Call should allow pass through of keep-alive packets.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 41f53feee7c57ed5e42753ad9e2a78075eb8a22c..2440cd79c41c09afd81c445d2ecc18250e99e99f 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -236,10 +236,10 @@ class Call : public webrtc::Call,
                                  MediaType media_type)
       SHARED_LOCKS_REQUIRED(receive_crit_);
 
-  rtc::Optional<RtpPacketReceived> ParseRtpPacket(const uint8_t* packet,
-                                                  size_t length,
-                                                  const PacketTime* packet_time)
-      SHARED_LOCKS_REQUIRED(receive_crit_);
+  rtc::Optional<RtpPacketReceived> ParseRtpPacket(
+      const uint8_t* packet,
+      size_t length,
+      const PacketTime* packet_time) const;
 
   void UpdateSendHistograms(int64_t first_sent_packet_ms)
       EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
@@ -485,7 +485,7 @@ Call::~Call() {
 rtc::Optional<RtpPacketReceived> Call::ParseRtpPacket(
     const uint8_t* packet,
     size_t length,
-    const PacketTime* packet_time) {
+    const PacketTime* packet_time) const {
   RtpPacketReceived parsed_packet;
   if (!parsed_packet.Parse(packet, length))
     return rtc::Optional<RtpPacketReceived>();
@@ -1299,17 +1299,20 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                 const PacketTime& packet_time) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtp");
 
-  RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO);
-
-  ReadLockScoped read_lock(*receive_crit_);
   // TODO(nisse): We should parse the RTP header only here, and pass
   // on parsed_packet to the receive streams.
   rtc::Optional<RtpPacketReceived> parsed_packet =
       ParseRtpPacket(packet, length, &packet_time);
+  const bool is_keep_alive_packet =

[pbos-webrtc, 2017/07/10 22:36:31]

Can you put some reference to keepalive packets to motivate this check? Should
this also be verifying a payload type?

[pbos-webrtc, 2017/07/10 22:39:16]

Reading the other CL, please check config_.rtp.keepalive.payload_type here too
(instead? does payload_size() matter here?), if this is signalled from the other
side and not just a send-payload-type attribute.

[sprang_webrtc, 2017/07/11 08:33:06]

This is based on https://tools.ietf.org/html/rfc6263#section-4.6
The intent is that the packets should have an explicitly unknown payload type.
RFC3550 states "a receiver MUST ignore packets with payload types that it does
not understand".
The packets should also have zero payload length, since we have a special case
for such keep-alive packets in the receiver; if the payload length is non-zero,
we will log a warning.

So we can't check the payload type on the receiver. (This method is receive side
only, right?)
I'll add a comment.

+      parsed_packet && parsed_packet->payload_size() == 0;
+
+  RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO ||
+             is_keep_alive_packet);
 
   if (!parsed_packet)
     return DELIVERY_PACKET_ERROR;
 
+  ReadLockScoped read_lock(*receive_crit_);
   auto it = receive_rtp_config_.find(parsed_packet->Ssrc());
   if (it == receive_rtp_config_.end()) {
     LOG(LS_ERROR) << "receive_rtp_config_ lookup failed for ssrc "