WebRTC Monitoring and Alerting
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1 11/25/2013 1
2 WebRTC Monitoring and Alerting David A. Bryan Assistant Professor, Computer Science St. Edward s University 2 11/25/2013
3 Speakers Chris Cavigioli Strategy Planning Intel MCG Varun Singh Co-founder callstats.io 3 11/25/2013
4 Latency is critical for conversational uses cases. Long latency results in double-talking, making conversation difficult. Chris Cavigioli Strategy Planning Intel Mobile and Communications Group (MCG) IMPACT OF TRANSCODING LATENCY TO USER EXPERIENCE CONNECTING WEBRTC TO 3GPP/IMS 4 11/25/2013
5 VoIP Latency Source: Understanding Latency in IP Telephony, Alan Percy, Brooktrout, Embedded Computing Design, July 2000, /25/2013
6 VoIP Latency Measurements Source (Table 9, Fig.3): Mouth-To-Ear Latency in Popular VoIP Clients, Agastya, Mechanic, and Kothari, Columbia University, July 2009, Source: (Fig.1) Understanding Latency in IP Telephony, Alan Percy, Brooktrout, Embedded Computing Design, July 2000, /25/2013
7 Latency Limits in 3GPP Source (Table 1): 3GPP TS (Rel.11), Nov 2012, Source (Fig.1): Understanding Latency in IP Telephony, Alan Percy, Brooktrout, Embedded Computing Design, July 2000, /25/2013
8 IMS Network Audio/Video AV Transcoding Opus Encode Opus Decode Opus Dec Opus Enc AMR-WR Enc AMR-WB Dec AMR-WB Dec AMR-WB Enc VP8 Encode VP8 Decode WebRTC Client VP8 Dec H.264 Enc VP8 Enc H.264 Dec WebRTC2IMS Gateway UE #1 H.264 Dec H.264 Enc IMS Client UE #2 UE #1 must support 12 codecs instead of 4 6 concurrent video codecs in parallel and 6 concurrent audio codecs simultaneously 8 11/25/2013
9 Codec Latencies add Extra RTD Implementation-agnostic numbers based on worst case to maintain frame rate Function H.264 assumes supports of up to Codec Delay (ms) 1080p60, then each frame has up to 16 ms max to complete. VP8 = 1080p30, thus 33 ms 320x240p30 Level p30 Level p30 Level 4.1 A = H.264 decode 16/27 (=0.6 ms) 7.1 = 16/ max B = VP8 encode = 33/ max C = VP8 decode max D = H.264 encode max H.264 decode + VP8 encode [A+B] =1.8 ms =22 ms =49 ms VP8 decode + H.264 encode [C+D] =1.8 ms =22 ms =49 ms Extra RTD, based purely on V codecs =3.6 ms =44 ms =98 ms 1 frame 30 fps 33 ms 33 ms 33 ms 1 frame 60 fps 16 ms 16 ms 16 ms Additional round-trip delay (RTD) incurred purely by transcoding 2x (decode + re-encode) Function Delay A = AMR-WB decode 20 ms B = Opus encode 20 C = Opus decode 20 D = AMR-WB encode 20 AMR-WB decode + Opus encode [A+B] =40 ms Opus decode + AMR-WB encode [C+D] =40 ms Extra RTD, based purely on A codecs =80 ms 9 11/25/2013
10 Call for Action Latency is #1 concern for best conversational user experience 3GPP, operators target ms for VoLTE (and IR.94 video) Transcoding adds Up to 49 ms video or 40 ms audio in parallel for codec delays Extra 49 or 40 ms frame slips to re-align audio-video mismatches Additional 30 ms delay for jitter buffer in network-based transcoder Users perceive Round Trip Delay (RTD) which doubles end-to-end delay To preserve end-user experience, AVOID transcoding altogether End points MUST be able to negotiate and pick identical codecs 11/25/
11 /25/2013
12 Varun Singh Co-founder, CEO callstats.io PERFORMANCE MONITORING OF MEDIA FLOWS IN WEBRTC 12 11/25/2013
13 Recap: WebRTC Signaling Protocol (e.g., SIP, Jingle, ) WebRTC Server APP WebRTC API APP WebRTC API Browser Internals (WebRTC Stack) 13 PeerConnection SRTP/DTLS/UDP Data/SCTP/DTLS/UDP Browser Internals (WebRTC Stack)
14 Latency kills 14
15 Delay Variation [ms] Berlin-Helsinki time [s] audio video 15 Calls between TU Berlin- Aalto Univ. Helsinki
16 Delay Variation [ms] but on a bad day time [s] and this is just audio
17 Monitoring Annoyances Call setup time, call failures, NAT traversal Transport quality Relayed or not Session throughput, delay and loss Per-stream media quality MoS, User feedback 17
18 Client Monitoring STATS API Monitoring Architecture WebRTC Server Monitoring Server APP APP WebRTC API Browser Internals (WebRTC Stack) PeerConnection HTTP, IPFIX, or NetFlow WebRTC API Browser Internals (WebRTC Stack) 18 TURN Gateway
19 Stats API Web app queries underlying RTP stack Per stream statistics (e.g., Audio and Video) In-bound and out-bound statistics Identifiers RTCP RR ICE candidates, Query at application defined intervals Typically, 1s
20 StatsAPI: Example var statcollector = setinterval(function () { if (pc && pc.getremotestreams() && active == true) { if (pc.getstats) { pc.getstats(onstatssuccess); } else { log('no stats function. Use Chrome > '); } } }, 1000); onstatssuccess = function (stats) { // parse the stats.result() // Audio and Video stats // Local and Remote stats } 20
21 RTCP Monitoring Needs support in the WebRTC Stack implement RTCP Extension Reports Send XRs to a performance monitoring server 21
22 Multiplexing helps TURN server WebRTC Gateway Detecting WebRTC flows observe 2 or more categories of packets STUN messages DTLS packets RTP packets Audio frames (ptime between 10-30ms) Video packets (at least 7 frames per second) Implemented in (Connection Monitor) 22
23 Receive UDP Packet DTLS Processing Packet Monitoring (1/2) H-> Parse first 8-bits of UDP Payload H < 2 19 < H < < H < 192 0b00 STUN Message 0b01 ChannelData Message (TURN) -> Parse Recursively SCTP DTLS Forward to RTP Processing Implemented in (Connection Monitor) 23
24 Parse PT Parse SSRC Parse PT SSRC #1 0 < PT < < PT < 128 RTP media FEC retx 191 < PT < 255 RTCP Receive RTP Packet Codec #1 SSRC #N same as above media FEC retx Codec #2 Codec #3 24
25 Tying it together Endpoints monitor their calls observe raw stats: loss, delay Build a QoE model Diagnosing by analysing transport G.1070 for video, P for audio Cloud monitoring across calls 25
26 Conclusions Tools for Performance monitoring WebRTC s StatsAPI Packet capture Bonus tip: Coupling getusermedia() constraints and stats APIs Bonus tip: better initial application settings 26
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