Skip to content

Commit 092f0d4

Browse files
sloanstersloanster
authored
Update media-quality-sdk.md
1 parent 05bea10 commit 092f0d4

File tree

1 file changed

+11
-10
lines changed

1 file changed

+11
-10
lines changed

‎articles/communication-services/concepts/voice-video-calling/media-quality-sdk.md

Lines changed: 11 additions & 10 deletions
Original file line numberDiff line numberDiff line change
@@ -7,20 +7,21 @@ ms.author: micahvivion
77
manager: nmurav
88

99
services: azure-communication-services
10-
ms.date: 11/30/2021
10+
ms.date: 11/30/2022
1111
ms.topic: conceptual
1212
ms.service: azure-communication-services
1313
ms.subservice: calling
1414
---
1515

1616
# Media quality statistics
17-
Do help understand media quality in VoIP and Video calls using Azure Communication Services, we have a feature called "Media quality statistics" that you can use to examine the low-level audio, video, and screen-sharing quality metrics for incoming and outgoing call metrics.
17+
To help understand media quality in VoIP and Video calls using Azure Communication Services, we have a feature called "Media quality statistics" that you can use to examine the low-level audio, video, and screen-sharing quality metrics for incoming and outgoing call metrics.
1818

1919
## Media quality statistics for ongoing call
20-
> **NOTE**
20+
> [!NOTE]
2121
> This API is provided as a preview ('beta') for developers and may change based on feedback that we receive. Do not use this API in a production environment.
22-
>
23-
> There is an API breaking change on MediaStats in the SDK beginning since version 1.8.0-beta.1
22+
23+
> [!IMPORTANT]
24+
> There is also an API breaking change on MediaStats in the SDK beginning since version 1.8.0-beta.1
2425
2526
Media quality statistics is an extended feature of the core `Call` API. You first need to obtain the MediaStats feature API object:
2627

@@ -145,7 +146,7 @@ The bandwidth metrics have changes to `availableBitrate` in Audio Send / Video S
145146
| packetsPerSecond | packet rate (packets/sec) | |
146147
| packetsLostPerSecond | packet loss rate (packets/sec) | Lower is better. |
147148
| pairRttInMs | round-trip time (milliseconds) | Lower is better. A round trip time of 200 ms or less is recommended. |
148-
| jitterBufferInMs | jitter buffer (milliseconds) | Lower is better. The jitter buffer is used for smooth playout. This value is the how long the packets of the frame stay in the jitter buffer. |
149+
| jitterBufferInMs | jitter buffer (milliseconds) | Lower is better. The jitter buffer is used for smooth playout. This value is how long the packets of the frame stay in the jitter buffer. |
149150
| streamId | stream id | The streamId value corresponds to id in VideoStreamCommon. It can be used to match the sender. |
150151
| frameRateOutput | frame rate output (frames/sec) | |
151152
| frameRateDecoded | frame rate correctly decoded for the RTP stream (frames/sec) | |
@@ -165,7 +166,7 @@ Currently stats fields are the same as *Video Send metrics*
165166
### ScreenShare Receive metrics
166167
Currently stats fields are the same as *Video Receive metrics*
167168

168-
### Using Media Quality Statics on SDK Version `< 1.8.0` or older
169+
### Using Media Quality Statistics on SDK Version `< 1.8.0` or older
169170
If you are using an ACS SDK version of 1.8.0 or older, please see below for documentation on how to use this functionality.
170171

171172
As a developer you can invoke the `startCollector` method of `mediaStatsApi` with optional `mediaStatsSubscriptionOptions`.
@@ -204,14 +205,14 @@ mediaStatsFeature.disposeAllCollectors();
204205
| audioSendBitrate | Sent bitrate | Send bitrate of audio (bits per second) | General values are in the 24 kbps range (36-128 kbps typical) |
205206
| audioRecvBitrate | Received audio bitrate | Received bitrate of audio received (bits per second) | |
206207
| audioSendPackets | Sent packets | The number of audio packets sent in last second (packets per second) | |
207-
| audioRecvJitterBufferMs | Jitter buffer delay | The jitter buffer is used for smooth playout. This value is the how long the packets of the samples stay in the jitter buffer. (in milliseconds (ms)) | Lower is better. |
208+
| audioRecvJitterBufferMs | Jitter buffer delay | The jitter buffer is used for smooth playout. This value is how long the packets of the samples stay in the jitter buffer. (in milliseconds (ms)) | Lower is better. |
208209
| audioRecvPacketsLost | Received packet loss | The number of audio packets that were to be received but were lost. Results are packets per second (over the last second). | Lower is better. |
209210
| audioSendPacketsLost | Sent packet loss | The number of audio packets sent that were lost (not received) in the last second. Results are packets per second (over the last second). | Lower is better. |
210211
| audioRecvPackets | Received packets | The number of audio packets received in the last second. Results are packets per second (over the last second). | Information only. |
211212
| audioSendCodecName | Sent codec | Audio codec used. | Information only. |
212213
| audioSendRtt | Send Round-Trip Time | Round trip time between your system and Azure Communication Services server. Results are in milliseconds (ms). | A round trip time of 200 ms or less is recommended. |
213214
| audioSendPairRtt | Send Pair Round-Trip Time | Round trip time for entire transport. Results are in milliseconds (ms). | A round trip time of 200 ms or less is recommended. |
214-
| audioRecvPairRtt | Receive Pair Round-Trip Time | Round trip time for entire transport Results are in milliseconds (ms). | A round trip time of 200 ms or less is recommended. |
215+
| audioRecvPairRtt | Receive Pair Round-Trip Time | Round trip time for entire transport. Results are in milliseconds (ms). | A round trip time of 200 ms or less is recommended. |
215216
| audioSendAudioInputLevel | Input level for microphone | Sent audio playout level. If source data is between 0-1, media stack multiplies it with 0xFFFF. Depends on microphone. Used to confirm if microphone is silent (no incoming energy). | Microphone input level. |
216217
| audioRecvAudioOutputLevel | Speaker output level. | Received audio playout level. If source data is between 0-1, media stack multiplies it with 0xFFFF. | Speaker output level. |
217218

@@ -225,7 +226,7 @@ mediaStatsFeature.disposeAllCollectors();
225226
| videoSendBitrate | Sent bitrate | Amount of video bitrate being sent. Results are bps (bits per second) | |
226227
| videoSendPackets | Sent packets | The number of video packets sent. Results are packets per second (over the last second). | Information only |
227228
| VideoSendCodecName | Sent codec | Video codec used for encoding video | VP8 (1:1 calls) and H264 |
228-
| videoRecvJitterBufferMs | Received Jitter | The jitter buffer is used for smooth playout. This value is the how long the packets of the frame stay in the jitter buffer. (in milliseconds (ms)) | Lower is better. |
229+
| videoRecvJitterBufferMs | Received Jitter | The jitter buffer is used for smooth playout. This value is how long the packets of the frame stay in the jitter buffer. (in milliseconds (ms)) | Lower is better. |
229230
| videoSendRtt | Send Round-Trip Time | Response time between your system and Azure Communication Services server. Lower is better | A round trip time of 200 ms or less is recommended. |
230231
| videoSendPairRtt | Send Pair Round-Trip Time | Response time between your system and Azure Communication Services server. Results are in milliseconds (ms). | A round trip time of 200 ms or less is recommended. |
231232
| videoRecvPairRtt | Receive Pair Round-Trip Time | Round trip time for entire transport. Results are in milliseconds (ms). | A round trip time of 200 ms or less is recommended. |

0 commit comments

Comments
 (0)