Stream Statistics Field Descriptions
The following tables list the available stream statistics fields.
General
Statistic | Description/Values |
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State | The current operating status of the stream, either Disconnected, Connecting, Connection established, or Connected. |
Mode | For UDP or RTP: Unicast or Multicast For HLS: HLS For SRT: Caller, Listener, or Rendezvous |
Uptime | The elapsed running time of the stream. |
Bitrate | The stream bitrate (in kbps). |
Received Packets (Source) Sent Packets (Destination) | Number of packets received/sent for that stream. |
Lost Packets | (RTP only) Number of packets lost from the RTP source stream.
Note
If RTP redundancy (SMPTE 2022.7) is enabled, this value only increments if every RTP stream loses the same packet. See the individual RTP stream statistics for their specific lost packet value.
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Used Bandwidth | Bandwidth used. |
Reconnections | Number of signal losses. |
User NICs | Number of network interfaces used. |
SRT
Statistic | Description/Values |
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Connections/Limit | Shows the number of SRT Caller clients connected to an SRT Listener stream and the connection limit defined in the SRT Listener destination. |
Path Redundancy | Shows the defined path redundancy mode for the route. |
Buffer | Decoder buffer in milliseconds. SRT decoder buffers are the received stream packets waiting to be decoded. This statistic shows the portion of the buffers up to the first missing packet. In other words, the remaining time to transmit the missing packet before it’s too late. The level of the buffer in absence of packet loss is just below the latency value. In presence of packets loss, it is between 0 and the latency value.
Tip
If the Buffer goes to 0 often, then there is most likely insufficient BW to support the desired bitrate. In this case, decrease your bitrate. If the Buffer occasionally goes to 0, then the SRT Latency should be increased.
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Latency | Maximum of the decoder and encoder configured Latency. For example:- Encoder Configured SRT Latency = 750 ms
- Decoder Configured SRT Latency = 20 ms
- The SRT Stats Latency (which is the current SRT connection applied Buffering Latency) = 750 (largest of the two).
At startup, handshake exchanges the value configured on both sides and the largest one is selected. The decoder default is set to the minimum (20ms), so the latency value can be completely controlled from the encoder side. |
RTT | Measured round-trip time. Round Trip Time (RTT) is the time it would take for a packet to travel from a specific source to a specific destination and back again. In SRT, this is measured as the time it takes for the destination device to send an acknowledgment (ACK) packet, and then receive a corresponding confirmation (ACKACK) packet. |
FEC/FEC+ARQ | (SRT Caller mode only) If either FEC mode is enabled: Active. |
FEC Layout | (SRT Caller mode only) FEC layout type: Even or Staircase. |
FEC Rows/Columns | (SRT Caller mode only) Number of FEC rows and columns. |
Other SRT statistics are available depending on whether the route is a source route or a destination route, as described below.
SRT Source
Statistic | Description/Values |
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Lost Rate | Rate at which the source route is receiving lost packets in bits/s. See the following Lost Packets description for more details. |
Lost Packets | Number of SRT packets reported missing on the UDP connection. For each "hole" detected in the packet sequence, a request to re-transmit the lost packet is sent to the sender. This lost packet may (or may not) be re-covered by the re-transmit request.
Note
This is the raw number of packets dropped by the network. Most are recovered by retransmission at the source and so do not necessarily result in any artifacts.
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Packet Loss Rate | SRT packet loss rate, expressed as a percentage of packets lost with respect to packets sent. The SRT on the sender side cannot deliver the packets within the defined latency time and dropped the packet. This occurs when the packets cannot be transmitted fast enough due to low latency, not enough bandwidth overhead, etc. |
Skipped Packets | Packets that have arrived too late, or that never arrive at all from the receiver. If the "time to play" for a packet has passed, and it has either not arrived or arrives after the content it is associated with has already played, that packet is reported as "skipped". Usually this results in some type of video artifact (a replayed frame or video blocking). This is the raw number of packets skipped and are not recoverable by the SRT protocol. - If this statistic increments slowly, the best thing to do is increase the SRT latency.
- If this statistic increments in large jumps, the best thing to do is lower your video bitrate or increase your overhead if you have available bandwidth.
Note
The skipped packets value from the receiver does not correlate to the dropped packets value from the sender, as they count different types of irrecoverable packets.
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Undecrypted Packets | The total number of packets that failed to be decrypted at the receiver side. |
Encryption | SRT encryption type: None, AES128, or AES256. |
Decryption | SRT decryption state: Active, Initializing, Inactive (no passphrase), Inactive (invalid passphrase). |
Network Interface | Name of network interface used. |
SRT Destination
Statistic | Description/Values |
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Retransmit Rate | Rate at which the destination route is resending lost packets in bits/s. |
Packet Loss Rate | SRT packet loss rate, expressed as a percentage of packets lost with respect to packets sent. The SRT on the sender side cannot deliver the packets within the defined latency time and dropped the packet. This occurs when the packets cannot be transmitted fast enough due to low latency, not enough bandwidth overhead, etc. |
Dropped Packets | Number of packets reported missing by the SRT sender, as the output queue has overflowed. The most likely cause for this is the system is overloaded and cannot process the data fast enough. This is the raw number of packets dropped and are not recoverable by the SRT protocol.
Note
The skipped packets value from the receiver does not correlate to the dropped packets value from the sender, as they count different types of irrecoverable packets.
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Max Bandwidth | Maximum bandwidth used by the source device for this SRT stream (i.e. the current total of audio/video bit rate plus ancillary data plus the SRT bandwidth overhead). |
Path Max Bandwidth | An estimate of the maximum path/link bandwidth as viewed from the destination. |
Encryption | SRT encryption type: None, AES128, or AES256. |
Network Interface | Name of network interface used. |
Pro-MPEG FEC
Note
PRO-MPEG FEC is available only on Haivision SRT Gateway.
Statistic | Description/Values |
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Lost Packets | Number of lost FEC packets. |
Recovered Packets | Number of recovered FEC packets. |
Unrecovered Packets | Number of unrecovered FEC packets. |
Reordered Packets | Number of reordered FEC packets. |
Level | The level of FEC protection:- A (Column only): uses the column FEC stream.
- B (Row and Column): uses both column and row FEC streams.
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Block Aligned | The type of FEC matrix scheme:- True: Sequential columns within a group start on the same row.
- False: Each column starts on the row below the row on which the previous column started.
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Columns/Rows | Number of columns and number of rows are the dimensions of the FEC matrix. |