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Make sure b lines appear before all the a lines. Per RFC 4566, the order of media description should be:

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m=  (media name and transport address)
  i=* (media title)
  c=* (connection information -- optional if included at
       session level)
  b=* (zero or more bandwidth information lines)
  k=* (encryption key)
  a=* (zero or more media attribute lines)

BUG=2260
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6708 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
wu@webrtc.org
2014-07-16 21:03:13 +00:00
parent 46fb331bc5
commit 4c3e9917e7
2 changed files with 90 additions and 106 deletions
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190
talk/app/webrtc/webrtcsdp.cc
View File

@@ -231,15 +231,12 @@ struct SsrcInfo {
typedef std::vector<SsrcInfo> SsrcInfoVec;
typedef std::vector<SsrcGroup> SsrcGroupVec;
// Serializes the passed in SessionDescription to a SDP string.
// desc - The SessionDescription object to be serialized.
static std::string SdpSerializeSessionDescription(
const JsepSessionDescription& jdesc);
template <class T>
static void AddFmtpLine(const T& codec, std::string* message);
static void BuildMediaDescription(const ContentInfo* content_info,
const TransportInfo* transport_info,
const MediaType media_type,
const std::vector<Candidate>& candidates,
std::string* message);
static void BuildSctpContentAttributes(std::string* message, int sctp_port);
static void BuildRtpContentAttributes(
@@ -712,22 +709,21 @@ static bool GetDefaultDestination(const std::vector<Candidate>& candidates,
return true;
}
// Update the media default destination.
// Update |mline|'s default destination and append a c line after it.
static void UpdateMediaDefaultDestination(
const std::vector<Candidate>& candidates, std::string* mline) {
const std::vector<Candidate>& candidates,
const std::string mline,
std::string* message) {
std::string new_lines;
AddLine(mline, &new_lines);
// RFC 4566
// m=<media> <port> <proto> <fmt> ...
std::vector<std::string> fields;
talk_base::split(*mline, kSdpDelimiterSpace, &fields);
talk_base::split(mline, kSdpDelimiterSpace, &fields);
if (fields.size() < 3) {
return;
}
bool is_rtp =
fields[2].empty() ||
talk_base::starts_with(fields[2].data(),
cricket::kMediaProtocolRtpPrefix);
std::ostringstream os;
std::string rtp_port, rtp_ip;
if (GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTP,
@@ -742,39 +738,43 @@ static void UpdateMediaDefaultDestination(
// Update the port in the m line.
// If this is a m-line with port equal to 0, we don't change it.
if (fields[1] != kMediaPortRejected) {
mline->replace(fields[0].size() + 1,
fields[1].size(),
rtp_port);
new_lines.replace(fields[0].size() + 1,
fields[1].size(),
rtp_port);
}
// Add the c line.
// RFC 4566
// c=<nettype> <addrtype> <connection-address>
InitLine(kLineTypeConnection, kConnectionNettype, &os);
os << " " << kConnectionAddrtype << " " << rtp_ip;
AddLine(os.str(), mline);
AddLine(os.str(), &new_lines);
}
message->append(new_lines);
}
if (is_rtp) {
std::string rtcp_port, rtcp_ip;
if (GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTCP,
&rtcp_port, &rtcp_ip)) {
// Found default RTCP candidate.
// RFC 5245
// If the agent is utilizing RTCP, it MUST encode the RTCP candidate
// using the a=rtcp attribute as defined in RFC 3605.
// Gets "a=rtcp" line if found default RTCP candidate from |candidates|.
static std::string GetRtcpLine(const std::vector<Candidate>& candidates) {
std::string rtcp_line, rtcp_port, rtcp_ip;
if (GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTCP,
&rtcp_port, &rtcp_ip)) {
// Found default RTCP candidate.
// RFC 5245
// If the agent is utilizing RTCP, it MUST encode the RTCP candidate
// using the a=rtcp attribute as defined in RFC 3605.
// RFC 3605
// rtcp-attribute = "a=rtcp:" port [nettype space addrtype space
// connection-address] CRLF
InitAttrLine(kAttributeRtcp, &os);
os << kSdpDelimiterColon
<< rtcp_port << " "
<< kConnectionNettype << " "
<< kConnectionAddrtype << " "
<< rtcp_ip;
AddLine(os.str(), mline);
}
// RFC 3605
// rtcp-attribute = "a=rtcp:" port [nettype space addrtype space
// connection-address] CRLF
std::ostringstream os;
InitAttrLine(kAttributeRtcp, &os);
os << kSdpDelimiterColon
<< rtcp_port << " "
<< kConnectionNettype << " "
<< kConnectionAddrtype << " "
<< rtcp_ip;
rtcp_line = os.str();
}
return rtcp_line;
}
// Get candidates according to the mline index from SessionDescriptionInterface.
@@ -792,36 +792,6 @@ static void GetCandidatesByMindex(const SessionDescriptionInterface& desci,
}
std::string SdpSerialize(const JsepSessionDescription& jdesc) {
std::string sdp = SdpSerializeSessionDescription(jdesc);
std::string sdp_with_candidates;
size_t pos = 0;
std::string line;
int mline_index = -1;
while (GetLine(sdp, &pos, &line)) {
if (IsLineType(line, kLineTypeMedia)) {
++mline_index;
std::vector<Candidate> candidates;
GetCandidatesByMindex(jdesc, mline_index, &candidates);
// Media line may append other lines inside the
// UpdateMediaDefaultDestination call, so add the kLineBreak here first.
line.append(kLineBreak);
UpdateMediaDefaultDestination(candidates, &line);
sdp_with_candidates.append(line);
// Build the a=candidate lines.
BuildCandidate(candidates, &sdp_with_candidates);
} else {
// Copy old line to new sdp without change.
AddLine(line, &sdp_with_candidates);
}
}
sdp = sdp_with_candidates;
return sdp;
}
std::string SdpSerializeSessionDescription(
const JsepSessionDescription& jdesc) {
const cricket::SessionDescription* desc = jdesc.description();
if (!desc) {
return "";
@@ -868,40 +838,36 @@ std::string SdpSerializeSessionDescription(
// MediaStream semantics
InitAttrLine(kAttributeMsidSemantics, &os);
os << kSdpDelimiterColon << " " << kMediaStreamSemantic;
std::set<std::string> media_stream_labels;
const ContentInfo* audio_content = GetFirstAudioContent(desc);
if (audio_content)
GetMediaStreamLabels(audio_content, &media_stream_labels);
const ContentInfo* video_content = GetFirstVideoContent(desc);
if (video_content)
GetMediaStreamLabels(video_content, &media_stream_labels);
for (std::set<std::string>::const_iterator it =
media_stream_labels.begin(); it != media_stream_labels.end(); ++it) {
os << " " << *it;
}
AddLine(os.str(), &message);
if (audio_content) {
BuildMediaDescription(audio_content,
desc->GetTransportInfoByName(audio_content->name),
cricket::MEDIA_TYPE_AUDIO, &message);
// Preserve the order of the media contents.
int mline_index = -1;
for (cricket::ContentInfos::const_iterator it = desc->contents().begin();
it != desc->contents().end(); ++it) {
const MediaContentDescription* mdesc =
static_cast<const MediaContentDescription*>(it->description);
std::vector<Candidate> candidates;
GetCandidatesByMindex(jdesc, ++mline_index, &candidates);
BuildMediaDescription(&*it,
desc->GetTransportInfoByName(it->name),
mdesc->type(),
candidates,
&message);
}
if (video_content) {
BuildMediaDescription(video_content,
desc->GetTransportInfoByName(video_content->name),
cricket::MEDIA_TYPE_VIDEO, &message);
}
const ContentInfo* data_content = GetFirstDataContent(desc);
if (data_content) {
BuildMediaDescription(data_content,
desc->GetTransportInfoByName(data_content->name),
cricket::MEDIA_TYPE_DATA, &message);
}
return message;
}
@@ -1157,6 +1123,7 @@ bool ParseExtmap(const std::string& line, RtpHeaderExtension* extmap,
void BuildMediaDescription(const ContentInfo* content_info,
const TransportInfo* transport_info,
const MediaType media_type,
const std::vector<Candidate>& candidates,
std::string* message) {
ASSERT(message != NULL);
if (content_info == NULL || message == NULL) {
@@ -1249,9 +1216,43 @@ void BuildMediaDescription(const ContentInfo* content_info,
talk_base::SSLFingerprint* fp = (transport_info) ?
transport_info->description.identity_fingerprint.get() : NULL;
// Add the m and c lines.
InitLine(kLineTypeMedia, type, &os);
os << " " << port << " " << media_desc->protocol() << fmt;
AddLine(os.str(), message);
std::string mline = os.str();
UpdateMediaDefaultDestination(candidates, mline, message);
// RFC 4566
// b=AS:<bandwidth>
// We should always use the default bandwidth for RTP-based data
// channels. Don't allow SDP to set the bandwidth, because that
// would give JS the opportunity to "break the Internet".
// TODO(pthatcher): But we need to temporarily allow the SDP to control
// this for backwards-compatibility. Once we don't need that any
// more, remove this.
bool support_dc_sdp_bandwidth_temporarily = true;
if (media_desc->bandwidth() >= 1000 &&
(media_type != cricket::MEDIA_TYPE_DATA ||
support_dc_sdp_bandwidth_temporarily)) {
InitLine(kLineTypeSessionBandwidth, kApplicationSpecificMaximum, &os);
os << kSdpDelimiterColon << (media_desc->bandwidth() / 1000);
AddLine(os.str(), message);
}
// Add the a=rtcp line.
bool is_rtp =
media_desc->protocol().empty() ||
talk_base::starts_with(media_desc->protocol().data(),
cricket::kMediaProtocolRtpPrefix);
if (is_rtp) {
std::string rtcp_line = GetRtcpLine(candidates);
if (!rtcp_line.empty()) {
AddLine(rtcp_line, message);
}
}
// Build the a=candidate lines.
BuildCandidate(candidates, message);
// Use the transport_info to build the media level ice-ufrag and ice-pwd.
if (transport_info) {
@@ -1363,23 +1364,6 @@ void BuildRtpContentAttributes(
}
AddLine(os.str(), message);
// RFC 4566
// b=AS:<bandwidth>
// We should always use the default bandwidth for RTP-based data
// channels. Don't allow SDP to set the bandwidth, because that
// would give JS the opportunity to "break the Internet".
// TODO(pthatcher): But we need to temporarily allow the SDP to control
// this for backwards-compatibility. Once we don't need that any
// more, remove this.
bool support_dc_sdp_bandwidth_temporarily = true;
if (media_desc->bandwidth() >= 1000 &&
(media_type != cricket::MEDIA_TYPE_DATA ||
support_dc_sdp_bandwidth_temporarily)) {
InitLine(kLineTypeSessionBandwidth, kApplicationSpecificMaximum, &os);
os << kSdpDelimiterColon << (media_desc->bandwidth() / 1000);
AddLine(os.str(), message);
}
// RFC 5761
// a=rtcp-mux
if (media_desc->rtcp_mux()) {

6
talk/app/webrtc/webrtcsdp_unittest.cc
View File

@@ -1408,10 +1408,10 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBandwidth) {
jdesc_.session_version()));
std::string message = webrtc::SdpSerialize(jdesc_);
std::string sdp_with_bandwidth = kSdpFullString;
InjectAfter("a=mid:video_content_name\r\na=sendrecv\r\n",
InjectAfter("c=IN IP4 74.125.224.39\r\n",
"b=AS:100\r\n",
&sdp_with_bandwidth);
InjectAfter("a=mid:audio_content_name\r\na=sendrecv\r\n",
InjectAfter("c=IN IP4 74.125.127.126\r\n",
"b=AS:50\r\n",
&sdp_with_bandwidth);
EXPECT_EQ(sdp_with_bandwidth, message);
@@ -1535,7 +1535,7 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithDataChannelAndBandwidth) {
// TODO(pthatcher): We need to temporarily allow the SDP to control
// this for backwards-compatibility. Once we don't need that any
// more, remove this.
InjectAfter("a=mid:data_content_name\r\na=sendrecv\r\n",
InjectAfter("m=application 1 RTP/SAVPF 101\r\nc=IN IP4 0.0.0.0\r\n",
"b=AS:100\r\n",
&expected_sdp);
EXPECT_EQ(expected_sdp, message);