Call should allow pass through of keep-alive packets.
Don't force requirement of media type for those packets.
BUG=webrtc:7964
Review-Url: https://codereview.webrtc.org/2973323002
Cr-Commit-Position: refs/heads/master@{#18966}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 3966d5e..0bddde9 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -236,10 +236,10 @@
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 @@
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>();
@@ -1300,17 +1300,24 @@
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);
+ // We might get RTP keep-alive packets in accordance with RFC6263 section 4.6.
+ // These are empty (zero length payload) RTP packets with an unsignaled
+ // payload type.
+ const bool is_keep_alive_packet =
+ 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 "
