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webrtc/talk/session/media/mediasession.cc
buildbot@webrtc.org a09a99950e (Auto)update libjingle 73222930-> 73226398 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 17:26:08 +00:00

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/*
* libjingle
* Copyright 2004 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/mediasession.h"
#include <functional>
#include <map>
#include <set>
#include <utility>
#include "talk/media/base/constants.h"
#include "talk/media/base/cryptoparams.h"
#include "talk/p2p/base/constants.h"
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/srtpfilter.h"
#include "talk/xmpp/constants.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/stringutils.h"
#ifdef HAVE_SCTP
#include "talk/media/sctp/sctpdataengine.h"
#else
static const uint32 kMaxSctpSid = 1023;
#endif
namespace {
const char kInline[] = "inline:";
}
namespace cricket {
using rtc::scoped_ptr;
// RTP Profile names
// http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xml
// RFC4585
const char kMediaProtocolAvpf[] = "RTP/AVPF";
// RFC5124
const char kMediaProtocolDtlsSavpf[] = "UDP/TLS/RTP/SAVPF";
// This should be replaced by "UDP/TLS/RTP/SAVPF", but we need to support it for
// now to be compatible with previous Chrome versions.
const char kMediaProtocolSavpf[] = "RTP/SAVPF";
const char kMediaProtocolRtpPrefix[] = "RTP/";
const char kMediaProtocolSctp[] = "SCTP";
const char kMediaProtocolDtlsSctp[] = "DTLS/SCTP";
static bool IsMediaContentOfType(const ContentInfo* content,
MediaType media_type) {
if (!IsMediaContent(content)) {
return false;
}
const MediaContentDescription* mdesc =
static_cast<const MediaContentDescription*>(content->description);
return mdesc && mdesc->type() == media_type;
}
static bool CreateCryptoParams(int tag, const std::string& cipher,
CryptoParams *out) {
std::string key;
key.reserve(SRTP_MASTER_KEY_BASE64_LEN);
if (!rtc::CreateRandomString(SRTP_MASTER_KEY_BASE64_LEN, &key)) {
return false;
}
out->tag = tag;
out->cipher_suite = cipher;
out->key_params = kInline;
out->key_params += key;
return true;
}
#ifdef HAVE_SRTP
static bool AddCryptoParams(const std::string& cipher_suite,
CryptoParamsVec *out) {
int size = static_cast<int>(out->size());
out->resize(size + 1);
return CreateCryptoParams(size, cipher_suite, &out->at(size));
}
void AddMediaCryptos(const CryptoParamsVec& cryptos,
MediaContentDescription* media) {
for (CryptoParamsVec::const_iterator crypto = cryptos.begin();
crypto != cryptos.end(); ++crypto) {
media->AddCrypto(*crypto);
}
}
bool CreateMediaCryptos(const std::vector<std::string>& crypto_suites,
MediaContentDescription* media) {
CryptoParamsVec cryptos;
for (std::vector<std::string>::const_iterator it = crypto_suites.begin();
it != crypto_suites.end(); ++it) {
if (!AddCryptoParams(*it, &cryptos)) {
return false;
}
}
AddMediaCryptos(cryptos, media);
return true;
}
#endif
const CryptoParamsVec* GetCryptos(const MediaContentDescription* media) {
if (!media) {
return NULL;
}
return &media->cryptos();
}
bool FindMatchingCrypto(const CryptoParamsVec& cryptos,
const CryptoParams& crypto,
CryptoParams* out) {
for (CryptoParamsVec::const_iterator it = cryptos.begin();
it != cryptos.end(); ++it) {
if (crypto.Matches(*it)) {
*out = *it;
return true;
}
}
return false;
}
// For audio, HMAC 32 is prefered because of the low overhead.
void GetSupportedAudioCryptoSuites(
std::vector<std::string>* crypto_suites) {
#ifdef HAVE_SRTP
crypto_suites->push_back(CS_AES_CM_128_HMAC_SHA1_32);
crypto_suites->push_back(CS_AES_CM_128_HMAC_SHA1_80);
#endif
}
void GetSupportedVideoCryptoSuites(
std::vector<std::string>* crypto_suites) {
GetSupportedDefaultCryptoSuites(crypto_suites);
}
void GetSupportedDataCryptoSuites(
std::vector<std::string>* crypto_suites) {
GetSupportedDefaultCryptoSuites(crypto_suites);
}
void GetSupportedDefaultCryptoSuites(
std::vector<std::string>* crypto_suites) {
#ifdef HAVE_SRTP
crypto_suites->push_back(CS_AES_CM_128_HMAC_SHA1_80);
#endif
}
// For video support only 80-bit SHA1 HMAC. For audio 32-bit HMAC is
// tolerated unless bundle is enabled because it is low overhead. Pick the
// crypto in the list that is supported.
static bool SelectCrypto(const MediaContentDescription* offer,
bool bundle,
CryptoParams *crypto) {
bool audio = offer->type() == MEDIA_TYPE_AUDIO;
const CryptoParamsVec& cryptos = offer->cryptos();
for (CryptoParamsVec::const_iterator i = cryptos.begin();
i != cryptos.end(); ++i) {
if (CS_AES_CM_128_HMAC_SHA1_80 == i->cipher_suite ||
(CS_AES_CM_128_HMAC_SHA1_32 == i->cipher_suite && audio && !bundle)) {
return CreateCryptoParams(i->tag, i->cipher_suite, crypto);
}
}
return false;
}
static const StreamParams* FindFirstStreamParamsByCname(
const StreamParamsVec& params_vec,
const std::string& cname) {
for (StreamParamsVec::const_iterator it = params_vec.begin();
it != params_vec.end(); ++it) {
if (cname == it->cname)
return &*it;
}
return NULL;
}
// Generates a new CNAME or the CNAME of an already existing StreamParams
// if a StreamParams exist for another Stream in streams with sync_label
// sync_label.
static bool GenerateCname(const StreamParamsVec& params_vec,
const MediaSessionOptions::Streams& streams,
const std::string& synch_label,
std::string* cname) {
ASSERT(cname != NULL);
if (!cname)
return false;
// Check if a CNAME exist for any of the other synched streams.
for (MediaSessionOptions::Streams::const_iterator stream_it = streams.begin();
stream_it != streams.end() ; ++stream_it) {
if (synch_label != stream_it->sync_label)
continue;
StreamParams param;
// groupid is empty for StreamParams generated using
// MediaSessionDescriptionFactory.
if (GetStreamByIds(params_vec, "", stream_it->id,
&param)) {
*cname = param.cname;
return true;
}
}
// No other stream seems to exist that we should sync with.
// Generate a random string for the RTCP CNAME, as stated in RFC 6222.
// This string is only used for synchronization, and therefore is opaque.
do {
if (!rtc::CreateRandomString(16, cname)) {
ASSERT(false);
return false;
}
} while (FindFirstStreamParamsByCname(params_vec, *cname));
return true;
}
// Generate random SSRC values that are not already present in |params_vec|.
// The generated values are added to |ssrcs|.
// |num_ssrcs| is the number of the SSRC will be generated.
static void GenerateSsrcs(const StreamParamsVec& params_vec,
int num_ssrcs,
std::vector<uint32>* ssrcs) {
for (int i = 0; i < num_ssrcs; i++) {
uint32 candidate;
do {
candidate = rtc::CreateRandomNonZeroId();
} while (GetStreamBySsrc(params_vec, candidate, NULL) ||
std::count(ssrcs->begin(), ssrcs->end(), candidate) > 0);
ssrcs->push_back(candidate);
}
}
// Returns false if we exhaust the range of SIDs.
static bool GenerateSctpSid(const StreamParamsVec& params_vec,
uint32* sid) {
if (params_vec.size() > kMaxSctpSid) {
LOG(LS_WARNING) <<
"Could not generate an SCTP SID: too many SCTP streams.";
return false;
}
while (true) {
uint32 candidate = rtc::CreateRandomNonZeroId() % kMaxSctpSid;
if (!GetStreamBySsrc(params_vec, candidate, NULL)) {
*sid = candidate;
return true;
}
}
}
static bool GenerateSctpSids(const StreamParamsVec& params_vec,
std::vector<uint32>* sids) {
uint32 sid;
if (!GenerateSctpSid(params_vec, &sid)) {
LOG(LS_WARNING) << "Could not generated an SCTP SID.";
return false;
}
sids->push_back(sid);
return true;
}
// Finds all StreamParams of all media types and attach them to stream_params.
static void GetCurrentStreamParams(const SessionDescription* sdesc,
StreamParamsVec* stream_params) {
if (!sdesc)
return;
const ContentInfos& contents = sdesc->contents();
for (ContentInfos::const_iterator content = contents.begin();
content != contents.end(); ++content) {
if (!IsMediaContent(&*content)) {
continue;
}
const MediaContentDescription* media =
static_cast<const MediaContentDescription*>(
content->description);
const StreamParamsVec& streams = media->streams();
for (StreamParamsVec::const_iterator it = streams.begin();
it != streams.end(); ++it) {
stream_params->push_back(*it);
}
}
}
// Filters the data codecs for the data channel type.
void FilterDataCodecs(std::vector<DataCodec>* codecs, bool sctp) {
// Filter RTP codec for SCTP and vice versa.
int codec_id = sctp ? kGoogleRtpDataCodecId : kGoogleSctpDataCodecId;
for (std::vector<DataCodec>::iterator iter = codecs->begin();
iter != codecs->end();) {
if (iter->id == codec_id) {
iter = codecs->erase(iter);
} else {
++iter;
}
}
}
template <typename IdStruct>
class UsedIds {
public:
UsedIds(int min_allowed_id, int max_allowed_id)
: min_allowed_id_(min_allowed_id),
max_allowed_id_(max_allowed_id),
next_id_(max_allowed_id) {
}
// Loops through all Id in |ids| and changes its id if it is
// already in use by another IdStruct. Call this methods with all Id
// in a session description to make sure no duplicate ids exists.
// Note that typename Id must be a type of IdStruct.
template <typename Id>
void FindAndSetIdUsed(std::vector<Id>* ids) {
for (typename std::vector<Id>::iterator it = ids->begin();
it != ids->end(); ++it) {
FindAndSetIdUsed(&*it);
}
}
// Finds and sets an unused id if the |idstruct| id is already in use.
void FindAndSetIdUsed(IdStruct* idstruct) {
const int original_id = idstruct->id;
int new_id = idstruct->id;
if (original_id > max_allowed_id_ || original_id < min_allowed_id_) {
// If the original id is not in range - this is an id that can't be
// dynamically changed.
return;
}
if (IsIdUsed(original_id)) {
new_id = FindUnusedId();
LOG(LS_WARNING) << "Duplicate id found. Reassigning from " << original_id
<< " to " << new_id;
idstruct->id = new_id;
}
SetIdUsed(new_id);
}
private:
// Returns the first unused id in reverse order.
// This hopefully reduce the risk of more collisions. We want to change the
// default ids as little as possible.
int FindUnusedId() {
while (IsIdUsed(next_id_) && next_id_ >= min_allowed_id_) {
--next_id_;
}
ASSERT(next_id_ >= min_allowed_id_);
return next_id_;
}
bool IsIdUsed(int new_id) {
return id_set_.find(new_id) != id_set_.end();
}
void SetIdUsed(int new_id) {
id_set_.insert(new_id);
}
const int min_allowed_id_;
const int max_allowed_id_;
int next_id_;
std::set<int> id_set_;
};
// Helper class used for finding duplicate RTP payload types among audio, video
// and data codecs. When bundle is used the payload types may not collide.
class UsedPayloadTypes : public UsedIds<Codec> {
public:
UsedPayloadTypes()
: UsedIds<Codec>(kDynamicPayloadTypeMin, kDynamicPayloadTypeMax) {
}
private:
static const int kDynamicPayloadTypeMin = 96;
static const int kDynamicPayloadTypeMax = 127;
};
// Helper class used for finding duplicate RTP Header extension ids among
// audio and video extensions.
class UsedRtpHeaderExtensionIds : public UsedIds<RtpHeaderExtension> {
public:
UsedRtpHeaderExtensionIds()
: UsedIds<RtpHeaderExtension>(kLocalIdMin, kLocalIdMax) {
}
private:
// Min and Max local identifier for one-byte header extensions, per RFC5285.
static const int kLocalIdMin = 1;
static const int kLocalIdMax = 14;
};
static bool IsSctp(const MediaContentDescription* desc) {
return ((desc->protocol() == kMediaProtocolSctp) ||
(desc->protocol() == kMediaProtocolDtlsSctp));
}
// Adds a StreamParams for each Stream in Streams with media type
// media_type to content_description.
// |current_params| - All currently known StreamParams of any media type.
template <class C>
static bool AddStreamParams(
MediaType media_type,
const MediaSessionOptions::Streams& streams,
StreamParamsVec* current_streams,
MediaContentDescriptionImpl<C>* content_description,
const bool add_legacy_stream) {
const bool include_rtx_stream =
ContainsRtxCodec(content_description->codecs());
if (streams.empty() && add_legacy_stream) {
// TODO(perkj): Remove this legacy stream when all apps use StreamParams.
std::vector<uint32> ssrcs;
if (IsSctp(content_description)) {
GenerateSctpSids(*current_streams, &ssrcs);
} else {
int num_ssrcs = include_rtx_stream ? 2 : 1;
GenerateSsrcs(*current_streams, num_ssrcs, &ssrcs);
}
if (include_rtx_stream) {
content_description->AddLegacyStream(ssrcs[0], ssrcs[1]);
content_description->set_multistream(true);
} else {
content_description->AddLegacyStream(ssrcs[0]);
}
return true;
}
MediaSessionOptions::Streams::const_iterator stream_it;
for (stream_it = streams.begin();
stream_it != streams.end(); ++stream_it) {
if (stream_it->type != media_type)
continue; // Wrong media type.
StreamParams param;
// groupid is empty for StreamParams generated using
// MediaSessionDescriptionFactory.
if (!GetStreamByIds(*current_streams, "", stream_it->id,
&param)) {
// This is a new stream.
// Get a CNAME. Either new or same as one of the other synched streams.
std::string cname;
if (!GenerateCname(*current_streams, streams, stream_it->sync_label,
&cname)) {
return false;
}
std::vector<uint32> ssrcs;
if (IsSctp(content_description)) {
GenerateSctpSids(*current_streams, &ssrcs);
} else {
GenerateSsrcs(*current_streams, stream_it->num_sim_layers, &ssrcs);
}
StreamParams stream_param;
stream_param.id = stream_it->id;
// Add the generated ssrc.
for (size_t i = 0; i < ssrcs.size(); ++i) {
stream_param.ssrcs.push_back(ssrcs[i]);
}
if (stream_it->num_sim_layers > 1) {
SsrcGroup group(kSimSsrcGroupSemantics, stream_param.ssrcs);
stream_param.ssrc_groups.push_back(group);
}
// Generate an extra ssrc for include_rtx_stream case.
if (include_rtx_stream) {
std::vector<uint32> rtx_ssrc;
GenerateSsrcs(*current_streams, 1, &rtx_ssrc);
stream_param.AddFidSsrc(ssrcs[0], rtx_ssrc[0]);
content_description->set_multistream(true);
}
stream_param.cname = cname;
stream_param.sync_label = stream_it->sync_label;
content_description->AddStream(stream_param);
// Store the new StreamParams in current_streams.
// This is necessary so that we can use the CNAME for other media types.
current_streams->push_back(stream_param);
} else {
content_description->AddStream(param);
}
}
return true;
}
// Updates the transport infos of the |sdesc| according to the given
// |bundle_group|. The transport infos of the content names within the
// |bundle_group| should be updated to use the ufrag and pwd of the first
// content within the |bundle_group|.
static bool UpdateTransportInfoForBundle(const ContentGroup& bundle_group,
SessionDescription* sdesc) {
// The bundle should not be empty.
if (!sdesc || !bundle_group.FirstContentName()) {
return false;
}
// We should definitely have a transport for the first content.
std::string selected_content_name = *bundle_group.FirstContentName();
const TransportInfo* selected_transport_info =
sdesc->GetTransportInfoByName(selected_content_name);
if (!selected_transport_info) {
return false;
}
// Set the other contents to use the same ICE credentials.
const std::string selected_ufrag =
selected_transport_info->description.ice_ufrag;
const std::string selected_pwd =
selected_transport_info->description.ice_pwd;
for (TransportInfos::iterator it =
sdesc->transport_infos().begin();
it != sdesc->transport_infos().end(); ++it) {
if (bundle_group.HasContentName(it->content_name) &&
it->content_name != selected_content_name) {
it->description.ice_ufrag = selected_ufrag;
it->description.ice_pwd = selected_pwd;
}
}
return true;
}
// Gets the CryptoParamsVec of the given |content_name| from |sdesc|, and
// sets it to |cryptos|.
static bool GetCryptosByName(const SessionDescription* sdesc,
const std::string& content_name,
CryptoParamsVec* cryptos) {
if (!sdesc || !cryptos) {
return false;
}
const ContentInfo* content = sdesc->GetContentByName(content_name);
if (!IsMediaContent(content) || !content->description) {
return false;
}
const MediaContentDescription* media_desc =
static_cast<const MediaContentDescription*>(content->description);
*cryptos = media_desc->cryptos();
return true;
}
// Predicate function used by the remove_if.
// Returns true if the |crypto|'s cipher_suite is not found in |filter|.
static bool CryptoNotFound(const CryptoParams crypto,
const CryptoParamsVec* filter) {
if (filter == NULL) {
return true;
}
for (CryptoParamsVec::const_iterator it = filter->begin();
it != filter->end(); ++it) {
if (it->cipher_suite == crypto.cipher_suite) {
return false;
}
}
return true;
}
// Prunes the |target_cryptos| by removing the crypto params (cipher_suite)
// which are not available in |filter|.
static void PruneCryptos(const CryptoParamsVec& filter,
CryptoParamsVec* target_cryptos) {
if (!target_cryptos) {
return;
}
target_cryptos->erase(std::remove_if(target_cryptos->begin(),
target_cryptos->end(),
bind2nd(ptr_fun(CryptoNotFound),
&filter)),
target_cryptos->end());
}
static bool IsRtpContent(SessionDescription* sdesc,
const std::string& content_name) {
bool is_rtp = false;
ContentInfo* content = sdesc->GetContentByName(content_name);
if (IsMediaContent(content)) {
MediaContentDescription* media_desc =
static_cast<MediaContentDescription*>(content->description);
if (!media_desc) {
return false;
}
is_rtp = media_desc->protocol().empty() ||
rtc::starts_with(media_desc->protocol().data(),
kMediaProtocolRtpPrefix);
}
return is_rtp;
}
// Updates the crypto parameters of the |sdesc| according to the given
// |bundle_group|. The crypto parameters of all the contents within the
// |bundle_group| should be updated to use the common subset of the
// available cryptos.
static bool UpdateCryptoParamsForBundle(const ContentGroup& bundle_group,
SessionDescription* sdesc) {
// The bundle should not be empty.
if (!sdesc || !bundle_group.FirstContentName()) {
return false;
}
bool common_cryptos_needed = false;
// Get the common cryptos.
const ContentNames& content_names = bundle_group.content_names();
CryptoParamsVec common_cryptos;
for (ContentNames::const_iterator it = content_names.begin();
it != content_names.end(); ++it) {
if (!IsRtpContent(sdesc, *it)) {
continue;
}
// The common cryptos are needed if any of the content does not have DTLS
// enabled.
if (!sdesc->GetTransportInfoByName(*it)->description.secure()) {
common_cryptos_needed = true;
}
if (it == content_names.begin()) {
// Initial the common_cryptos with the first content in the bundle group.
if (!GetCryptosByName(sdesc, *it, &common_cryptos)) {
return false;
}
if (common_cryptos.empty()) {
// If there's no crypto params, we should just return.
return true;
}
} else {
CryptoParamsVec cryptos;
if (!GetCryptosByName(sdesc, *it, &cryptos)) {
return false;
}
PruneCryptos(cryptos, &common_cryptos);
}
}
if (common_cryptos.empty() && common_cryptos_needed) {
return false;
}
// Update to use the common cryptos.
for (ContentNames::const_iterator it = content_names.begin();
it != content_names.end(); ++it) {
if (!IsRtpContent(sdesc, *it)) {
continue;
}
ContentInfo* content = sdesc->GetContentByName(*it);
if (IsMediaContent(content)) {
MediaContentDescription* media_desc =
static_cast<MediaContentDescription*>(content->description);
if (!media_desc) {
return false;
}
media_desc->set_cryptos(common_cryptos);
}
}
return true;
}
template <class C>
static bool ContainsRtxCodec(const std::vector<C>& codecs) {
typename std::vector<C>::const_iterator it;
for (it = codecs.begin(); it != codecs.end(); ++it) {
if (IsRtxCodec(*it)) {
return true;
}
}
return false;
}
template <class C>
static bool IsRtxCodec(const C& codec) {
return stricmp(codec.name.c_str(), kRtxCodecName) == 0;
}
// Create a media content to be offered in a session-initiate,
// according to the given options.rtcp_mux, options.is_muc,
// options.streams, codecs, secure_transport, crypto, and streams. If we don't
// currently have crypto (in current_cryptos) and it is enabled (in
// secure_policy), crypto is created (according to crypto_suites). If
// add_legacy_stream is true, and current_streams is empty, a legacy
// stream is created. The created content is added to the offer.
template <class C>
static bool CreateMediaContentOffer(
const MediaSessionOptions& options,
const std::vector<C>& codecs,
const SecurePolicy& secure_policy,
const CryptoParamsVec* current_cryptos,
const std::vector<std::string>& crypto_suites,
const RtpHeaderExtensions& rtp_extensions,
bool add_legacy_stream,
StreamParamsVec* current_streams,
MediaContentDescriptionImpl<C>* offer) {
offer->AddCodecs(codecs);
offer->SortCodecs();
if (secure_policy == SEC_REQUIRED) {
offer->set_crypto_required(CT_SDES);
}
offer->set_rtcp_mux(options.rtcp_mux_enabled);
offer->set_multistream(options.is_muc);
offer->set_rtp_header_extensions(rtp_extensions);
if (!AddStreamParams(
offer->type(), options.streams, current_streams,
offer, add_legacy_stream)) {
return false;
}
#ifdef HAVE_SRTP
if (secure_policy != SEC_DISABLED) {
if (current_cryptos) {
AddMediaCryptos(*current_cryptos, offer);
}
if (offer->cryptos().empty()) {
if (!CreateMediaCryptos(crypto_suites, offer)) {
return false;
}
}
}
#endif
if (offer->crypto_required() == CT_SDES && offer->cryptos().empty()) {
return false;
}
return true;
}
template <class C>
static void NegotiateCodecs(const std::vector<C>& local_codecs,
const std::vector<C>& offered_codecs,
std::vector<C>* negotiated_codecs) {
typename std::vector<C>::const_iterator ours;
for (ours = local_codecs.begin();
ours != local_codecs.end(); ++ours) {
typename std::vector<C>::const_iterator theirs;
for (theirs = offered_codecs.begin();
theirs != offered_codecs.end(); ++theirs) {
if (ours->Matches(*theirs)) {
C negotiated = *ours;
negotiated.IntersectFeedbackParams(*theirs);
if (IsRtxCodec(negotiated)) {
// Only negotiate RTX if kCodecParamAssociatedPayloadType has been
// set.
std::string apt_value;
if (!theirs->GetParam(kCodecParamAssociatedPayloadType, &apt_value)) {
LOG(LS_WARNING) << "RTX missing associated payload type.";
continue;
}
negotiated.SetParam(kCodecParamAssociatedPayloadType, apt_value);
}
negotiated.id = theirs->id;
// RFC3264: Although the answerer MAY list the formats in their desired
// order of preference, it is RECOMMENDED that unless there is a
// specific reason, the answerer list formats in the same relative order
// they were present in the offer.
negotiated.preference = theirs->preference;
negotiated_codecs->push_back(negotiated);
}
}
}
}
template <class C>
static bool FindMatchingCodec(const std::vector<C>& codecs,
const C& codec_to_match,
C* found_codec) {
for (typename std::vector<C>::const_iterator it = codecs.begin();
it != codecs.end(); ++it) {
if (it->Matches(codec_to_match)) {
if (found_codec != NULL) {
*found_codec= *it;
}
return true;
}
}
return false;
}
// Adds all codecs from |reference_codecs| to |offered_codecs| that dont'
// already exist in |offered_codecs| and ensure the payload types don't
// collide.
template <class C>
static void FindCodecsToOffer(
const std::vector<C>& reference_codecs,
std::vector<C>* offered_codecs,
UsedPayloadTypes* used_pltypes) {
typedef std::map<int, C> RtxCodecReferences;
RtxCodecReferences new_rtx_codecs;
// Find all new RTX codecs.
for (typename std::vector<C>::const_iterator it = reference_codecs.begin();
it != reference_codecs.end(); ++it) {
if (!FindMatchingCodec<C>(*offered_codecs, *it, NULL) && IsRtxCodec(*it)) {
C rtx_codec = *it;
int referenced_pl_type =
rtc::FromString<int>(0,
rtx_codec.params[kCodecParamAssociatedPayloadType]);
new_rtx_codecs.insert(std::pair<int, C>(referenced_pl_type,
rtx_codec));
}
}
// Add all new codecs that are not RTX codecs.
for (typename std::vector<C>::const_iterator it = reference_codecs.begin();
it != reference_codecs.end(); ++it) {
if (!FindMatchingCodec<C>(*offered_codecs, *it, NULL) && !IsRtxCodec(*it)) {
C codec = *it;
int original_payload_id = codec.id;
used_pltypes->FindAndSetIdUsed(&codec);
offered_codecs->push_back(codec);
// If this codec is referenced by a new RTX codec, update the reference
// in the RTX codec with the new payload type.
typename RtxCodecReferences::iterator rtx_it =
new_rtx_codecs.find(original_payload_id);
if (rtx_it != new_rtx_codecs.end()) {
C& rtx_codec = rtx_it->second;
rtx_codec.params[kCodecParamAssociatedPayloadType] =
rtc::ToString(codec.id);
}
}
}
// Add all new RTX codecs.
for (typename RtxCodecReferences::iterator it = new_rtx_codecs.begin();
it != new_rtx_codecs.end(); ++it) {
C& rtx_codec = it->second;
used_pltypes->FindAndSetIdUsed(&rtx_codec);
offered_codecs->push_back(rtx_codec);
}
}
static bool FindByUri(const RtpHeaderExtensions& extensions,
const RtpHeaderExtension& ext_to_match,
RtpHeaderExtension* found_extension) {
for (RtpHeaderExtensions::const_iterator it = extensions.begin();
it != extensions.end(); ++it) {
// We assume that all URIs are given in a canonical format.
if (it->uri == ext_to_match.uri) {
if (found_extension != NULL) {
*found_extension = *it;
}
return true;
}
}
return false;
}
static void FindAndSetRtpHdrExtUsed(
const RtpHeaderExtensions& reference_extensions,
RtpHeaderExtensions* offered_extensions,
const RtpHeaderExtensions& other_extensions,
UsedRtpHeaderExtensionIds* used_extensions) {
for (RtpHeaderExtensions::const_iterator it = reference_extensions.begin();
it != reference_extensions.end(); ++it) {
if (!FindByUri(*offered_extensions, *it, NULL)) {
RtpHeaderExtension ext;
if (!FindByUri(other_extensions, *it, &ext)) {
ext = *it;
used_extensions->FindAndSetIdUsed(&ext);
}
offered_extensions->push_back(ext);
}
}
}
static void NegotiateRtpHeaderExtensions(
const RtpHeaderExtensions& local_extensions,
const RtpHeaderExtensions& offered_extensions,
RtpHeaderExtensions* negotiated_extenstions) {
RtpHeaderExtensions::const_iterator ours;
for (ours = local_extensions.begin();
ours != local_extensions.end(); ++ours) {
RtpHeaderExtension theirs;
if (FindByUri(offered_extensions, *ours, &theirs)) {
// We respond with their RTP header extension id.
negotiated_extenstions->push_back(theirs);
}
}
}
static void StripCNCodecs(AudioCodecs* audio_codecs) {
AudioCodecs::iterator iter = audio_codecs->begin();
while (iter != audio_codecs->end()) {
if (stricmp(iter->name.c_str(), kComfortNoiseCodecName) == 0) {
iter = audio_codecs->erase(iter);
} else {
++iter;
}
}
}
// Create a media content to be answered in a session-accept,
// according to the given options.rtcp_mux, options.streams, codecs,
// crypto, and streams. If we don't currently have crypto (in
// current_cryptos) and it is enabled (in secure_policy), crypto is
// created (according to crypto_suites). If add_legacy_stream is
// true, and current_streams is empty, a legacy stream is created.
// The codecs, rtcp_mux, and crypto are all negotiated with the offer
// from the incoming session-initiate. If the negotiation fails, this
// method returns false. The created content is added to the offer.
template <class C>
static bool CreateMediaContentAnswer(
const MediaContentDescriptionImpl<C>* offer,
const MediaSessionOptions& options,
const std::vector<C>& local_codecs,
const SecurePolicy& sdes_policy,
const CryptoParamsVec* current_cryptos,
const RtpHeaderExtensions& local_rtp_extenstions,
StreamParamsVec* current_streams,
bool add_legacy_stream,
bool bundle_enabled,
MediaContentDescriptionImpl<C>* answer) {
std::vector<C> negotiated_codecs;
NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs);
answer->AddCodecs(negotiated_codecs);
answer->SortCodecs();
answer->set_protocol(offer->protocol());
RtpHeaderExtensions negotiated_rtp_extensions;
NegotiateRtpHeaderExtensions(local_rtp_extenstions,
offer->rtp_header_extensions(),
&negotiated_rtp_extensions);
answer->set_rtp_header_extensions(negotiated_rtp_extensions);
answer->set_rtcp_mux(options.rtcp_mux_enabled && offer->rtcp_mux());
if (sdes_policy != SEC_DISABLED) {
CryptoParams crypto;
if (SelectCrypto(offer, bundle_enabled, &crypto)) {
if (current_cryptos) {
FindMatchingCrypto(*current_cryptos, crypto, &crypto);
}
answer->AddCrypto(crypto);
}
}
if (answer->cryptos().empty() &&
(offer->crypto_required() == CT_SDES || sdes_policy == SEC_REQUIRED)) {
return false;
}
if (!AddStreamParams(
answer->type(), options.streams, current_streams,
answer, add_legacy_stream)) {
return false; // Something went seriously wrong.
}
// Make sure the answer media content direction is per default set as
// described in RFC3264 section 6.1.
switch (offer->direction()) {
case MD_INACTIVE:
answer->set_direction(MD_INACTIVE);
break;
case MD_SENDONLY:
answer->set_direction(MD_RECVONLY);
break;
case MD_RECVONLY:
answer->set_direction(MD_SENDONLY);
break;
case MD_SENDRECV:
answer->set_direction(MD_SENDRECV);
break;
default:
break;
}
return true;
}
static bool IsMediaProtocolSupported(MediaType type,
const std::string& protocol,
bool secure_transport) {
// Data channels can have a protocol of SCTP or SCTP/DTLS.
if (type == MEDIA_TYPE_DATA &&
((protocol == kMediaProtocolSctp && !secure_transport)||
(protocol == kMediaProtocolDtlsSctp && secure_transport))) {
return true;
}
// Since not all applications serialize and deserialize the media protocol,
// we will have to accept |protocol| to be empty.
return protocol == kMediaProtocolAvpf || protocol.empty() ||
protocol == kMediaProtocolSavpf ||
(protocol == kMediaProtocolDtlsSavpf && secure_transport);
}
static void SetMediaProtocol(bool secure_transport,
MediaContentDescription* desc) {
if (!desc->cryptos().empty() || secure_transport)
desc->set_protocol(kMediaProtocolSavpf);
else
desc->set_protocol(kMediaProtocolAvpf);
}
// Gets the TransportInfo of the given |content_name| from the
// |current_description|. If doesn't exist, returns a new one.
static const TransportDescription* GetTransportDescription(
const std::string& content_name,
const SessionDescription* current_description) {
const TransportDescription* desc = NULL;
if (current_description) {
const TransportInfo* info =
current_description->GetTransportInfoByName(content_name);
if (info) {
desc = &info->description;
}
}
return desc;
}
// Gets the current DTLS state from the transport description.
static bool IsDtlsActive(
const std::string& content_name,
const SessionDescription* current_description) {
if (!current_description)
return false;
const ContentInfo* content =
current_description->GetContentByName(content_name);
if (!content)
return false;
const TransportDescription* current_tdesc =
GetTransportDescription(content_name, current_description);
if (!current_tdesc)
return false;
return current_tdesc->secure();
}
std::string MediaTypeToString(MediaType type) {
std::string type_str;
switch (type) {
case MEDIA_TYPE_AUDIO:
type_str = "audio";
break;
case MEDIA_TYPE_VIDEO:
type_str = "video";
break;
case MEDIA_TYPE_DATA:
type_str = "data";
break;
default:
ASSERT(false);
break;
}
return type_str;
}
void MediaSessionOptions::AddStream(MediaType type,
const std::string& id,
const std::string& sync_label) {
AddStreamInternal(type, id, sync_label, 1);
}
void MediaSessionOptions::AddVideoStream(
const std::string& id,
const std::string& sync_label,
int num_sim_layers) {
AddStreamInternal(MEDIA_TYPE_VIDEO, id, sync_label, num_sim_layers);
}
void MediaSessionOptions::AddStreamInternal(
MediaType type,
const std::string& id,
const std::string& sync_label,
int num_sim_layers) {
streams.push_back(Stream(type, id, sync_label, num_sim_layers));
if (type == MEDIA_TYPE_VIDEO)
has_video = true;
else if (type == MEDIA_TYPE_AUDIO)
has_audio = true;
// If we haven't already set the data_channel_type, and we add a
// stream, we assume it's an RTP data stream.
else if (type == MEDIA_TYPE_DATA && data_channel_type == DCT_NONE)
data_channel_type = DCT_RTP;
}
void MediaSessionOptions::RemoveStream(MediaType type,
const std::string& id) {
Streams::iterator stream_it = streams.begin();
for (; stream_it != streams.end(); ++stream_it) {
if (stream_it->type == type && stream_it->id == id) {
streams.erase(stream_it);
return;
}
}
ASSERT(false);
}
MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
const TransportDescriptionFactory* transport_desc_factory)
: secure_(SEC_DISABLED),
add_legacy_(true),
transport_desc_factory_(transport_desc_factory) {
}
MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
ChannelManager* channel_manager,
const TransportDescriptionFactory* transport_desc_factory)
: secure_(SEC_DISABLED),
add_legacy_(true),
transport_desc_factory_(transport_desc_factory) {
channel_manager->GetSupportedAudioCodecs(&audio_codecs_);
channel_manager->GetSupportedAudioRtpHeaderExtensions(&audio_rtp_extensions_);
channel_manager->GetSupportedVideoCodecs(&video_codecs_);
channel_manager->GetSupportedVideoRtpHeaderExtensions(&video_rtp_extensions_);
channel_manager->GetSupportedDataCodecs(&data_codecs_);
}
SessionDescription* MediaSessionDescriptionFactory::CreateOffer(
const MediaSessionOptions& options,
const SessionDescription* current_description) const {
scoped_ptr<SessionDescription> offer(new SessionDescription());
StreamParamsVec current_streams;
GetCurrentStreamParams(current_description, &current_streams);
AudioCodecs audio_codecs;
VideoCodecs video_codecs;
DataCodecs data_codecs;
GetCodecsToOffer(current_description, &audio_codecs, &video_codecs,
&data_codecs);
if (!options.vad_enabled) {
// If application doesn't want CN codecs in offer.
StripCNCodecs(&audio_codecs);
}
RtpHeaderExtensions audio_rtp_extensions;
RtpHeaderExtensions video_rtp_extensions;
GetRtpHdrExtsToOffer(current_description, &audio_rtp_extensions,
&video_rtp_extensions);
bool audio_added = false;
bool video_added = false;
bool data_added = false;
// Iterate through the contents of |current_description| to maintain the order
// of the m-lines in the new offer.
if (current_description) {
ContentInfos::const_iterator it = current_description->contents().begin();
for (; it != current_description->contents().end(); ++it) {
if (IsMediaContentOfType(&*it, MEDIA_TYPE_AUDIO) && options.has_audio) {
if (!AddAudioContentForOffer(options, current_description,
audio_rtp_extensions, audio_codecs,
&current_streams, offer.get())) {
return NULL;
}
audio_added = true;
} else if (IsMediaContentOfType(&*it, MEDIA_TYPE_VIDEO) &&
options.has_video) {
if (!AddVideoContentForOffer(options, current_description,
video_rtp_extensions, video_codecs,
&current_streams, offer.get())) {
return NULL;
}
video_added = true;
} else if (IsMediaContentOfType(&*it, MEDIA_TYPE_DATA) &&
options.has_data()) {
if (!AddDataContentForOffer(options, current_description, &data_codecs,
&current_streams, offer.get())) {
return NULL;
}
data_added = true;
}
}
}
// Append contents that are not in |current_description|.
if (!audio_added && options.has_audio &&
!AddAudioContentForOffer(options, current_description,
audio_rtp_extensions, audio_codecs,
&current_streams, offer.get())) {
return NULL;
}
if (!video_added && options.has_video &&
!AddVideoContentForOffer(options, current_description,
video_rtp_extensions, video_codecs,
&current_streams, offer.get())) {
return NULL;
}
if (!data_added && options.has_data() &&
!AddDataContentForOffer(options, current_description, &data_codecs,
&current_streams, offer.get())) {
return NULL;
}
// Bundle the contents together, if we've been asked to do so, and update any
// parameters that need to be tweaked for BUNDLE.
if (options.bundle_enabled) {
ContentGroup offer_bundle(GROUP_TYPE_BUNDLE);
for (ContentInfos::const_iterator content = offer->contents().begin();
content != offer->contents().end(); ++content) {
offer_bundle.AddContentName(content->name);
}
offer->AddGroup(offer_bundle);
if (!UpdateTransportInfoForBundle(offer_bundle, offer.get())) {
LOG(LS_ERROR) << "CreateOffer failed to UpdateTransportInfoForBundle.";
return NULL;
}
if (!UpdateCryptoParamsForBundle(offer_bundle, offer.get())) {
LOG(LS_ERROR) << "CreateOffer failed to UpdateCryptoParamsForBundle.";
return NULL;
}
}
return offer.release();
}
SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
const SessionDescription* offer, const MediaSessionOptions& options,
const SessionDescription* current_description) const {
// The answer contains the intersection of the codecs in the offer with the
// codecs we support, ordered by our local preference. As indicated by
// XEP-0167, we retain the same payload ids from the offer in the answer.
scoped_ptr<SessionDescription> answer(new SessionDescription());
StreamParamsVec current_streams;
GetCurrentStreamParams(current_description, &current_streams);
if (offer) {
ContentInfos::const_iterator it = offer->contents().begin();
for (; it != offer->contents().end(); ++it) {
if (IsMediaContentOfType(&*it, MEDIA_TYPE_AUDIO)) {
if (!AddAudioContentForAnswer(offer, options, current_description,
&current_streams, answer.get())) {
return NULL;
}
} else if (IsMediaContentOfType(&*it, MEDIA_TYPE_VIDEO)) {
if (!AddVideoContentForAnswer(offer, options, current_description,
&current_streams, answer.get())) {
return NULL;
}
} else {
ASSERT(IsMediaContentOfType(&*it, MEDIA_TYPE_DATA));
if (!AddDataContentForAnswer(offer, options, current_description,
&current_streams, answer.get())) {
return NULL;
}
}
}
}
// If the offer supports BUNDLE, and we want to use it too, create a BUNDLE
// group in the answer with the appropriate content names.
if (offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled) {
const ContentGroup* offer_bundle = offer->GetGroupByName(GROUP_TYPE_BUNDLE);
ContentGroup answer_bundle(GROUP_TYPE_BUNDLE);
for (ContentInfos::const_iterator content = answer->contents().begin();
content != answer->contents().end(); ++content) {
if (!content->rejected && offer_bundle->HasContentName(content->name)) {
answer_bundle.AddContentName(content->name);
}
}
if (answer_bundle.FirstContentName()) {
answer->AddGroup(answer_bundle);
// Share the same ICE credentials and crypto params across all contents,
// as BUNDLE requires.
if (!UpdateTransportInfoForBundle(answer_bundle, answer.get())) {
LOG(LS_ERROR) << "CreateAnswer failed to UpdateTransportInfoForBundle.";
return NULL;
}
if (!UpdateCryptoParamsForBundle(answer_bundle, answer.get())) {
LOG(LS_ERROR) << "CreateAnswer failed to UpdateCryptoParamsForBundle.";
return NULL;
}
}
}
return answer.release();
}
void MediaSessionDescriptionFactory::GetCodecsToOffer(
const SessionDescription* current_description,
AudioCodecs* audio_codecs,
VideoCodecs* video_codecs,
DataCodecs* data_codecs) const {
UsedPayloadTypes used_pltypes;
audio_codecs->clear();
video_codecs->clear();
data_codecs->clear();
// First - get all codecs from the current description if the media type
// is used.
// Add them to |used_pltypes| so the payloadtype is not reused if a new media
// type is added.
if (current_description) {
const AudioContentDescription* audio =
GetFirstAudioContentDescription(current_description);
if (audio) {
*audio_codecs = audio->codecs();
used_pltypes.FindAndSetIdUsed<AudioCodec>(audio_codecs);
}
const VideoContentDescription* video =
GetFirstVideoContentDescription(current_description);
if (video) {
*video_codecs = video->codecs();
used_pltypes.FindAndSetIdUsed<VideoCodec>(video_codecs);
}
const DataContentDescription* data =
GetFirstDataContentDescription(current_description);
if (data) {
*data_codecs = data->codecs();
used_pltypes.FindAndSetIdUsed<DataCodec>(data_codecs);
}
}
// Add our codecs that are not in |current_description|.
FindCodecsToOffer<AudioCodec>(audio_codecs_, audio_codecs, &used_pltypes);
FindCodecsToOffer<VideoCodec>(video_codecs_, video_codecs, &used_pltypes);
FindCodecsToOffer<DataCodec>(data_codecs_, data_codecs, &used_pltypes);
}
void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer(
const SessionDescription* current_description,
RtpHeaderExtensions* audio_extensions,
RtpHeaderExtensions* video_extensions) const {
// All header extensions allocated from the same range to avoid potential
// issues when using BUNDLE.
UsedRtpHeaderExtensionIds used_ids;
audio_extensions->clear();
video_extensions->clear();
// First - get all extensions from the current description if the media type
// is used.
// Add them to |used_ids| so the local ids are not reused if a new media
// type is added.
if (current_description) {
const AudioContentDescription* audio =
GetFirstAudioContentDescription(current_description);
if (audio) {
*audio_extensions = audio->rtp_header_extensions();
used_ids.FindAndSetIdUsed(audio_extensions);
}
const VideoContentDescription* video =
GetFirstVideoContentDescription(current_description);
if (video) {
*video_extensions = video->rtp_header_extensions();
used_ids.FindAndSetIdUsed(video_extensions);
}
}
// Add our default RTP header extensions that are not in
// |current_description|.
FindAndSetRtpHdrExtUsed(audio_rtp_header_extensions(), audio_extensions,
*video_extensions, &used_ids);
FindAndSetRtpHdrExtUsed(video_rtp_header_extensions(), video_extensions,
*audio_extensions, &used_ids);
}
bool MediaSessionDescriptionFactory::AddTransportOffer(
const std::string& content_name,
const TransportOptions& transport_options,
const SessionDescription* current_desc,
SessionDescription* offer_desc) const {
if (!transport_desc_factory_)
return false;
const TransportDescription* current_tdesc =
GetTransportDescription(content_name, current_desc);
rtc::scoped_ptr<TransportDescription> new_tdesc(
transport_desc_factory_->CreateOffer(transport_options, current_tdesc));
bool ret = (new_tdesc.get() != NULL &&
offer_desc->AddTransportInfo(TransportInfo(content_name, *new_tdesc)));
if (!ret) {
LOG(LS_ERROR)
<< "Failed to AddTransportOffer, content name=" << content_name;
}
return ret;
}
TransportDescription* MediaSessionDescriptionFactory::CreateTransportAnswer(
const std::string& content_name,
const SessionDescription* offer_desc,
const TransportOptions& transport_options,
const SessionDescription* current_desc) const {
if (!transport_desc_factory_)
return NULL;
const TransportDescription* offer_tdesc =
GetTransportDescription(content_name, offer_desc);
const TransportDescription* current_tdesc =
GetTransportDescription(content_name, current_desc);
return
transport_desc_factory_->CreateAnswer(offer_tdesc, transport_options,
current_tdesc);
}
bool MediaSessionDescriptionFactory::AddTransportAnswer(
const std::string& content_name,
const TransportDescription& transport_desc,
SessionDescription* answer_desc) const {
if (!answer_desc->AddTransportInfo(TransportInfo(content_name,
transport_desc))) {
LOG(LS_ERROR)
<< "Failed to AddTransportAnswer, content name=" << content_name;
return false;
}
return true;
}
bool MediaSessionDescriptionFactory::AddAudioContentForOffer(
const MediaSessionOptions& options,
const SessionDescription* current_description,
const RtpHeaderExtensions& audio_rtp_extensions,
const AudioCodecs& audio_codecs,
StreamParamsVec* current_streams,
SessionDescription* desc) const {
cricket::SecurePolicy sdes_policy =
IsDtlsActive(CN_AUDIO, current_description) ?
cricket::SEC_DISABLED : secure();
scoped_ptr<AudioContentDescription> audio(new AudioContentDescription());
std::vector<std::string> crypto_suites;
GetSupportedAudioCryptoSuites(&crypto_suites);
if (!CreateMediaContentOffer(
options,
audio_codecs,
sdes_policy,
GetCryptos(GetFirstAudioContentDescription(current_description)),
crypto_suites,
audio_rtp_extensions,
add_legacy_,
current_streams,
audio.get())) {
return false;
}
audio->set_lang(lang_);
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
SetMediaProtocol(secure_transport, audio.get());
desc->AddContent(CN_AUDIO, NS_JINGLE_RTP, audio.release());
if (!AddTransportOffer(CN_AUDIO, options.transport_options,
current_description, desc)) {
return false;
}
return true;
}
bool MediaSessionDescriptionFactory::AddVideoContentForOffer(
const MediaSessionOptions& options,
const SessionDescription* current_description,
const RtpHeaderExtensions& video_rtp_extensions,
const VideoCodecs& video_codecs,
StreamParamsVec* current_streams,
SessionDescription* desc) const {
cricket::SecurePolicy sdes_policy =
IsDtlsActive(CN_VIDEO, current_description) ?
cricket::SEC_DISABLED : secure();
scoped_ptr<VideoContentDescription> video(new VideoContentDescription());
std::vector<std::string> crypto_suites;
GetSupportedVideoCryptoSuites(&crypto_suites);
if (!CreateMediaContentOffer(
options,
video_codecs,
sdes_policy,
GetCryptos(GetFirstVideoContentDescription(current_description)),
crypto_suites,
video_rtp_extensions,
add_legacy_,
current_streams,
video.get())) {
return false;
}
video->set_bandwidth(options.video_bandwidth);
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
SetMediaProtocol(secure_transport, video.get());
desc->AddContent(CN_VIDEO, NS_JINGLE_RTP, video.release());
if (!AddTransportOffer(CN_VIDEO, options.transport_options,
current_description, desc)) {
return false;
}
return true;
}
bool MediaSessionDescriptionFactory::AddDataContentForOffer(
const MediaSessionOptions& options,
const SessionDescription* current_description,
DataCodecs* data_codecs,
StreamParamsVec* current_streams,
SessionDescription* desc) const {
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
scoped_ptr<DataContentDescription> data(new DataContentDescription());
bool is_sctp = (options.data_channel_type == DCT_SCTP);
FilterDataCodecs(data_codecs, is_sctp);
cricket::SecurePolicy sdes_policy =
IsDtlsActive(CN_DATA, current_description) ?
cricket::SEC_DISABLED : secure();
std::vector<std::string> crypto_suites;
if (is_sctp) {
// SDES doesn't make sense for SCTP, so we disable it, and we only
// get SDES crypto suites for RTP-based data channels.
sdes_policy = cricket::SEC_DISABLED;
// Unlike SetMediaProtocol below, we need to set the protocol
// before we call CreateMediaContentOffer. Otherwise,
// CreateMediaContentOffer won't know this is SCTP and will
// generate SSRCs rather than SIDs.
data->set_protocol(
secure_transport ? kMediaProtocolDtlsSctp : kMediaProtocolSctp);
} else {
GetSupportedDataCryptoSuites(&crypto_suites);
}
if (!CreateMediaContentOffer(
options,
*data_codecs,
sdes_policy,
GetCryptos(GetFirstDataContentDescription(current_description)),
crypto_suites,
RtpHeaderExtensions(),
add_legacy_,
current_streams,
data.get())) {
return false;
}
if (is_sctp) {
desc->AddContent(CN_DATA, NS_JINGLE_DRAFT_SCTP, data.release());
} else {
data->set_bandwidth(options.data_bandwidth);
SetMediaProtocol(secure_transport, data.get());
desc->AddContent(CN_DATA, NS_JINGLE_RTP, data.release());
}
if (!AddTransportOffer(CN_DATA, options.transport_options,
current_description, desc)) {
return false;
}
return true;
}
bool MediaSessionDescriptionFactory::AddAudioContentForAnswer(
const SessionDescription* offer,
const MediaSessionOptions& options,
const SessionDescription* current_description,
StreamParamsVec* current_streams,
SessionDescription* answer) const {
const ContentInfo* audio_content = GetFirstAudioContent(offer);
scoped_ptr<TransportDescription> audio_transport(
CreateTransportAnswer(audio_content->name, offer,
options.transport_options,
current_description));
if (!audio_transport) {
return false;
}
AudioCodecs audio_codecs = audio_codecs_;
if (!options.vad_enabled) {
StripCNCodecs(&audio_codecs);
}
bool bundle_enabled =
offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled;
scoped_ptr<AudioContentDescription> audio_answer(
new AudioContentDescription());
// Do not require or create SDES cryptos if DTLS is used.
cricket::SecurePolicy sdes_policy =
audio_transport->secure() ? cricket::SEC_DISABLED : secure();
if (!CreateMediaContentAnswer(
static_cast<const AudioContentDescription*>(
audio_content->description),
options,
audio_codecs,
sdes_policy,
GetCryptos(GetFirstAudioContentDescription(current_description)),
audio_rtp_extensions_,
current_streams,
add_legacy_,
bundle_enabled,
audio_answer.get())) {
return false; // Fails the session setup.
}
bool rejected = !options.has_audio || audio_content->rejected ||
!IsMediaProtocolSupported(MEDIA_TYPE_AUDIO,
audio_answer->protocol(),
audio_transport->secure());
if (!rejected) {
AddTransportAnswer(audio_content->name, *(audio_transport.get()), answer);
} else {
// RFC 3264
// The answer MUST contain the same number of m-lines as the offer.
LOG(LS_INFO) << "Audio is not supported in the answer.";
}
answer->AddContent(audio_content->name, audio_content->type, rejected,
audio_answer.release());
return true;
}
bool MediaSessionDescriptionFactory::AddVideoContentForAnswer(
const SessionDescription* offer,
const MediaSessionOptions& options,
const SessionDescription* current_description,
StreamParamsVec* current_streams,
SessionDescription* answer) const {
const ContentInfo* video_content = GetFirstVideoContent(offer);
scoped_ptr<TransportDescription> video_transport(
CreateTransportAnswer(video_content->name, offer,
options.transport_options,
current_description));
if (!video_transport) {
return false;
}
scoped_ptr<VideoContentDescription> video_answer(
new VideoContentDescription());
// Do not require or create SDES cryptos if DTLS is used.
cricket::SecurePolicy sdes_policy =
video_transport->secure() ? cricket::SEC_DISABLED : secure();
bool bundle_enabled =
offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled;
if (!CreateMediaContentAnswer(
static_cast<const VideoContentDescription*>(
video_content->description),
options,
video_codecs_,
sdes_policy,
GetCryptos(GetFirstVideoContentDescription(current_description)),
video_rtp_extensions_,
current_streams,
add_legacy_,
bundle_enabled,
video_answer.get())) {
return false;
}
bool rejected = !options.has_video || video_content->rejected ||
!IsMediaProtocolSupported(MEDIA_TYPE_VIDEO,
video_answer->protocol(),
video_transport->secure());
if (!rejected) {
if (!AddTransportAnswer(video_content->name, *(video_transport.get()),
answer)) {
return false;
}
video_answer->set_bandwidth(options.video_bandwidth);
} else {
// RFC 3264
// The answer MUST contain the same number of m-lines as the offer.
LOG(LS_INFO) << "Video is not supported in the answer.";
}
answer->AddContent(video_content->name, video_content->type, rejected,
video_answer.release());
return true;
}
bool MediaSessionDescriptionFactory::AddDataContentForAnswer(
const SessionDescription* offer,
const MediaSessionOptions& options,
const SessionDescription* current_description,
StreamParamsVec* current_streams,
SessionDescription* answer) const {
const ContentInfo* data_content = GetFirstDataContent(offer);
scoped_ptr<TransportDescription> data_transport(
CreateTransportAnswer(data_content->name, offer,
options.transport_options,
current_description));
if (!data_transport) {
return false;
}
bool is_sctp = (options.data_channel_type == DCT_SCTP);
std::vector<DataCodec> data_codecs(data_codecs_);
FilterDataCodecs(&data_codecs, is_sctp);
scoped_ptr<DataContentDescription> data_answer(
new DataContentDescription());
// Do not require or create SDES cryptos if DTLS is used.
cricket::SecurePolicy sdes_policy =
data_transport->secure() ? cricket::SEC_DISABLED : secure();
bool bundle_enabled =
offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled;
if (!CreateMediaContentAnswer(
static_cast<const DataContentDescription*>(
data_content->description),
options,
data_codecs_,
sdes_policy,
GetCryptos(GetFirstDataContentDescription(current_description)),
RtpHeaderExtensions(),
current_streams,
add_legacy_,
bundle_enabled,
data_answer.get())) {
return false; // Fails the session setup.
}
bool rejected = !options.has_data() || data_content->rejected ||
!IsMediaProtocolSupported(MEDIA_TYPE_DATA,
data_answer->protocol(),
data_transport->secure());
if (!rejected) {
data_answer->set_bandwidth(options.data_bandwidth);
if (!AddTransportAnswer(data_content->name, *(data_transport.get()),
answer)) {
return false;
}
} else {
// RFC 3264
// The answer MUST contain the same number of m-lines as the offer.
LOG(LS_INFO) << "Data is not supported in the answer.";
}
answer->AddContent(data_content->name, data_content->type, rejected,
data_answer.release());
return true;
}
bool IsMediaContent(const ContentInfo* content) {
return (content &&
(content->type == NS_JINGLE_RTP ||
content->type == NS_JINGLE_DRAFT_SCTP));
}
bool IsAudioContent(const ContentInfo* content) {
return IsMediaContentOfType(content, MEDIA_TYPE_AUDIO);
}
bool IsVideoContent(const ContentInfo* content) {
return IsMediaContentOfType(content, MEDIA_TYPE_VIDEO);
}
bool IsDataContent(const ContentInfo* content) {
return IsMediaContentOfType(content, MEDIA_TYPE_DATA);
}
static const ContentInfo* GetFirstMediaContent(const ContentInfos& contents,
MediaType media_type) {
for (ContentInfos::const_iterator content = contents.begin();
content != contents.end(); content++) {
if (IsMediaContentOfType(&*content, media_type)) {
return &*content;
}
}
return NULL;
}
const ContentInfo* GetFirstAudioContent(const ContentInfos& contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_AUDIO);
}
const ContentInfo* GetFirstVideoContent(const ContentInfos& contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_VIDEO);
}
const ContentInfo* GetFirstDataContent(const ContentInfos& contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_DATA);
}
static const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc,
MediaType media_type) {
if (sdesc == NULL)
return NULL;
return GetFirstMediaContent(sdesc->contents(), media_type);
}
const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc) {
return GetFirstMediaContent(sdesc, MEDIA_TYPE_AUDIO);
}
const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc) {
return GetFirstMediaContent(sdesc, MEDIA_TYPE_VIDEO);
}
const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc) {
return GetFirstMediaContent(sdesc, MEDIA_TYPE_DATA);
}
const MediaContentDescription* GetFirstMediaContentDescription(
const SessionDescription* sdesc, MediaType media_type) {
const ContentInfo* content = GetFirstMediaContent(sdesc, media_type);
const ContentDescription* description = content ? content->description : NULL;
return static_cast<const MediaContentDescription*>(description);
}
const AudioContentDescription* GetFirstAudioContentDescription(
const SessionDescription* sdesc) {
return static_cast<const AudioContentDescription*>(
GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_AUDIO));
}
const VideoContentDescription* GetFirstVideoContentDescription(
const SessionDescription* sdesc) {
return static_cast<const VideoContentDescription*>(
GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_VIDEO));
}
const DataContentDescription* GetFirstDataContentDescription(
const SessionDescription* sdesc) {
return static_cast<const DataContentDescription*>(
GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA));
}
bool GetMediaChannelNameFromComponent(
int component, MediaType media_type, std::string* channel_name) {
if (media_type == MEDIA_TYPE_AUDIO) {
if (component == ICE_CANDIDATE_COMPONENT_RTP) {
*channel_name = GICE_CHANNEL_NAME_RTP;
return true;
} else if (component == ICE_CANDIDATE_COMPONENT_RTCP) {
*channel_name = GICE_CHANNEL_NAME_RTCP;
return true;
}
} else if (media_type == MEDIA_TYPE_VIDEO) {
if (component == ICE_CANDIDATE_COMPONENT_RTP) {
*channel_name = GICE_CHANNEL_NAME_VIDEO_RTP;
return true;
} else if (component == ICE_CANDIDATE_COMPONENT_RTCP) {
*channel_name = GICE_CHANNEL_NAME_VIDEO_RTCP;
return true;
}
} else if (media_type == MEDIA_TYPE_DATA) {
if (component == ICE_CANDIDATE_COMPONENT_RTP) {
*channel_name = GICE_CHANNEL_NAME_DATA_RTP;
return true;
} else if (component == ICE_CANDIDATE_COMPONENT_RTCP) {
*channel_name = GICE_CHANNEL_NAME_DATA_RTCP;
return true;
}
}
return false;
}
bool GetMediaComponentFromChannelName(
const std::string& channel_name, int* component) {
if (channel_name == GICE_CHANNEL_NAME_RTP ||
channel_name == GICE_CHANNEL_NAME_VIDEO_RTP ||
channel_name == GICE_CHANNEL_NAME_DATA_RTP) {
*component = ICE_CANDIDATE_COMPONENT_RTP;
return true;
} else if (channel_name == GICE_CHANNEL_NAME_RTCP ||
channel_name == GICE_CHANNEL_NAME_VIDEO_RTCP ||
channel_name == GICE_CHANNEL_NAME_DATA_RTP) {
*component = ICE_CANDIDATE_COMPONENT_RTCP;
return true;
}
return false;
}
bool GetMediaTypeFromChannelName(
const std::string& channel_name, MediaType* media_type) {
if (channel_name == GICE_CHANNEL_NAME_RTP ||
channel_name == GICE_CHANNEL_NAME_RTCP) {
*media_type = MEDIA_TYPE_AUDIO;
return true;
} else if (channel_name == GICE_CHANNEL_NAME_VIDEO_RTP ||
channel_name == GICE_CHANNEL_NAME_VIDEO_RTCP) {
*media_type = MEDIA_TYPE_VIDEO;
return true;
} else if (channel_name == GICE_CHANNEL_NAME_DATA_RTP ||
channel_name == GICE_CHANNEL_NAME_DATA_RTCP) {
*media_type = MEDIA_TYPE_DATA;
return true;
}
return false;
}
} // namespace cricket
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