Wire up transport sequence number / send time callbacks to webrtc via libjingle.

webrtc/base/asynctcpsocket.cc

Index: webrtc/base/asynctcpsocket.cc
diff --git a/webrtc/base/asynctcpsocket.cc b/webrtc/base/asynctcpsocket.cc
index 97d1e171761dfcd9eb2a1a4724ffa24516a525d9..c774b5cca32988eae51f04aa1c6d8d361266e9f2 100644
--- a/webrtc/base/asynctcpsocket.cc
+++ b/webrtc/base/asynctcpsocket.cc
@@ -268,6 +268,9 @@ int AsyncTCPSocket::Send(const void *pv, size_t cb,
     return res;
   }
 
+  rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
+  SignalPacketSent(this, GetRemoteAddress(), sent_packet);

[pthatcher1, 2015/10/05 18:30:14]

We have SignalPacketSent, but that doesn't signal when the packet gets dropped
above and is never sent.  Should we have a SignalPacketDropped or a field on
sent_packet that's ".dropped" (of course, then SentPacket would be a misnomer,
and we'd have to think of a different name)?  In other words, would webrtc::Call
want to know that we dropped a packet before it went out?

[stefan-webrtc, 2015/10/06 13:00:19]

This is not something I'm trying to achieve right now, but could potentially be
something we want to do in the future. I'd suggest that we make those changes
when/if we need them.

We could already infer that by identifying that a certain packet id didn't get a
SignalPacketSent().

[pthatcher1, 2015/10/06 20:23:55]

Makes sense.

+
   // We claim to have sent the whole thing, even if we only sent partial
   return static_cast<int>(cb);
 }