Fixing segfault caused by TurnServer.

webrtc/p2p/base/turnserver.cc

Index: webrtc/p2p/base/turnserver.cc
diff --git a/webrtc/p2p/base/turnserver.cc b/webrtc/p2p/base/turnserver.cc
index 7dd9e19f63d0f9abd5268218d3210d84f01ab33b..25d090c0800edc375aeae4895c31d95fe0d65ecd 100644
--- a/webrtc/p2p/base/turnserver.cc
+++ b/webrtc/p2p/base/turnserver.cc
@@ -525,10 +525,14 @@ void TurnServer::DestroyInternalSocket(rtc::AsyncPacketSocket* socket) {
   InternalSocketMap::iterator iter = server_sockets_.find(socket);
   if (iter != server_sockets_.end()) {
     rtc::AsyncPacketSocket* socket = iter->first;
-    // We must destroy the socket async to avoid invalidating the sigslot
-    // callback list iterator inside a sigslot callback.
-    rtc::Thread::Current()->Dispose(socket);
     server_sockets_.erase(iter);
+    // We must destroy the socket async to avoid invalidating the sigslot
+    // callback list iterator inside a sigslot callback. (In other words,
+    // deleting an object from within a callback from that object).
+    pending_delete_sockets_.push_back(
+        std::unique_ptr<rtc::AsyncPacketSocket>(socket));
+    invoker_.AsyncInvoke<void>(RTC_FROM_HERE, rtc::Thread::Current(),
+                               [this] { pending_delete_sockets_.clear(); });

[pthatcher1, 2016/08/23 00:36:55]

I think the style guide said not to store lambdas.

[Taylor Brandstetter, 2016/08/23 01:11:20]

The Chromium style guide says to never store *capturing* lambdas because
non-capturing lambdas used with base::Bind provide added protection:

invoker_.AsyncInvoke<void>(RTC_FROM_HERE, rtc::Thread::Current(),
  base::Bind([] (TurnServer* self) {
    self->pending_delete_sockets_.clear();
  }, this));

But we don't have Chromium's base::Bind. So what would you do here? Is a lambda
basically out of the question, and we have add a 1 line method instead?

[pthatcher1, 2016/08/23 01:15:32]

Ah, thanks for correcting my poor memory.  
That's what we've been doing elsewhere.  We could probably be less strict for
code that doesn't go in Chromium (like this one).  But it's probably worth
checking with the rest of the team via webrtc-eng@.

[Taylor Brandstetter, 2016/08/23 23:31:12]

I'll just bite the bullet, it's only a few extra lines of code.

   }
 }