Audio_Sample_Rate is not aligned between spec and real life

Realtime Media Streams
1

Hello,
Iam doing RTMS for ZCC, about the audio sample rate, we have this from the spec, which would be applied for raw L16 audio
enum AUDIO_SAMPLE_RATE
{
SR_8K = 0,
SR_16K = 1,
SR_32K = 2,
SR_48K = 3
}

However, in reality, 8000Hz is not available, only from 16000Hz, and it is 0 instead, and 1 and 2 for 32K and 48K respectively
This is observed from Data Hanshake Response from the server, i.e if it says "sample_rate": 0 then the audio it sends is 16K

Please confirm if this is know issue and would it be keep this way, otherwise future fix will break all existing implementation
Thanks

2

Hi @MinhCm,

Thank you for bringing this up. I will attempt to verify what you’re seeing on my end and share with the engineering team. I’ll share any relevant updates with you in this thread.

Best,

Rehema

3

Hi @rehema.zoom
Do you have any update on this?
Thanks

4

Hi @MinhCm,

Still looking into this, sorry for the delay in an update. I will have something to you by EOW.

Best,

Rehema

5

hi @MinhCm,

could you share a screenshot of the server response you received, mentioned earlier? and also a screenshot of your server connection config settings.

Best,

Rehema

6

Hi Rehema,
Sorry for the late response:
Here is the media handshake request from our application:
{"media_params":{"audio":{"channel":2,"codec":1,"content_type":2,"data_opt":2,"sample_rate":0,"send_rate":500}},"media_type":1,"meeting_uuid":"s-k3_S73TjW63dvp8YUkxw","msg_type":3,"protocol_version":1,"rtms_stream_id":"909090f3-d866-432e-8911-ddb1a2353d47","sequence":0,"signature":"xxxxx"}
We have codec:1 which is L16, sample_rate:0 which should be 8Khz according to the spec

and here is the response from the server:
{"media_params":{"audio":{"channel":2,"codec":1,"content_type":2,"data_opt":2,"sample_rate":0,"send_rate":500}},"msg_type":4,"payload_encrypted":false,"protocol_version":1,"reason":"OK","sequence":0,"status_code":0}

The server response sample_rate:0 which should be 8Khz, but the incoming audio is actually 16Khz

I did saved all the websocket messages between our application and Zoom’s RTMS for this session, including raw audio(in its base64 form) in a text file. Please DM me if you want to get a copy
Thanks