dump raw rtp packets in text2pcap format

guarded by a new field trial flag WebRTC-Debugging-RtpDump.
Packets have a RTP_DUMP postfix for easy grep-ing.

BUG=webrtc:10675

Change-Id: I73c0e0db47dca1079cd303c41a8b80fd7ae4a902
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196087
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32775}

pc/srtp_session.cc

@@ -10,11 +10,16 @@
 
 #include "pc/srtp_session.h"
 
+#include <iomanip>
+
 #include "absl/base/attributes.h"
 #include "media/base/rtp_utils.h"
 #include "pc/external_hmac.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/ssl_stream_adapter.h"
+#include "rtc_base/string_encode.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/metrics.h"
 #include "third_party/libsrtp/include/srtp.h"
 #include "third_party/libsrtp/include/srtp_priv.h"
@@ -26,7 +31,9 @@ namespace cricket {
 // in srtp.h.
 constexpr int kSrtpErrorCodeBoundary = 28;
 
-SrtpSession::SrtpSession() {}
+SrtpSession::SrtpSession() {
+  dump_plain_rtp_ = webrtc::field_trial::IsEnabled("WebRTC-Debugging-RtpDump");
+}
 
 SrtpSession::~SrtpSession() {
   if (session_) {
@@ -79,6 +86,9 @@ bool SrtpSession::ProtectRtp(void* p, int in_len, int max_len, int* out_len) {
                         << max_len << " is less than the needed " << need_len;
     return false;
   }
+  if (dump_plain_rtp_) {
+    DumpPacket(p, in_len, /*outbound=*/true);
+  }
 
   *out_len = in_len;
   int err = srtp_protect(session_, p, out_len);
@@ -118,6 +128,9 @@ bool SrtpSession::ProtectRtcp(void* p, int in_len, int max_len, int* out_len) {
                         << max_len << " is less than the needed " << need_len;
     return false;
   }
+  if (dump_plain_rtp_) {
+    DumpPacket(p, in_len, /*outbound=*/true);
+  }
 
   *out_len = in_len;
   int err = srtp_protect_rtcp(session_, p, out_len);
@@ -151,6 +164,9 @@ bool SrtpSession::UnprotectRtp(void* p, int in_len, int* out_len) {
                               static_cast<int>(err), kSrtpErrorCodeBoundary);
     return false;
   }
+  if (dump_plain_rtp_) {
+    DumpPacket(p, *out_len, /*outbound=*/false);
+  }
   return true;
 }
 
@@ -169,6 +185,9 @@ bool SrtpSession::UnprotectRtcp(void* p, int in_len, int* out_len) {
                               static_cast<int>(err), kSrtpErrorCodeBoundary);
     return false;
   }
+  if (dump_plain_rtp_) {
+    DumpPacket(p, *out_len, /*outbound=*/false);
+  }
   return true;
 }
 
@@ -444,4 +463,25 @@ void SrtpSession::HandleEventThunk(srtp_event_data_t* ev) {
   }
 }
 
+// Logs the unencrypted packet in text2pcap format. This can then be
+// extracted by searching for RTP_DUMP
+//   grep RTP_DUMP chrome_debug.log > in.txt
+// and converted to pcap using
+//   text2pcap -D -u 1000,2000 -t %H:%M:%S. in.txt out.pcap
+// The resulting file can be replayed using the WebRTC video_replay tool and
+// be inspected in Wireshark using the RTP, VP8 and H264 dissectors.
+void SrtpSession::DumpPacket(const void* buf, int len, bool outbound) {
+  int64_t time_of_day = rtc::TimeUTCMillis() % (24 * 3600 * 1000);
+  int64_t hours = time_of_day / (3600 * 1000);
+  int64_t minutes = (time_of_day / (60 * 1000)) % 60;
+  int64_t seconds = (time_of_day / 1000) % 60;
+  int64_t millis = time_of_day % 1000;
+  RTC_LOG(LS_VERBOSE) << "\n" << (outbound ? "O" : "I") << " "
+    << std::setw(2) << hours << ":" << std::setw(2) << minutes << ":"
+    << std::setw(2) << seconds << "." << std::setw(3)
+    << std::setfill('0') << millis << " "
+    << "000000 " << rtc::hex_encode_with_delimiter((const char *)buf, len, ' ')
+    << " # RTP_DUMP";
+}
+
 }  // namespace cricket

pc/srtp_session.h

@@ -109,6 +109,10 @@ class SrtpSession {
   // Returns send stream current packet index from srtp db.
   bool GetSendStreamPacketIndex(void* data, int in_len, int64_t* index);
 
+  // Writes unencrypted packets in text2pcap format to the log file
+  // for debugging.
+  void DumpPacket(const void* buf, int len, bool outbound);
+
   // These methods are responsible for initializing libsrtp (if the usage count
   // is incremented from 0 to 1) or deinitializing it (when decremented from 1
   // to 0).
@@ -130,6 +134,7 @@ class SrtpSession {
   bool external_auth_active_ = false;
   bool external_auth_enabled_ = false;
   int decryption_failure_count_ = 0;
+  bool dump_plain_rtp_ = false;
   RTC_DISALLOW_COPY_AND_ASSIGN(SrtpSession);
 };