On 17.05.20 at 01:28 Joshua C. Colp wrote:
> On Sat, May 16, 2020 at 10:58 AM Michael Maier <[email protected]> wrote:
>
>> => How are the RTT values exactly calculated? Which values are actually
>> used for?
>>
>
> The value is calculated according to the logic in the RFC[1]. Specifically
> using embedded timestamps in the RTCP packets and calculated delays. The
> value is presented in seconds I believe in the output.
Thanks Joshua!
>> => What about the processing time between the inbound leg and the outbound
>> leg (transcoding, ...)?
>>
>
> That has no impact within this, since it's calculated using the RTCP
> traffic which is not used for media. It's really just for round trip time
> of a UDP packet, not for end to end time of a single audio packet/frame
> through the system.
Let's try to sum it up on base of the given easy example how to get the
complete delay between those two speakers:
A calls B:
...........Receive.........
.........Transmit..........
BridgeId ChannelId ........ UpTime.. Codec. Count Lost Pct Jitter
Count Lost Pct Jitter RTT....
===========================================================================================================
c8137221 327-00000004 03:22:42 g722 608K 0 0 0.000
608K 0 0 0.000 0.000
c8137221 providePJSIP-xxx-0 03:22:42 alaw 608K 0 0 0.000
608K 0 0 0.000 0.023
A says something.
1. quantization: 20 ms
2. processing time on the phone base / DECT: ?
3. way from phone base to asterisk: 0 ms
4. processing time on asterisk / transcoding: ?
5. transport to destination: 11.5 ms
6. processing time on destination side: ?
Therefore it would take about 35 ms until B can here A. Is this basically a
correct estimation or did I miss(understand) something?
Thanks
Michael
> [1] https://tools.ietf.org/html/rfc3550
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