Comments (4)
The intent of the spec (I think this was hashed out in the Washington interim, a few years back) is that in the normal case (RTCP-mux), there would be only one transport object; in the exceptional case of unbundled RTCP, there would be two (has to be, given that selectedCandidateId will be different), but the only way to find the RTCP one would be through the RTP one.
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That's what I'd assumed. In that case I think the summary just needs to be updated to make this clear. I'll make a PR.
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PR is #73
from webrtc-stats.
Fixed in #73
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Related Issues (20)
- Conformance criteria incompatible with standards-track HOT 7
- RTX and FEC stats are incomplete. HOT 12
- Should we flatten the hierarchy? HOT 5
- Change playoutdelay to jitterbuffertarget HOT 2
- Add clarification that inbound-rtp.bytesReceived includes retransmissions. HOT 6
- RTCAudioSourceStats.audioLevel - what algorithm described in totalAudioEnergy? HOT 3
- Delete the obsolete section, "track" and "stream" stats - they don't exist anymore HOT 3
- Improve test coverage HOT 1
- add framesInput stats to rtcoutboundrtpstreamstats HOT 6
- Stats example uses confusing variable names
- Exposing audio interruption metrics to JavaScript
- Define a mechanism for setting the trigger duration for a video freeze
- Sender-side packetsReceived can be negative from garbage RRs HOT 1
- Change "Does not" to "must not"
- Make "not present" reference "map/exist"
- Is codec.sdpFmtpLine present when there is no fmtp line in the SDP? HOT 2
- Lifetime of outbound-rtp should start BEFORE first packet is sent HOT 3
- Should RTCRtptransceiver.stop() cause inbound-rtp stats to disappear?
- RTCCodecStats.clockRate - media sampling rate or the codec clock rate?
- RTCStats.timestamp - fingerprinting and since epoch
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