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platform-external-webrtc/webrtc/pc/mediasession.cc
magjed f823ededce Negotiate H264 profiles in SDP 
This CL will start to distinguish H264 profiles during SDP negotiation.
We currently don't look at the H264 profile at all and assume they are
all Constrained Baseline Level 3.1. This CL will start to check profiles
for equality when matching, and will generate the correct answer H264
level.

Each local supported H264 profile needs to be listed explicitly in the
list of local supported codecs, even if they are redundant. For example,
Baseline profile should be listed explicitly even though another profile
that is a superset of Baseline is also listed. The reason for this is to
simplify the code and avoid profile intersection during matching. So
VideoCodec::Matches will check for profile equality, and not check if
one codec is a subset of the other. This also leads to the nice property
that VideoCodec::Matches is symmetric, i.e. iif a.Matches(b) then
b.Matches(a).

BUG=webrtc:6337
TBR=tkchin@webrtc.org

Review-Url: https://codereview.webrtc.org/2483173002
Cr-Commit-Position: refs/heads/master@{#15051}
2016-11-12 17:53:08 +00:00

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/*
* Copyright 2004 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/pc/mediasession.h"
#include <algorithm> // For std::find_if, std::sort.
#include <functional>
#include <map>
#include <memory>
#include <set>
#include <unordered_map>
#include <utility>
#include "webrtc/base/base64.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/common_video/h264/profile_level_id.h"
#include "webrtc/media/base/cryptoparams.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/p2p/base/p2pconstants.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/pc/srtpfilter.h"
namespace {
const char kInline[] = "inline:";
void GetSupportedCryptoSuiteNames(void (*func)(const rtc::CryptoOptions&,
std::vector<int>*),
const rtc::CryptoOptions& crypto_options,
std::vector<std::string>* names) {
#ifdef HAVE_SRTP
std::vector<int> crypto_suites;
func(crypto_options, &crypto_suites);
for (const auto crypto : crypto_suites) {
names->push_back(rtc::SrtpCryptoSuiteToName(crypto));
}
#endif
}
} // namespace
namespace cricket {
// 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";
// We always generate offers with "UDP/TLS/RTP/SAVPF" when using DTLS-SRTP,
// but we tolerate "RTP/SAVPF" in offers we receive, for compatibility.
const char kMediaProtocolSavpf[] = "RTP/SAVPF";
const char kMediaProtocolRtpPrefix[] = "RTP/";
const char kMediaProtocolSctp[] = "SCTP";
const char kMediaProtocolDtlsSctp[] = "DTLS/SCTP";
const char kMediaProtocolUdpDtlsSctp[] = "UDP/DTLS/SCTP";
const char kMediaProtocolTcpDtlsSctp[] = "TCP/DTLS/SCTP";
RtpTransceiverDirection RtpTransceiverDirection::FromMediaContentDirection(
MediaContentDirection md) {
const bool send = (md == MD_SENDRECV || md == MD_SENDONLY);
const bool recv = (md == MD_SENDRECV || md == MD_RECVONLY);
return RtpTransceiverDirection(send, recv);
}
MediaContentDirection RtpTransceiverDirection::ToMediaContentDirection() const {
if (send && recv) {
return MD_SENDRECV;
} else if (send) {
return MD_SENDONLY;
} else if (recv) {
return MD_RECVONLY;
}
return MD_INACTIVE;
}
RtpTransceiverDirection
NegotiateRtpTransceiverDirection(RtpTransceiverDirection offer,
RtpTransceiverDirection wants) {
return RtpTransceiverDirection(offer.recv && wants.send,
offer.send && wants.recv);
}
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) {
int key_len;
int salt_len;
if (!rtc::GetSrtpKeyAndSaltLengths(
rtc::SrtpCryptoSuiteFromName(cipher), &key_len, &salt_len)) {
return false;
}
int master_key_len = key_len + salt_len;
std::string master_key;
if (!rtc::CreateRandomData(master_key_len, &master_key)) {
return false;
}
RTC_CHECK_EQ(static_cast<size_t>(master_key_len), master_key.size());
std::string key = rtc::Base64::Encode(master_key);
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 over HMAC 80 because of the low overhead.
void GetSupportedAudioCryptoSuites(const rtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites) {
#ifdef HAVE_SRTP
if (crypto_options.enable_gcm_crypto_suites) {
crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
}
crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_32);
crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
#endif
}
void GetSupportedAudioCryptoSuiteNames(const rtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names) {
GetSupportedCryptoSuiteNames(GetSupportedAudioCryptoSuites,
crypto_options, crypto_suite_names);
}
void GetSupportedVideoCryptoSuites(const rtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites) {
GetDefaultSrtpCryptoSuites(crypto_options, crypto_suites);
}
void GetSupportedVideoCryptoSuiteNames(const rtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names) {
GetSupportedCryptoSuiteNames(GetSupportedVideoCryptoSuites,
crypto_options, crypto_suite_names);
}
void GetSupportedDataCryptoSuites(const rtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites) {
GetDefaultSrtpCryptoSuites(crypto_options, crypto_suites);
}
void GetSupportedDataCryptoSuiteNames(const rtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names) {
GetSupportedCryptoSuiteNames(GetSupportedDataCryptoSuites,
crypto_options, crypto_suite_names);
}
void GetDefaultSrtpCryptoSuites(const rtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites) {
#ifdef HAVE_SRTP
if (crypto_options.enable_gcm_crypto_suites) {
crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
}
crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
#endif
}
void GetDefaultSrtpCryptoSuiteNames(const rtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names) {
GetSupportedCryptoSuiteNames(GetDefaultSrtpCryptoSuites,
crypto_options, crypto_suite_names);
}
// Support any GCM cipher (if enabled through options). 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,
const rtc::CryptoOptions& crypto_options,
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 ((crypto_options.enable_gcm_crypto_suites &&
rtc::IsGcmCryptoSuiteName(i->cipher_suite)) ||
rtc::CS_AES_CM_128_HMAC_SHA1_80 == i->cipher_suite ||
(rtc::CS_AES_CM_128_HMAC_SHA1_32 == i->cipher_suite && audio &&
!bundle)) {
return CreateCryptoParams(i->tag, i->cipher_suite, crypto);
}
}
return false;
}
// 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_t>* ssrcs) {
for (int i = 0; i < num_ssrcs; i++) {
uint32_t candidate;
do {
candidate = rtc::CreateRandomNonZeroId();
} while (GetStreamBySsrc(params_vec, candidate) ||
std::count(ssrcs->begin(), ssrcs->end(), candidate) > 0);
ssrcs->push_back(candidate);
}
}
// 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.
const char* codec_name =
sctp ? kGoogleRtpDataCodecName : kGoogleSctpDataCodecName;
for (std::vector<DataCodec>::iterator iter = codecs->begin();
iter != codecs->end();) {
if (CodecNamesEq(iter->name, codec_name)) {
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<webrtc::RtpExtension> {
public:
UsedRtpHeaderExtensionIds()
: UsedIds<webrtc::RtpExtension>(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& options,
StreamParamsVec* current_streams,
MediaContentDescriptionImpl<C>* content_description,
const bool add_legacy_stream) {
// SCTP streams are not negotiated using SDP/ContentDescriptions.
if (IsSctp(content_description)) {
return true;
}
const bool include_rtx_streams =
ContainsRtxCodec(content_description->codecs());
const MediaSessionOptions::Streams& streams = options.streams;
if (streams.empty() && add_legacy_stream) {
// TODO(perkj): Remove this legacy stream when all apps use StreamParams.
std::vector<uint32_t> ssrcs;
int num_ssrcs = include_rtx_streams ? 2 : 1;
GenerateSsrcs(*current_streams, num_ssrcs, &ssrcs);
if (include_rtx_streams) {
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.
const StreamParams* param =
GetStreamByIds(*current_streams, "", stream_it->id);
// groupid is empty for StreamParams generated using
// MediaSessionDescriptionFactory.
if (!param) {
// This is a new stream.
std::vector<uint32_t> ssrcs;
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 extra ssrcs for include_rtx_streams case.
if (include_rtx_streams) {
// Generate an RTX ssrc for every ssrc in the group.
std::vector<uint32_t> rtx_ssrcs;
GenerateSsrcs(*current_streams, static_cast<int>(ssrcs.size()),
&rtx_ssrcs);
for (size_t i = 0; i < ssrcs.size(); ++i) {
stream_param.AddFidSsrc(ssrcs[i], rtx_ssrcs[i]);
}
content_description->set_multistream(true);
}
stream_param.cname = options.rtcp_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, pwd and DTLS role 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.
const 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;
ConnectionRole selected_connection_role =
selected_transport_info->description.connection_role;
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;
it->description.connection_role = selected_connection_role;
}
}
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 IsRtpProtocol(const std::string& protocol) {
return protocol.empty() ||
(protocol.find(cricket::kMediaProtocolRtpPrefix) != std::string::npos);
}
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 = IsRtpProtocol(media_desc->protocol());
}
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;
}
static TransportOptions GetTransportOptions(const MediaSessionOptions& options,
const std::string& content_name) {
TransportOptions transport_options;
auto it = options.transport_options.find(content_name);
if (it != options.transport_options.end()) {
transport_options = it->second;
}
transport_options.enable_ice_renomination = options.enable_ice_renomination;
return transport_options;
}
// 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);
if (secure_policy == SEC_REQUIRED) {
offer->set_crypto_required(CT_SDES);
}
offer->set_rtcp_mux(options.rtcp_mux_enabled);
if (offer->type() == cricket::MEDIA_TYPE_VIDEO) {
offer->set_rtcp_reduced_size(true);
}
offer->set_multistream(options.is_muc);
offer->set_rtp_header_extensions(rtp_extensions);
if (!AddStreamParams(offer->type(), options, 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 bool ReferencedCodecsMatch(const std::vector<C>& codecs1,
const int codec1_id,
const std::vector<C>& codecs2,
const int codec2_id) {
const C* codec1 = FindCodecById(codecs1, codec1_id);
const C* codec2 = FindCodecById(codecs2, codec2_id);
return codec1 != nullptr && codec2 != nullptr && codec1->Matches(*codec2);
}
template <class C>
static void NegotiateCodecs(const std::vector<C>& local_codecs,
const std::vector<C>& offered_codecs,
std::vector<C>* negotiated_codecs) {
for (const C& ours : local_codecs) {
C theirs;
// Note that we intentionally only find one matching codec for each of our
// local codecs, in case the remote offer contains duplicate codecs.
if (FindMatchingCodec(local_codecs, offered_codecs, ours, &theirs)) {
C negotiated = ours;
negotiated.IntersectFeedbackParams(theirs);
if (IsRtxCodec(negotiated)) {
const auto apt_it =
theirs.params.find(kCodecParamAssociatedPayloadType);
// FindMatchingCodec shouldn't return something with no apt value.
RTC_DCHECK(apt_it != theirs.params.end());
negotiated.SetParam(kCodecParamAssociatedPayloadType, apt_it->second);
}
if (CodecNamesEq(ours.name.c_str(), kH264CodecName)) {
webrtc::H264::GenerateProfileLevelIdForAnswer(
ours.params, theirs.params, &negotiated.params);
}
negotiated.id = theirs.id;
negotiated.name = theirs.name;
negotiated_codecs->push_back(std::move(negotiated));
}
}
// 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.
std::unordered_map<int, int> payload_type_preferences;
int preference = static_cast<int>(offered_codecs.size() + 1);
for (const C& codec : offered_codecs) {
payload_type_preferences[codec.id] = preference--;
}
std::sort(negotiated_codecs->begin(), negotiated_codecs->end(),
[&payload_type_preferences](const C& a, const C& b) {
return payload_type_preferences[a.id] >
payload_type_preferences[b.id];
});
}
// Finds a codec in |codecs2| that matches |codec_to_match|, which is
// a member of |codecs1|. If |codec_to_match| is an RTX codec, both
// the codecs themselves and their associated codecs must match.
template <class C>
static bool FindMatchingCodec(const std::vector<C>& codecs1,
const std::vector<C>& codecs2,
const C& codec_to_match,
C* found_codec) {
for (const C& potential_match : codecs2) {
if (potential_match.Matches(codec_to_match)) {
if (IsRtxCodec(codec_to_match)) {
int apt_value_1 = 0;
int apt_value_2 = 0;
if (!codec_to_match.GetParam(kCodecParamAssociatedPayloadType,
&apt_value_1) ||
!potential_match.GetParam(kCodecParamAssociatedPayloadType,
&apt_value_2)) {
LOG(LS_WARNING) << "RTX missing associated payload type.";
continue;
}
if (!ReferencedCodecsMatch(codecs1, apt_value_1, codecs2,
apt_value_2)) {
continue;
}
}
if (found_codec) {
*found_codec = potential_match;
}
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) {
// Add all new codecs that are not RTX codecs.
for (const C& reference_codec : reference_codecs) {
if (!IsRtxCodec(reference_codec) &&
!FindMatchingCodec<C>(reference_codecs, *offered_codecs,
reference_codec, nullptr)) {
C codec = reference_codec;
used_pltypes->FindAndSetIdUsed(&codec);
offered_codecs->push_back(codec);
}
}
// Add all new RTX codecs.
for (const C& reference_codec : reference_codecs) {
if (IsRtxCodec(reference_codec) &&
!FindMatchingCodec<C>(reference_codecs, *offered_codecs,
reference_codec, nullptr)) {
C rtx_codec = reference_codec;
std::string associated_pt_str;
if (!rtx_codec.GetParam(kCodecParamAssociatedPayloadType,
&associated_pt_str)) {
LOG(LS_WARNING) << "RTX codec " << rtx_codec.name
<< " is missing an associated payload type.";
continue;
}
int associated_pt;
if (!rtc::FromString(associated_pt_str, &associated_pt)) {
LOG(LS_WARNING) << "Couldn't convert payload type " << associated_pt_str
<< " of RTX codec " << rtx_codec.name
<< " to an integer.";
continue;
}
// Find the associated reference codec for the reference RTX codec.
const C* associated_codec =
FindCodecById(reference_codecs, associated_pt);
if (!associated_codec) {
LOG(LS_WARNING) << "Couldn't find associated codec with payload type "
<< associated_pt << " for RTX codec " << rtx_codec.name
<< ".";
continue;
}
// Find a codec in the offered list that matches the reference codec.
// Its payload type may be different than the reference codec.
C matching_codec;
if (!FindMatchingCodec<C>(reference_codecs, *offered_codecs,
*associated_codec, &matching_codec)) {
LOG(LS_WARNING) << "Couldn't find matching " << associated_codec->name
<< " codec.";
continue;
}
rtx_codec.params[kCodecParamAssociatedPayloadType] =
rtc::ToString(matching_codec.id);
used_pltypes->FindAndSetIdUsed(&rtx_codec);
offered_codecs->push_back(rtx_codec);
}
}
}
static bool FindByUri(const RtpHeaderExtensions& extensions,
const webrtc::RtpExtension& ext_to_match,
webrtc::RtpExtension* 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;
}
// Iterates through |offered_extensions|, adding each one to |all_extensions|
// and |used_ids|, and resolving ID conflicts. If an offered extension has the
// same URI as one in |all_extensions|, it will re-use the same ID and won't be
// treated as a conflict.
static void FindAndSetRtpHdrExtUsed(RtpHeaderExtensions* offered_extensions,
RtpHeaderExtensions* all_extensions,
UsedRtpHeaderExtensionIds* used_ids) {
for (auto& extension : *offered_extensions) {
webrtc::RtpExtension existing;
if (FindByUri(*all_extensions, extension, &existing)) {
extension.id = existing.id;
} else {
used_ids->FindAndSetIdUsed(&extension);
all_extensions->push_back(extension);
}
}
}
// Adds |reference_extensions| to |offered_extensions|, while updating
// |all_extensions| and |used_ids|.
static void FindRtpHdrExtsToOffer(
const RtpHeaderExtensions& reference_extensions,
RtpHeaderExtensions* offered_extensions,
RtpHeaderExtensions* all_extensions,
UsedRtpHeaderExtensionIds* used_ids) {
for (auto reference_extension : reference_extensions) {
if (!FindByUri(*offered_extensions, reference_extension, NULL)) {
webrtc::RtpExtension existing;
if (FindByUri(*all_extensions, reference_extension, &existing)) {
offered_extensions->push_back(existing);
} else {
used_ids->FindAndSetIdUsed(&reference_extension);
all_extensions->push_back(reference_extension);
offered_extensions->push_back(reference_extension);
}
}
}
}
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) {
webrtc::RtpExtension 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->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 (answer->type() == cricket::MEDIA_TYPE_VIDEO) {
answer->set_rtcp_reduced_size(offer->rtcp_reduced_size());
}
if (sdes_policy != SEC_DISABLED) {
CryptoParams crypto;
if (SelectCrypto(offer, bundle_enabled, options.crypto_options, &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, 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.
const bool is_data = !IsRtpProtocol(answer->protocol());
const bool has_send_streams = !answer->streams().empty();
const bool wants_send = has_send_streams || is_data;
const bool recv_audio =
answer->type() == cricket::MEDIA_TYPE_AUDIO && options.recv_audio;
const bool recv_video =
answer->type() == cricket::MEDIA_TYPE_VIDEO && options.recv_video;
const bool recv_data =
answer->type() == cricket::MEDIA_TYPE_DATA;
const bool wants_receive = recv_audio || recv_video || recv_data;
auto offer_rtd =
RtpTransceiverDirection::FromMediaContentDirection(offer->direction());
auto wants_rtd = RtpTransceiverDirection(wants_send, wants_receive);
answer->set_direction(NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd)
.ToMediaContentDirection());
return true;
}
static bool IsDtlsRtp(const std::string& protocol) {
// Most-likely values first.
return protocol == "UDP/TLS/RTP/SAVPF" || protocol == "TCP/TLS/RTP/SAVPF" ||
protocol == "UDP/TLS/RTP/SAVP" || protocol == "TCP/TLS/RTP/SAVP";
}
static bool IsPlainRtp(const std::string& protocol) {
// Most-likely values first.
return protocol == "RTP/SAVPF" || protocol == "RTP/AVPF" ||
protocol == "RTP/SAVP" || protocol == "RTP/AVP";
}
static bool IsDtlsSctp(const std::string& protocol) {
return protocol == "DTLS/SCTP";
}
static bool IsPlainSctp(const std::string& protocol) {
return protocol == "SCTP";
}
static bool IsMediaProtocolSupported(MediaType type,
const std::string& protocol,
bool secure_transport) {
// Since not all applications serialize and deserialize the media protocol,
// we will have to accept |protocol| to be empty.
if (protocol.empty()) {
return true;
}
if (type == MEDIA_TYPE_DATA) {
// Check for SCTP, but also for RTP for RTP-based data channels.
// TODO(pthatcher): Remove RTP once RTP-based data channels are gone.
if (secure_transport) {
// Most likely scenarios first.
return IsDtlsSctp(protocol) || IsDtlsRtp(protocol) ||
IsPlainRtp(protocol);
} else {
return IsPlainSctp(protocol) || IsPlainRtp(protocol);
}
}
// Allow for non-DTLS RTP protocol even when using DTLS because that's what
// JSEP specifies.
if (secure_transport) {
// Most likely scenarios first.
return IsDtlsRtp(protocol) || IsPlainRtp(protocol);
} else {
return IsPlainRtp(protocol);
}
}
static void SetMediaProtocol(bool secure_transport,
MediaContentDescription* desc) {
if (!desc->cryptos().empty())
desc->set_protocol(kMediaProtocolSavpf);
else if (secure_transport)
desc->set_protocol(kMediaProtocolDtlsSavpf);
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;
}
std::string MediaContentDirectionToString(MediaContentDirection direction) {
std::string dir_str;
switch (direction) {
case MD_INACTIVE:
dir_str = "inactive";
break;
case MD_SENDONLY:
dir_str = "sendonly";
break;
case MD_RECVONLY:
dir_str = "recvonly";
break;
case MD_SENDRECV:
dir_str = "sendrecv";
break;
default:
ASSERT(false);
break;
}
return dir_str;
}
void MediaSessionOptions::AddSendStream(MediaType type,
const std::string& id,
const std::string& sync_label) {
AddSendStreamInternal(type, id, sync_label, 1);
}
void MediaSessionOptions::AddSendVideoStream(
const std::string& id,
const std::string& sync_label,
int num_sim_layers) {
AddSendStreamInternal(MEDIA_TYPE_VIDEO, id, sync_label, num_sim_layers);
}
void MediaSessionOptions::AddSendStreamInternal(
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 we haven't already set the data_channel_type, and we add a
// stream, we assume it's an RTP data stream.
if (type == MEDIA_TYPE_DATA && data_channel_type == DCT_NONE)
data_channel_type = DCT_RTP;
}
void MediaSessionOptions::RemoveSendStream(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);
}
bool MediaSessionOptions::HasSendMediaStream(MediaType type) const {
Streams::const_iterator stream_it = streams.begin();
for (; stream_it != streams.end(); ++stream_it) {
if (stream_it->type == type) {
return true;
}
}
return 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->GetSupportedAudioSendCodecs(&audio_send_codecs_);
channel_manager->GetSupportedAudioReceiveCodecs(&audio_recv_codecs_);
channel_manager->GetSupportedAudioSendCodecs(&audio_send_codecs_);
channel_manager->GetSupportedAudioRtpHeaderExtensions(&audio_rtp_extensions_);
channel_manager->GetSupportedVideoCodecs(&video_codecs_);
channel_manager->GetSupportedVideoRtpHeaderExtensions(&video_rtp_extensions_);
channel_manager->GetSupportedDataCodecs(&data_codecs_);
NegotiateCodecs(audio_recv_codecs_, audio_send_codecs_,
&audio_sendrecv_codecs_);
}
const AudioCodecs& MediaSessionDescriptionFactory::audio_sendrecv_codecs()
const {
return audio_sendrecv_codecs_;
}
const AudioCodecs& MediaSessionDescriptionFactory::audio_send_codecs() const {
return audio_send_codecs_;
}
const AudioCodecs& MediaSessionDescriptionFactory::audio_recv_codecs() const {
return audio_recv_codecs_;
}
void MediaSessionDescriptionFactory::set_audio_codecs(
const AudioCodecs& send_codecs, const AudioCodecs& recv_codecs) {
audio_send_codecs_ = send_codecs;
audio_recv_codecs_ = recv_codecs;
audio_sendrecv_codecs_.clear();
// Use NegotiateCodecs to merge our codec lists, since the operation is
// essentially the same. Put send_codecs as the offered_codecs, which is the
// order we'd like to follow. The reasoning is that encoding is usually more
// expensive than decoding, and prioritizing a codec in the send list probably
// means it's a codec we can handle efficiently.
NegotiateCodecs(recv_codecs, send_codecs, &audio_sendrecv_codecs_);
}
SessionDescription* MediaSessionDescriptionFactory::CreateOffer(
const MediaSessionOptions& options,
const SessionDescription* current_description) const {
std::unique_ptr<SessionDescription> offer(new SessionDescription());
StreamParamsVec current_streams;
GetCurrentStreamParams(current_description, &current_streams);
const bool wants_send =
options.HasSendMediaStream(MEDIA_TYPE_AUDIO) || add_legacy_;
const AudioCodecs& supported_audio_codecs =
GetAudioCodecsForOffer({wants_send, options.recv_audio});
AudioCodecs audio_codecs;
VideoCodecs video_codecs;
DataCodecs data_codecs;
GetCodecsToOffer(current_description, supported_audio_codecs,
video_codecs_, data_codecs_,
&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)) {
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)) {
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)) {
MediaSessionOptions options_copy(options);
if (IsSctp(static_cast<const MediaContentDescription*>(
it->description))) {
options_copy.data_channel_type = DCT_SCTP;
}
if (!AddDataContentForOffer(options_copy, current_description,
&data_codecs, &current_streams,
offer.get())) {
return NULL;
}
data_added = true;
} else {
ASSERT(false);
}
}
}
// 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. As indicated by XEP-0167, we retain the same payload ids
// from the offer in the answer.
std::unique_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();
}
const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForOffer(
const RtpTransceiverDirection& direction) const {
// If stream is inactive - generate list as if sendrecv.
if (direction.send == direction.recv) {
return audio_sendrecv_codecs_;
} else if (direction.send) {
return audio_send_codecs_;
} else {
return audio_recv_codecs_;
}
}
const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForAnswer(
const RtpTransceiverDirection& offer,
const RtpTransceiverDirection& answer) const {
// For inactive and sendrecv answers, generate lists as if we were to accept
// the offer's direction. See RFC 3264 Section 6.1.
if (answer.send == answer.recv) {
if (offer.send == offer.recv) {
return audio_sendrecv_codecs_;
} else if (offer.send) {
return audio_recv_codecs_;
} else {
return audio_send_codecs_;
}
} else if (answer.send) {
return audio_send_codecs_;
} else {
return audio_recv_codecs_;
}
}
void MediaSessionDescriptionFactory::GetCodecsToOffer(
const SessionDescription* current_description,
const AudioCodecs& supported_audio_codecs,
const VideoCodecs& supported_video_codecs,
const DataCodecs& supported_data_codecs,
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>(supported_audio_codecs, audio_codecs,
&used_pltypes);
FindCodecsToOffer<VideoCodec>(supported_video_codecs, video_codecs,
&used_pltypes);
FindCodecsToOffer<DataCodec>(supported_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;
RtpHeaderExtensions all_extensions;
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();
FindAndSetRtpHdrExtUsed(audio_extensions, &all_extensions, &used_ids);
}
const VideoContentDescription* video =
GetFirstVideoContentDescription(current_description);
if (video) {
*video_extensions = video->rtp_header_extensions();
FindAndSetRtpHdrExtUsed(video_extensions, &all_extensions, &used_ids);
}
}
// Add our default RTP header extensions that are not in
// |current_description|.
FindRtpHdrExtsToOffer(audio_rtp_header_extensions(), audio_extensions,
&all_extensions, &used_ids);
FindRtpHdrExtsToOffer(video_rtp_header_extensions(), video_extensions,
&all_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);
std::unique_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 {
const ContentInfo* current_audio_content =
GetFirstAudioContent(current_description);
std::string content_name =
current_audio_content ? current_audio_content->name : CN_AUDIO;
cricket::SecurePolicy sdes_policy =
IsDtlsActive(content_name, current_description) ? cricket::SEC_DISABLED
: secure();
std::unique_ptr<AudioContentDescription> audio(new AudioContentDescription());
std::vector<std::string> crypto_suites;
GetSupportedAudioCryptoSuiteNames(options.crypto_options, &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());
auto offer_rtd =
RtpTransceiverDirection(!audio->streams().empty(), options.recv_audio);
audio->set_direction(offer_rtd.ToMediaContentDirection());
desc->AddContent(content_name, NS_JINGLE_RTP, audio.release());
if (!AddTransportOffer(content_name,
GetTransportOptions(options, content_name),
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 {
const ContentInfo* current_video_content =
GetFirstVideoContent(current_description);
std::string content_name =
current_video_content ? current_video_content->name : CN_VIDEO;
cricket::SecurePolicy sdes_policy =
IsDtlsActive(content_name, current_description) ? cricket::SEC_DISABLED
: secure();
std::unique_ptr<VideoContentDescription> video(new VideoContentDescription());
std::vector<std::string> crypto_suites;
GetSupportedVideoCryptoSuiteNames(options.crypto_options, &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());
if (!video->streams().empty()) {
if (options.recv_video) {
video->set_direction(MD_SENDRECV);
} else {
video->set_direction(MD_SENDONLY);
}
} else {
if (options.recv_video) {
video->set_direction(MD_RECVONLY);
} else {
video->set_direction(MD_INACTIVE);
}
}
desc->AddContent(content_name, NS_JINGLE_RTP, video.release());
if (!AddTransportOffer(content_name,
GetTransportOptions(options, content_name),
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);
std::unique_ptr<DataContentDescription> data(new DataContentDescription());
bool is_sctp = (options.data_channel_type == DCT_SCTP);
FilterDataCodecs(data_codecs, is_sctp);
const ContentInfo* current_data_content =
GetFirstDataContent(current_description);
std::string content_name =
current_data_content ? current_data_content->name : CN_DATA;
cricket::SecurePolicy sdes_policy =
IsDtlsActive(content_name, 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 {
GetSupportedDataCryptoSuiteNames(options.crypto_options, &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(content_name, NS_JINGLE_DRAFT_SCTP, data.release());
} else {
data->set_bandwidth(options.data_bandwidth);
SetMediaProtocol(secure_transport, data.get());
desc->AddContent(content_name, NS_JINGLE_RTP, data.release());
}
if (!AddTransportOffer(content_name,
GetTransportOptions(options, content_name),
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);
const AudioContentDescription* offer_audio =
static_cast<const AudioContentDescription*>(audio_content->description);
std::unique_ptr<TransportDescription> audio_transport(CreateTransportAnswer(
audio_content->name, offer,
GetTransportOptions(options, audio_content->name), current_description));
if (!audio_transport) {
return false;
}
// Pick codecs based on the requested communications direction in the offer.
const bool wants_send =
options.HasSendMediaStream(MEDIA_TYPE_AUDIO) || add_legacy_;
auto wants_rtd = RtpTransceiverDirection(wants_send, options.recv_audio);
auto offer_rtd =
RtpTransceiverDirection::FromMediaContentDirection(
offer_audio->direction());
auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
AudioCodecs audio_codecs = GetAudioCodecsForAnswer(offer_rtd, answer_rtd);
if (!options.vad_enabled) {
StripCNCodecs(&audio_codecs);
}
bool bundle_enabled =
offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled;
std::unique_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(
offer_audio,
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);
std::unique_ptr<TransportDescription> video_transport(CreateTransportAnswer(
video_content->name, offer,
GetTransportOptions(options, video_content->name), current_description));
if (!video_transport) {
return false;
}
std::unique_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);
std::unique_ptr<TransportDescription> data_transport(CreateTransportAnswer(
data_content->name, offer,
GetTransportOptions(options, data_content->name), 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);
std::unique_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);
}
const ContentInfo* GetFirstMediaContent(const ContentInfos& contents,
MediaType media_type) {
for (const ContentInfo& content : contents) {
if (IsMediaContentOfType(&content, media_type)) {
return &content;
}
}
return nullptr;
}
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 == nullptr) {
return nullptr;
}
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));
}
//
// Non-const versions of the above functions.
//
ContentInfo* GetFirstMediaContent(ContentInfos& contents,
MediaType media_type) {
for (ContentInfo& content : contents) {
if (IsMediaContentOfType(&content, media_type)) {
return &content;
}
}
return nullptr;
}
ContentInfo* GetFirstAudioContent(ContentInfos& contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_AUDIO);
}
ContentInfo* GetFirstVideoContent(ContentInfos& contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_VIDEO);
}
ContentInfo* GetFirstDataContent(ContentInfos& contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_DATA);
}
static ContentInfo* GetFirstMediaContent(SessionDescription* sdesc,
MediaType media_type) {
if (sdesc == nullptr) {
return nullptr;
}
return GetFirstMediaContent(sdesc->contents(), media_type);
}
ContentInfo* GetFirstAudioContent(SessionDescription* sdesc) {
return GetFirstMediaContent(sdesc, MEDIA_TYPE_AUDIO);
}
ContentInfo* GetFirstVideoContent(SessionDescription* sdesc) {
return GetFirstMediaContent(sdesc, MEDIA_TYPE_VIDEO);
}
ContentInfo* GetFirstDataContent(SessionDescription* sdesc) {
return GetFirstMediaContent(sdesc, MEDIA_TYPE_DATA);
}
MediaContentDescription* GetFirstMediaContentDescription(
SessionDescription* sdesc,
MediaType media_type) {
ContentInfo* content = GetFirstMediaContent(sdesc, media_type);
ContentDescription* description = content ? content->description : NULL;
return static_cast<MediaContentDescription*>(description);
}
AudioContentDescription* GetFirstAudioContentDescription(
SessionDescription* sdesc) {
return static_cast<AudioContentDescription*>(
GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_AUDIO));
}
VideoContentDescription* GetFirstVideoContentDescription(
SessionDescription* sdesc) {
return static_cast<VideoContentDescription*>(
GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_VIDEO));
}
DataContentDescription* GetFirstDataContentDescription(
SessionDescription* sdesc) {
return static_cast<DataContentDescription*>(
GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA));
}
} // namespace cricket
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