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Call should allow pass through of keep-alive packets.

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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}
This commit is contained in:
sprang 2017-07-11 03:56:21 -07:00 committed by Commit Bot
parent 7fc3f15686
commit c1abde7e8e
1 changed files with 15 additions and 8 deletions
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23
webrtc/call/call.cc
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@ -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>();
@ -1300,17 +1300,24 @@ 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);
// 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 "