Re: pjsua -- non-interactive mode for quality monitoring?

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Hello Franz,

On 06/02/2020 17:54, Skale Franz wrote:
I really do think you didn't make a call at all.
You didn't attach a wav file nor did you supply a tone generation at the commandline.

I definitely did a call, but you are correct to assume the worst :)

I even did a tcpdump capture, and using wireshark I see RTP traffic going both ways. Even more - wireshark is able to show me the waveform of the voice communication and play it out right from the pcap (how cool is that?!).

Apparently pjsua does not need a tone instruction nor it require any kind of sound sample - with the command line I used, it simply "generates" silence and sends it as a G.711 stream.

If you look at the TX/RX stats I posted in my initial message, you will see that it says "RX = total 336 pkts, TX = total 460 pkts".

But why it is unable to compute the RTT is a mystery to me. The PBX server is 45ms away from me, so I'd expect this to show up in the stats...

Anyway, my three initial questions still hold :)

1. Is there a "proper" way to fetch call stats from pjsua? (other than savagely grepping, seding and cutting its console output)

2. How should I instruct pjsua to quit when the call is over? Currently I run it through the Linux "timeout" command so it gets killed after a few seconds, but that's a really dirty way of doing business.

3. What may be the reason pjsua outputs "0.000" in its RTT stats output?

The stats are for media quality, but will show 0 when no RDP session was established !

Do you confirm that you do get some meaningful "RTT msec" values in pjsua's output? Do you use vanilla pjsua code, as found in pjsip 2.9?

best,
Mateusz




________________________________________
Von: pjsip <pjsip-bounces@xxxxxxxxxxxxxxx> im Auftrag von Mateusz Viste <mateusz@xxxxxxxx>
Gesendet: Donnerstag, 06. Februar 2020 16:49
An: pjsip@xxxxxxxxxxxxxxx
Betreff:  pjsua -- non-interactive mode for quality monitoring?

Hello list,

I am fiddling with the pjsua tool from pjsip 2.9, because I'd like to
use it to monitor the quality of my VoIP provider at home. I am able to
run a call all right, but pjsua does not exit once the call is
terminated - I have to press 'q' to quit the interactive shell.

Is there any trick I could use so pjsua performs the call, and exit once
the call ends? This is the command line I use now:

./pjsua-armv7l-unknown-linux-gnueabihf
--id='sip:mylogin@xxxxxxxxxxxxxxx' --realm='*' --username='mylogin'
--password='xxxxxx' --disable-stun --no-tcp --null-audio --no-vad
--max-calls=1 --duration=10 'sip:123@xxxxxxxxxxxxxxx' --no-color

Additionally, I'd need to get some stats from it. Currently I am
grepping the logs that pjsua outputs, to extract some data from this:

   RX pt=0, last update:00h:00m:03.841s ago
      total 336pkt 53.7KB (67.2KB +IP hdr) @avg=46.7Kbps/58.4Kbps
      pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
            (msec)    min     avg     max     last    dev
      loss period:   0.000   0.000   0.000   0.000   0.000
      jitter     :   0.000   0.119   0.750   0.125   0.151
   TX pt=0, ptime=20, last update:never
      total 460pkt 73.6KB (92.0KB +IP hdr) @avg=64.0Kbps/80.0Kbps
      pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
            (msec)    min     avg     max     last    dev
      loss period:   0.000   0.000   0.000   0.000   0.000
      jitter     :   0.000   0.000   0.000   0.000   0.000
   RTT msec      :   0.000   0.000   0.000   0.000   0.000

...but maybe there is some easier way? I noticed that 'RTT msec' always
shows 0.000, which is kind of strange. Any idea?

best,
Mateusz

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