Index: content/browser/renderer_host/p2p/socket_host.cc |
diff --git a/content/browser/renderer_host/p2p/socket_host.cc b/content/browser/renderer_host/p2p/socket_host.cc |
index 4b45e99adad087881aa5f9bff69a1004e727a39f..e927f2d42e0d8a84855e834c525af6e368a8a064 100644 |
--- a/content/browser/renderer_host/p2p/socket_host.cc |
+++ b/content/browser/renderer_host/p2p/socket_host.cc |
@@ -8,9 +8,56 @@ |
#include "content/browser/renderer_host/p2p/socket_host_tcp.h" |
#include "content/browser/renderer_host/p2p/socket_host_tcp_server.h" |
#include "content/browser/renderer_host/p2p/socket_host_udp.h" |
+#include "crypto/hmac.h" |
+#include "third_party/libjingle/source/talk/base/asyncpacketsocket.h" |
+#include "third_party/libjingle/source/talk/base/byteorder.h" |
+#include "third_party/libjingle/source/talk/base/messagedigest.h" |
+#include "third_party/libjingle/source/talk/p2p/base/stun.h" |
namespace { |
+ |
const uint32 kStunMagicCookie = 0x2112A442; |
+const int kMinRtpHdrLen = 12; |
+const int kRtpExtnHdrLen = 4; |
+const int kDtlsRecordHeaderLen = 13; |
+const int kTurnChannelHdrLen = 4; |
+const int kAbsSendTimeExtnLen = 3; |
+const int kOneByteHdrLen = 1; |
+ |
+// Fake auth tag written by the render process if external authentication is |
+// enabled. HMAC in packet will be compared against this value before updating |
+// packet with actual HMAC value. |
+static const unsigned char kFakeAuthTag[10] = { |
+ 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd |
+}; |
+ |
+bool IsTurnChannelData(const char* data) { |
+ return ((*data & 0xC0) == 0x40); |
+} |
+ |
+bool IsDtlsPacket(const char* data, int len) { |
+ const uint8* u = reinterpret_cast<const uint8*>(data); |
+ return (len >= kDtlsRecordHeaderLen && (u[0] > 19 && u[0] < 64)); |
+} |
+ |
+bool IsRtcpPacket(const char* data) { |
+ int type = (static_cast<uint8>(data[1]) & 0x7F); |
+ return (type >= 64 && type < 96); |
+} |
+ |
+bool IsStunPacket(const char* data) { |
+ return ((*data & 0xC0) == 0); |
+} |
+ |
+bool IsTurnSendIndicationPacket(const char* data) { |
+ uint16 type = talk_base::GetBE16(data); |
+ return (type == cricket::TURN_SEND_INDICATION); |
+} |
+ |
+bool IsRtpPacket(const char* data, int len) { |
+ return ((*data & 0xC0) == 0x80); |
+} |
+ |
} // namespace |
namespace content { |
@@ -102,4 +149,252 @@ P2PSocketHost* P2PSocketHost::Create( |
return NULL; |
} |
+void P2PSocketHost::MaybeUpdatePacketSendTimeExtn( |
juberti2
2014/02/25 00:48:24
Check for zero length packets (will cause IsDtlsPa
Mallinath (Gone from Chromium)
2014/02/25 22:56:44
Done.
|
+ char* data, int length, const talk_base::PacketOptions& options) { |
+ // if there is no valid |rtp_sendtime_extension_id| and |srtp_auth_key| in |
+ // PacketOptions, nothing to be updated in this packet. |
+ if (options.packet_time_params.rtp_sendtime_extension_id == -1 && |
+ options.packet_time_params.srtp_auth_key.empty()) { |
+ return; |
+ } |
+ |
+ DCHECK(!IsDtlsPacket(data, length)); |
+ DCHECK(!IsRtcpPacket(data)); |
+ |
+ // If there is a srtp auth key present then packet must be a RTP packet. |
+ // RTP packet may have been wrapped in a TURN Channel Data or |
+ // TURN send indication. |
+ int rtp_start_pos; |
+ int rtp_length; |
+ if (!GetRtpPacketStartPositionAndLength( |
+ data, length, &rtp_start_pos, &rtp_length)) { |
+ // This method should never return false. |
+ NOTREACHED(); |
+ return; |
+ } |
+ |
+ // Skip to rtp packet. |
+ char* start = data + rtp_start_pos; |
+ int rtp_packet_length = length - rtp_start_pos; |
+ // If there a non negetive sendtime extension id present in packet options, |
+ // then we should parse the rtp packet to update the timestamp. Otherwise |
+ // just calculate HMAC and update packet with it. |
+ if (options.packet_time_params.rtp_sendtime_extension_id != -1) { |
+ if (!MaybeUpdateRtpSendTimeExtn( |
+ start, rtp_packet_length, |
+ options.packet_time_params.rtp_sendtime_extension_id)) { |
+ // We should find an extension id in the rtp packet. |
+ NOTREACHED(); |
+ return; |
+ } |
+ } |
+ |
+ MaybeUpdateRtpAuthTag(start, rtp_packet_length, options); |
+} |
+ |
+bool P2PSocketHost::GetRtpPacketStartPositionAndLength( |
+ char* packet, int length, int* rtp_start_pos, int* rtp_packet_length) { |
+ if (IsTurnChannelData(packet)) { |
+ // Turn Channel Message header format. |
+ // 0 1 2 3 |
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // | Channel Number | Length | |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // | | |
+ // / Application Data / |
+ // / / |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ *rtp_start_pos = kTurnChannelHdrLen; |
+ *rtp_packet_length = talk_base::GetBE16(&packet[2]); |
juberti2
2014/02/25 00:48:24
need to check len before reading this field, and a
Mallinath (Gone from Chromium)
2014/02/25 22:56:44
Done.
|
+ } else if (IsTurnSendIndicationPacket(packet)) { |
+ // First skip mandatory stun header which is of 20 bytes. |
+ *rtp_start_pos = kStunHeaderSize; |
+ // Loop through STUN attributes until we find STUN DATA attribute. |
+ char* start = packet + *rtp_start_pos; |
+ while ((packet + *rtp_start_pos) - start < length) { |
+ // Keep reading STUN attributes until we hit DATA attribute. |
+ // Attribute will be a TLV structure. |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // | Type | Length | |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // | Value (variable) .... |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // The value in the length field MUST contain the length of the Value |
+ // part of the attribute, prior to padding, measured in bytes. Since |
+ // STUN aligns attributes on 32-bit boundaries, attributes whose content |
+ // is not a multiple of 4 bytes are padded with 1, 2, or 3 bytes of |
+ // padding so that its value contains a multiple of 4 bytes. The |
+ // padding bits are ignored, and may be any value. |
+ uint16 attr_type, attr_length; |
+ // Getting attribute type and length. |
+ attr_type = talk_base::GetBE16(&packet[*rtp_start_pos]); |
juberti2
2014/02/25 00:48:24
need to check len here
Mallinath (Gone from Chromium)
2014/02/25 22:56:44
Done. Also there is check in while condition.
|
+ attr_length = talk_base::GetBE16( |
+ &packet[*rtp_start_pos + sizeof(attr_type)]); |
+ if (attr_type != cricket::STUN_ATTR_DATA) { |
+ *rtp_start_pos += sizeof(attr_type) + sizeof(attr_length) + attr_length; |
+ if ((attr_length % 4) != 0) { |
+ *rtp_start_pos += (4- (attr_length % 4)); |
juberti2
2014/02/25 00:48:24
4 -
Mallinath (Gone from Chromium)
2014/02/25 22:56:44
Done.
|
+ } |
+ continue; |
+ } |
+ *rtp_start_pos += 4; // Skip STUN_DATA_ATTR header. |
+ *rtp_packet_length = attr_length; |
juberti2
2014/02/25 00:48:24
need to verify length is legit
Mallinath (Gone from Chromium)
2014/02/25 22:56:44
Done.
|
+ } |
+ } else { |
+ // This is a raw RTP packet. |
+ *rtp_start_pos = 0; |
+ *rtp_packet_length = length; |
+ } |
+ |
+ return IsRtpPacket(packet + *rtp_start_pos, length - *rtp_start_pos); |
+} |
+ |
+bool P2PSocketHost::MaybeUpdateRtpSendTimeExtn(char* rtp, int length, |
+ int extension_id) { |
+ // 0 1 2 3 |
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // |V=2|P|X| CC |M| PT | sequence number | |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // | timestamp | |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // | synchronization source (SSRC) identifier | |
+ // +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
+ // | contributing source (CSRC) identifiers | |
+ // | .... | |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ |
+ int cc_count = rtp[0] & 0x0F; |
+ int rtp_hdr_len_without_extn = kMinRtpHdrLen + 4 * cc_count; |
+ if (rtp_hdr_len_without_extn > length) { |
+ NOTREACHED(); |
+ return false; |
+ } |
+ |
+ bool X = (rtp[0] & 0x10); |
+ if (!X) // Return if extension bit is not set. |
+ return true; |
+ |
+ rtp += rtp_hdr_len_without_extn; |
+ |
+ // Getting extension profile ID and length. |
+ uint16 profile_id = talk_base::GetBE16(rtp); |
juberti2
2014/02/25 00:48:24
need to validate this length
Mallinath (Gone from Chromium)
2014/02/25 22:56:44
It's done @ line 287.
|
+ // Length is in 32 bit words. |
+ uint16 extn_length = talk_base::GetBE16(rtp + 2) * 4; |
+ |
+ // Verify input length against total header size. |
+ if (rtp_hdr_len_without_extn + kRtpExtnHdrLen + extn_length > length) { |
+ NOTREACHED(); |
+ return false; |
+ } |
+ |
+ rtp += kRtpExtnHdrLen; // Moving past extn header. |
+ |
+ bool found = false; |
juberti2
2014/02/25 00:48:24
not sure we need this - i think we can return true
Mallinath (Gone from Chromium)
2014/02/25 22:56:44
Agree. I added for testing this method when we pas
|
+ // WebRTC is using one byte header extension. |
+ // TODO(mallinath) - Handle two byte header extension. |
+ if (profile_id == 0xBEDE) { // OneByte extension header |
+ // 0 |
+ // 0 1 2 3 4 5 6 7 |
+ // +-+-+-+-+-+-+-+-+ |
+ // | ID | len | |
+ // +-+-+-+-+-+-+-+-+ |
+ |
+ // 0 1 2 3 |
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // | 0xBE | 0xDE | length=3 | |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // | ID | L=0 | data | ID | L=1 | data... |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // ...data | 0 (pad) | 0 (pad) | ID | L=3 | |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // | data | |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ char* extn_start = rtp; |
+ while (rtp - extn_start < extn_length) { |
+ const int id = (*rtp & 0xF0) >> 4; |
+ const int len = (*rtp & 0x0F) + 1; |
+ // The 4-bit length is the number minus one of data bytes of this header |
+ // extension element following the one-byte header. |
+ if (id == extension_id) { |
+ UpdateSendTimeExtnValue(rtp + kOneByteHdrLen, len); |
+ found = true; |
+ break; |
+ } |
+ rtp += kOneByteHdrLen + len; |
+ // Counting padding bytes |
+ while ((*rtp != 0) && (rtp - extn_start < extn_length)) { |
+ ++rtp; |
+ } |
+ } |
+ } |
+ return found; |
+} |
+ |
+void P2PSocketHost::UpdateSendTimeExtnValue(char* extn_data, int len) { |
+ // Absolute send time in RTP streams. |
+ // |
+ // The absolute send time is signaled to the receiver in-band using the |
+ // general mechanism for RTP header extensions [RFC5285]. The payload |
+ // of this extension (the transmitted value) is a 24-bit unsigned integer |
+ // containing the sender's current time in seconds as a fixed point number |
+ // with 18 bits fractional part. |
+ // |
+ // The form of the absolute send time extension block: |
+ // |
+ // 0 1 2 3 |
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // | ID | len=2 | absolute send time | |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ DCHECK_EQ(len, kAbsSendTimeExtnLen); |
+ // Now() has resolution ~1-15ms, using HighResNow(). But it is warned not to |
+ // use it unless necessary, as it is expensive than Now(). |
+ uint64 now_us = |
+ (base::TimeTicks::HighResNow() - base::TimeTicks()).InMicroseconds(); |
+ // Convert second to 24-bit unsigned with 18 bit fractional part |
+ uint32 now_second = ((now_us << 18) / base::Time::kMicrosecondsPerSecond) & |
+ 0x00FFFFFF; |
+ // TODO(mallinath) - Add SetBE24 to byteorder.h in libjingle. |
+ extn_data[0] = static_cast<uint8>(now_second >> 16); |
+ extn_data[1] = static_cast<uint8>(now_second >> 8); |
+ extn_data[2] = static_cast<uint8>(now_second); |
+} |
+ |
+// Assumes |len| is actual packet length + tag length. |
+void P2PSocketHost::MaybeUpdateRtpAuthTag( |
+ char* rtp, int len, const talk_base::PacketOptions& options) { |
+ size_t tag_length = options.packet_time_params.srtp_auth_tag_len; |
+ char* auth_tag = rtp + (len - tag_length); |
+ |
+ // We should have a fake HMAC value @ auth_tag. |
+ DCHECK_EQ(0, memcmp(auth_tag, kFakeAuthTag, tag_length)); |
+ |
+ crypto::HMAC hmac(crypto::HMAC::SHA1); |
+ if (!hmac.Init(reinterpret_cast<const unsigned char*>( |
+ &options.packet_time_params.srtp_auth_key[0]), |
+ options.packet_time_params.srtp_auth_key.size())) { |
+ NOTREACHED(); |
+ return; |
+ } |
+ |
+ if (hmac.DigestLength() < tag_length) { |
+ NOTREACHED(); |
+ return; |
+ } |
+ |
+ // Copy ROC after end of rtp packet. |
+ memcpy(auth_tag, &options.packet_time_params.srtp_packet_index, 4); |
+ |
+ unsigned char output[64]; |
+ size_t out_len = sizeof(output); |
+ DCHECK(!hmac.Sign(rtp, output, out_len)); |
juberti2
2014/02/25 00:48:24
how does Sign know how many bytes to hash?
Mallinath (Gone from Chromium)
2014/02/25 22:56:44
It calculates on whole |rtp|. Yes, there will be a
|
+ |
+ // Copy HMAC from output to packet. This is required as auth tag length |
+ // may not be equal to the actual HMAC length. |
+ memcpy(auth_tag, output, tag_length); |
+} |
+ |
} // namespace content |