Advice on QoS/RTCP logging for later analysis

Asterisk Asterisk SIP
1

Hello,

In a pure SIP environment, I’m wondering if I should log some stats into CDR for later analysis.

Reading about this subject, I realized there are several things I don’t know or understand:

  1. I came to read RFC8079.
    Do current Asterisk 17 RTCP stats qualify as “end to end”, ie summing both inbound and outbound channels stats together ?

  2. Which stats should be preferred for Call Quality troubleshooting ?
    I’m currently using CHANNEL(rtpqos,audio,all).

[1] https://tools.ietf.org/html/rfc8079

Best regards

2

It is not end to end, each channel is independent and unrelated. The stats you mention would be the ones for that specific channel.

3

So, basically, if I find a way (with handlers) to log the ITSP trunk leg stats, the scope of my stats would exactly cover the path between Asterisk and ITSP’s SBC, leaving aside anything between this SBC and end caller or callee, right ?

4

Asterisk itself only cares about the RTCP report it receives, it would depend on what exactly the SBC does.

5

In a setup, I’m seeing Asterisk sending reports but nothing coming back from ITSP.
Either, ITSP is sending reports that are lost somewhere (NAT, …) in between Asterisk and ITSP SBC (Oracle SBC), or ITSP is not sending anything.

I’ve just discovered rtcp_mux=yes option in chan_sip (from 13.5.0 and later).
Maybe, it would solve report loss.

6

RTCP Mux has to be supported by the remote side, if it’s not then it won’t be done.

7

Yes, of course.
Life is hard, anyway ;-)))

8

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