Support encrypted RTP extensions (RFC 6904)

webrtc/pc/channel.cc

Index: webrtc/pc/channel.cc
diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc
index ecc1c9a633c6d4a935a146f048483256ccdc5d49..09a4d8360a927c6189620d53dfae5f4150058fec 100644
--- a/webrtc/pc/channel.cc
+++ b/webrtc/pc/channel.cc
@@ -43,20 +43,6 @@ struct SendPacketMessageData : public rtc::MessageData {
   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 {
@@ -1150,26 +1136,39 @@ 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;
   }
 
@@ -1196,6 +1195,7 @@ bool BaseChannel::CheckSrtpConfig_n(const std::vector<CryptoParams>& cryptos,
 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) {
@@ -1208,6 +1208,7 @@ bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos,
   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
@@ -1470,7 +1471,8 @@ void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension_w(
 // 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>(
@@ -1804,7 +1806,8 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
     return false;
   }
 
-  if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_LOCAL,
+      audio->rtp_header_extensions(), error_desc)) {
     return false;
   }
 
@@ -1849,7 +1852,8 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
     return false;
   }
 
-  if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_REMOTE,
+      audio->rtp_header_extensions(), error_desc)) {
     return false;
   }
 
@@ -2082,7 +2086,8 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
     return false;
   }
 
-  if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_LOCAL,
+      video->rtp_header_extensions(), error_desc)) {
     return false;
   }
 
@@ -2127,7 +2132,8 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
     return false;
   }
 
-  if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_REMOTE,
+      video->rtp_header_extensions(), error_desc)) {
     return false;
   }
 
@@ -2282,7 +2288,8 @@ bool RtpDataChannel::SetLocalContent_w(const MediaContentDescription* content,
     return false;
   }
 
-  if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_LOCAL,
+      data->rtp_header_extensions(), error_desc)) {
     return false;
   }
 
@@ -2337,7 +2344,8 @@ bool RtpDataChannel::SetRemoteContent_w(const MediaContentDescription* content,
   }
 
   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;
   }