Completed the functionalities of SrtpTransport.

webrtc/pc/channel.cc

 /*
  *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 140 matching lines @@
                          rtc::Thread* signaling_thread,
                          MediaChannel* media_channel,
                          const std::string& content_name,
                          bool rtcp_mux_required,
                          bool srtp_required)
     : worker_thread_(worker_thread),
       network_thread_(network_thread),
       signaling_thread_(signaling_thread),
       content_name_(content_name),
       rtcp_mux_required_(rtcp_mux_required),
-      rtp_transport_(
-          srtp_required
-              ? rtc::WrapUnique<webrtc::RtpTransportInternal>(
-                    new webrtc::SrtpTransport(rtcp_mux_required, content_name))
-              : rtc::MakeUnique<webrtc::RtpTransport>(rtcp_mux_required)),
       srtp_required_(srtp_required),
       media_channel_(media_channel),
       selected_candidate_pair_(nullptr) {
   RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
+  if (srtp_required) {
+    auto transport =
+        rtc::MakeUnique<webrtc::SrtpTransport>(rtcp_mux_required, content_name);
+    srtp_transport_ = transport.get();
+    rtp_transport_ = std::move(transport);
 #if defined(ENABLE_EXTERNAL_AUTH)
-  srtp_filter_.EnableExternalAuth();
+    srtp_transport_->EnableExternalAuth();
 #endif
+  } else {
+    rtp_transport_ = rtc::MakeUnique<webrtc::RtpTransport>(rtcp_mux_required);
+    srtp_transport_ = nullptr;
+  }
   rtp_transport_->SignalReadyToSend.connect(
       this, &BaseChannel::OnTransportReadyToSend);
   // TODO(zstein):  RtpTransport::SignalPacketReceived will probably be replaced
   // with a callback interface later so that the demuxer can select which
   // channel to signal.
   rtp_transport_->SignalPacketReceived.connect(this,
                                                &BaseChannel::OnPacketReceived);
   LOG(LS_INFO) << "Created channel for " << content_name;
 }
 
@@ skipping 131 matching lines @@
     return;
   }
 
   // When using DTLS-SRTP, we must reset the SrtpFilter every time the transport
   // changes and wait until the DTLS handshake is complete to set the newly
   // negotiated parameters.
   if (ShouldSetupDtlsSrtp_n()) {
     // Set |writable_| to false such that UpdateWritableState_w can set up
     // DTLS-SRTP when |writable_| becomes true again.
     writable_ = false;
     srtp_filter_.ResetParams();

[Taylor Brandstetter, 2017/08/23 22:13:29]

nit: Don't need to actually reset srtp_filter_ here. If DTLS is being used,
srtp_filter_ is unused, right?

[Zhi Huang, 2017/08/24 23:38:07]

Right. Previously, srtp_filter_.IsActive() can also imply the DTLS keys are
successfully installed. So I think the method |secure()| should be renamed to
|secure_sdes()| as well.

+    if (srtp_transport_) {
+      srtp_transport_->ResetParams();
+    }
   }
 
   // If this BaseChannel doesn't require RTCP mux and we haven't fully
   // negotiated RTCP mux, we need an RTCP transport.
   if (rtcp_packet_transport) {
     LOG(LS_INFO) << "Setting RTCP Transport for " << content_name() << " on "
                  << debug_name << " transport " << rtcp_packet_transport;
     SetTransport_n(true, rtcp_dtls_transport, rtcp_packet_transport);
   }
 
@@ skipping 247 matching lines @@
     return;
   }
 
   // Reset the srtp filter if it's not the CONNECTED state. For the CONNECTED
   // state, setting up DTLS-SRTP context is deferred to ChannelWritable_w to
   // cover other scenarios like the whole transport is writable (not just this
   // TransportChannel) or when TransportChannel is attached after DTLS is
   // negotiated.
   if (state != DTLS_TRANSPORT_CONNECTED) {
     srtp_filter_.ResetParams();
+    if (srtp_transport_) {
+      srtp_transport_->ResetParams();
+    }
   }
 }
 
 void BaseChannel::OnSelectedCandidatePairChanged(
     IceTransportInternal* ice_transport,
     CandidatePairInterface* selected_candidate_pair,
     int last_sent_packet_id,
     bool ready_to_send) {
   RTC_DCHECK((rtp_dtls_transport_ &&
               ice_transport == rtp_dtls_transport_->ice_transport()) ||
@@ skipping 53 matching lines @@
   }
 
   // Protect ourselves against crazy data.
   if (!ValidPacket(rtcp, packet)) {
     LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " "
                   << RtpRtcpStringLiteral(rtcp)
                   << " packet: wrong size=" << packet->size();
     return false;
   }
 
-  rtc::PacketOptions updated_options;
-  updated_options = options;
-  // Protect if needed.
-  if (srtp_filter_.IsActive()) {
-    TRACE_EVENT0("webrtc", "SRTP Encode");
-    bool res;
-    uint8_t* data = packet->data();
-    int len = static_cast<int>(packet->size());
-    if (!rtcp) {
-    // If ENABLE_EXTERNAL_AUTH flag is on then packet authentication is not done
-    // inside libsrtp for a RTP packet. A external HMAC module will be writing
-    // a fake HMAC value. This is ONLY done for a RTP packet.
-    // Socket layer will update rtp sendtime extension header if present in
-    // packet with current time before updating the HMAC.
-#if !defined(ENABLE_EXTERNAL_AUTH)
-      res = srtp_filter_.ProtectRtp(
-          data, len, static_cast<int>(packet->capacity()), &len);
-#else
-      if (!srtp_filter_.IsExternalAuthActive()) {
-        res = srtp_filter_.ProtectRtp(
-            data, len, static_cast<int>(packet->capacity()), &len);
-      } else {
-        updated_options.packet_time_params.rtp_sendtime_extension_id =
-            rtp_abs_sendtime_extn_id_;
-        res = srtp_filter_.ProtectRtp(
-            data, len, static_cast<int>(packet->capacity()), &len,
-            &updated_options.packet_time_params.srtp_packet_index);
-        // If protection succeeds, let's get auth params from srtp.
-        if (res) {
-          uint8_t* auth_key = NULL;
-          int key_len;
-          res = srtp_filter_.GetRtpAuthParams(
-              &auth_key, &key_len,
-              &updated_options.packet_time_params.srtp_auth_tag_len);
-          if (res) {
-            updated_options.packet_time_params.srtp_auth_key.resize(key_len);
-            updated_options.packet_time_params.srtp_auth_key.assign(
-                auth_key, auth_key + key_len);
-          }
-        }
-      }
-#endif
-      if (!res) {
-        int seq_num = -1;
-        uint32_t ssrc = 0;
-        GetRtpSeqNum(data, len, &seq_num);
-        GetRtpSsrc(data, len, &ssrc);
-        LOG(LS_ERROR) << "Failed to protect " << content_name_
-                      << " RTP packet: size=" << len
-                      << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
-        return false;
-      }
-    } else {
-      res = srtp_filter_.ProtectRtcp(data, len,
-                                     static_cast<int>(packet->capacity()),
-                                     &len);
-      if (!res) {
-        int type = -1;
-        GetRtcpType(data, len, &type);
-        LOG(LS_ERROR) << "Failed to protect " << content_name_
-                      << " RTCP packet: size=" << len << ", type=" << type;
-        return false;
-      }
-    }
-
-    // Update the length of the packet now that we've added the auth tag.
-    packet->SetSize(len);
-  } else if (srtp_required_) {
+  if (!srtp_filter_.IsActive() && srtp_required_) {
     // The audio/video engines may attempt to send RTCP packets as soon as the
     // streams are created, so don't treat this as an error for RTCP.
     // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6809
     if (rtcp) {
       return false;
     }
     // However, there shouldn't be any RTP packets sent before SRTP is set up
     // (and SetSend(true) is called).
     LOG(LS_ERROR) << "Can't send outgoing RTP packet when SRTP is inactive"
                   << " and crypto is required";
     RTC_NOTREACHED();
     return false;
   }
 
+  rtc::PacketOptions updated_options;
+  updated_options = options;

[Taylor Brandstetter, 2017/08/23 22:13:29]

Since the options are updated inside of srtp_transport_->SendPacket, don't need
to make a copy here.

[Zhi Huang, 2017/08/24 23:38:07]

Done.

+  if (srtp_filter_.IsActive()) {
+    RTC_DCHECK(srtp_transport_);
+    RTC_DCHECK(srtp_transport_->IsActive());
+    // Bon voyage.
+    int flags = secure_dtls() ? PF_SRTP_BYPASS : PF_NORMAL;
+    return srtp_transport_->SendPacket(rtcp, packet, updated_options, flags);
+  }
+
   // Bon voyage.
-  int flags = (secure() && secure_dtls()) ? PF_SRTP_BYPASS : PF_NORMAL;
-  return rtp_transport_->SendPacket(rtcp, packet, updated_options, flags);
+  return rtp_transport_->SendPacket(rtcp, packet, updated_options, PF_NORMAL);
 }
 
 bool BaseChannel::HandlesPayloadType(int packet_type) const {
   return rtp_transport_->HandlesPayloadType(packet_type);
 }
 
 void BaseChannel::OnPacketReceived(bool rtcp,
                                    rtc::CopyOnWriteBuffer* packet,
                                    const rtc::PacketTime& packet_time) {
   if (!has_received_packet_ && !rtcp) {
     has_received_packet_ = true;
     signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FIRSTPACKETRECEIVED);
   }
 
-  // Unprotect the packet, if needed.
-  if (srtp_filter_.IsActive()) {
-    TRACE_EVENT0("webrtc", "SRTP Decode");
-    char* data = packet->data<char>();
-    int len = static_cast<int>(packet->size());
-    bool res;
-    if (!rtcp) {
-      res = srtp_filter_.UnprotectRtp(data, len, &len);
-      if (!res) {
-        int seq_num = -1;
-        uint32_t ssrc = 0;
-        GetRtpSeqNum(data, len, &seq_num);
-        GetRtpSsrc(data, len, &ssrc);
-        LOG(LS_ERROR) << "Failed to unprotect " << content_name_
-                      << " RTP packet: size=" << len << ", seqnum=" << seq_num
-                      << ", SSRC=" << ssrc;
-        return;
-      }
-    } else {
-      res = srtp_filter_.UnprotectRtcp(data, len, &len);
-      if (!res) {
-        int type = -1;
-        GetRtcpType(data, len, &type);
-        LOG(LS_ERROR) << "Failed to unprotect " << content_name_
-                      << " RTCP packet: size=" << len << ", type=" << type;
-        return;
-      }
-    }
-
-    packet->SetSize(len);
-  } else if (srtp_required_) {
+  if (!srtp_filter_.IsActive() && srtp_required_) {
     // Our session description indicates that SRTP is required, but we got a
     // packet before our SRTP filter is active. This means either that
     // a) we got SRTP packets before we received the SDES keys, in which case
     //    we can't decrypt it anyway, or
     // b) we got SRTP packets before DTLS completed on both the RTP and RTCP
     //    transports, so we haven't yet extracted keys, even if DTLS did
     //    complete on the transport that the packets are being sent on. It's
     //    really good practice to wait for both RTP and RTCP to be good to go
     //    before sending  media, to prevent weird failure modes, so it's fine
     //    for us to just eat packets here. This is all sidestepped if RTCP mux
@@ skipping 185 matching lines @@
   if (role == rtc::SSL_SERVER) {
     send_key = &server_write_key;
     recv_key = &client_write_key;
   } else {
     send_key = &client_write_key;
     recv_key = &server_write_key;
   }
 
   if (!srtp_filter_.IsActive()) {
     if (rtcp) {
-      ret = srtp_filter_.SetRtcpParams(selected_crypto_suite, &(*send_key)[0],
-                                       static_cast<int>(send_key->size()),
-                                       selected_crypto_suite, &(*recv_key)[0],
-                                       static_cast<int>(recv_key->size()));
+      RTC_DCHECK(srtp_transport_);
+      ret = srtp_transport_->SetRtcpParams(
+          selected_crypto_suite, &(*send_key)[0],
+          static_cast<int>(send_key->size()), selected_crypto_suite,
+          &(*recv_key)[0], static_cast<int>(recv_key->size()));
     } else {
-      ret = srtp_filter_.SetRtpParams(selected_crypto_suite, &(*send_key)[0],
-                                      static_cast<int>(send_key->size()),
-                                      selected_crypto_suite, &(*recv_key)[0],
-                                      static_cast<int>(recv_key->size()));
+      RTC_DCHECK(srtp_transport_);
+      // If the SRTP crypto keys are from DTLS handshake, explicitly call
+      // |EnableDtlsSrtp| to activate SrtpFilter.
+      srtp_filter_.EnableDtlsSrtp();

[Taylor Brandstetter, 2017/08/23 22:13:29]

Why does srtp_filter_ even need to know if DTLS-SRTP is used? If DTLS-SRTP is
used, srtp_filter_ isn't really used, right?

[Zhi Huang, 2017/08/24 23:38:07]

Done.

+      ret = srtp_transport_->SetRtpParams(
+          selected_crypto_suite, &(*send_key)[0],
+          static_cast<int>(send_key->size()), selected_crypto_suite,
+          &(*recv_key)[0], static_cast<int>(recv_key->size()));
     }
   } else {
     if (rtcp) {
       // RTCP doesn't need to be updated because UpdateRtpParams is only used
       // to update the set of encrypted RTP header extension IDs.
       ret = true;
     } else {
-      ret = srtp_filter_.UpdateRtpParams(
-          selected_crypto_suite,
-          &(*send_key)[0], static_cast<int>(send_key->size()),
-          selected_crypto_suite,
+      ret = srtp_transport_->UpdateRtpParams(
+          selected_crypto_suite, &(*send_key)[0],
+          static_cast<int>(send_key->size()), selected_crypto_suite,
           &(*recv_key)[0], static_cast<int>(recv_key->size()));
     }
   }
 
   if (!ret) {
     LOG(LS_WARNING) << "DTLS-SRTP key installation failed";
   } else {
     dtls_keyed_ = true;
     UpdateTransportOverhead();
   }
   return ret;
 }
 
 void BaseChannel::MaybeSetupDtlsSrtp_n() {
   if (srtp_filter_.IsActive()) {
     return;
   }
 
   if (!ShouldSetupDtlsSrtp_n()) {
     return;
   }
 
+  if (!srtp_transport_) {
+    EnableSrtpTransport_n();
+  }
+
   if (!SetupDtlsSrtp_n(false)) {
     SignalDtlsSrtpSetupFailure_n(false);
     return;
   }
 
   if (rtcp_dtls_transport_) {
     if (!SetupDtlsSrtp_n(true)) {
       SignalDtlsSrtpSetupFailure_n(true);
       return;
     }
@@ skipping 63 matching lines @@
                                     bool* dtls,
                                     std::string* error_desc) {
   *dtls = rtp_dtls_transport_ && rtp_dtls_transport_->IsDtlsActive();
   if (*dtls && !cryptos.empty()) {
     SafeSetError("Cryptos must be empty when DTLS is active.", error_desc);
     return false;
   }
   return true;
 }
 
+void BaseChannel::EnableSrtpTransport_n() {
+  if (srtp_transport_ == nullptr) {
+    rtp_transport_->SignalReadyToSend.disconnect(this);
+    rtp_transport_->SignalPacketReceived.disconnect(this);
+
+    auto transport = rtc::MakeUnique<webrtc::SrtpTransport>(
+        std::move(rtp_transport_), content_name_);
+    srtp_transport_ = transport.get();
+    rtp_transport_ = std::move(transport);
+
+    rtp_transport_->SignalReadyToSend.connect(
+        this, &BaseChannel::OnTransportReadyToSend);
+    rtp_transport_->SignalPacketReceived.connect(
+        this, &BaseChannel::OnPacketReceived);
+
+    if (rtp_abs_sendtime_extn_id_ != -1) {
+      srtp_transport_->CacheRtpAbsSendTimeHeaderExtension(
+          rtp_abs_sendtime_extn_id_);
+    }
+    LOG(LS_INFO) << "Wrapping RtpTransport in SrtpTransport.";

[Taylor Brandstetter, 2017/08/23 22:13:29]

I don't think upgrading from plain RTP to SRTP is something we need to support.
RFC4568 says:

   When the offerer receives the answer, the offerer MUST verify that
   one of the initially offered crypto suites and its accompanying tag
   were accepted and echoed in the answer.

So I'd say we should drop support for this, for both simplicity and standards
compliance. Probably safest to do so in a separate CL.

This means that we'd only create the RTP transport object once the first
description is set, and the "proto" string (e.g., "RTP", "RTP/SAVPF" or
"UDP/TLS/RTP/SAVPF") is known.

[Zhi Huang, 2017/08/24 23:38:07]

I would like to talk about this a little bit more.

+  }
+}
+
 bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos,
                             ContentAction action,
                             ContentSource src,
                             const std::vector<int>& encrypted_extension_ids,
                             std::string* error_desc) {
   TRACE_EVENT0("webrtc", "BaseChannel::SetSrtp_w");
   if (action == CA_UPDATE) {
     // no crypto params.
     return true;
   }
   bool ret = false;
   bool dtls = false;
   ret = CheckSrtpConfig_n(cryptos, &dtls, error_desc);
   if (!ret) {
     return false;
   }
-  srtp_filter_.SetEncryptedHeaderExtensionIds(src, encrypted_extension_ids);
+  if (!srtp_transport_ && !dtls) {
+    EnableSrtpTransport_n();
+  }
+  if (srtp_transport_) {
+    srtp_transport_->SetEncryptedHeaderExtensionIds(src,
+                                                    encrypted_extension_ids);
+  }
   switch (action) {
     case CA_OFFER:
       // If DTLS is already active on the channel, we could be renegotiating
       // here. We don't update the srtp filter.
       if (!dtls) {
         ret = srtp_filter_.SetOffer(cryptos, src);
       }
       break;
     case CA_PRANSWER:
       // If we're doing DTLS-SRTP, we don't want to update the filter
       // with an answer, because we already have SRTP parameters.
       if (!dtls) {
         ret = srtp_filter_.SetProvisionalAnswer(cryptos, src);
+        if (ret) {
+          if (srtp_filter_.send_cipher_suite() &&
+              srtp_filter_.recv_cipher_suite()) {
+            auto send_key = srtp_filter_.send_key();
+            auto recv_key = srtp_filter_.recv_key();
+            ret = srtp_transport_->SetRtpParams(
+                *(srtp_filter_.send_cipher_suite()), &(*send_key)[0],
+                static_cast<int>(send_key->size()),
+                *(srtp_filter_.recv_cipher_suite()), &(*recv_key)[0],
+                static_cast<int>(recv_key->size()));
+          } else {
+            LOG(LS_INFO) << "No crypto keys are provided for SDES.";
+          }
+        }
       }
       break;
     case CA_ANSWER:
       // If we're doing DTLS-SRTP, we don't want to update the filter
       // with an answer, because we already have SRTP parameters.
       if (!dtls) {
         ret = srtp_filter_.SetAnswer(cryptos, src);
+        if (ret) {
+          if (srtp_filter_.send_cipher_suite() &&
+              srtp_filter_.recv_cipher_suite()) {
+            auto send_key = srtp_filter_.send_key();
+            auto recv_key = srtp_filter_.recv_key();
+            ret = srtp_transport_->SetRtpParams(
+                *(srtp_filter_.send_cipher_suite()), &(*send_key)[0],
+                static_cast<int>(send_key->size()),
+                *(srtp_filter_.recv_cipher_suite()), &(*recv_key)[0],
+                static_cast<int>(recv_key->size()));
+          } else {
+            // Explicitly reset the |srtp_transport_| if no crypto param is
+            // provided in the answer. No need to call |ResetParams()| for
+            // |srtp_filter_| because it resets the params inside |SetAnswer|.
+            srtp_transport_->ResetParams();
+            LOG(LS_INFO) << "No crypto keys are provided for SDES.";

[Taylor Brandstetter, 2017/08/23 22:13:29]

This looks the same as the code above; could it be moved out of the switch
statement?

// If setting an SDES answer succeeded, apply the negotiated parameters
// to the SRTP transport.
if ((action == CA_PRANSWER || action == CA_ANSWER) && !dlts && ret) {
  // ...
}

[Zhi Huang, 2017/08/24 23:38:07]

Yes, that would be cleaner.

+          }
+        }
       }
       break;
     default:
       break;
   }
   // Only update SRTP filter if using DTLS. SDES is handled internally
   // by the SRTP filter.
   // TODO(jbauch): Only update if encrypted extension ids have changed.
   if (ret && dtls_keyed_ && rtp_dtls_transport_ &&
       rtp_dtls_transport_->dtls_state() == DTLS_TRANSPORT_CONNECTED) {
@@ skipping 32 matching lines @@
       break;
     case CA_ANSWER:
       ret = rtcp_mux_filter_.SetAnswer(enable, src);
       if (ret && rtcp_mux_filter_.IsActive()) {
         // We permanently activated RTCP muxing; signal that we no longer need
         // the RTCP transport.
         std::string debug_name =
             transport_name_.empty()
                 ? rtp_transport_->rtp_packet_transport()->debug_name()
                 : transport_name_;
-        ;
         LOG(LS_INFO) << "Enabling rtcp-mux for " << content_name()
                      << "; no longer need RTCP transport for " << debug_name;
         if (rtp_transport_->rtcp_packet_transport()) {
           SetTransport_n(true, nullptr, nullptr);
           SignalRtcpMuxFullyActive(transport_name_);
         }
         UpdateWritableState_n();
       }
       break;
     case CA_UPDATE:
@@ skipping 208 matching lines @@
       send_time_extension ? send_time_extension->id : -1;
   invoker_.AsyncInvoke<void>(
       RTC_FROM_HERE, network_thread_,
       Bind(&BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n, this,
            rtp_abs_sendtime_extn_id));
 #endif
 }
 
 void BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n(
     int rtp_abs_sendtime_extn_id) {
   rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id;

[Taylor Brandstetter, 2017/08/23 22:13:29]

If this were always called after the SRTP transport is created, it wouldn't need
to be cached in both places.

[Zhi Huang, 2017/08/24 23:38:07]

This is called after SetTransportParameters which create the srtptransport. I'll
add a DCHECK here then.

+  if (srtp_transport_) {
+    srtp_transport_->CacheRtpAbsSendTimeHeaderExtension(
+        rtp_abs_sendtime_extn_id_);
+  }
 }
 
 void BaseChannel::OnMessage(rtc::Message *pmsg) {
   TRACE_EVENT0("webrtc", "BaseChannel::OnMessage");
   switch (pmsg->message_id) {
     case MSG_SEND_RTP_PACKET:
     case MSG_SEND_RTCP_PACKET: {
       RTC_DCHECK(network_thread_->IsCurrent());
       SendPacketMessageData* data =
           static_cast<SendPacketMessageData*>(pmsg->pdata);
@@ skipping 265 matching lines @@
   constexpr int kUdpOverhaed = 8;
   constexpr int kTcpOverhaed = 20;
   transport_overhead_per_packet +=
       selected_candidate_pair_->local_candidate().protocol() ==
               TCP_PROTOCOL_NAME
           ? kTcpOverhaed
           : kUdpOverhaed;
 
   if (secure()) {
     int srtp_overhead = 0;
-    if (srtp_filter_.GetSrtpOverhead(&srtp_overhead))
+    if (srtp_transport_->GetSrtpOverhead(&srtp_overhead))
       transport_overhead_per_packet += srtp_overhead;
   }
 
   return transport_overhead_per_packet;
 }
 
 void BaseChannel::UpdateTransportOverhead() {
   int transport_overhead_per_packet = GetTransportOverheadPerPacket();
   if (transport_overhead_per_packet)
     invoker_.AsyncInvoke<void>(
@@ skipping 714 matching lines @@
 
 void RtpDataChannel::OnDataChannelReadyToSend(bool writable) {
   // This is usded for congestion control to indicate that the stream is ready
   // to send by the MediaChannel, as opposed to OnReadyToSend, which indicates
   // that the transport channel is ready.
   signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA,
                            new DataChannelReadyToSendMessageData(writable));
 }
 
 }  // namespace cricket