Support encrypted RTP extensions (RFC 6904)

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 25 matching lines @@
                        std::unique_ptr<webrtc::AudioSinkInterface>* sink) {
   channel->SetRawAudioSink(ssrc, std::move(*sink));
   return true;
 }
 
 struct SendPacketMessageData : public rtc::MessageData {
   rtc::CopyOnWriteBuffer packet;
   rtc::PacketOptions options;
 };
 
-#if defined(ENABLE_EXTERNAL_AUTH)
-// Returns the named header extension if found among all extensions,
-// nullptr otherwise.
-const webrtc::RtpExtension* FindHeaderExtension(
-    const std::vector<webrtc::RtpExtension>& extensions,
-    const std::string& uri) {
-  for (const auto& extension : extensions) {
-    if (extension.uri == uri)
-      return &extension;
-  }
-  return nullptr;
-}
-#endif
-
 }  // namespace
 
 enum {
   MSG_EARLYMEDIATIMEOUT = 1,
   MSG_SEND_RTP_PACKET,
   MSG_SEND_RTCP_PACKET,
   MSG_CHANNEL_ERROR,
   MSG_READYTOSENDDATA,
   MSG_DATARECEIVED,
   MSG_FIRSTPACKETRECEIVED,
@@ skipping 1073 matching lines @@
 
   LOG(LS_INFO) << "Channel not writable (" << content_name_ << ")";
   writable_ = false;
   UpdateMediaSendRecvState();
 }
 
 bool BaseChannel::SetRtpTransportParameters(
     const MediaContentDescription* content,
     ContentAction action,
     ContentSource src,
+    const RtpHeaderExtensions& extensions,
     std::string* error_desc) {
   if (action == CA_UPDATE) {
     // These parameters never get changed by a CA_UDPATE.
     return true;
   }
 
+  RtpHeaderExtensions encrypted_extensions;
+  for (const webrtc::RtpExtension& extension : extensions) {
+    if (extension.encrypt) {
+      LOG(LS_INFO) << "Using " << src << " encrypted extension: "

[Taylor Brandstetter, 2017/03/23 20:10:56]

nit: "src" will print as "0" or "1" (I believe), which isn't ideal. Maybe add a
stream insertion operator, or just do:

src == CS_LOCAL ? "local" : "remote"

[joachim, 2017/03/30 22:43:49]

Done.

+          << extension.ToString();
+      encrypted_extensions.push_back(extension);
+    }
+  }
+
   // Cache srtp_required_ for belt and suspenders check on SendPacket
   return network_thread_->Invoke<bool>(
       RTC_FROM_HERE, Bind(&BaseChannel::SetRtpTransportParameters_n, this,
-                          content, action, src, error_desc));
+                          content, action, src, encrypted_extensions,
+                          error_desc));
 }
 
 bool BaseChannel::SetRtpTransportParameters_n(
     const MediaContentDescription* content,
     ContentAction action,
     ContentSource src,
+    const RtpHeaderExtensions& encrypted_extensions,
     std::string* error_desc) {
   RTC_DCHECK(network_thread_->IsCurrent());
 
-  if (!SetSrtp_n(content->cryptos(), action, src, error_desc)) {
+  if (!SetSrtp_n(content->cryptos(), action, src, encrypted_extensions,
+      error_desc)) {
     return false;
   }
 
   if (!SetRtcpMux_n(content->rtcp_mux(), action, src, error_desc)) {
     return false;
   }
 
   return true;
 }
 
 // |dtls| will be set to true if DTLS is active for transport and crypto is
 // empty.
 bool BaseChannel::CheckSrtpConfig_n(const std::vector<CryptoParams>& cryptos,
                                     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;
 }
 
 bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos,
                             ContentAction action,
                             ContentSource src,
+                            const RtpHeaderExtensions& encrypted_extensions,
                             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_.SetEncryptedHeaderExtensions(src, encrypted_extensions);
   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
@@ skipping 242 matching lines @@
   return ret;
 }
 
 void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension_w(
     const std::vector<webrtc::RtpExtension>& extensions) {
 // Absolute Send Time extension id is used only with external auth,
 // so do not bother searching for it and making asyncronious call to set
 // something that is not used.
 #if defined(ENABLE_EXTERNAL_AUTH)
   const webrtc::RtpExtension* send_time_extension =
-      FindHeaderExtension(extensions, webrtc::RtpExtension::kAbsSendTimeUri);
+      webrtc::RtpExtension::FindHeaderExtensionByUri(
+          extensions, webrtc::RtpExtension::kAbsSendTimeUri);
   int rtp_abs_sendtime_extn_id =
       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(
@@ skipping 313 matching lines @@
   LOG(LS_INFO) << "Setting local voice description";
 
   const AudioContentDescription* audio =
       static_cast<const AudioContentDescription*>(content);
   RTC_DCHECK(audio != NULL);
   if (!audio) {
     SafeSetError("Can't find audio content in local description.", error_desc);
     return false;
   }
 
-  if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_LOCAL,
+      audio->rtp_header_extensions(), error_desc)) {
     return false;
   }
 
   AudioRecvParameters recv_params = last_recv_params_;
   RtpParametersFromMediaDescription(audio, &recv_params);
   if (!media_channel()->SetRecvParameters(recv_params)) {
     SafeSetError("Failed to set local audio description recv parameters.",
                  error_desc);
     return false;
   }
@@ skipping 24 matching lines @@
   LOG(LS_INFO) << "Setting remote voice description";
 
   const AudioContentDescription* audio =
       static_cast<const AudioContentDescription*>(content);
   RTC_DCHECK(audio != NULL);
   if (!audio) {
     SafeSetError("Can't find audio content in remote description.", error_desc);
     return false;
   }
 
-  if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_REMOTE,
+      audio->rtp_header_extensions(), error_desc)) {
     return false;
   }
 
   AudioSendParameters send_params = last_send_params_;
   RtpSendParametersFromMediaDescription(audio, &send_params);
   if (audio->agc_minus_10db()) {
     send_params.options.adjust_agc_delta = rtc::Optional<int>(kAgcMinus10db);
   }
 
   bool parameters_applied = media_channel()->SetSendParameters(send_params);
@@ skipping 212 matching lines @@
   LOG(LS_INFO) << "Setting local video description";
 
   const VideoContentDescription* video =
       static_cast<const VideoContentDescription*>(content);
   RTC_DCHECK(video != NULL);
   if (!video) {
     SafeSetError("Can't find video content in local description.", error_desc);
     return false;
   }
 
-  if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_LOCAL,
+      video->rtp_header_extensions(), error_desc)) {
     return false;
   }
 
   VideoRecvParameters recv_params = last_recv_params_;
   RtpParametersFromMediaDescription(video, &recv_params);
   if (!media_channel()->SetRecvParameters(recv_params)) {
     SafeSetError("Failed to set local video description recv parameters.",
                  error_desc);
     return false;
   }
@@ skipping 24 matching lines @@
   LOG(LS_INFO) << "Setting remote video description";
 
   const VideoContentDescription* video =
       static_cast<const VideoContentDescription*>(content);
   RTC_DCHECK(video != NULL);
   if (!video) {
     SafeSetError("Can't find video content in remote description.", error_desc);
     return false;
   }
 
-  if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_REMOTE,
+      video->rtp_header_extensions(), error_desc)) {
     return false;
   }
 
   VideoSendParameters send_params = last_send_params_;
   RtpSendParametersFromMediaDescription(video, &send_params);
   if (video->conference_mode()) {
     send_params.conference_mode = true;
   }
 
   bool parameters_applied = media_channel()->SetSendParameters(send_params);
@@ skipping 134 matching lines @@
   RTC_DCHECK(data != NULL);
   if (!data) {
     SafeSetError("Can't find data content in local description.", error_desc);
     return false;
   }
 
   if (!CheckDataChannelTypeFromContent(data, error_desc)) {
     return false;
   }
 
-  if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_LOCAL,
+      data->rtp_header_extensions(), error_desc)) {
     return false;
   }
 
   DataRecvParameters recv_params = last_recv_params_;
   RtpParametersFromMediaDescription(data, &recv_params);
   if (!media_channel()->SetRecvParameters(recv_params)) {
     SafeSetError("Failed to set remote data description recv parameters.",
                  error_desc);
     return false;
   }
@@ skipping 34 matching lines @@
   // must be empty, so ignore it.
   if (!data->has_codecs() && action != CA_UPDATE) {
     return true;
   }
 
   if (!CheckDataChannelTypeFromContent(data, error_desc)) {
     return false;
   }
 
   LOG(LS_INFO) << "Setting remote data description";
-  if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_REMOTE,
+      data->rtp_header_extensions(), error_desc)) {
     return false;
   }
 
   DataSendParameters send_params = last_send_params_;
   RtpSendParametersFromMediaDescription<DataCodec>(data, &send_params);
   if (!media_channel()->SetSendParameters(send_params)) {
     SafeSetError("Failed to set remote data description send parameters.",
                  error_desc);
     return false;
   }
@@ skipping 114 matching lines @@
   signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA,
                            new DataChannelReadyToSendMessageData(writable));
 }
 
 void RtpDataChannel::GetSrtpCryptoSuites_n(
     std::vector<int>* crypto_suites) const {
   GetSupportedDataCryptoSuites(crypto_options(), crypto_suites);
 }
 
 }  // namespace cricket