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patch from issue 25469004

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7517 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pthatcher@webrtc.org 2014-10-23 23:37:22 +00:00
parent 8fe75ee234
commit c9d6d14020
2 changed files with 88 additions and 79 deletions

111
talk/app/webrtc/webrtcsdp.cc

@ -197,13 +197,17 @@ static const char kSessionName[] = "s=-";
static const char kTimeDescription[] = "t=0 0";
static const char kAttrGroup[] = "a=group:BUNDLE";
static const char kConnectionNettype[] = "IN";
static const char kConnectionAddrtype[] = "IP4";
static const char kConnectionIpv4Addrtype[] = "IP4";
static const char kConnectionIpv6Addrtype[] = "IP6";
static const char kMediaTypeVideo[] = "video";
static const char kMediaTypeAudio[] = "audio";
static const char kMediaTypeData[] = "application";
static const char kMediaPortRejected[] = "0";
static const char kDefaultAddress[] = "0.0.0.0";
static const char kDefaultPort[] = "1";
// draft-ietf-mmusic-trickle-ice-01
// When no candidates have been gathered, set the connection
// address to IP6 ::.
static const char kDummyAddress[] = "::";
static const char kDummyPort[] = "9";
// RFC 3556
static const char kApplicationSpecificMaximum[] = "AS";
@ -667,10 +671,13 @@ static int GetCandidatePreferenceFromType(const std::string& type) {
// The value of |component_id| currently supported are 1 (RTP) and 2 (RTCP).
// TODO: Decide the default destination in webrtcsession and
// pass it down via SessionDescription.
static bool GetDefaultDestination(const std::vector<Candidate>& candidates,
int component_id, std::string* port, std::string* ip) {
*port = kDefaultPort;
*ip = kDefaultAddress;
static void GetDefaultDestination(
const std::vector<Candidate>& candidates,
int component_id, std::string* port,
std::string* ip, std::string* addr_type) {
*addr_type = kConnectionIpv6Addrtype;
*port = kDummyPort;
*ip = kDummyAddress;
int current_preference = kPreferenceUnknown;
for (std::vector<Candidate>::const_iterator it = candidates.begin();
it != candidates.end(); ++it) {
@ -685,8 +692,13 @@ static bool GetDefaultDestination(const std::vector<Candidate>& candidates,
current_preference = preference;
*port = it->address().PortAsString();
*ip = it->address().ipaddr().ToString();
int family = it->address().ipaddr().family();
if (family == AF_INET) {
addr_type->assign(kConnectionIpv4Addrtype);
} else if (family == AF_INET6) {
addr_type->assign(kConnectionIpv6Addrtype);
}
}
return true;
}
// Update |mline|'s default destination and append a c line after it.
@ -705,55 +717,52 @@ static void UpdateMediaDefaultDestination(
}
std::ostringstream os;
std::string rtp_port, rtp_ip;
if (GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTP,
&rtp_port, &rtp_ip)) {
// Found default RTP candidate.
// RFC 5245
// The default candidates are added to the SDP as the default
// destination for media. For streams based on RTP, this is done by
// placing the IP address and port of the RTP candidate into the c and m
// lines, respectively.
// 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) {
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(), &new_lines);
std::string rtp_port, rtp_ip, addr_type;
GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTP,
&rtp_port, &rtp_ip, &addr_type);
// Found default RTP candidate.
// RFC 5245
// The default candidates are added to the SDP as the default
// destination for media. For streams based on RTP, this is done by
// placing the IP address and port of the RTP candidate into the c and m
// lines, respectively.
// 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) {
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 << " " << addr_type << " " << rtp_ip;
AddLine(os.str(), &new_lines);
message->append(new_lines);
}
// 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.
std::string rtcp_line, rtcp_port, rtcp_ip, addr_type;
GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTCP,
&rtcp_port, &rtcp_ip, &addr_type);
// 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
std::ostringstream os;
InitAttrLine(kAttributeRtcp, &os);
os << kSdpDelimiterColon
<< rtcp_port << " "
<< kConnectionNettype << " "
<< kConnectionAddrtype << " "
<< rtcp_ip;
rtcp_line = os.str();
}
// 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 << " "
<< addr_type << " "
<< rtcp_ip;
rtcp_line = os.str();
return rtcp_line;
}
@ -1243,7 +1252,7 @@ void BuildMediaDescription(const ContentInfo* content_info,
// To reject an offered stream, the port number in the corresponding stream in
// the answer MUST be set to zero.
const std::string port = content_info->rejected ?
kMediaPortRejected : kDefaultPort;
kMediaPortRejected : kDummyPort;
rtc::SSLFingerprint* fp = (transport_info) ?
transport_info->description.identity_fingerprint.get() : NULL;

56
talk/app/webrtc/webrtcsdp_unittest.cc

@ -212,9 +212,9 @@ static const char kSdpString[] =
"s=-\r\n"
"t=0 0\r\n"
"a=msid-semantic: WMS local_stream_1 local_stream_2\r\n"
"m=audio 1 RTP/SAVPF 111 103 104\r\n"
"c=IN IP4 0.0.0.0\r\n"
"a=rtcp:1 IN IP4 0.0.0.0\r\n"
"m=audio 9 RTP/SAVPF 111 103 104\r\n"
"c=IN IP6 ::\r\n"
"a=rtcp:9 IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_voice\r\na=ice-pwd:pwd_voice\r\n"
"a=mid:audio_content_name\r\n"
"a=sendrecv\r\n"
@ -233,9 +233,9 @@ static const char kSdpString[] =
"a=ssrc:4 msid:local_stream_2 audio_track_id_2\r\n"
"a=ssrc:4 mslabel:local_stream_2\r\n"
"a=ssrc:4 label:audio_track_id_2\r\n"
"m=video 1 RTP/SAVPF 120\r\n"
"c=IN IP4 0.0.0.0\r\n"
"a=rtcp:1 IN IP4 0.0.0.0\r\n"
"m=video 9 RTP/SAVPF 120\r\n"
"c=IN IP6 ::\r\n"
"a=rtcp:9 IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_video\r\na=ice-pwd:pwd_video\r\n"
"a=mid:video_content_name\r\n"
"a=sendrecv\r\n"
@ -261,9 +261,9 @@ static const char kSdpString[] =
"a=ssrc:6 label:video_track_id_3\r\n";
static const char kSdpRtpDataChannelString[] =
"m=application 1 RTP/SAVPF 101\r\n"
"c=IN IP4 0.0.0.0\r\n"
"a=rtcp:1 IN IP4 0.0.0.0\r\n"
"m=application 9 RTP/SAVPF 101\r\n"
"c=IN IP6 ::\r\n"
"a=rtcp:9 IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_data\r\n"
"a=ice-pwd:pwd_data\r\n"
"a=mid:data_content_name\r\n"
@ -277,8 +277,8 @@ static const char kSdpRtpDataChannelString[] =
"a=ssrc:10 label:data_channeld0\r\n";
static const char kSdpSctpDataChannelString[] =
"m=application 1 DTLS/SCTP 5000\r\n"
"c=IN IP4 0.0.0.0\r\n"
"m=application 9 DTLS/SCTP 5000\r\n"
"c=IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_data\r\n"
"a=ice-pwd:pwd_data\r\n"
"a=mid:data_content_name\r\n"
@ -286,10 +286,10 @@ static const char kSdpSctpDataChannelString[] =
// draft-ietf-mmusic-sctp-sdp-07
static const char kSdpSctpDataChannelStringWithSctpPort[] =
"m=application 1 DTLS/SCTP webrtc-datachannel\r\n"
"m=application 9 DTLS/SCTP webrtc-datachannel\r\n"
"a=fmtp:webrtc-datachannel max-message-size=100000\r\n"
"a=sctp-port 5000\r\n"
"c=IN IP4 0.0.0.0\r\n"
"c=IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_data\r\n"
"a=ice-pwd:pwd_data\r\n"
"a=mid:data_content_name\r\n";
@ -315,11 +315,11 @@ static const char kSdpConferenceString[] =
"s=-\r\n"
"t=0 0\r\n"
"a=msid-semantic: WMS\r\n"
"m=audio 1 RTP/SAVPF 111 103 104\r\n"
"c=IN IP4 0.0.0.0\r\n"
"m=audio 9 RTP/SAVPF 111 103 104\r\n"
"c=IN IP6 ::\r\n"
"a=x-google-flag:conference\r\n"
"m=video 1 RTP/SAVPF 120\r\n"
"c=IN IP4 0.0.0.0\r\n"
"m=video 9 RTP/SAVPF 120\r\n"
"c=IN IP6 ::\r\n"
"a=x-google-flag:conference\r\n";
static const char kSdpSessionString[] =
@ -330,9 +330,9 @@ static const char kSdpSessionString[] =
"a=msid-semantic: WMS local_stream\r\n";
static const char kSdpAudioString[] =
"m=audio 1 RTP/SAVPF 111\r\n"
"c=IN IP4 0.0.0.0\r\n"
"a=rtcp:1 IN IP4 0.0.0.0\r\n"
"m=audio 9 RTP/SAVPF 111\r\n"
"c=IN IP6 ::\r\n"
"a=rtcp:9 IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_voice\r\na=ice-pwd:pwd_voice\r\n"
"a=mid:audio_content_name\r\n"
"a=sendrecv\r\n"
@ -343,9 +343,9 @@ static const char kSdpAudioString[] =
"a=ssrc:1 label:audio_track_id_1\r\n";
static const char kSdpVideoString[] =
"m=video 1 RTP/SAVPF 120\r\n"
"c=IN IP4 0.0.0.0\r\n"
"a=rtcp:1 IN IP4 0.0.0.0\r\n"
"m=video 9 RTP/SAVPF 120\r\n"
"c=IN IP6 ::\r\n"
"a=rtcp:9 IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_video\r\na=ice-pwd:pwd_video\r\n"
"a=mid:video_content_name\r\n"
"a=sendrecv\r\n"
@ -1195,7 +1195,7 @@ class WebRtcSdpTest : public testing::Test {
// description.
"a=msid-semantic: WMS\r\n"
// Pl type 111 preferred.
"m=audio 1 RTP/SAVPF 111 104 103 102\r\n"
"m=audio 9 RTP/SAVPF 111 104 103 102\r\n"
// Pltype 111 listed before 103 and 104 in the map.
"a=rtpmap:111 opus/48000/2\r\n"
// Pltype 103 listed before 104.
@ -1217,7 +1217,7 @@ class WebRtcSdpTest : public testing::Test {
os.clear();
os.str("");
// Pl type 100 preferred.
os << "m=video 1 RTP/SAVPF 99 95\r\n"
os << "m=video 9 RTP/SAVPF 99 95\r\n"
<< "a=rtpmap:99 VP8/90000\r\n"
<< "a=rtpmap:95 RTX/90000\r\n"
<< "a=fmtp:95 apt=99;rtx-time=1000\r\n";
@ -1279,7 +1279,7 @@ class WebRtcSdpTest : public testing::Test {
// this parser, and will be added to the SDP when serializing a session
// description.
"a=msid-semantic: WMS\r\n"
"m=audio 1 RTP/SAVPF 111\r\n"
"m=audio 9 RTP/SAVPF 111\r\n"
"a=rtpmap:111 opus/48000/2\r\n"
"a=rtcp-fb:111 nack\r\n"
"m=video 3457 RTP/SAVPF 101\r\n"
@ -1596,7 +1596,7 @@ 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("m=application 1 RTP/SAVPF 101\r\nc=IN IP4 0.0.0.0\r\n",
InjectAfter("m=application 9 RTP/SAVPF 101\r\nc=IN IP6 ::\r\n",
"b=AS:100\r\n",
&expected_sdp);
EXPECT_EQ(expected_sdp, message);
@ -2308,7 +2308,7 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithReorderedPltypes) {
"o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"m=audio 1 RTP/SAVPF 104 103\r\n" // Pl type 104 preferred.
"m=audio 9 RTP/SAVPF 104 103\r\n" // Pl type 104 preferred.
"a=rtpmap:111 opus/48000/2\r\n" // Pltype 111 listed before 103 and 104
// in the map.
"a=rtpmap:103 ISAC/16000\r\n" // Pltype 103 listed before 104 in the map.