I am implementing Call statistics over Javascrit CSDK.
I see that following parameters are not mentioned in JSCSDK documentation:
For AudioDetails;
Packetization, Local Jitter, Current and Preffered buffer size, Packet loss,Preemptive Rate , Accelerate Rate, RTPTransport.

Common for Audio and Video Details:
Call Encryption


Here is the reference documentation link
AudioDetails:
https://www.devconnectprogram.com/site/global/products_resources/avaya_client_sdk/programming_docs/current/javascript/communication/api_refs/AvayaClientServices/AvayaClientServices.Services.Call.AudioDetails.html


VideoDetails
https://www.devconnectprogram.com/site/global/products_resources/avaya_client_sdk/programming_docs/current/javascript/communication/api_refs/AvayaClientServices/AvayaClientServices.Services.Call.VideoDetails.html


Please let me know how I can get these parameters.

Hello AlokKulkarni,

Audio packet loss stats:

Audio packetLossTotalReceived loss https://www.devconnectprogram.com/site/global/products_resources/avaya_client_sdk/programming_docs/current/javascript/communication/api_refs/AvayaClientServices/AvayaClientServices.Services.Call.AudioDetails.html#toc11

Audio packetLossTotalTransmitted loss https://www.devconnectprogram.com/site/global/products_resources/avaya_client_sdk/programming_docs/current/javascript/communication/api_refs/AvayaClientServices/AvayaClientServices.Services.Call.AudioDetails.html#toc12

Regarding the rest of the statistics you mention, they are not available from the JSCSDK at the moment.
Since some of them cannot be found in webRTC's definitions, in case you want to make a request for those statistics to be considered for future releases, we ask you to please attach references to webRTC's official documentation of the statistics you are looking for.

Leandro,
Avaya DevConnect support team.