Voice and Video Tests
A variety of voice and video tests are supported using the latest codecs.
Voice
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Voice from->to |
| Description |
Test voice quality from E1 to E2, with G711, G729, AMR codecs.
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| Purpose |
Voice Media stream is generated in one direction and key transport metrics are validated. At the same time a Mean Opinion Score is computed to validate the transport quality.
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| Available Options |
Voice codec: G711, G729, AMR
DSCP setting.
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| Metrics |
- One way delay (ms)
- jitter (ms)
- jitter max (ms)
- packet loss (%)
- voice MOS score on path
- packet loss burst
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| Advanced Information |
This test will generate unidirectional Voice traffic with selected codec. |
| Results interpretation |
The voice pair generates RTP stream according to selected codec. The content of the voice traffic itself is not actual voice. This test qualifies network transport capability for realistic voice traffic. |
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Potential
Errors
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If test result pair cannot be collected:
Check that your E2 endpoint can open 10115 port toward E1 for management (this is needed for stream pairs).
Check that synchronization port between pairs is opened (management UDP 10115).
Check that port in use is in range (see "preferences/Test Controller/Destination Port Conf Option" and "preferences/Test Controller/First Destination Port" so that correct destination port for test traffic is used.
If results cannot be collected (some pairs failures complete error):
Check that your E2 endpoint can open 10115 port toward E1 for management (this is needed for stream pairs).
Check that port for traffic is opened (no firewall).
Check that synchronization port between pairs is opened (management UDP 10115).
Check that port in use is in range (see "preferences/Test Controller/Destination Port Conf Option" and "preferences/Test Controller/First Destination Port" so that correct destination port for test traffic is used.
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Voice bidirectional |
| Description |
Test voice quality from E1 to E2 AND E2 to E1, with G711, G729, AMR codecs.
Metrics are gathered by direction.
This reproduces a realistic voice media call.
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| Purpose |
See Voice from-> to.
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| Available Options |
| Metrics |
| Advanced Information |
| Results interpretation |
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Potential
Errors
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Voice n pairs unidirectional |
| Description |
Test voice quality from E1 to E2, with G711, G729, AMR codecs, with N calls simultaneously.
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| Purpose |
Voice Media stream is generated in one direction and key transport metrics are validated. At the same time a Mean Opinion Score is computed to validate the transport quality.
This test is performed for N calls.
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| Available Options |
Voice codec: G711, G729, AMR
DSCP setting
, Number of calls. |
| Metrics |
Aggregated over the N pairs:
- One way delay (ms)
- jitter (ms)
- jitter max (ms)
- packet loss (%)
- voice MOS score on path
- packet loss burst
Total Voice traffic throughput (layer 4)
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| Advanced Information |
This test will generate N pairs.
Test results are aggregated and there are no detailed results. |
| Results interpretation |
See Voice from->to. |
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Potential
Errors
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If the number of pairs selected is greater than available licenses the test can be stuck in queue. Note the number of pairs is N (one pair per voice direction).
The maximum number of simultaneous pairs (N value) each individual test can perform is also controlled from an advanced setting. Only Administrator can modify this setting:
In Main Section, Max Concurrent Tests in GUI
See voice from->to for other potential issues.
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Voice n pairs bidirectional |
| Description |
Test voice quality from E1 to E2 AND E2 to E1, with G711, G729, AMR codecs.
Metrics are gathered by direction.
This reproduces a realistic voice media call.
This test does N calls.
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| Purpose |
See Voice N pairs unidirectional.
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| Available Options |
| Metrics |
| Advanced Information |
| Results interpretation |
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Potential
Errors
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Video Stream
For public endpoint to private endpoint tests, complete the following steps:
- Enable a range of source and destination ports for traffic in Preferences > Traffic Management.
- Open the ports in the firewall for the private endpoint.
If a CGN gateway is present for the private endpoint, it may not be possible to open the necessary ports which will prevent Video traffic. This will impact the following test types: Netflix, Youtube, Adaptive Video, Adaptive video with quality, Flash RTMP, and Multicast Video over RTMP.
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Video Stream |
| Description |
Define a video stream from E1 to E2 with defined bit rate and MPEG2 or customizable codec. The client is the “From” endpoint and the server is the “To” endpoint. |
| Purpose |
Generate a unidirectional video flow from E1 to E2. This can simulate video based on RTP/UDP flows going through the network.
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| Available Options |
- Generated bitrate
- DSCP setting
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| Metrics |
- One way delay (ms)
- jitter (ms)
- jitter max (ms)
- packet loss (%)
- max loss burst
- video MDI scores (Media Delivery Index, Delay Factor)
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| Advanced Information |
This test allows generating video rtp streams - bitrate will define corresponding codec (HD or SD). |
| Results interpretation |
This test generate Video stream on the selected path.
It can be optimized with setting up send low jitter in the preferences to be 1 if the endpoints support it.
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Potential
Errors
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If test result pair cannot be collected:
Check that your E2 endpoint can open 10115 port toward E1 for management (this is needed for stream pairs).
Check that synchronization port between pairs is opened (management UDP 10115).
Check that port in use is in range (see "preferences/Test Controller/Destination Port Conf Option" and "preferences/Test Controller/First Destination Port" so that correct destination port for test traffic is used.
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Multicast Video over RTP |
| Description |
Multicast Flow. Emulates the standard streaming video codec. Measures video metrics. The client is the “From” endpoint and the server is the “To” endpoint.
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| Purpose |
Generate a multicast video flows from E1 to (E2…..EN). Where N the number of endpoints supported in the Mesh. This can simulate video based on RTP/UDP flows going through the network.
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| Available Options |
Test duration ,
Test bit rate (Mbps),
DSCP Setting,
Multicast Address: Port Video from->to.
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| Metrics |
- Datagrams Out of Order
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Delay (ms)
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Jitter (ms)
-
Jitter Max (ms)
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Loss
- Max loss burst
- Source and destination ports
- Throughput Video (Mbps)
- Total Bytes Lost
Video
- MDI Delay Factor (Media Delivery Index, Factor)
- Video MDI Media Loss Rate (Media Delivery Index, Loss Rate)
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| Advanced Information |
This test allows generating video rtp streams - bitrate will define corresponding codec (HD or SD).
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| Results interpretation |
This test generate Video stream on the selected path. It can be optimized with setting up send low jitter in the preferences to be 1 if the endpoints support it.
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Potential
Errors
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If test result pair cannot be collected:
- Ensure that your E2 endpoint can open 10115 port toward E1 for management (this is needed for stream pairs).
- Ensure that synchronization port between pairs is opened (management UDP 10115).
- Ensure that the port in use is in range. See "preferences/Test Controller/Destination Port Conf Option" and "preferences/Test Controller/First Destination Port" so that correct destination port for test traffic is used.
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Adaptive video |
| Description |
Emulates Adaptive Video Streaming over HTTP. The client is the “From” endpoint and the server is the “To” endpoint. |
| Purpose |
In the case of this video streaming technique, the video is encoded on a server at several bitrates and divided into video segments of 2 to 10 seconds.
The client first learns the available bitrates by downloading a video manifest, then starts requesting video segments from the server at a specific bitrate. The client starts requesting video segments at a medium bitrate, but this can be changed by altering the Starting Bitrate parameter of any Adaptive Video over HTTP flows.
Depending on the network bandwidth and the local speed of video decoding and rendering, the client can decide to increase (upshift) or decrease (downshift) the requested bitrate for future video segments as compared to that requested for the previous video segments.
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| Available Options |
Test duration ,
Test bit rate list,
DSCP Setting,
Number of Parallel Streams. |
| Metrics |
- Avg Video Playback Rate (bps)
- Avg Video Pre-buffering duration (ms)
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Jitter Buffer Full Count
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Jitter Buffer Full Duration (ms)
- Throughput from->to (Mbps)
- Video Playback Downshifts
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Video Playback Upshifts
- Video Quality Segments - Very High
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Video Quality Segments - High
- Video Quality Segments – Medium
- Video Quality Segments - Low
- Video Quality Segments - Very Low
- Video Stopped Count
- Video Stopped Duration (ms)
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| Advanced Information |
Upshifting provides better video quality, but at a greater risk of playback interruptions, while downshifting increases playback smoothness at the expense of video quality.
Through its adapting capability, this video streaming technique provides the best trade-off between video quality and smooth video streaming.
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| Results interpretation |
Where no jitter buffer is used (there are no video download limitations), the video is downloaded at the maximum speed allowed by network conditions. This means that there are no pauses between GET requests.
One in which the jitter buffer is used and as such, the download of the video segments is paused when the buffer becomes full. On average, the throughput will not exceed the maximum video bitrate.
Video Playback Downshift is a decrease in the requested bit rate for future video segments as compared to that requested for the previous video segments.
Video Playback Upshift is an increase in the requested bit rate for future video segments as compared to that requested for the previous video segments.
Video quality segments indicate the distribution of video playback quality
Video stopped count and duration indicate the video playback interruptions and its duration.
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Potential
Errors
|
If test result pair cannot be collected:
- Ensure that your E2 endpoint can open 10115 port toward E1 for management (this is needed for stream pairs).
- Ensure that synchronization port between pairs is opened (management UDP 10115).
- Ensure that port in use is in range. See "preferences/Test Controller/Destination Port Conf Option" and "preferences/Test Controller/First Destination Port" so that correct destination port for test traffic is used.
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Adaptive video with network quality |
| Description |
Emulates Adaptive Video Streaming over HTTP and measure network quality with a separate stream. The client is the “From” endpoint and the server is the “To” endpoint. |
| Purpose |
In the case of this video streaming technique, the video is encoded on a server at several bitrates and divided into video segments of 2 to 10 seconds. The client first learns the available bitrates by downloading a video manifest, then starts requesting video segments from the server at a specific bitrate.
The client starts requesting video segments at a medium bitrate, but this can be changed by altering the Starting Bitrate parameter of any Adaptive Video over HTTP flows. Depending on the network bandwidth and the local speed of video decoding and rendering, the client can decide to increase (upshift) or decrease (downshift) the requested bitrate for future video segments as compared to that requested for the previous video segments.
Additionally this also tests the Network KPI along with video streaming on both directions : it adds a bidirectional very low footprint packet heartbeat to measure the network kpis (loss/delay/jitter) on the segment.
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| Available Options |
Test duration , Test bit rate list, DSCP Setting, Number of Parallel Streams. |
| Metrics |
- Avg Video Playback Rate (bps)
- Avg Video Pre-buffering duration (ms)
-
Jitter Buffer Full Count
-
Jitter Buffer Full Duration (ms)
- Throughput from->to (Mbps)
- Video Playback Downshifts
-
Video Playback Upshifts
- Video Stopped Count
- Video Stopped Duration (ms)
Network KPI Metric:
- Delay (Ms):
- Jitter (Ms):
- Jitter Max (Ms):
- Loss:
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| Advanced Information |
Upshifting provides better video quality, but at a greater risk of playback interruptions, while downshifting increases playback smoothness at the expense of video quality.
Through its adapting capability, this video streaming technique provides the best trade-off between video quality and smooth video streaming.
|
| Results interpretation |
This test is realistic and since jitter buffer is used and as such, the download of the video segments is paused when the buffer becomes full. On average, the throughput will not exceed the maximum video bitrate.
Video Playback Downshift is a decrease in the requested bit rate for future video segments as compared to that requested for the previous video segments. Video Playback Upshift is an increase in the requested bit rate for future video segments as compared to that requested for the previous video segments.
Video quality segments indicate the distribution of video playback quality Video stopped count and duration indicate the video playback interruptions and its duration.
Video Stopped Count Video Stopped Duration will tell for any user experience having video stalling during the watch.
The number of Downshift and Upshifts indicate how often the adaptive rate needed to change: lots of changes indicate an instable network condition.
The Avg Playback rate will indicate the overall video quality experience: high rate will mean the high quality segments could be transported, lower rates mean the video had to adapt to lower quality segments.
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Potential
Errors
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This test can only work from public->public, private->private and private->public. It is NOT supported for public->private direction.
If test result pair cannot be collected:
- Ensure that port in use is in range. See "preferences/Test Controller/Destination Port Conf Option" and "preferences/Test Controller/First Destination Port" so that correct destination port for test traffic is used.
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Flash RTMP |
| Description |
Emulates Adaptive Video Streaming With default RTMP bit rates sizes. “The client is the “From” endpoint and the server is the “To” endpoint”. |
| Purpose |
In the case of this video streaming technique, the video is encoded on a server at several bitrates and divided into video segments of 2 to 10 seconds.
The client first learns the available bitrates by downloading a video manifest, then starts requesting video segments from the server at a specific bitrate.
The client starts requesting video segments at a medium bitrate, but this can be changed by altering the Starting Bitrate parameter of any Adaptive Video over HTTP flows.
Depending on the network bandwidth and the local speed of video decoding and rendering, the client can decide to increase (upshift) or decrease (downshift) the requested bitrate for future video segments as compared to that requested for the previous video segments.
Additionally this also tests the Network KPI along with video streaming.
|
| Available Options |
Test duration , Test bit rate list, DSCP Setting, Number of Parallel Streams. |
| Metrics |
- Avg Video Playback Rate (bps)
- Avg Video Pre-buffering duration (ms)
-
Jitter Buffer Full Count
-
Jitter Buffer Full Duration (ms)
- Throughput from->to (Mbps)
- Video Playback Downshifts
-
Video Playback Upshifts
- Video Stopped Count
- Video Stopped Duration (ms)
|
| Advanced Information |
Upshifting provides better video quality, but at a greater risk of playback interruptions, while downshifting increases playback smoothness at the expense of video quality.
Through its adapting capability, this video streaming technique provides the best trade-off between video quality and smooth video streaming.
|
| Results interpretation |
The download of the video segments is paused when the buffer becomes full. On average, the throughput will not exceed the maximum video bitrate.
Video Playback Downshift is a decrease in the requested bit rate for future video segments as compared to that requested for the previous video segments. Video Playback Upshift is an increase in the requested bit rate for future video segments as compared to that requested for the previous video segments.
Video quality segments indicate the distribution of video playback quality Video stopped count and duration indicate the video playback interruptions and its duration.
Video Stopped Count Video Stopped Duration will tell for any user experience having video stalling during the watch.
The number of Downshift and Upshifts indicate how often the adaptive rate needed to change: lots of changes indicate an instable network condition.
The Avg Playback rate will indicate the overall video quality experience: high rate will mean the high quality segments could be transported, lower rates mean the video had to adapt to lower quality segments.
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Potential
Errors
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This test can only work from public->public, private->private and private->public. It is NOT supported for public->private direction
If test result pair cannot be collected:
- Ensure that port in use is in range. See "preferences/Test Controller/Destination Port Conf Option" and "preferences/Test Controller/First Destination Port" so that correct destination port for test traffic is used.
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Netflix |
| Description |
Emulates Netflix Adaptive Video. The client is the “From” endpoint and the server is the “To” endpoint. |
| Purpose |
In the case of this video streaming technique, the video is encoded on a server at several bitrates and divided into video segments of 2 to 10 seconds.
The client first learns the available bitrates by downloading a video manifest, then starts requesting video segments from the server at a specific bitrate.
The client starts requesting video segments at a medium bitrate, but this can be changed by altering the Starting Bitrate parameter of any Adaptive Video over HTTP flows.
Depending on the network bandwidth and the local speed of video decoding and rendering, the client can decide to increase (upshift) or decrease (downshift) the requested bitrate for future video segments as compared to that requested for the previous video segments. Additionally this also tests the Network KPI along with video streaming.
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| Available Options |
Test duration , Video Rates Video Stream, Audio Rates Audio Stream, quality for video (HD or SD), DSCP Setting, Number of Parallel Streams. |
| Metrics |
- Avg Video Playback Rate (bps)
- Avg Video Pre-buffering duration (ms)
-
Jitter Buffer Full Count
-
Jitter Buffer Full Duration (ms)
- Throughput from->to (Mbps)
- Video Playback Downshifts
-
Video Playback Upshifts
- Video Stopped Count
- Video Stopped Duration (ms)
|
| Advanced Information |
Upshifting provides better video quality, but at a greater risk of playback interruptions, while downshifting increases playback smoothness at the expense of video quality.
Through its adapting capability, this video streaming technique provides the best trade-off between video quality and smooth video streaming.
|
| Results interpretation |
Where no jitter buffer is used (there are no video download limitations), the video is downloaded at the maximum speed allowed by network conditions. This means that there are no pauses between GET requests.
One in which the jitter buffer is used and as such, the download of the video segments is paused when the buffer becomes full. On average, the throughput will not exceed the maximum video bitrate.
Video Playback Downshift is a decrease in the requested bit rate for future video segments as compared to that requested for the previous video segments.
Video Playback Upshift is an increase in the requested bit rate for future video segments as compared to that requested for the previous video segments.
Video quality segments indicate the distribution of video playback quality Video stopped count and duration indicate the video playback interruptions and its duration.
Video Stopped Count Video Stopped Duration will tell for any user experience having video stalling during the watch.
The number of Downshift and Upshifts indicate how often the adaptive rate needed to change: lots of changes indicate an instable network condition.
The Avg Playback rate will indicate the overall video quality experience: high rate will mean the high quality segments could be transported, lower rates mean the video had to adapt to lower quality segments.
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Potential
Errors
|
This test can only work from public->public, private->private and private->public. It is NOT supported for public->private direction.
If test result pair cannot be collected:
- Ensure that port in use is in range. See "preferences/Test Controller/Destination Port Conf Option" and "preferences/Test Controller/First Destination Port" so that correct destination port for test traffic is used.
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Youtube |
| Description |
Emulates Youtube Adaptive Video. The client is the “From” endpoint and the server is the “To” endpoint. |
| Purpose |
In the case of this video streaming technique, the video is encoded on a server at several bitrates and divided into video segments of 2 to 10 seconds. The client first learns the available bitrates by downloading a video manifest, then starts requesting video segments from the server at a specific bitrate.
The client starts requesting video segments at a medium bitrate, but this can be changed by altering the Starting Bitrate parameter of any Adaptive Video over HTTP flows.
Depending on the network bandwidth and the local speed of video decoding and rendering, the client can decide to increase (upshift) or decrease (downshift) the requested bitrate for future video segments as compared to that requested for the previous video segments. Additionally this also tests the Network KPI along with video streaming.
|
| Available Options |
Test duration, Video Rates Video Stream, Test bit rate list, DSCP Setting, Number of Parallel Streams. |
| Metrics |
- Avg Video Playback Rate (bps)
- Avg Video Pre-buffering duration (ms)
-
Jitter Buffer Full Count
-
Jitter Buffer Full Duration (ms)
- Throughput from->to (Mbps)
- Video Playback Downshifts
-
Video Playback Upshifts
- Video Stopped Count
- Video Stopped Duration (ms)
|
| Advanced Information |
Upshifting provides better video quality, but at a greater risk of playback interruptions, while downshifting increases playback smoothness at the expense of video quality.
Through its adapting capability, this video streaming technique provides the best trade-off between video quality and smooth video streaming.
|
| Results interpretation |
Where no jitter buffer is used (there are no video download limitations), the video is downloaded at the maximum speed allowed by network conditions. This means that there are no pauses between GET requests. One in which the jitter buffer is used and as such, the download of the video segments is paused when the buffer becomes full. On average, the throughput will not exceed the maximum video bitrate.
Video Playback Downshift is a decrease in the requested bit rate for future video segments as compared to that requested for the previous video segments. Video Playback Upshift is an increase in the requested bit rate for future video segments as compared to that requested for the previous video segments.
Video quality segments indicate the distribution of video playback quality Video stopped count and duration indicate the video playback interruptions and its duration.
Video Stopped Count Video Stopped Duration will tell for any user experience having video stalling during the watch.
The number of Downshift and Upshifts indicate how often the adaptive rate needed to change: lots of changes indicate an instable network condition.
The Avg Playback rate will indicate the overall video quality experience: high rate will mean the high quality segments could be transported, lower rates mean the video had to adapt to lower quality segments.
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|
Potential
Errors
|
This test can only work from public->public, private->private and private->public. It is NOT supported for public->private direction.
If test result pair cannot be collected:
- Ensure that port in use is in range. See "preferences/Test Controller/Destination Port Conf Option" and "preferences/Test Controller/First Destination Port" so that correct destination port for test traffic is used.
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