdateTransportInfoForBundle(const ContentGroup& bundle_group, |
| 525 | SessionDescription* sdesc) { |
| 526 | // The bundle should not be empty. |
| 527 | if (!sdesc || !bundle_group.FirstContentName()) { |
| 528 | return false; |
| 529 | } |
| 530 | |
| 531 | // We should definitely have a transport for the first content. |
| 532 | std::string selected_content_name = *bundle_group.FirstContentName(); |
| 533 | const TransportInfo* selected_transport_info = |
| 534 | sdesc->GetTransportInfoByName(selected_content_name); |
| 535 | if (!selected_transport_info) { |
| 536 | return false; |
| 537 | } |
| 538 | |
| 539 | // Set the other contents to use the same ICE credentials. |
| 540 | const std::string selected_ufrag = |
| 541 | selected_transport_info->description.ice_ufrag; |
| 542 | const std::string selected_pwd = |
| 543 | selected_transport_info->description.ice_pwd; |
| 544 | for (TransportInfos::iterator it = |
| 545 | sdesc->transport_infos().begin(); |
| 546 | it != sdesc->transport_infos().end(); ++it) { |
| 547 | if (bundle_group.HasContentName(it->content_name) && |
| 548 | it->content_name != selected_content_name) { |
| 549 | it->description.ice_ufrag = selected_ufrag; |
| 550 | it->description.ice_pwd = selected_pwd; |
| 551 | } |
| 552 | } |
| 553 | return true; |
| 554 | } |
| 555 | |
| 556 | // Gets the CryptoParamsVec of the given |content_name| from |sdesc|, and |
| 557 | // sets it to |cryptos|. |
| 558 | static bool GetCryptosByName(const SessionDescription* sdesc, |
| 559 | const std::string& content_name, |
| 560 | CryptoParamsVec* cryptos) { |
| 561 | if (!sdesc || !cryptos) { |
| 562 | return false; |
| 563 | } |
| 564 | |
| 565 | const ContentInfo* content = sdesc->GetContentByName(content_name); |
| 566 | if (!IsMediaContent(content) || !content->description) { |
| 567 | return false; |
| 568 | } |
| 569 | |
| 570 | const MediaContentDescription* media_desc = |
| 571 | static_cast<const MediaContentDescription*>(content->description); |
| 572 | *cryptos = media_desc->cryptos(); |
| 573 | return true; |
| 574 | } |
| 575 | |
| 576 | // Predicate function used by the remove_if. |
| 577 | // Returns true if the |crypto|'s cipher_suite is not found in |filter|. |
| 578 | static bool CryptoNotFound(const CryptoParams crypto, |
| 579 | const CryptoParamsVec* filter) { |
| 580 | if (filter == NULL) { |
| 581 | return true; |
| 582 | } |
| 583 | for (CryptoParamsVec::const_iterator it = filter->begin(); |
| 584 | it != filter->end(); ++it) { |
| 585 | if (it->cipher_suite == crypto.cipher_suite) { |
| 586 | return false; |
| 587 | } |
| 588 | } |
| 589 | return true; |
| 590 | } |
| 591 | |
| 592 | // Prunes the |target_cryptos| by removing the crypto params (cipher_suite) |
| 593 | // which are not available in |filter|. |
| 594 | static void PruneCryptos(const CryptoParamsVec& filter, |
| 595 | CryptoParamsVec* target_cryptos) { |
| 596 | if (!target_cryptos) { |
| 597 | return; |
| 598 | } |
| 599 | target_cryptos->erase(std::remove_if(target_cryptos->begin(), |
| 600 | target_cryptos->end(), |
| 601 | bind2nd(ptr_fun(CryptoNotFound), |
| 602 | &filter)), |
| 603 | target_cryptos->end()); |
| 604 | } |
| 605 | |
| 606 | static bool IsRtpContent(SessionDescription* sdesc, |
| 607 | const std::string& content_name) { |
| 608 | bool is_rtp = false; |
| 609 | ContentInfo* content = sdesc->GetContentByName(content_name); |
| 610 | if (IsMediaContent(content)) { |
| 611 | MediaContentDescription* media_desc = |
| 612 | static_cast<MediaContentDescription*>(content->description); |
| 613 | if (!media_desc) { |
| 614 | return false; |
| 615 | } |
| 616 | is_rtp = media_desc->protocol().empty() || |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 617 | rtc::starts_with(media_desc->protocol().data(), |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 618 | kMediaProtocolRtpPrefix); |
| 619 | } |
| 620 | return is_rtp; |
| 621 | } |
| 622 | |
| 623 | // Updates the crypto parameters of the |sdesc| according to the given |
| 624 | // |bundle_group|. The crypto parameters of all the contents within the |
| 625 | // |bundle_group| should be updated to use the common subset of the |
| 626 | // available cryptos. |
| 627 | static bool UpdateCryptoParamsForBundle(const ContentGroup& bundle_group, |
| 628 | SessionDescription* sdesc) { |
| 629 | // The bundle should not be empty. |
| 630 | if (!sdesc || !bundle_group.FirstContentName()) { |
| 631 | return false; |
| 632 | } |
| 633 | |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 634 | bool common_cryptos_needed = false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 635 | // Get the common cryptos. |
| 636 | const ContentNames& content_names = bundle_group.content_names(); |
| 637 | CryptoParamsVec common_cryptos; |
| 638 | for (ContentNames::const_iterator it = content_names.begin(); |
| 639 | it != content_names.end(); ++it) { |
| 640 | if (!IsRtpContent(sdesc, *it)) { |
| 641 | continue; |
| 642 | } |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 643 | // The common cryptos are needed if any of the content does not have DTLS |
| 644 | // enabled. |
| 645 | if (!sdesc->GetTransportInfoByName(*it)->description.secure()) { |
| 646 | common_cryptos_needed = true; |
| 647 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 648 | if (it == content_names.begin()) { |
| 649 | // Initial the common_cryptos with the first content in the bundle group. |
| 650 | if (!GetCryptosByName(sdesc, *it, &common_cryptos)) { |
| 651 | return false; |
| 652 | } |
| 653 | if (common_cryptos.empty()) { |
| 654 | // If there's no crypto params, we should just return. |
| 655 | return true; |
| 656 | } |
| 657 | } else { |
| 658 | CryptoParamsVec cryptos; |
| 659 | if (!GetCryptosByName(sdesc, *it, &cryptos)) { |
| 660 | return false; |
| 661 | } |
| 662 | PruneCryptos(cryptos, &common_cryptos); |
| 663 | } |
| 664 | } |
| 665 | |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 666 | if (common_cryptos.empty() && common_cryptos_needed) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 667 | return false; |
| 668 | } |
| 669 | |
| 670 | // Update to use the common cryptos. |
| 671 | for (ContentNames::const_iterator it = content_names.begin(); |
| 672 | it != content_names.end(); ++it) { |
| 673 | if (!IsRtpContent(sdesc, *it)) { |
| 674 | continue; |
| 675 | } |
| 676 | ContentInfo* content = sdesc->GetContentByName(*it); |
| 677 | if (IsMediaContent(content)) { |
| 678 | MediaContentDescription* media_desc = |
| 679 | static_cast<MediaContentDescription*>(content->description); |
| 680 | if (!media_desc) { |
| 681 | return false; |
| 682 | } |
| 683 | media_desc->set_cryptos(common_cryptos); |
| 684 | } |
| 685 | } |
| 686 | return true; |
| 687 | } |
| 688 | |
| 689 | template <class C> |
| 690 | static bool ContainsRtxCodec(const std::vector<C>& codecs) { |
| 691 | typename std::vector<C>::const_iterator it; |
| 692 | for (it = codecs.begin(); it != codecs.end(); ++it) { |
| 693 | if (IsRtxCodec(*it)) { |
| 694 | return true; |
| 695 | } |
| 696 | } |
| 697 | return false; |
| 698 | } |
| 699 | |
| 700 | template <class C> |
| 701 | static bool IsRtxCodec(const C& codec) { |
| 702 | return stricmp(codec.name.c_str(), kRtxCodecName) == 0; |
| 703 | } |
| 704 | |
| 705 | // Create a media content to be offered in a session-initiate, |
| 706 | // according to the given options.rtcp_mux, options.is_muc, |
| 707 | // options.streams, codecs, secure_transport, crypto, and streams. If we don't |
| 708 | // currently have crypto (in current_cryptos) and it is enabled (in |
| 709 | // secure_policy), crypto is created (according to crypto_suites). If |
| 710 | // add_legacy_stream is true, and current_streams is empty, a legacy |
| 711 | // stream is created. The created content is added to the offer. |
| 712 | template <class C> |
| 713 | static bool CreateMediaContentOffer( |
| 714 | const MediaSessionOptions& options, |
| 715 | const std::vector<C>& codecs, |
henrike@webrtc.org | b90991d | 2014-03-04 19:54:57 +0000 | [diff] [blame] | 716 | const SecurePolicy& secure_policy, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 717 | const CryptoParamsVec* current_cryptos, |
| 718 | const std::vector<std::string>& crypto_suites, |
| 719 | const RtpHeaderExtensions& rtp_extensions, |
| 720 | bool add_legacy_stream, |
| 721 | StreamParamsVec* current_streams, |
| 722 | MediaContentDescriptionImpl<C>* offer) { |
| 723 | offer->AddCodecs(codecs); |
| 724 | offer->SortCodecs(); |
| 725 | |
henrike@webrtc.org | b90991d | 2014-03-04 19:54:57 +0000 | [diff] [blame] | 726 | if (secure_policy == SEC_REQUIRED) { |
| 727 | offer->set_crypto_required(CT_SDES); |
| 728 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 729 | offer->set_rtcp_mux(options.rtcp_mux_enabled); |
| 730 | offer->set_multistream(options.is_muc); |
| 731 | offer->set_rtp_header_extensions(rtp_extensions); |
| 732 | |
| 733 | if (!AddStreamParams( |
| 734 | offer->type(), options.streams, current_streams, |
| 735 | offer, add_legacy_stream)) { |
| 736 | return false; |
| 737 | } |
| 738 | |
| 739 | #ifdef HAVE_SRTP |
| 740 | if (secure_policy != SEC_DISABLED) { |
| 741 | if (current_cryptos) { |
| 742 | AddMediaCryptos(*current_cryptos, offer); |
| 743 | } |
| 744 | if (offer->cryptos().empty()) { |
| 745 | if (!CreateMediaCryptos(crypto_suites, offer)) { |
| 746 | return false; |
| 747 | } |
| 748 | } |
| 749 | } |
| 750 | #endif |
| 751 | |
henrike@webrtc.org | b90991d | 2014-03-04 19:54:57 +0000 | [diff] [blame] | 752 | if (offer->crypto_required() == CT_SDES && offer->cryptos().empty()) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 753 | return false; |
| 754 | } |
| 755 | return true; |
| 756 | } |
| 757 | |
| 758 | template <class C> |
changbin.shao@webrtc.org | 2d25b44 | 2015-03-16 04:14:34 +0000 | [diff] [blame] | 759 | static bool ReferencedCodecsMatch(const std::vector<C>& codecs1, |
| 760 | const std::string& codec1_id_str, |
| 761 | const std::vector<C>& codecs2, |
| 762 | const std::string& codec2_id_str) { |
| 763 | int codec1_id; |
| 764 | int codec2_id; |
| 765 | C codec1; |
| 766 | C codec2; |
| 767 | if (!rtc::FromString(codec1_id_str, &codec1_id) || |
| 768 | !rtc::FromString(codec2_id_str, &codec2_id) || |
| 769 | !FindCodecById(codecs1, codec1_id, &codec1) || |
| 770 | !FindCodecById(codecs2, codec2_id, &codec2)) { |
| 771 | return false; |
| 772 | } |
| 773 | return codec1.Matches(codec2); |
| 774 | } |
| 775 | |
| 776 | template <class C> |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 777 | static void NegotiateCodecs(const std::vector<C>& local_codecs, |
| 778 | const std::vector<C>& offered_codecs, |
| 779 | std::vector<C>* negotiated_codecs) { |
| 780 | typename std::vector<C>::const_iterator ours; |
| 781 | for (ours = local_codecs.begin(); |
| 782 | ours != local_codecs.end(); ++ours) { |
| 783 | typename std::vector<C>::const_iterator theirs; |
| 784 | for (theirs = offered_codecs.begin(); |
| 785 | theirs != offered_codecs.end(); ++theirs) { |
| 786 | if (ours->Matches(*theirs)) { |
| 787 | C negotiated = *ours; |
| 788 | negotiated.IntersectFeedbackParams(*theirs); |
| 789 | if (IsRtxCodec(negotiated)) { |
changbin.shao@webrtc.org | 2d25b44 | 2015-03-16 04:14:34 +0000 | [diff] [blame] | 790 | std::string offered_apt_value; |
| 791 | std::string local_apt_value; |
| 792 | if (!ours->GetParam(kCodecParamAssociatedPayloadType, |
| 793 | &local_apt_value) || |
| 794 | !theirs->GetParam(kCodecParamAssociatedPayloadType, |
| 795 | &offered_apt_value)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 796 | LOG(LS_WARNING) << "RTX missing associated payload type."; |
| 797 | continue; |
| 798 | } |
changbin.shao@webrtc.org | 2d25b44 | 2015-03-16 04:14:34 +0000 | [diff] [blame] | 799 | // Only negotiate RTX if kCodecParamAssociatedPayloadType has been |
| 800 | // set in local and remote codecs, and they match. |
| 801 | if (!ReferencedCodecsMatch(local_codecs, local_apt_value, |
| 802 | offered_codecs, offered_apt_value)) { |
| 803 | LOG(LS_WARNING) << "RTX associated codecs don't match."; |
| 804 | continue; |
| 805 | } |
| 806 | negotiated.SetParam(kCodecParamAssociatedPayloadType, |
| 807 | offered_apt_value); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 808 | } |
changbin.shao@webrtc.org | 2d25b44 | 2015-03-16 04:14:34 +0000 | [diff] [blame] | 809 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 810 | negotiated.id = theirs->id; |
wu@webrtc.org | ff1b1bf | 2014-06-20 20:57:42 +0000 | [diff] [blame] | 811 | // RFC3264: Although the answerer MAY list the formats in their desired |
| 812 | // order of preference, it is RECOMMENDED that unless there is a |
| 813 | // specific reason, the answerer list formats in the same relative order |
| 814 | // they were present in the offer. |
| 815 | negotiated.preference = theirs->preference; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 816 | negotiated_codecs->push_back(negotiated); |
| 817 | } |
| 818 | } |
| 819 | } |
| 820 | } |
| 821 | |
| 822 | template <class C> |
| 823 | static bool FindMatchingCodec(const std::vector<C>& codecs, |
| 824 | const C& codec_to_match, |
| 825 | C* found_codec) { |
| 826 | for (typename std::vector<C>::const_iterator it = codecs.begin(); |
| 827 | it != codecs.end(); ++it) { |
| 828 | if (it->Matches(codec_to_match)) { |
| 829 | if (found_codec != NULL) { |
| 830 | *found_codec= *it; |
| 831 | } |
| 832 | return true; |
| 833 | } |
| 834 | } |
| 835 | return false; |
| 836 | } |
| 837 | |
| 838 | // Adds all codecs from |reference_codecs| to |offered_codecs| that dont' |
| 839 | // already exist in |offered_codecs| and ensure the payload types don't |
| 840 | // collide. |
| 841 | template <class C> |
| 842 | static void FindCodecsToOffer( |
| 843 | const std::vector<C>& reference_codecs, |
| 844 | std::vector<C>* offered_codecs, |
| 845 | UsedPayloadTypes* used_pltypes) { |
| 846 | |
| 847 | typedef std::map<int, C> RtxCodecReferences; |
| 848 | RtxCodecReferences new_rtx_codecs; |
| 849 | |
| 850 | // Find all new RTX codecs. |
| 851 | for (typename std::vector<C>::const_iterator it = reference_codecs.begin(); |
| 852 | it != reference_codecs.end(); ++it) { |
| 853 | if (!FindMatchingCodec<C>(*offered_codecs, *it, NULL) && IsRtxCodec(*it)) { |
| 854 | C rtx_codec = *it; |
| 855 | int referenced_pl_type = |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 856 | rtc::FromString<int>(0, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 857 | rtx_codec.params[kCodecParamAssociatedPayloadType]); |
| 858 | new_rtx_codecs.insert(std::pair<int, C>(referenced_pl_type, |
| 859 | rtx_codec)); |
| 860 | } |
| 861 | } |
| 862 | |
| 863 | // Add all new codecs that are not RTX codecs. |
| 864 | for (typename std::vector<C>::const_iterator it = reference_codecs.begin(); |
| 865 | it != reference_codecs.end(); ++it) { |
| 866 | if (!FindMatchingCodec<C>(*offered_codecs, *it, NULL) && !IsRtxCodec(*it)) { |
| 867 | C codec = *it; |
| 868 | int original_payload_id = codec.id; |
| 869 | used_pltypes->FindAndSetIdUsed(&codec); |
| 870 | offered_codecs->push_back(codec); |
| 871 | |
| 872 | // If this codec is referenced by a new RTX codec, update the reference |
| 873 | // in the RTX codec with the new payload type. |
| 874 | typename RtxCodecReferences::iterator rtx_it = |
| 875 | new_rtx_codecs.find(original_payload_id); |
| 876 | if (rtx_it != new_rtx_codecs.end()) { |
| 877 | C& rtx_codec = rtx_it->second; |
| 878 | rtx_codec.params[kCodecParamAssociatedPayloadType] = |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 879 | rtc::ToString(codec.id); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 880 | } |
| 881 | } |
| 882 | } |
| 883 | |
| 884 | // Add all new RTX codecs. |
| 885 | for (typename RtxCodecReferences::iterator it = new_rtx_codecs.begin(); |
| 886 | it != new_rtx_codecs.end(); ++it) { |
| 887 | C& rtx_codec = it->second; |
| 888 | used_pltypes->FindAndSetIdUsed(&rtx_codec); |
| 889 | offered_codecs->push_back(rtx_codec); |
| 890 | } |
| 891 | } |
| 892 | |
| 893 | |
| 894 | static bool FindByUri(const RtpHeaderExtensions& extensions, |
| 895 | const RtpHeaderExtension& ext_to_match, |
| 896 | RtpHeaderExtension* found_extension) { |
| 897 | for (RtpHeaderExtensions::const_iterator it = extensions.begin(); |
| 898 | it != extensions.end(); ++it) { |
| 899 | // We assume that all URIs are given in a canonical format. |
| 900 | if (it->uri == ext_to_match.uri) { |
| 901 | if (found_extension != NULL) { |
henrike@webrtc.org | 79047f9 | 2014-03-06 23:46:59 +0000 | [diff] [blame] | 902 | *found_extension = *it; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 903 | } |
| 904 | return true; |
| 905 | } |
| 906 | } |
| 907 | return false; |
| 908 | } |
| 909 | |
| 910 | static void FindAndSetRtpHdrExtUsed( |
| 911 | const RtpHeaderExtensions& reference_extensions, |
| 912 | RtpHeaderExtensions* offered_extensions, |
henrike@webrtc.org | 79047f9 | 2014-03-06 23:46:59 +0000 | [diff] [blame] | 913 | const RtpHeaderExtensions& other_extensions, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 914 | UsedRtpHeaderExtensionIds* used_extensions) { |
| 915 | for (RtpHeaderExtensions::const_iterator it = reference_extensions.begin(); |
| 916 | it != reference_extensions.end(); ++it) { |
| 917 | if (!FindByUri(*offered_extensions, *it, NULL)) { |
henrike@webrtc.org | 79047f9 | 2014-03-06 23:46:59 +0000 | [diff] [blame] | 918 | RtpHeaderExtension ext; |
| 919 | if (!FindByUri(other_extensions, *it, &ext)) { |
| 920 | ext = *it; |
| 921 | used_extensions->FindAndSetIdUsed(&ext); |
| 922 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 923 | offered_extensions->push_back(ext); |
| 924 | } |
| 925 | } |
| 926 | } |
| 927 | |
| 928 | static void NegotiateRtpHeaderExtensions( |
| 929 | const RtpHeaderExtensions& local_extensions, |
| 930 | const RtpHeaderExtensions& offered_extensions, |
| 931 | RtpHeaderExtensions* negotiated_extenstions) { |
| 932 | RtpHeaderExtensions::const_iterator ours; |
| 933 | for (ours = local_extensions.begin(); |
| 934 | ours != local_extensions.end(); ++ours) { |
| 935 | RtpHeaderExtension theirs; |
| 936 | if (FindByUri(offered_extensions, *ours, &theirs)) { |
| 937 | // We respond with their RTP header extension id. |
| 938 | negotiated_extenstions->push_back(theirs); |
| 939 | } |
| 940 | } |
| 941 | } |
| 942 | |
| 943 | static void StripCNCodecs(AudioCodecs* audio_codecs) { |
| 944 | AudioCodecs::iterator iter = audio_codecs->begin(); |
| 945 | while (iter != audio_codecs->end()) { |
| 946 | if (stricmp(iter->name.c_str(), kComfortNoiseCodecName) == 0) { |
| 947 | iter = audio_codecs->erase(iter); |
| 948 | } else { |
| 949 | ++iter; |
| 950 | } |
| 951 | } |
| 952 | } |
| 953 | |
| 954 | // Create a media content to be answered in a session-accept, |
| 955 | // according to the given options.rtcp_mux, options.streams, codecs, |
| 956 | // crypto, and streams. If we don't currently have crypto (in |
| 957 | // current_cryptos) and it is enabled (in secure_policy), crypto is |
| 958 | // created (according to crypto_suites). If add_legacy_stream is |
| 959 | // true, and current_streams is empty, a legacy stream is created. |
| 960 | // The codecs, rtcp_mux, and crypto are all negotiated with the offer |
| 961 | // from the incoming session-initiate. If the negotiation fails, this |
| 962 | // method returns false. The created content is added to the offer. |
| 963 | template <class C> |
| 964 | static bool CreateMediaContentAnswer( |
| 965 | const MediaContentDescriptionImpl<C>* offer, |
| 966 | const MediaSessionOptions& options, |
| 967 | const std::vector<C>& local_codecs, |
henrike@webrtc.org | b90991d | 2014-03-04 19:54:57 +0000 | [diff] [blame] | 968 | const SecurePolicy& sdes_policy, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 969 | const CryptoParamsVec* current_cryptos, |
| 970 | const RtpHeaderExtensions& local_rtp_extenstions, |
| 971 | StreamParamsVec* current_streams, |
| 972 | bool add_legacy_stream, |
| 973 | bool bundle_enabled, |
| 974 | MediaContentDescriptionImpl<C>* answer) { |
| 975 | std::vector<C> negotiated_codecs; |
| 976 | NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs); |
| 977 | answer->AddCodecs(negotiated_codecs); |
| 978 | answer->SortCodecs(); |
| 979 | answer->set_protocol(offer->protocol()); |
| 980 | RtpHeaderExtensions negotiated_rtp_extensions; |
| 981 | NegotiateRtpHeaderExtensions(local_rtp_extenstions, |
| 982 | offer->rtp_header_extensions(), |
| 983 | &negotiated_rtp_extensions); |
| 984 | answer->set_rtp_header_extensions(negotiated_rtp_extensions); |
| 985 | |
| 986 | answer->set_rtcp_mux(options.rtcp_mux_enabled && offer->rtcp_mux()); |
| 987 | |
| 988 | if (sdes_policy != SEC_DISABLED) { |
| 989 | CryptoParams crypto; |
| 990 | if (SelectCrypto(offer, bundle_enabled, &crypto)) { |
| 991 | if (current_cryptos) { |
| 992 | FindMatchingCrypto(*current_cryptos, crypto, &crypto); |
| 993 | } |
| 994 | answer->AddCrypto(crypto); |
| 995 | } |
| 996 | } |
| 997 | |
| 998 | if (answer->cryptos().empty() && |
henrike@webrtc.org | b90991d | 2014-03-04 19:54:57 +0000 | [diff] [blame] | 999 | (offer->crypto_required() == CT_SDES || sdes_policy == SEC_REQUIRED)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1000 | return false; |
| 1001 | } |
| 1002 | |
| 1003 | if (!AddStreamParams( |
| 1004 | answer->type(), options.streams, current_streams, |
| 1005 | answer, add_legacy_stream)) { |
| 1006 | return false; // Something went seriously wrong. |
| 1007 | } |
| 1008 | |
| 1009 | // Make sure the answer media content direction is per default set as |
| 1010 | // described in RFC3264 section 6.1. |
| 1011 | switch (offer->direction()) { |
| 1012 | case MD_INACTIVE: |
| 1013 | answer->set_direction(MD_INACTIVE); |
| 1014 | break; |
| 1015 | case MD_SENDONLY: |
| 1016 | answer->set_direction(MD_RECVONLY); |
| 1017 | break; |
| 1018 | case MD_RECVONLY: |
| 1019 | answer->set_direction(MD_SENDONLY); |
| 1020 | break; |
| 1021 | case MD_SENDRECV: |
| 1022 | answer->set_direction(MD_SENDRECV); |
| 1023 | break; |
| 1024 | default: |
| 1025 | break; |
| 1026 | } |
| 1027 | |
| 1028 | return true; |
| 1029 | } |
| 1030 | |
| 1031 | static bool IsMediaProtocolSupported(MediaType type, |
jiayl@webrtc.org | 8dcd43c | 2014-05-29 22:07:59 +0000 | [diff] [blame] | 1032 | const std::string& protocol, |
| 1033 | bool secure_transport) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1034 | // Data channels can have a protocol of SCTP or SCTP/DTLS. |
| 1035 | if (type == MEDIA_TYPE_DATA && |
jiayl@webrtc.org | 8dcd43c | 2014-05-29 22:07:59 +0000 | [diff] [blame] | 1036 | ((protocol == kMediaProtocolSctp && !secure_transport)|| |
| 1037 | (protocol == kMediaProtocolDtlsSctp && secure_transport))) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1038 | return true; |
| 1039 | } |
jiayl@webrtc.org | 8dcd43c | 2014-05-29 22:07:59 +0000 | [diff] [blame] | 1040 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1041 | // Since not all applications serialize and deserialize the media protocol, |
| 1042 | // we will have to accept |protocol| to be empty. |
jiayl@webrtc.org | 8dcd43c | 2014-05-29 22:07:59 +0000 | [diff] [blame] | 1043 | return protocol == kMediaProtocolAvpf || protocol.empty() || |
| 1044 | protocol == kMediaProtocolSavpf || |
| 1045 | (protocol == kMediaProtocolDtlsSavpf && secure_transport); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1046 | } |
| 1047 | |
| 1048 | static void SetMediaProtocol(bool secure_transport, |
| 1049 | MediaContentDescription* desc) { |
| 1050 | if (!desc->cryptos().empty() || secure_transport) |
| 1051 | desc->set_protocol(kMediaProtocolSavpf); |
| 1052 | else |
| 1053 | desc->set_protocol(kMediaProtocolAvpf); |
| 1054 | } |
| 1055 | |
mallinath@webrtc.org | 19f27e6 | 2013-10-13 17:18:27 +0000 | [diff] [blame] | 1056 | // Gets the TransportInfo of the given |content_name| from the |
| 1057 | // |current_description|. If doesn't exist, returns a new one. |
| 1058 | static const TransportDescription* GetTransportDescription( |
| 1059 | const std::string& content_name, |
| 1060 | const SessionDescription* current_description) { |
| 1061 | const TransportDescription* desc = NULL; |
| 1062 | if (current_description) { |
| 1063 | const TransportInfo* info = |
| 1064 | current_description->GetTransportInfoByName(content_name); |
| 1065 | if (info) { |
| 1066 | desc = &info->description; |
| 1067 | } |
| 1068 | } |
| 1069 | return desc; |
| 1070 | } |
| 1071 | |
| 1072 | // Gets the current DTLS state from the transport description. |
| 1073 | static bool IsDtlsActive( |
| 1074 | const std::string& content_name, |
| 1075 | const SessionDescription* current_description) { |
| 1076 | if (!current_description) |
| 1077 | return false; |
| 1078 | |
| 1079 | const ContentInfo* content = |
| 1080 | current_description->GetContentByName(content_name); |
| 1081 | if (!content) |
| 1082 | return false; |
| 1083 | |
| 1084 | const TransportDescription* current_tdesc = |
| 1085 | GetTransportDescription(content_name, current_description); |
| 1086 | if (!current_tdesc) |
| 1087 | return false; |
| 1088 | |
| 1089 | return current_tdesc->secure(); |
| 1090 | } |
| 1091 | |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1092 | std::string MediaTypeToString(MediaType type) { |
| 1093 | std::string type_str; |
| 1094 | switch (type) { |
| 1095 | case MEDIA_TYPE_AUDIO: |
| 1096 | type_str = "audio"; |
| 1097 | break; |
| 1098 | case MEDIA_TYPE_VIDEO: |
| 1099 | type_str = "video"; |
| 1100 | break; |
| 1101 | case MEDIA_TYPE_DATA: |
| 1102 | type_str = "data"; |
| 1103 | break; |
| 1104 | default: |
| 1105 | ASSERT(false); |
| 1106 | break; |
| 1107 | } |
| 1108 | return type_str; |
| 1109 | } |
| 1110 | |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1111 | void MediaSessionOptions::AddSendStream(MediaType type, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1112 | const std::string& id, |
| 1113 | const std::string& sync_label) { |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1114 | AddSendStreamInternal(type, id, sync_label, 1); |
wu@webrtc.org | cecfd18 | 2013-10-30 05:18:12 +0000 | [diff] [blame] | 1115 | } |
| 1116 | |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1117 | void MediaSessionOptions::AddSendVideoStream( |
wu@webrtc.org | cecfd18 | 2013-10-30 05:18:12 +0000 | [diff] [blame] | 1118 | const std::string& id, |
| 1119 | const std::string& sync_label, |
| 1120 | int num_sim_layers) { |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1121 | AddSendStreamInternal(MEDIA_TYPE_VIDEO, id, sync_label, num_sim_layers); |
wu@webrtc.org | cecfd18 | 2013-10-30 05:18:12 +0000 | [diff] [blame] | 1122 | } |
| 1123 | |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1124 | void MediaSessionOptions::AddSendStreamInternal( |
wu@webrtc.org | cecfd18 | 2013-10-30 05:18:12 +0000 | [diff] [blame] | 1125 | MediaType type, |
| 1126 | const std::string& id, |
| 1127 | const std::string& sync_label, |
| 1128 | int num_sim_layers) { |
| 1129 | streams.push_back(Stream(type, id, sync_label, num_sim_layers)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1130 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1131 | // If we haven't already set the data_channel_type, and we add a |
| 1132 | // stream, we assume it's an RTP data stream. |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1133 | if (type == MEDIA_TYPE_DATA && data_channel_type == DCT_NONE) |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1134 | data_channel_type = DCT_RTP; |
| 1135 | } |
| 1136 | |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1137 | void MediaSessionOptions::RemoveSendStream(MediaType type, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1138 | const std::string& id) { |
| 1139 | Streams::iterator stream_it = streams.begin(); |
| 1140 | for (; stream_it != streams.end(); ++stream_it) { |
| 1141 | if (stream_it->type == type && stream_it->id == id) { |
| 1142 | streams.erase(stream_it); |
| 1143 | return; |
| 1144 | } |
| 1145 | } |
| 1146 | ASSERT(false); |
| 1147 | } |
| 1148 | |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1149 | bool MediaSessionOptions::HasSendMediaStream(MediaType type) const { |
| 1150 | Streams::const_iterator stream_it = streams.begin(); |
| 1151 | for (; stream_it != streams.end(); ++stream_it) { |
| 1152 | if (stream_it->type == type) { |
| 1153 | return true; |
| 1154 | } |
| 1155 | } |
| 1156 | return false; |
| 1157 | } |
| 1158 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1159 | MediaSessionDescriptionFactory::MediaSessionDescriptionFactory( |
| 1160 | const TransportDescriptionFactory* transport_desc_factory) |
| 1161 | : secure_(SEC_DISABLED), |
| 1162 | add_legacy_(true), |
| 1163 | transport_desc_factory_(transport_desc_factory) { |
| 1164 | } |
| 1165 | |
| 1166 | MediaSessionDescriptionFactory::MediaSessionDescriptionFactory( |
| 1167 | ChannelManager* channel_manager, |
| 1168 | const TransportDescriptionFactory* transport_desc_factory) |
| 1169 | : secure_(SEC_DISABLED), |
| 1170 | add_legacy_(true), |
| 1171 | transport_desc_factory_(transport_desc_factory) { |
| 1172 | channel_manager->GetSupportedAudioCodecs(&audio_codecs_); |
| 1173 | channel_manager->GetSupportedAudioRtpHeaderExtensions(&audio_rtp_extensions_); |
| 1174 | channel_manager->GetSupportedVideoCodecs(&video_codecs_); |
| 1175 | channel_manager->GetSupportedVideoRtpHeaderExtensions(&video_rtp_extensions_); |
| 1176 | channel_manager->GetSupportedDataCodecs(&data_codecs_); |
| 1177 | } |
| 1178 | |
| 1179 | SessionDescription* MediaSessionDescriptionFactory::CreateOffer( |
| 1180 | const MediaSessionOptions& options, |
| 1181 | const SessionDescription* current_description) const { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1182 | scoped_ptr<SessionDescription> offer(new SessionDescription()); |
| 1183 | |
| 1184 | StreamParamsVec current_streams; |
| 1185 | GetCurrentStreamParams(current_description, ¤t_streams); |
| 1186 | |
| 1187 | AudioCodecs audio_codecs; |
| 1188 | VideoCodecs video_codecs; |
| 1189 | DataCodecs data_codecs; |
| 1190 | GetCodecsToOffer(current_description, &audio_codecs, &video_codecs, |
| 1191 | &data_codecs); |
| 1192 | |
| 1193 | if (!options.vad_enabled) { |
| 1194 | // If application doesn't want CN codecs in offer. |
| 1195 | StripCNCodecs(&audio_codecs); |
| 1196 | } |
| 1197 | |
| 1198 | RtpHeaderExtensions audio_rtp_extensions; |
| 1199 | RtpHeaderExtensions video_rtp_extensions; |
| 1200 | GetRtpHdrExtsToOffer(current_description, &audio_rtp_extensions, |
| 1201 | &video_rtp_extensions); |
| 1202 | |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1203 | bool audio_added = false; |
| 1204 | bool video_added = false; |
| 1205 | bool data_added = false; |
mallinath@webrtc.org | 19f27e6 | 2013-10-13 17:18:27 +0000 | [diff] [blame] | 1206 | |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1207 | // Iterate through the contents of |current_description| to maintain the order |
| 1208 | // of the m-lines in the new offer. |
| 1209 | if (current_description) { |
| 1210 | ContentInfos::const_iterator it = current_description->contents().begin(); |
| 1211 | for (; it != current_description->contents().end(); ++it) { |
jiayl@webrtc.org | 7d4891d | 2014-09-09 21:43:15 +0000 | [diff] [blame] | 1212 | if (IsMediaContentOfType(&*it, MEDIA_TYPE_AUDIO)) { |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1213 | if (!AddAudioContentForOffer(options, current_description, |
| 1214 | audio_rtp_extensions, audio_codecs, |
| 1215 | ¤t_streams, offer.get())) { |
| 1216 | return NULL; |
| 1217 | } |
| 1218 | audio_added = true; |
jiayl@webrtc.org | 7d4891d | 2014-09-09 21:43:15 +0000 | [diff] [blame] | 1219 | } else if (IsMediaContentOfType(&*it, MEDIA_TYPE_VIDEO)) { |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1220 | if (!AddVideoContentForOffer(options, current_description, |
| 1221 | video_rtp_extensions, video_codecs, |
| 1222 | ¤t_streams, offer.get())) { |
| 1223 | return NULL; |
| 1224 | } |
| 1225 | video_added = true; |
jiayl@webrtc.org | 7d4891d | 2014-09-09 21:43:15 +0000 | [diff] [blame] | 1226 | } else if (IsMediaContentOfType(&*it, MEDIA_TYPE_DATA)) { |
tommi@webrtc.org | f15dee6 | 2014-10-27 22:15:04 +0000 | [diff] [blame] | 1227 | MediaSessionOptions options_copy(options); |
| 1228 | if (IsSctp(static_cast<const MediaContentDescription*>( |
| 1229 | it->description))) { |
| 1230 | options_copy.data_channel_type = DCT_SCTP; |
| 1231 | } |
| 1232 | if (!AddDataContentForOffer(options_copy, current_description, |
| 1233 | &data_codecs, ¤t_streams, |
| 1234 | offer.get())) { |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1235 | return NULL; |
| 1236 | } |
| 1237 | data_added = true; |
jiayl@webrtc.org | 7d4891d | 2014-09-09 21:43:15 +0000 | [diff] [blame] | 1238 | } else { |
| 1239 | ASSERT(false); |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1240 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1241 | } |
| 1242 | } |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1243 | |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1244 | // Append contents that are not in |current_description|. |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1245 | if (!audio_added && options.has_audio() && |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1246 | !AddAudioContentForOffer(options, current_description, |
| 1247 | audio_rtp_extensions, audio_codecs, |
| 1248 | ¤t_streams, offer.get())) { |
| 1249 | return NULL; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1250 | } |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1251 | if (!video_added && options.has_video() && |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1252 | !AddVideoContentForOffer(options, current_description, |
| 1253 | video_rtp_extensions, video_codecs, |
| 1254 | ¤t_streams, offer.get())) { |
| 1255 | return NULL; |
| 1256 | } |
| 1257 | if (!data_added && options.has_data() && |
| 1258 | !AddDataContentForOffer(options, current_description, &data_codecs, |
| 1259 | ¤t_streams, offer.get())) { |
| 1260 | return NULL; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1261 | } |
| 1262 | |
| 1263 | // Bundle the contents together, if we've been asked to do so, and update any |
| 1264 | // parameters that need to be tweaked for BUNDLE. |
| 1265 | if (options.bundle_enabled) { |
| 1266 | ContentGroup offer_bundle(GROUP_TYPE_BUNDLE); |
| 1267 | for (ContentInfos::const_iterator content = offer->contents().begin(); |
| 1268 | content != offer->contents().end(); ++content) { |
| 1269 | offer_bundle.AddContentName(content->name); |
| 1270 | } |
| 1271 | offer->AddGroup(offer_bundle); |
| 1272 | if (!UpdateTransportInfoForBundle(offer_bundle, offer.get())) { |
| 1273 | LOG(LS_ERROR) << "CreateOffer failed to UpdateTransportInfoForBundle."; |
| 1274 | return NULL; |
| 1275 | } |
| 1276 | if (!UpdateCryptoParamsForBundle(offer_bundle, offer.get())) { |
| 1277 | LOG(LS_ERROR) << "CreateOffer failed to UpdateCryptoParamsForBundle."; |
| 1278 | return NULL; |
| 1279 | } |
| 1280 | } |
| 1281 | |
| 1282 | return offer.release(); |
| 1283 | } |
| 1284 | |
| 1285 | SessionDescription* MediaSessionDescriptionFactory::CreateAnswer( |
| 1286 | const SessionDescription* offer, const MediaSessionOptions& options, |
| 1287 | const SessionDescription* current_description) const { |
| 1288 | // The answer contains the intersection of the codecs in the offer with the |
| 1289 | // codecs we support, ordered by our local preference. As indicated by |
| 1290 | // XEP-0167, we retain the same payload ids from the offer in the answer. |
| 1291 | scoped_ptr<SessionDescription> answer(new SessionDescription()); |
| 1292 | |
| 1293 | StreamParamsVec current_streams; |
| 1294 | GetCurrentStreamParams(current_description, ¤t_streams); |
| 1295 | |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1296 | if (offer) { |
| 1297 | ContentInfos::const_iterator it = offer->contents().begin(); |
| 1298 | for (; it != offer->contents().end(); ++it) { |
| 1299 | if (IsMediaContentOfType(&*it, MEDIA_TYPE_AUDIO)) { |
| 1300 | if (!AddAudioContentForAnswer(offer, options, current_description, |
| 1301 | ¤t_streams, answer.get())) { |
| 1302 | return NULL; |
| 1303 | } |
| 1304 | } else if (IsMediaContentOfType(&*it, MEDIA_TYPE_VIDEO)) { |
| 1305 | if (!AddVideoContentForAnswer(offer, options, current_description, |
| 1306 | ¤t_streams, answer.get())) { |
| 1307 | return NULL; |
| 1308 | } |
| 1309 | } else { |
| 1310 | ASSERT(IsMediaContentOfType(&*it, MEDIA_TYPE_DATA)); |
| 1311 | if (!AddDataContentForAnswer(offer, options, current_description, |
| 1312 | ¤t_streams, answer.get())) { |
| 1313 | return NULL; |
| 1314 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1315 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1316 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1317 | } |
| 1318 | |
| 1319 | // If the offer supports BUNDLE, and we want to use it too, create a BUNDLE |
| 1320 | // group in the answer with the appropriate content names. |
| 1321 | if (offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled) { |
| 1322 | const ContentGroup* offer_bundle = offer->GetGroupByName(GROUP_TYPE_BUNDLE); |
| 1323 | ContentGroup answer_bundle(GROUP_TYPE_BUNDLE); |
| 1324 | for (ContentInfos::const_iterator content = answer->contents().begin(); |
| 1325 | content != answer->contents().end(); ++content) { |
| 1326 | if (!content->rejected && offer_bundle->HasContentName(content->name)) { |
| 1327 | answer_bundle.AddContentName(content->name); |
| 1328 | } |
| 1329 | } |
| 1330 | if (answer_bundle.FirstContentName()) { |
| 1331 | answer->AddGroup(answer_bundle); |
| 1332 | |
| 1333 | // Share the same ICE credentials and crypto params across all contents, |
| 1334 | // as BUNDLE requires. |
| 1335 | if (!UpdateTransportInfoForBundle(answer_bundle, answer.get())) { |
| 1336 | LOG(LS_ERROR) << "CreateAnswer failed to UpdateTransportInfoForBundle."; |
| 1337 | return NULL; |
| 1338 | } |
| 1339 | |
| 1340 | if (!UpdateCryptoParamsForBundle(answer_bundle, answer.get())) { |
| 1341 | LOG(LS_ERROR) << "CreateAnswer failed to UpdateCryptoParamsForBundle."; |
| 1342 | return NULL; |
| 1343 | } |
| 1344 | } |
| 1345 | } |
| 1346 | |
| 1347 | return answer.release(); |
| 1348 | } |
| 1349 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1350 | void MediaSessionDescriptionFactory::GetCodecsToOffer( |
| 1351 | const SessionDescription* current_description, |
| 1352 | AudioCodecs* audio_codecs, |
| 1353 | VideoCodecs* video_codecs, |
| 1354 | DataCodecs* data_codecs) const { |
| 1355 | UsedPayloadTypes used_pltypes; |
| 1356 | audio_codecs->clear(); |
| 1357 | video_codecs->clear(); |
| 1358 | data_codecs->clear(); |
| 1359 | |
| 1360 | |
| 1361 | // First - get all codecs from the current description if the media type |
| 1362 | // is used. |
| 1363 | // Add them to |used_pltypes| so the payloadtype is not reused if a new media |
| 1364 | // type is added. |
| 1365 | if (current_description) { |
| 1366 | const AudioContentDescription* audio = |
| 1367 | GetFirstAudioContentDescription(current_description); |
| 1368 | if (audio) { |
| 1369 | *audio_codecs = audio->codecs(); |
| 1370 | used_pltypes.FindAndSetIdUsed<AudioCodec>(audio_codecs); |
| 1371 | } |
| 1372 | const VideoContentDescription* video = |
| 1373 | GetFirstVideoContentDescription(current_description); |
| 1374 | if (video) { |
| 1375 | *video_codecs = video->codecs(); |
| 1376 | used_pltypes.FindAndSetIdUsed<VideoCodec>(video_codecs); |
| 1377 | } |
| 1378 | const DataContentDescription* data = |
| 1379 | GetFirstDataContentDescription(current_description); |
| 1380 | if (data) { |
| 1381 | *data_codecs = data->codecs(); |
| 1382 | used_pltypes.FindAndSetIdUsed<DataCodec>(data_codecs); |
| 1383 | } |
| 1384 | } |
| 1385 | |
| 1386 | // Add our codecs that are not in |current_description|. |
| 1387 | FindCodecsToOffer<AudioCodec>(audio_codecs_, audio_codecs, &used_pltypes); |
| 1388 | FindCodecsToOffer<VideoCodec>(video_codecs_, video_codecs, &used_pltypes); |
| 1389 | FindCodecsToOffer<DataCodec>(data_codecs_, data_codecs, &used_pltypes); |
| 1390 | } |
| 1391 | |
| 1392 | void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer( |
| 1393 | const SessionDescription* current_description, |
| 1394 | RtpHeaderExtensions* audio_extensions, |
| 1395 | RtpHeaderExtensions* video_extensions) const { |
henrike@webrtc.org | 79047f9 | 2014-03-06 23:46:59 +0000 | [diff] [blame] | 1396 | // All header extensions allocated from the same range to avoid potential |
| 1397 | // issues when using BUNDLE. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1398 | UsedRtpHeaderExtensionIds used_ids; |
| 1399 | audio_extensions->clear(); |
| 1400 | video_extensions->clear(); |
| 1401 | |
| 1402 | // First - get all extensions from the current description if the media type |
| 1403 | // is used. |
| 1404 | // Add them to |used_ids| so the local ids are not reused if a new media |
| 1405 | // type is added. |
| 1406 | if (current_description) { |
| 1407 | const AudioContentDescription* audio = |
| 1408 | GetFirstAudioContentDescription(current_description); |
| 1409 | if (audio) { |
| 1410 | *audio_extensions = audio->rtp_header_extensions(); |
| 1411 | used_ids.FindAndSetIdUsed(audio_extensions); |
| 1412 | } |
| 1413 | const VideoContentDescription* video = |
| 1414 | GetFirstVideoContentDescription(current_description); |
| 1415 | if (video) { |
| 1416 | *video_extensions = video->rtp_header_extensions(); |
| 1417 | used_ids.FindAndSetIdUsed(video_extensions); |
| 1418 | } |
| 1419 | } |
| 1420 | |
| 1421 | // Add our default RTP header extensions that are not in |
| 1422 | // |current_description|. |
| 1423 | FindAndSetRtpHdrExtUsed(audio_rtp_header_extensions(), audio_extensions, |
henrike@webrtc.org | 79047f9 | 2014-03-06 23:46:59 +0000 | [diff] [blame] | 1424 | *video_extensions, &used_ids); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1425 | FindAndSetRtpHdrExtUsed(video_rtp_header_extensions(), video_extensions, |
henrike@webrtc.org | 79047f9 | 2014-03-06 23:46:59 +0000 | [diff] [blame] | 1426 | *audio_extensions, &used_ids); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1427 | } |
| 1428 | |
| 1429 | bool MediaSessionDescriptionFactory::AddTransportOffer( |
| 1430 | const std::string& content_name, |
| 1431 | const TransportOptions& transport_options, |
| 1432 | const SessionDescription* current_desc, |
| 1433 | SessionDescription* offer_desc) const { |
| 1434 | if (!transport_desc_factory_) |
| 1435 | return false; |
| 1436 | const TransportDescription* current_tdesc = |
| 1437 | GetTransportDescription(content_name, current_desc); |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 1438 | rtc::scoped_ptr<TransportDescription> new_tdesc( |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1439 | transport_desc_factory_->CreateOffer(transport_options, current_tdesc)); |
| 1440 | bool ret = (new_tdesc.get() != NULL && |
| 1441 | offer_desc->AddTransportInfo(TransportInfo(content_name, *new_tdesc))); |
| 1442 | if (!ret) { |
| 1443 | LOG(LS_ERROR) |
| 1444 | << "Failed to AddTransportOffer, content name=" << content_name; |
| 1445 | } |
| 1446 | return ret; |
| 1447 | } |
| 1448 | |
| 1449 | TransportDescription* MediaSessionDescriptionFactory::CreateTransportAnswer( |
| 1450 | const std::string& content_name, |
| 1451 | const SessionDescription* offer_desc, |
| 1452 | const TransportOptions& transport_options, |
| 1453 | const SessionDescription* current_desc) const { |
| 1454 | if (!transport_desc_factory_) |
| 1455 | return NULL; |
| 1456 | const TransportDescription* offer_tdesc = |
| 1457 | GetTransportDescription(content_name, offer_desc); |
| 1458 | const TransportDescription* current_tdesc = |
| 1459 | GetTransportDescription(content_name, current_desc); |
| 1460 | return |
| 1461 | transport_desc_factory_->CreateAnswer(offer_tdesc, transport_options, |
| 1462 | current_tdesc); |
| 1463 | } |
| 1464 | |
| 1465 | bool MediaSessionDescriptionFactory::AddTransportAnswer( |
| 1466 | const std::string& content_name, |
| 1467 | const TransportDescription& transport_desc, |
| 1468 | SessionDescription* answer_desc) const { |
| 1469 | if (!answer_desc->AddTransportInfo(TransportInfo(content_name, |
| 1470 | transport_desc))) { |
| 1471 | LOG(LS_ERROR) |
| 1472 | << "Failed to AddTransportAnswer, content name=" << content_name; |
| 1473 | return false; |
| 1474 | } |
| 1475 | return true; |
| 1476 | } |
| 1477 | |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1478 | bool MediaSessionDescriptionFactory::AddAudioContentForOffer( |
| 1479 | const MediaSessionOptions& options, |
| 1480 | const SessionDescription* current_description, |
| 1481 | const RtpHeaderExtensions& audio_rtp_extensions, |
| 1482 | const AudioCodecs& audio_codecs, |
| 1483 | StreamParamsVec* current_streams, |
| 1484 | SessionDescription* desc) const { |
| 1485 | cricket::SecurePolicy sdes_policy = |
| 1486 | IsDtlsActive(CN_AUDIO, current_description) ? |
| 1487 | cricket::SEC_DISABLED : secure(); |
| 1488 | |
| 1489 | scoped_ptr<AudioContentDescription> audio(new AudioContentDescription()); |
| 1490 | std::vector<std::string> crypto_suites; |
| 1491 | GetSupportedAudioCryptoSuites(&crypto_suites); |
| 1492 | if (!CreateMediaContentOffer( |
| 1493 | options, |
| 1494 | audio_codecs, |
| 1495 | sdes_policy, |
| 1496 | GetCryptos(GetFirstAudioContentDescription(current_description)), |
| 1497 | crypto_suites, |
| 1498 | audio_rtp_extensions, |
| 1499 | add_legacy_, |
| 1500 | current_streams, |
| 1501 | audio.get())) { |
| 1502 | return false; |
| 1503 | } |
| 1504 | audio->set_lang(lang_); |
| 1505 | |
| 1506 | bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED); |
| 1507 | SetMediaProtocol(secure_transport, audio.get()); |
jiayl@webrtc.org | 7d4891d | 2014-09-09 21:43:15 +0000 | [diff] [blame] | 1508 | |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1509 | if (!options.recv_audio) { |
| 1510 | audio->set_direction(MD_SENDONLY); |
| 1511 | } |
| 1512 | |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1513 | desc->AddContent(CN_AUDIO, NS_JINGLE_RTP, audio.release()); |
| 1514 | if (!AddTransportOffer(CN_AUDIO, options.transport_options, |
| 1515 | current_description, desc)) { |
| 1516 | return false; |
| 1517 | } |
| 1518 | |
| 1519 | return true; |
| 1520 | } |
| 1521 | |
| 1522 | bool MediaSessionDescriptionFactory::AddVideoContentForOffer( |
| 1523 | const MediaSessionOptions& options, |
| 1524 | const SessionDescription* current_description, |
| 1525 | const RtpHeaderExtensions& video_rtp_extensions, |
| 1526 | const VideoCodecs& video_codecs, |
| 1527 | StreamParamsVec* current_streams, |
| 1528 | SessionDescription* desc) const { |
| 1529 | cricket::SecurePolicy sdes_policy = |
| 1530 | IsDtlsActive(CN_VIDEO, current_description) ? |
| 1531 | cricket::SEC_DISABLED : secure(); |
| 1532 | |
| 1533 | scoped_ptr<VideoContentDescription> video(new VideoContentDescription()); |
| 1534 | std::vector<std::string> crypto_suites; |
| 1535 | GetSupportedVideoCryptoSuites(&crypto_suites); |
| 1536 | if (!CreateMediaContentOffer( |
| 1537 | options, |
| 1538 | video_codecs, |
| 1539 | sdes_policy, |
| 1540 | GetCryptos(GetFirstVideoContentDescription(current_description)), |
| 1541 | crypto_suites, |
| 1542 | video_rtp_extensions, |
| 1543 | add_legacy_, |
| 1544 | current_streams, |
| 1545 | video.get())) { |
| 1546 | return false; |
| 1547 | } |
| 1548 | |
| 1549 | video->set_bandwidth(options.video_bandwidth); |
| 1550 | |
| 1551 | bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED); |
| 1552 | SetMediaProtocol(secure_transport, video.get()); |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1553 | |
| 1554 | if (!options.recv_video) { |
| 1555 | video->set_direction(MD_SENDONLY); |
| 1556 | } |
| 1557 | |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1558 | desc->AddContent(CN_VIDEO, NS_JINGLE_RTP, video.release()); |
| 1559 | if (!AddTransportOffer(CN_VIDEO, options.transport_options, |
| 1560 | current_description, desc)) { |
| 1561 | return false; |
| 1562 | } |
| 1563 | |
| 1564 | return true; |
| 1565 | } |
| 1566 | |
| 1567 | bool MediaSessionDescriptionFactory::AddDataContentForOffer( |
| 1568 | const MediaSessionOptions& options, |
| 1569 | const SessionDescription* current_description, |
| 1570 | DataCodecs* data_codecs, |
| 1571 | StreamParamsVec* current_streams, |
| 1572 | SessionDescription* desc) const { |
| 1573 | bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED); |
| 1574 | |
| 1575 | scoped_ptr<DataContentDescription> data(new DataContentDescription()); |
| 1576 | bool is_sctp = (options.data_channel_type == DCT_SCTP); |
| 1577 | |
| 1578 | FilterDataCodecs(data_codecs, is_sctp); |
| 1579 | |
| 1580 | cricket::SecurePolicy sdes_policy = |
| 1581 | IsDtlsActive(CN_DATA, current_description) ? |
| 1582 | cricket::SEC_DISABLED : secure(); |
| 1583 | std::vector<std::string> crypto_suites; |
| 1584 | if (is_sctp) { |
| 1585 | // SDES doesn't make sense for SCTP, so we disable it, and we only |
| 1586 | // get SDES crypto suites for RTP-based data channels. |
| 1587 | sdes_policy = cricket::SEC_DISABLED; |
| 1588 | // Unlike SetMediaProtocol below, we need to set the protocol |
| 1589 | // before we call CreateMediaContentOffer. Otherwise, |
| 1590 | // CreateMediaContentOffer won't know this is SCTP and will |
| 1591 | // generate SSRCs rather than SIDs. |
| 1592 | data->set_protocol( |
| 1593 | secure_transport ? kMediaProtocolDtlsSctp : kMediaProtocolSctp); |
| 1594 | } else { |
| 1595 | GetSupportedDataCryptoSuites(&crypto_suites); |
| 1596 | } |
| 1597 | |
| 1598 | if (!CreateMediaContentOffer( |
| 1599 | options, |
| 1600 | *data_codecs, |
| 1601 | sdes_policy, |
| 1602 | GetCryptos(GetFirstDataContentDescription(current_description)), |
| 1603 | crypto_suites, |
| 1604 | RtpHeaderExtensions(), |
| 1605 | add_legacy_, |
| 1606 | current_streams, |
| 1607 | data.get())) { |
| 1608 | return false; |
| 1609 | } |
| 1610 | |
| 1611 | if (is_sctp) { |
| 1612 | desc->AddContent(CN_DATA, NS_JINGLE_DRAFT_SCTP, data.release()); |
| 1613 | } else { |
| 1614 | data->set_bandwidth(options.data_bandwidth); |
| 1615 | SetMediaProtocol(secure_transport, data.get()); |
| 1616 | desc->AddContent(CN_DATA, NS_JINGLE_RTP, data.release()); |
| 1617 | } |
| 1618 | if (!AddTransportOffer(CN_DATA, options.transport_options, |
| 1619 | current_description, desc)) { |
| 1620 | return false; |
| 1621 | } |
| 1622 | return true; |
| 1623 | } |
| 1624 | |
| 1625 | bool MediaSessionDescriptionFactory::AddAudioContentForAnswer( |
| 1626 | const SessionDescription* offer, |
| 1627 | const MediaSessionOptions& options, |
| 1628 | const SessionDescription* current_description, |
| 1629 | StreamParamsVec* current_streams, |
| 1630 | SessionDescription* answer) const { |
| 1631 | const ContentInfo* audio_content = GetFirstAudioContent(offer); |
| 1632 | |
| 1633 | scoped_ptr<TransportDescription> audio_transport( |
| 1634 | CreateTransportAnswer(audio_content->name, offer, |
| 1635 | options.transport_options, |
| 1636 | current_description)); |
| 1637 | if (!audio_transport) { |
| 1638 | return false; |
| 1639 | } |
| 1640 | |
| 1641 | AudioCodecs audio_codecs = audio_codecs_; |
| 1642 | if (!options.vad_enabled) { |
| 1643 | StripCNCodecs(&audio_codecs); |
| 1644 | } |
| 1645 | |
| 1646 | bool bundle_enabled = |
| 1647 | offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled; |
| 1648 | scoped_ptr<AudioContentDescription> audio_answer( |
| 1649 | new AudioContentDescription()); |
| 1650 | // Do not require or create SDES cryptos if DTLS is used. |
| 1651 | cricket::SecurePolicy sdes_policy = |
| 1652 | audio_transport->secure() ? cricket::SEC_DISABLED : secure(); |
| 1653 | if (!CreateMediaContentAnswer( |
| 1654 | static_cast<const AudioContentDescription*>( |
| 1655 | audio_content->description), |
| 1656 | options, |
| 1657 | audio_codecs, |
| 1658 | sdes_policy, |
| 1659 | GetCryptos(GetFirstAudioContentDescription(current_description)), |
| 1660 | audio_rtp_extensions_, |
| 1661 | current_streams, |
| 1662 | add_legacy_, |
| 1663 | bundle_enabled, |
| 1664 | audio_answer.get())) { |
| 1665 | return false; // Fails the session setup. |
| 1666 | } |
| 1667 | |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1668 | bool rejected = !options.has_audio() || audio_content->rejected || |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1669 | !IsMediaProtocolSupported(MEDIA_TYPE_AUDIO, |
| 1670 | audio_answer->protocol(), |
| 1671 | audio_transport->secure()); |
| 1672 | if (!rejected) { |
| 1673 | AddTransportAnswer(audio_content->name, *(audio_transport.get()), answer); |
| 1674 | } else { |
| 1675 | // RFC 3264 |
| 1676 | // The answer MUST contain the same number of m-lines as the offer. |
| 1677 | LOG(LS_INFO) << "Audio is not supported in the answer."; |
| 1678 | } |
| 1679 | |
| 1680 | answer->AddContent(audio_content->name, audio_content->type, rejected, |
| 1681 | audio_answer.release()); |
| 1682 | return true; |
| 1683 | } |
| 1684 | |
| 1685 | bool MediaSessionDescriptionFactory::AddVideoContentForAnswer( |
| 1686 | const SessionDescription* offer, |
| 1687 | const MediaSessionOptions& options, |
| 1688 | const SessionDescription* current_description, |
| 1689 | StreamParamsVec* current_streams, |
| 1690 | SessionDescription* answer) const { |
| 1691 | const ContentInfo* video_content = GetFirstVideoContent(offer); |
| 1692 | scoped_ptr<TransportDescription> video_transport( |
| 1693 | CreateTransportAnswer(video_content->name, offer, |
| 1694 | options.transport_options, |
| 1695 | current_description)); |
| 1696 | if (!video_transport) { |
| 1697 | return false; |
| 1698 | } |
| 1699 | |
| 1700 | scoped_ptr<VideoContentDescription> video_answer( |
| 1701 | new VideoContentDescription()); |
| 1702 | // Do not require or create SDES cryptos if DTLS is used. |
| 1703 | cricket::SecurePolicy sdes_policy = |
| 1704 | video_transport->secure() ? cricket::SEC_DISABLED : secure(); |
| 1705 | bool bundle_enabled = |
| 1706 | offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled; |
| 1707 | if (!CreateMediaContentAnswer( |
| 1708 | static_cast<const VideoContentDescription*>( |
| 1709 | video_content->description), |
| 1710 | options, |
| 1711 | video_codecs_, |
| 1712 | sdes_policy, |
| 1713 | GetCryptos(GetFirstVideoContentDescription(current_description)), |
| 1714 | video_rtp_extensions_, |
| 1715 | current_streams, |
| 1716 | add_legacy_, |
| 1717 | bundle_enabled, |
| 1718 | video_answer.get())) { |
| 1719 | return false; |
| 1720 | } |
jiayl@webrtc.org | 742922b | 2014-10-07 21:32:43 +0000 | [diff] [blame] | 1721 | bool rejected = !options.has_video() || video_content->rejected || |
jiayl@webrtc.org | e7d47a1 | 2014-08-05 19:19:05 +0000 | [diff] [blame] | 1722 | !IsMediaProtocolSupported(MEDIA_TYPE_VIDEO, |
| 1723 | video_answer->protocol(), |
| 1724 | video_transport->secure()); |
| 1725 | if (!rejected) { |
| 1726 | if (!AddTransportAnswer(video_content->name, *(video_transport.get()), |
| 1727 | answer)) { |
| 1728 | return false; |
| 1729 | } |
| 1730 | video_answer->set_bandwidth(options.video_bandwidth); |
| 1731 | } else { |
| 1732 | // RFC 3264 |
| 1733 | // The answer MUST contain the same number of m-lines as the offer. |
| 1734 | LOG(LS_INFO) << "Video is not supported in the answer."; |
| 1735 | } |
| 1736 | answer->AddContent(video_content->name, video_content->type, rejected, |
| 1737 | video_answer.release()); |
| 1738 | return true; |
| 1739 | } |
| 1740 | |
| 1741 | bool MediaSessionDescriptionFactory::AddDataContentForAnswer( |
| 1742 | const SessionDescription* offer, |
| 1743 | const MediaSessionOptions& options, |
| 1744 | const SessionDescription* current_description, |
| 1745 | StreamParamsVec* current_streams, |
| 1746 | SessionDescription* answer) const { |
| 1747 | const ContentInfo* data_content = GetFirstDataContent(offer); |
| 1748 | scoped_ptr<TransportDescription> data_transport( |
| 1749 | CreateTransportAnswer(data_content->name, offer, |
| 1750 | options.transport_options, |
| 1751 | current_description)); |
| 1752 | if (!data_transport) { |
| 1753 | return false; |
| 1754 | } |
| 1755 | bool is_sctp = (options.data_channel_type == DCT_SCTP); |
| 1756 | std::vector<DataCodec> data_codecs(data_codecs_); |
| 1757 | FilterDataCodecs(&data_codecs, is_sctp); |
| 1758 | |
| 1759 | scoped_ptr<DataContentDescription> data_answer( |
| 1760 | new DataContentDescription()); |
| 1761 | // Do not require or create SDES cryptos if DTLS is used. |
| 1762 | cricket::SecurePolicy sdes_policy = |
| 1763 | data_transport->secure() ? cricket::SEC_DISABLED : secure(); |
| 1764 | bool bundle_enabled = |
| 1765 | offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled; |
| 1766 | if (!CreateMediaContentAnswer( |
| 1767 | static_cast<const DataContentDescription*>( |
| 1768 | data_content->description), |
| 1769 | options, |
| 1770 | data_codecs_, |
| 1771 | sdes_policy, |
| 1772 | GetCryptos(GetFirstDataContentDescription(current_description)), |
| 1773 | RtpHeaderExtensions(), |
| 1774 | current_streams, |
| 1775 | add_legacy_, |
| 1776 | bundle_enabled, |
| 1777 | data_answer.get())) { |
| 1778 | return false; // Fails the session setup. |
| 1779 | } |
| 1780 | |
| 1781 | bool rejected = !options.has_data() || data_content->rejected || |
| 1782 | !IsMediaProtocolSupported(MEDIA_TYPE_DATA, |
| 1783 | data_answer->protocol(), |
| 1784 | data_transport->secure()); |
| 1785 | if (!rejected) { |
| 1786 | data_answer->set_bandwidth(options.data_bandwidth); |
| 1787 | if (!AddTransportAnswer(data_content->name, *(data_transport.get()), |
| 1788 | answer)) { |
| 1789 | return false; |
| 1790 | } |
| 1791 | } else { |
| 1792 | // RFC 3264 |
| 1793 | // The answer MUST contain the same number of m-lines as the offer. |
| 1794 | LOG(LS_INFO) << "Data is not supported in the answer."; |
| 1795 | } |
| 1796 | answer->AddContent(data_content->name, data_content->type, rejected, |
| 1797 | data_answer.release()); |
| 1798 | return true; |
| 1799 | } |
| 1800 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1801 | bool IsMediaContent(const ContentInfo* content) { |
| 1802 | return (content && |
| 1803 | (content->type == NS_JINGLE_RTP || |
| 1804 | content->type == NS_JINGLE_DRAFT_SCTP)); |
| 1805 | } |
| 1806 | |
| 1807 | bool IsAudioContent(const ContentInfo* content) { |
| 1808 | return IsMediaContentOfType(content, MEDIA_TYPE_AUDIO); |
| 1809 | } |
| 1810 | |
| 1811 | bool IsVideoContent(const ContentInfo* content) { |
| 1812 | return IsMediaContentOfType(content, MEDIA_TYPE_VIDEO); |
| 1813 | } |
| 1814 | |
| 1815 | bool IsDataContent(const ContentInfo* content) { |
| 1816 | return IsMediaContentOfType(content, MEDIA_TYPE_DATA); |
| 1817 | } |
| 1818 | |
| 1819 | static const ContentInfo* GetFirstMediaContent(const ContentInfos& contents, |
| 1820 | MediaType media_type) { |
| 1821 | for (ContentInfos::const_iterator content = contents.begin(); |
| 1822 | content != contents.end(); content++) { |
| 1823 | if (IsMediaContentOfType(&*content, media_type)) { |
| 1824 | return &*content; |
| 1825 | } |
| 1826 | } |
| 1827 | return NULL; |
| 1828 | } |
| 1829 | |
| 1830 | const ContentInfo* GetFirstAudioContent(const ContentInfos& contents) { |
| 1831 | return GetFirstMediaContent(contents, MEDIA_TYPE_AUDIO); |
| 1832 | } |
| 1833 | |
| 1834 | const ContentInfo* GetFirstVideoContent(const ContentInfos& contents) { |
| 1835 | return GetFirstMediaContent(contents, MEDIA_TYPE_VIDEO); |
| 1836 | } |
| 1837 | |
| 1838 | const ContentInfo* GetFirstDataContent(const ContentInfos& contents) { |
| 1839 | return GetFirstMediaContent(contents, MEDIA_TYPE_DATA); |
| 1840 | } |
| 1841 | |
| 1842 | static const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc, |
| 1843 | MediaType media_type) { |
| 1844 | if (sdesc == NULL) |
| 1845 | return NULL; |
| 1846 | |
| 1847 | return GetFirstMediaContent(sdesc->contents(), media_type); |
| 1848 | } |
| 1849 | |
| 1850 | const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc) { |
| 1851 | return GetFirstMediaContent(sdesc, MEDIA_TYPE_AUDIO); |
| 1852 | } |
| 1853 | |
| 1854 | const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc) { |
| 1855 | return GetFirstMediaContent(sdesc, MEDIA_TYPE_VIDEO); |
| 1856 | } |
| 1857 | |
| 1858 | const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc) { |
| 1859 | return GetFirstMediaContent(sdesc, MEDIA_TYPE_DATA); |
| 1860 | } |
| 1861 | |
| 1862 | const MediaContentDescription* GetFirstMediaContentDescription( |
| 1863 | const SessionDescription* sdesc, MediaType media_type) { |
| 1864 | const ContentInfo* content = GetFirstMediaContent(sdesc, media_type); |
| 1865 | const ContentDescription* description = content ? content->description : NULL; |
| 1866 | return static_cast<const MediaContentDescription*>(description); |
| 1867 | } |
| 1868 | |
| 1869 | const AudioContentDescription* GetFirstAudioContentDescription( |
| 1870 | const SessionDescription* sdesc) { |
| 1871 | return static_cast<const AudioContentDescription*>( |
| 1872 | GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_AUDIO)); |
| 1873 | } |
| 1874 | |
| 1875 | const VideoContentDescription* GetFirstVideoContentDescription( |
| 1876 | const SessionDescription* sdesc) { |
| 1877 | return static_cast<const VideoContentDescription*>( |
| 1878 | GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_VIDEO)); |
| 1879 | } |
| 1880 | |
| 1881 | const DataContentDescription* GetFirstDataContentDescription( |
| 1882 | const SessionDescription* sdesc) { |
| 1883 | return static_cast<const DataContentDescription*>( |
| 1884 | GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA)); |
| 1885 | } |
| 1886 | |
| 1887 | bool GetMediaChannelNameFromComponent( |
| 1888 | int component, MediaType media_type, std::string* channel_name) { |
| 1889 | if (media_type == MEDIA_TYPE_AUDIO) { |
| 1890 | if (component == ICE_CANDIDATE_COMPONENT_RTP) { |
| 1891 | *channel_name = GICE_CHANNEL_NAME_RTP; |
| 1892 | return true; |
| 1893 | } else if (component == ICE_CANDIDATE_COMPONENT_RTCP) { |
| 1894 | *channel_name = GICE_CHANNEL_NAME_RTCP; |
| 1895 | return true; |
| 1896 | } |
| 1897 | } else if (media_type == MEDIA_TYPE_VIDEO) { |
| 1898 | if (component == ICE_CANDIDATE_COMPONENT_RTP) { |
| 1899 | *channel_name = GICE_CHANNEL_NAME_VIDEO_RTP; |
| 1900 | return true; |
| 1901 | } else if (component == ICE_CANDIDATE_COMPONENT_RTCP) { |
| 1902 | *channel_name = GICE_CHANNEL_NAME_VIDEO_RTCP; |
| 1903 | return true; |
| 1904 | } |
| 1905 | } else if (media_type == MEDIA_TYPE_DATA) { |
| 1906 | if (component == ICE_CANDIDATE_COMPONENT_RTP) { |
| 1907 | *channel_name = GICE_CHANNEL_NAME_DATA_RTP; |
| 1908 | return true; |
| 1909 | } else if (component == ICE_CANDIDATE_COMPONENT_RTCP) { |
| 1910 | *channel_name = GICE_CHANNEL_NAME_DATA_RTCP; |
| 1911 | return true; |
| 1912 | } |
| 1913 | } |
| 1914 | |
| 1915 | return false; |
| 1916 | } |
| 1917 | |
| 1918 | bool GetMediaComponentFromChannelName( |
| 1919 | const std::string& channel_name, int* component) { |
| 1920 | if (channel_name == GICE_CHANNEL_NAME_RTP || |
| 1921 | channel_name == GICE_CHANNEL_NAME_VIDEO_RTP || |
| 1922 | channel_name == GICE_CHANNEL_NAME_DATA_RTP) { |
| 1923 | *component = ICE_CANDIDATE_COMPONENT_RTP; |
| 1924 | return true; |
| 1925 | } else if (channel_name == GICE_CHANNEL_NAME_RTCP || |
| 1926 | channel_name == GICE_CHANNEL_NAME_VIDEO_RTCP || |
| 1927 | channel_name == GICE_CHANNEL_NAME_DATA_RTP) { |
| 1928 | *component = ICE_CANDIDATE_COMPONENT_RTCP; |
| 1929 | return true; |
| 1930 | } |
| 1931 | |
| 1932 | return false; |
| 1933 | } |
| 1934 | |
| 1935 | bool GetMediaTypeFromChannelName( |
| 1936 | const std::string& channel_name, MediaType* media_type) { |
| 1937 | if (channel_name == GICE_CHANNEL_NAME_RTP || |
| 1938 | channel_name == GICE_CHANNEL_NAME_RTCP) { |
| 1939 | *media_type = MEDIA_TYPE_AUDIO; |
| 1940 | return true; |
| 1941 | } else if (channel_name == GICE_CHANNEL_NAME_VIDEO_RTP || |
| 1942 | channel_name == GICE_CHANNEL_NAME_VIDEO_RTCP) { |
| 1943 | *media_type = MEDIA_TYPE_VIDEO; |
| 1944 | return true; |
| 1945 | } else if (channel_name == GICE_CHANNEL_NAME_DATA_RTP || |
| 1946 | channel_name == GICE_CHANNEL_NAME_DATA_RTCP) { |
| 1947 | *media_type = MEDIA_TYPE_DATA; |
| 1948 | return true; |
| 1949 | } |
| 1950 | |
| 1951 | return false; |
| 1952 | } |
| 1953 | |
| 1954 | } // namespace cricket |