How can I get the "remote-outbound-rtp" using getStats()?

[bigbean17]

I'm trying to get the value of "remote-outbound-rtp" of a video stream. I found that "remote-outbound-rtp" only occur when it is the audio stream. I'm iterating the result of RTCPeerConnection.getStatus(null), not sure what went wrong. Any help would be appreciated.

[niyatim23]

Hi @bigbean17, the rtp-outbound stats can be found below:

amazon-kinesis-video-streams-webrtc-sdk-c/src/include/com/amazonaws/kinesis/video/webrtcclient/Stats.h

Lines 345 to 390 in 0ae9ec6

	typedef struct {
	RTCSentRtpStreamStats sent; //!< Comprises of information such as packetsSent and bytesSent
	BOOL voiceActivityFlag; //!< Only valid for audio. Whether the last RTP packet sent contained voice activity or not based on the presence
	//!< of the V bit in the extension header
	DOMString trackId; //!< ID representing current track attached to the sender of the stream
	DOMString mediaSourceId; //!< TODO ID representing the current media source
	DOMString senderId; //!< TODO The stats ID used to look up the RTCAudioSenderStats or RTCVideoSenderStats object sending this stream
	DOMString remoteId; //!< TODO ID to look up the remote RTCRemoteInboundRtpStreamStats object for the same SSRC
	DOMString rid; //!< TODO Exposes the rid encoding parameter of this RTP stream if it has been set, otherwise it is undefined
	DOMString encoderImplementation; //!< Identifies the encoder implementation used.
	UINT32 packetsDiscardedOnSend; //!< Total number of RTP packets for this SSRC that have been discarded due to socket errors
	UINT32 framesSent; //!< Only valid for video. Represents the total number of frames sent on this RTP stream
	UINT32 hugeFramesSent; //!< Only valid for video. Represents the total number of huge frames sent by this RTP stream
	//!< Huge frames have an encoded size at least 2.5 times the average size of the frames
	UINT32 framesEncoded; //!< Only valid for video. It represents the total number of frames successfully encoded for this RTP media stream
	UINT32 keyFramesEncoded; //!< Only valid for video. It represents the total number of key frames encoded successfully in the RTP Stream
	UINT32 framesDiscardedOnSend; //!< Total number of video frames that have been discarded for this SSRC due to socket errors
	UINT32 frameWidth; //!< Only valid for video. Represents the width of the last encoded frame
	UINT32 frameHeight; //!< Only valid for video. Represents the height of the last encoded frame
	UINT32 frameBitDepth; //!< Only valid for video. Represents the bit depth per pixel of the last encoded frame. Typical values: 24, 30, 36
	UINT32 nackCount; //!< Count the total number of Negative ACKnowledgement (NACK) packets received by this sender.
	UINT32 firCount; //!< Only valid for video. Count the total number of Full Intra Request (FIR) packets received by this sender
	UINT32 pliCount; //!< Only valid for video. Count the total number of Picture Loss Indication (PLI) packets received by this sender
	UINT32 sliCount; //!< Only valid for video. Count the total number of Slice Loss Indication (SLI) packets received by this sender
	UINT32 qualityLimitationResolutionChanges; //!< Only valid for video. The number of times that the resolution has changed because we are quality
	//!< limited
	INT32 fecPacketsSent; //!< TODO Total number of RTP FEC packets sent for this SSRC. Can also be incremented while sending FEC packets in band
	UINT64 lastPacketSentTimestamp; //!< The timestamp in milliseconds at which the last packet was sent for this SSRC
	UINT64 headerBytesSent; //!< Total number of RTP header and padding bytes sent for this SSRC
	UINT64 bytesDiscardedOnSend; //!< Total number of bytes for this SSRC that have been discarded due to socket errors
	UINT64 retransmittedPacketsSent; //!< The total number of packets that were retransmitted for this SSRC
	UINT64 retransmittedBytesSent; //!< The total number of PAYLOAD bytes retransmitted for this SSRC
	UINT64 targetBitrate; //!< Current target TIAS bitrate configured for this particular SSRC
	UINT64 totalEncodedBytesTarget; //!< Increased by the target frame size in bytes every time a frame has been encoded
	DOUBLE framesPerSecond; //!< Only valid for video. The number of encoded frames during the last second
	UINT64 qpSum; //!< TODO Only valid for video. The sum of the QP values of frames encoded by this sender. QP value depends on the codec
	UINT64 totalSamplesSent; //!< TODO Only valid for audio. The total number of samples that have been sent over this RTP stream
	UINT64 samplesEncodedWithSilk; //!< TODO Only valid for audio and when the audio codec is Opus. Represnets only SILK portion of codec
	UINT64 samplesEncodedWithCelt; //!< TODO Only valid for audio and when the audio codec is Opus. Represnets only CELT portion of codec
	UINT64 totalEncodeTime; //!< Total number of milliseconds that has been spent encoding the framesEncoded frames of the stream
	UINT64 totalPacketSendDelay; //!< Total time (seconds) packets have spent buffered locally before being transmitted onto the network
	UINT64 averageRtcpInterval; //!< The average RTCP interval between two consecutive compound RTCP packets
	QualityLimitationDurationsRecord qualityLimitationDurations; //!< Total time (seconds) spent in each reason state
	DscpPacketsSentRecord perDscpPacketsSent; //!< Total number of packets sent for this SSRC, per DSCP
	RTC_QUALITY_LIMITATION_REASON qualityLimitationReason; //!< Only valid for video.
	} RtcOutboundRtpStreamStats, *PRtcOutboundRtpStreamStats;

You can use the function below with the correct rtcMetrics type:

amazon-kinesis-video-streams-webrtc-sdk-c/src/source/Metrics/Metrics.c

Line 204 in 0ae9ec6

STATUS rtcPeerConnectionGetMetrics(PRtcPeerConnection pRtcPeerConnection, PRtcRtpTransceiver pRtcRtpTransceiver, PRtcStats pRtcMetrics)

An example of its usage is:

amazon-kinesis-video-streams-webrtc-sdk-c/samples/Common.c

Lines 135 to 143 in 0ae9ec6

	rtcMetrics.requestedTypeOfStats = RTC_STATS_TYPE_LOCAL_CANDIDATE;
	CHK_STATUS(rtcPeerConnectionGetMetrics(pSampleStreamingSession->pPeerConnection, NULL, &rtcMetrics));
	DLOGD("Local Candidate IP Address: %s", rtcMetrics.rtcStatsObject.localIceCandidateStats.address);
	DLOGD("Local Candidate type: %s", rtcMetrics.rtcStatsObject.localIceCandidateStats.candidateType);
	DLOGD("Local Candidate port: %d", rtcMetrics.rtcStatsObject.localIceCandidateStats.port);
	DLOGD("Local Candidate priority: %d", rtcMetrics.rtcStatsObject.localIceCandidateStats.priority);
	DLOGD("Local Candidate transport protocol: %s", rtcMetrics.rtcStatsObject.localIceCandidateStats.protocol);
	DLOGD("Local Candidate relay protocol: %s", rtcMetrics.rtcStatsObject.localIceCandidateStats.relayProtocol);
	DLOGD("Local Candidate Ice server source: %s", rtcMetrics.rtcStatsObject.localIceCandidateStats.url);

[bigbean17]

Hi @niyatim23, thanks for the solution, I saw the "remote-outbound-rtp" is empty, is that the reason why I cannot get the "remote-outbound-rtp" using RTCPeerConnection.getStatus(null) at frontend for video streaming? And I'm using the amazon-kinesis-video-streams-webrtc-sdk-js as the viewer at frontend.

[niyatim23]

You mean you want to see video-related stats on the viewer side (the browser)? And you want to understand if we send them from the SDK or not?

[bigbean17]

You mean you want to see video-related stats on the viewer side (the browser)?

Yes
I'm trying to display the information in "REMOTE_OUTBOUND_RTP" but "REMOTE_OUTBOUND_RTP" doesn't present in the result of getStats().

[disa6302]

REMOTE_OUTBOUND_RTP is unsupported at the moment. If you would like to view REMOTE_OUTBOUND_RTP, you need to follow the instructions @niyatim23 mentioned. Our samples have that example. getStats() is not a WebRTC C SDK API and you need to use rtcPeerConnectionGetMetrics and pass in the necessary parameters to get the stats.

[bigbean17]

Ok I will have a try, thanks~