Receiving a live stream using srt-live-transmit

[AnWaKZ]

What parameters should be specified to receive a live stream? When I increase the latency parameter, packets are recovered, but video artifacts appear. Also, in the statistics I don’t understand the channel bandwidth parameter — it is always lower than the incoming stream bitrate.

[AnWaKZ]

======= SRT STATS: sid=24592161
PACKETS SENT: 0 RECEIVED: 344669
LOST PKT SENT: 0 RECEIVED: 6
REXMIT SENT: 0 RECEIVED: 6
DROP PKT SENT: 0 RECEIVED: 4
FILTER EXTRA TX: 0 RX: 0
FILTER RX SUPPL: 0 RX LOSS: 0
RATE SENDING: 0 RECEIVING: 3.28983
BELATED RECEIVED: 9 AVG TIME: 6.64083e+15
REORDER DISTANCE: 1
WINDOW FLOW: 8192 CONGESTION: 8192 FLIGHT: 0
LINK RTT: 2.667ms BANDWIDTH: 3.144Mb/s
BUFFERLEFT: SND: 12288000 RCV: 12283500

[ethouris]

If you don't have the dropped packets reported in the stats on the receiving side, artifacts in the video cannot be related to SRT.

In that above shot I can see 4 packets dropped, so your recovery still isn't perfect.

The bandwidth measurement in the live mode is a bit tricky. This was initially implemented for the file transmission mode and uses an opportunity to ignore the bandwidth limitation for every 16th packet. But to see the difference you would have to have any bandwidth limitation. In the current implementation the formulas used cannot carry over unlimited bandwidth, hence there is a 1Gbps limit for live mode (although it can be still set higher through the SRTO_MAXBW socket option). The bandwidth measurement can show a higher bandwidth only if it exceeds this limitation - simply because the 16th packet, even if sent intentionally without a delay, will still have a delay towards the 15th resulting from the live stream nature.

[AnWaKZ]

If the latency is large, can CC errors appear in the UDP?