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@sip.provider.pl' --realm='*' --username='mylogin'
--password='xxxxxx' --disable-stun --no-tcp --null-audio --no-vad
--max-calls=1 --duration=10 'sip:123@sip.provider.pl' --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
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.
Also keep in mind, that your receipient has CONFIRM the call and then the media channel can transmit the audio.
I programmed a call loop plugin for our VOIP infratructure sytem which is using a berofix GSM->TIP gateway for callin.
The stats are for media quality, but will show 0 when no RDP session was established !
Best regards
Franz
Von: pjsip pjsip-bounces@lists.pjsip.org im Auftrag von Mateusz Viste mateusz@viste.fr
Gesendet: Donnerstag, 06. Februar 2020 16:49
An: pjsip@lists.pjsip.org
Betreff: [pjsip] 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@sip.provider.pl' --realm='*' --username='mylogin'
--password='xxxxxx' --disable-stun --no-tcp --null-audio --no-vad
--max-calls=1 --duration=10 'sip:123@sip.provider.pl' --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
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
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 :)
-
Is there a "proper" way to fetch call stats from pjsua? (other than
savagely grepping, seding and cutting its console output) -
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. -
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@lists.pjsip.org im Auftrag von Mateusz Viste mateusz@viste.fr
Gesendet: Donnerstag, 06. Februar 2020 16:49
An: pjsip@lists.pjsip.org
Betreff: [pjsip] 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@sip.provider.pl' --realm='*' --username='mylogin'
--password='xxxxxx' --disable-stun --no-tcp --null-audio --no-vad
--max-calls=1 --duration=10 'sip:123@sip.provider.pl' --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
Hi,
i use version 2.9 only with the patch i commited (FFMPEG) as well as one from Andreas.
My check_call_loop connects a wav file (with text2speech) to the conference bridge and it will start to play when CONFIRM (200) has been sent. So, my wav file is 14s long. RTT is 20ms via SIP->GSM->SIP.
So basically, the client is also a server when using pjsua callbacks.
For the mainloop i use the superb EV Library (http://software.schmorp.de/pkg/EV.html).
pjsua uses and endless for loop, so the program won't be closed.
My guess is, that when you send silence, there are only protocol level packets but no stream, therefore the stats show "0".
My 14s call loop output: (keep in mind, that it calls a number which will be rerouted to SIP using another subscriber and calls back the app. So i have call 0 TX and call1 RX):
Here one call (obfuscated IP addresses and contacts etc)
21:00:01.421 pjsua_core.c !.pjsua version 2.9 for Linux-4.19.91/x86_64/glibc-2.28 initialized
21:00:01.422 init_app.c WAV player conf port id: 1
21:00:01.422 init_app.c Setting clock rate to wav file properties: 22050 Hz
21:00:01.422 init_app.c Conference port info: Port #01[22KHz/20ms/1]
21:00:01.422 init_app.c Wav properties: Sample rate: 22KHz Bits per sample: 16 Channel count: 1 File length: 624546 bytes
21:00:01.422 init_app.c Estimated play duration: 14 seconds
21:00:01.422 init_app.c Enabling NULL audio
21:00:01.426 module.c ....INVITE: From: xxxxxxxxxxxx
21:00:01.426 callbacks.c .......Call 0 state changed to CALLING
21:00:01.426 ev.c Initialize read event watcher
21:00:01.426 ev.c Start I/O event watcher
21:00:01.426 ev.c Init timeout event watcher
21:00:01.426 ev.c Init write event watcher
21:00:01.426 ev.c Start timeout event watcher
21:00:01.426 ev.c Start write event watcher
21:00:01.426 ev.c Start main event loop
21:00:01.436 module.c !.......INVITE: From: xxxxxxxxxxxxxx
21:00:01.436 module.c !.......INVITE: From: 0720258001
21:00:01.439 pjsua_acc.c ....sip:xxxxxxxxxxx@xxxxxxxxxxxxxxxx: registration success, status=200 (OK), will re-register in 600 seconds
21:00:05.200 callbacks.c ..Incoming call successfully verified
21:00:05.200 callbacks.c ..Incoming call from sip:xxxxxxxxxxxx@xxxxxxxxxxxxxxxxxxx
21:00:05.200 callbacks.c ..Call-ID: 1 Call-ID-string: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxx Active: 1
21:00:05.200 callbacks.c ..Local contact: sip:xxxxxxxxx@xxx.xxx.xxx.xxx:5060;ob
21:00:05.200 callbacks.c ..Remote contact: sip:xxxxxxxxxxxxxx@xxx.xxx.xxx.xxx
21:00:05.200 callbacks.c .......Media stream created: 1 0
21:00:05.201 callbacks.c .....Call media active: 1 Slot: 2
21:00:05.201 callbacks.c .........Call 1 state changed to CONNECTING
21:00:05.223 callbacks.c ...Successfully connected wav conf port: 1 to conf slot: 2
21:00:05.223 callbacks.c ...Call 1 state changed to CONFIRMED
21:00:05.292 callbacks.c .....Call 0 state changed to EARLY
21:00:05.496 callbacks.c .....Call 0 state changed to CONNECTING
21:00:05.496 callbacks.c .......Media stream created: 0 0
21:00:05.497 callbacks.c .....Call media active: 0 Slot: 3
21:00:05.497 callbacks.c .....Call 0 state changed to CONFIRMED
21:00:19.648 callbacks.c .....Call 1 is DISCONNECTED [reason=200 (Normal call clearing)]
21:00:19.648 callbacks.c .....Call 1 disconnected, dumping media stats..
21:00:19.648 common.c .....
[DISCONNCTD] t: sip:xxxxxxxxxxxxxxx@xxxxxxxxxx;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
Call time: 00h:00m:14s, 1st res in 1 ms, conn in 24ms
#0 audio PCMA @8kHz, sendrecv, peer=xxx.xxx.xxx.xxx:37724
SRTP status: Not active Crypto-suite:
RX pt=8, last update:00h:00m:00.226s ago
total 717pkt 114.7KB (143.4KB +IP hdr) @avg=63.5Kbps/79.4Kbps
pkt loss=1 (0.1%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
(msec) min avg max last dev
loss period: 20.000 20.000 20.000 20.000 0.000
jitter : 0.000 0.011 0.125 0.000 0.035
TX pt=8, ptime=20, last update:never
total 714pkt 114.2KB (142.8KB +IP hdr) @avg=63.2Kbps/79.0Kbps
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
21:00:19.681 callbacks.c .....Call 0 is DISCONNECTED [reason=200 (Normal call clearing)]
21:00:19.681 callbacks.c .....Call 0 disconnected, dumping media stats..
21:00:19.681 common.c .....
[DISCONNCTD] t: sip:xxxxxxxxxxxxxxxxxx@xxxxxxxxxxxxxx;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxx
Call time: 00h:00m:14s, 1st res in 3868 ms, conn in 4073ms
#0 audio PCMA @8kHz, sendrecv, peer=xxxxxxxxxxxxx:37764
SRTP status: Not active Crypto-suite:
RX pt=8, last update:00h:00m:03.418s ago
total 707pkt 113.1KB (141.4KB +IP hdr) @avg=63.7Kbps/79.7Kbps
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 0.285 0.500 0.375 0.124
TX pt=8, ptime=20, last update:00h:00m:05.044s ago
total 33pkt 5.2KB (6.6KB +IP hdr) @avg=2.9Kbps/3.7Kbps
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 : 12.802 16.518 20.233 12.802 3.715
21:00:20.428 ev.c !LOG DUMP DONE
21:00:20.428 common.c Connect duration (msec): 14184 Total durarion (msec): 18257 Setup time (seconds): 4.073
21:00:20.428 common.c Nagios output: connect time: 14.184s total time: 18.257s setup time: 4.073s jitter avg: 0.285 msecs rtt avg: 16.518 msec loss: 0.0% codec: PCMA @8kHz
21:00:20.428 common.c warning: 15 Critical: 20
21:00:20.428 common.c Call Loop finished:: Success [status=0]
21:00:20.428 ev.c Stopping outstanding event loops
21:00:20.428 common.c Releasing memory pool
21:00:20.428 common.c Destroy pjsua
21:00:20.442 pjsua_acc.c .....xxxxxxxxxxx@xxxxxxxxxxxxxxxxx: unregistration success
Best regards
Franz
Von: pjsip pjsip-bounces@lists.pjsip.org im Auftrag von Mateusz Viste mateusz@viste.fr
Gesendet: Donnerstag, 06. Februar 2020 21:21
An: pjsip@lists.pjsip.org
Betreff: Re: [pjsip] pjsua -- non-interactive mode for quality monitoring?
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 :)
-
Is there a "proper" way to fetch call stats from pjsua? (other than
savagely grepping, seding and cutting its console output) -
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. -
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@lists.pjsip.org im Auftrag von Mateusz Viste mateusz@viste.fr
Gesendet: Donnerstag, 06. Februar 2020 16:49
An: pjsip@lists.pjsip.org
Betreff: [pjsip] 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@sip.provider.pl' --realm='*' --username='mylogin'
--password='xxxxxx' --disable-stun --no-tcp --null-audio --no-vad
--max-calls=1 --duration=10 'sip:123@sip.provider.pl' --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
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Hi,
The 'silence' I send is being actually sent over RTP (ie. it truly is a
stream, I can see it in Wireshark), but just to be sure I added
"--auto-loop" to my pjsua command-line - so now my call echoes back
whatever the remote end says. And again - I can see in Wireshark that it
works, pjsua receives a stream, and sends it back with a slight delay.
But still -- the RTT stats shows 0.
I compared your console output with mine (thanks for sharing it!). I see
that in your case, there are two RX/TX dumps, and the first one displays
a RTT with a 0.000 value - just like mine. The big difference being that
you have a second RX/TX stats dump that follows the first one, and this
second dump does have a non-zero RTT. Myself, I have only one dumpe
(presumably because I am processing a single connection, while you
perform some kind of conferencing, I think). Would you mind telling what
is the difference between the two calls you have?
Mateusz
On 06/02/2020 21:42, Skale Franz wrote:
Hi,
i use version 2.9 only with the patch i commited (FFMPEG) as well as one from Andreas.
My check_call_loop connects a wav file (with text2speech) to the conference bridge and it will start to play when CONFIRM (200) has been sent. So, my wav file is 14s long. RTT is 20ms via SIP->GSM->SIP.
So basically, the client is also a server when using pjsua callbacks.
For the mainloop i use the superb EV Library (http://software.schmorp.de/pkg/EV.html).
pjsua uses and endless for loop, so the program won't be closed.
My guess is, that when you send silence, there are only protocol level packets but no stream, therefore the stats show "0".
My 14s call loop output: (keep in mind, that it calls a number which will be rerouted to SIP using another subscriber and calls back the app. So i have call 0 TX and call1 RX):
Here one call (obfuscated IP addresses and contacts etc)
21:00:01.421 pjsua_core.c !.pjsua version 2.9 for Linux-4.19.91/x86_64/glibc-2.28 initialized
21:00:01.422 init_app.c WAV player conf port id: 1
21:00:01.422 init_app.c Setting clock rate to wav file properties: 22050 Hz
21:00:01.422 init_app.c Conference port info: Port #01[22KHz/20ms/1]
21:00:01.422 init_app.c Wav properties: Sample rate: 22KHz Bits per sample: 16 Channel count: 1 File length: 624546 bytes
21:00:01.422 init_app.c Estimated play duration: 14 seconds
21:00:01.422 init_app.c Enabling NULL audio
21:00:01.426 module.c ....INVITE: From: xxxxxxxxxxxx
21:00:01.426 callbacks.c .......Call 0 state changed to CALLING
21:00:01.426 ev.c Initialize read event watcher
21:00:01.426 ev.c Start I/O event watcher
21:00:01.426 ev.c Init timeout event watcher
21:00:01.426 ev.c Init write event watcher
21:00:01.426 ev.c Start timeout event watcher
21:00:01.426 ev.c Start write event watcher
21:00:01.426 ev.c Start main event loop
21:00:01.436 module.c !.......INVITE: From: xxxxxxxxxxxxxx
21:00:01.436 module.c !.......INVITE: From: 0720258001
21:00:01.439 pjsua_acc.c ....sip:xxxxxxxxxxx@xxxxxxxxxxxxxxxx: registration success, status=200 (OK), will re-register in 600 seconds
21:00:05.200 callbacks.c ..Incoming call successfully verified
21:00:05.200 callbacks.c ..Incoming call from sip:xxxxxxxxxxxx@xxxxxxxxxxxxxxxxxxx
21:00:05.200 callbacks.c ..Call-ID: 1 Call-ID-string: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxx Active: 1
21:00:05.200 callbacks.c ..Local contact: sip:xxxxxxxxx@xxx.xxx.xxx.xxx:5060;ob
21:00:05.200 callbacks.c ..Remote contact: sip:xxxxxxxxxxxxxx@xxx.xxx.xxx.xxx
21:00:05.200 callbacks.c .......Media stream created: 1 0
21:00:05.201 callbacks.c .....Call media active: 1 Slot: 2
21:00:05.201 callbacks.c .........Call 1 state changed to CONNECTING
21:00:05.223 callbacks.c ...Successfully connected wav conf port: 1 to conf slot: 2
21:00:05.223 callbacks.c ...Call 1 state changed to CONFIRMED
21:00:05.292 callbacks.c .....Call 0 state changed to EARLY
21:00:05.496 callbacks.c .....Call 0 state changed to CONNECTING
21:00:05.496 callbacks.c .......Media stream created: 0 0
21:00:05.497 callbacks.c .....Call media active: 0 Slot: 3
21:00:05.497 callbacks.c .....Call 0 state changed to CONFIRMED
21:00:19.648 callbacks.c .....Call 1 is DISCONNECTED [reason=200 (Normal call clearing)]
21:00:19.648 callbacks.c .....Call 1 disconnected, dumping media stats..
21:00:19.648 common.c .....
[DISCONNCTD] t: sip:xxxxxxxxxxxxxxx@xxxxxxxxxx;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
Call time: 00h:00m:14s, 1st res in 1 ms, conn in 24ms
#0 audio PCMA @8kHz, sendrecv, peer=xxx.xxx.xxx.xxx:37724
SRTP status: Not active Crypto-suite:
RX pt=8, last update:00h:00m:00.226s ago
total 717pkt 114.7KB (143.4KB +IP hdr) @avg=63.5Kbps/79.4Kbps
pkt loss=1 (0.1%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
(msec) min avg max last dev
loss period: 20.000 20.000 20.000 20.000 0.000
jitter : 0.000 0.011 0.125 0.000 0.035
TX pt=8, ptime=20, last update:never
total 714pkt 114.2KB (142.8KB +IP hdr) @avg=63.2Kbps/79.0Kbps
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
21:00:19.681 callbacks.c .....Call 0 is DISCONNECTED [reason=200 (Normal call clearing)]
21:00:19.681 callbacks.c .....Call 0 disconnected, dumping media stats..
21:00:19.681 common.c .....
[DISCONNCTD] t: sip:xxxxxxxxxxxxxxxxxx@xxxxxxxxxxxxxx;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxx
Call time: 00h:00m:14s, 1st res in 3868 ms, conn in 4073ms
#0 audio PCMA @8kHz, sendrecv, peer=xxxxxxxxxxxxx:37764
SRTP status: Not active Crypto-suite:
RX pt=8, last update:00h:00m:03.418s ago
total 707pkt 113.1KB (141.4KB +IP hdr) @avg=63.7Kbps/79.7Kbps
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 0.285 0.500 0.375 0.124
TX pt=8, ptime=20, last update:00h:00m:05.044s ago
total 33pkt 5.2KB (6.6KB +IP hdr) @avg=2.9Kbps/3.7Kbps
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 : 12.802 16.518 20.233 12.802 3.715
21:00:20.428 ev.c !LOG DUMP DONE
21:00:20.428 common.c Connect duration (msec): 14184 Total durarion (msec): 18257 Setup time (seconds): 4.073
21:00:20.428 common.c Nagios output: connect time: 14.184s total time: 18.257s setup time: 4.073s jitter avg: 0.285 msecs rtt avg: 16.518 msec loss: 0.0% codec: PCMA @8kHz
21:00:20.428 common.c warning: 15 Critical: 20
21:00:20.428 common.c Call Loop finished:: Success [status=0]
21:00:20.428 ev.c Stopping outstanding event loops
21:00:20.428 common.c Releasing memory pool
21:00:20.428 common.c Destroy pjsua
21:00:20.442 pjsua_acc.c .....xxxxxxxxxxx@xxxxxxxxxxxxxxxxx: unregistration success
Best regards
Franz
Von: pjsip pjsip-bounces@lists.pjsip.org im Auftrag von Mateusz Viste mateusz@viste.fr
Gesendet: Donnerstag, 06. Februar 2020 21:21
An: pjsip@lists.pjsip.org
Betreff: Re: [pjsip] pjsua -- non-interactive mode for quality monitoring?
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 :)
-
Is there a "proper" way to fetch call stats from pjsua? (other than
savagely grepping, seding and cutting its console output) -
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. -
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@lists.pjsip.org im Auftrag von Mateusz Viste mateusz@viste.fr
Gesendet: Donnerstag, 06. Februar 2020 16:49
An: pjsip@lists.pjsip.org
Betreff: [pjsip] 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@sip.provider.pl' --realm='*' --username='mylogin'
--password='xxxxxx' --disable-stun --no-tcp --null-audio --no-vad
--max-calls=1 --duration=10 'sip:123@sip.provider.pl' --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
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Franz, I spotted yet another difference in our logs: yours includes this
very cool line:
common.c Nagios output: connect time: 14.184s total time: 18.257s setup
time: 4.073s jitter avg: 0.285 msecs rtt avg: 16.518 msec loss: 0.0%
codec: PCMA @8kHz
This is exactly what I would need - on my build it is missing. I also
grepped the entire sources looking for the phrase "Nagios output", but
found no match. Is this something you have added yourself?
cheers,
Mateusz
On 06/02/2020 23:22, Mateusz Viste wrote:
Hi,
The 'silence' I send is being actually sent over RTP (ie. it truly is a
stream, I can see it in Wireshark), but just to be sure I added
"--auto-loop" to my pjsua command-line - so now my call echoes back
whatever the remote end says. And again - I can see in Wireshark that it
works, pjsua receives a stream, and sends it back with a slight delay.
But still -- the RTT stats shows 0.
I compared your console output with mine (thanks for sharing it!). I see
that in your case, there are two RX/TX dumps, and the first one displays
a RTT with a 0.000 value - just like mine. The big difference being that
you have a second RX/TX stats dump that follows the first one, and this
second dump does have a non-zero RTT. Myself, I have only one dumpe
(presumably because I am processing a single connection, while you
perform some kind of conferencing, I think). Would you mind telling what
is the difference between the two calls you have?
Mateusz
On 06/02/2020 21:42, Skale Franz wrote:
Hi,
i use version 2.9 only with the patch i commited (FFMPEG) as well as
one from Andreas.
My check_call_loop connects a wav file (with text2speech) to the
conference bridge and it will start to play when CONFIRM (200) has
been sent. So, my wav file is 14s long. RTT is 20ms via SIP->GSM->SIP.
So basically, the client is also a server when using pjsua callbacks.
For the mainloop i use the superb EV Library
(http://software.schmorp.de/pkg/EV.html).
pjsua uses and endless for loop, so the program won't be closed.
My guess is, that when you send silence, there are only protocol level
packets but no stream, therefore the stats show "0".
My 14s call loop output: (keep in mind, that it calls a number which
will be rerouted to SIP using another subscriber and calls back the
app. So i have call 0 TX and call1 RX):
Here one call (obfuscated IP addresses and contacts etc)
21:00:01.421 pjsua_core.c !.pjsua version 2.9 for
Linux-4.19.91/x86_64/glibc-2.28 initialized
21:00:01.422 init_app.c WAV player conf port id: 1
21:00:01.422 init_app.c Setting clock rate to wav file
properties: 22050 Hz
21:00:01.422 init_app.c Conference port info: Port
#01[22KHz/20ms/1]
21:00:01.422 init_app.c Wav properties: Sample rate:
22KHz Bits per sample: 16 Channel count: 1 File length: 624546 bytes
21:00:01.422 init_app.c Estimated play duration: 14 seconds
21:00:01.422 init_app.c Enabling NULL audio
21:00:01.426 module.c ....INVITE: From: xxxxxxxxxxxx
21:00:01.426 callbacks.c .......Call 0 state changed to
CALLING
21:00:01.426 ev.c Initialize read event watcher
21:00:01.426 ev.c Start I/O event watcher
21:00:01.426 ev.c Init timeout event watcher
21:00:01.426 ev.c Init write event watcher
21:00:01.426 ev.c Start timeout event watcher
21:00:01.426 ev.c Start write event watcher
21:00:01.426 ev.c Start main event loop
21:00:01.436 module.c !.......INVITE: From: xxxxxxxxxxxxxx
21:00:01.436 module.c !.......INVITE: From: 0720258001
21:00:01.439 pjsua_acc.c
....sip:xxxxxxxxxxx@xxxxxxxxxxxxxxxx: registration success, status=200
(OK), will re-register in 600 seconds
21:00:05.200 callbacks.c ..Incoming call successfully
verified
21:00:05.200 callbacks.c ..Incoming call from
sip:xxxxxxxxxxxx@xxxxxxxxxxxxxxxxxxx
21:00:05.200 callbacks.c ..Call-ID: 1 Call-ID-string:
xxxxxxxxxxxxxxxxxxxxxxxxxxxxxx Active: 1
21:00:05.200 callbacks.c ..Local contact:
sip:xxxxxxxxx@xxx.xxx.xxx.xxx:5060;ob
21:00:05.200 callbacks.c ..Remote contact:
sip:xxxxxxxxxxxxxx@xxx.xxx.xxx.xxx
21:00:05.200 callbacks.c .......Media stream created: 1 0
21:00:05.201 callbacks.c .....Call media active: 1 Slot: 2
21:00:05.201 callbacks.c .........Call 1 state changed to
CONNECTING
21:00:05.223 callbacks.c ...Successfully connected wav
conf port: 1 to conf slot: 2
21:00:05.223 callbacks.c ...Call 1 state changed to CONFIRMED
21:00:05.292 callbacks.c .....Call 0 state changed to EARLY
21:00:05.496 callbacks.c .....Call 0 state changed to
CONNECTING
21:00:05.496 callbacks.c .......Media stream created: 0 0
21:00:05.497 callbacks.c .....Call media active: 0 Slot: 3
21:00:05.497 callbacks.c .....Call 0 state changed to
CONFIRMED
21:00:19.648 callbacks.c .....Call 1 is DISCONNECTED
[reason=200 (Normal call clearing)]
21:00:19.648 callbacks.c .....Call 1 disconnected, dumping
media stats..
21:00:19.648 common.c .....
[DISCONNCTD] t:
sip:xxxxxxxxxxxxxxx@xxxxxxxxxx;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
Call time: 00h:00m:14s, 1st res in 1 ms, conn in 24ms
#0 audio PCMA @8kHz, sendrecv, peer=xxx.xxx.xxx.xxx:37724
SRTP status: Not active Crypto-suite:
RX pt=8, last update:00h:00m:00.226s ago
total 717pkt 114.7KB (143.4KB +IP hdr) @avg=63.5Kbps/79.4Kbps
pkt loss=1 (0.1%), discrd=0 (0.0%), dup=0 (0.0%), reord=0
(0.0%)
(msec) min avg max last dev
loss period: 20.000 20.000 20.000 20.000 0.000
jitter : 0.000 0.011 0.125 0.000 0.035
TX pt=8, ptime=20, last update:never
total 714pkt 114.2KB (142.8KB +IP hdr) @avg=63.2Kbps/79.0Kbps
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
21:00:19.681 callbacks.c .....Call 0 is DISCONNECTED
[reason=200 (Normal call clearing)]
21:00:19.681 callbacks.c .....Call 0 disconnected, dumping
media stats..
21:00:19.681 common.c .....
[DISCONNCTD] t:
sip:xxxxxxxxxxxxxxxxxx@xxxxxxxxxxxxxx;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxx
Call time: 00h:00m:14s, 1st res in 3868 ms, conn in 4073ms
#0 audio PCMA @8kHz, sendrecv, peer=xxxxxxxxxxxxx:37764
SRTP status: Not active Crypto-suite:
RX pt=8, last update:00h:00m:03.418s ago
total 707pkt 113.1KB (141.4KB +IP hdr) @avg=63.7Kbps/79.7Kbps
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 0.285 0.500 0.375 0.124
TX pt=8, ptime=20, last update:00h:00m:05.044s ago
total 33pkt 5.2KB (6.6KB +IP hdr) @avg=2.9Kbps/3.7Kbps
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 : 12.802 16.518 20.233 12.802 3.715
21:00:20.428 ev.c !LOG DUMP DONE
21:00:20.428 common.c Connect duration (msec): 14184
Total durarion (msec): 18257 Setup time (seconds): 4.073
21:00:20.428 common.c Nagios output: connect time:
14.184s total time: 18.257s setup time: 4.073s jitter avg: 0.285 msecs
rtt avg: 16.518 msec loss: 0.0% codec: PCMA @8kHz
21:00:20.428 common.c warning: 15 Critical: 20
21:00:20.428 common.c Call Loop finished:: Success
[status=0]
21:00:20.428 ev.c Stopping outstanding event loops
21:00:20.428 common.c Releasing memory pool
21:00:20.428 common.c Destroy pjsua
21:00:20.442 pjsua_acc.c
.....xxxxxxxxxxx@xxxxxxxxxxxxxxxxx: unregistration success
Best regards
Franz
Von: pjsip pjsip-bounces@lists.pjsip.org im Auftrag von Mateusz
Viste mateusz@viste.fr
Gesendet: Donnerstag, 06. Februar 2020 21:21
An: pjsip@lists.pjsip.org
Betreff: Re: [pjsip] pjsua -- non-interactive mode for quality
monitoring?
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 :)
-
Is there a "proper" way to fetch call stats from pjsua? (other than
savagely grepping, seding and cutting its console output) -
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. -
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@lists.pjsip.org im Auftrag von Mateusz
Viste mateusz@viste.fr
Gesendet: Donnerstag, 06. Februar 2020 16:49
An: pjsip@lists.pjsip.org
Betreff: [pjsip] 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@sip.provider.pl' --realm='*' --username='mylogin'
--password='xxxxxx' --disable-stun --no-tcp --null-audio --no-vad
--max-calls=1 --duration=10 'sip:123@sip.provider.pl' --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
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Hi Mateusz,
as i stated in my last post, i programmed a call check plugin which:
1.) Calls a number which terminates on a beronet SIP Gateway (GSM -> SIP).
2.) The beronet SIP account is registered to the C5 so when a call reached the SIP Gateway, it routes the call to the internal C5 SIP account.
The program verifies the calling contact and then connects the WAV file to the conference bridge.
So i have one outgoing and one incoming call which terminates in my program, running in a different event loop.
I favour EV as eventloop interface though.
This is a special plugin for SIP providers !
When you sent nothing, there will be no jitter. Try to attach a wav file to the conference bridge and check the stats.
There's a bug when enabling the extended stats which has not been addressed by any of the core developers so far.
Perhaps, the stats my be wrong with regards to only calling a party but not reveicing a call on the same loop.
I will, later this year, publish the program on github.
Best regards
Franz
Von: pjsip pjsip-bounces@lists.pjsip.org im Auftrag von Mateusz Viste mateusz@viste.fr
Gesendet: Donnerstag, 06. Februar 2020 23:54
An: pjsip@lists.pjsip.org
Betreff: POSSIBLE SPAM Re: [pjsip] pjsua -- non-interactive mode for quality monitoring?
Franz, I spotted yet another difference in our logs: yours includes this
very cool line:
common.c Nagios output: connect time: 14.184s total time: 18.257s setup
time: 4.073s jitter avg: 0.285 msecs rtt avg: 16.518 msec loss: 0.0%
codec: PCMA @8kHz
This is exactly what I would need - on my build it is missing. I also
grepped the entire sources looking for the phrase "Nagios output", but
found no match. Is this something you have added yourself?
cheers,
Mateusz
On 06/02/2020 23:22, Mateusz Viste wrote:
Hi,
The 'silence' I send is being actually sent over RTP (ie. it truly is a
stream, I can see it in Wireshark), but just to be sure I added
"--auto-loop" to my pjsua command-line - so now my call echoes back
whatever the remote end says. And again - I can see in Wireshark that it
works, pjsua receives a stream, and sends it back with a slight delay.
But still -- the RTT stats shows 0.
I compared your console output with mine (thanks for sharing it!). I see
that in your case, there are two RX/TX dumps, and the first one displays
a RTT with a 0.000 value - just like mine. The big difference being that
you have a second RX/TX stats dump that follows the first one, and this
second dump does have a non-zero RTT. Myself, I have only one dumpe
(presumably because I am processing a single connection, while you
perform some kind of conferencing, I think). Would you mind telling what
is the difference between the two calls you have?
Mateusz
On 06/02/2020 21:42, Skale Franz wrote:
Hi,
i use version 2.9 only with the patch i commited (FFMPEG) as well as
one from Andreas.
My check_call_loop connects a wav file (with text2speech) to the
conference bridge and it will start to play when CONFIRM (200) has
been sent. So, my wav file is 14s long. RTT is 20ms via SIP->GSM->SIP.
So basically, the client is also a server when using pjsua callbacks.
For the mainloop i use the superb EV Library
(http://software.schmorp.de/pkg/EV.html).
pjsua uses and endless for loop, so the program won't be closed.
My guess is, that when you send silence, there are only protocol level
packets but no stream, therefore the stats show "0".
My 14s call loop output: (keep in mind, that it calls a number which
will be rerouted to SIP using another subscriber and calls back the
app. So i have call 0 TX and call1 RX):
Here one call (obfuscated IP addresses and contacts etc)
21:00:01.421 pjsua_core.c !.pjsua version 2.9 for
Linux-4.19.91/x86_64/glibc-2.28 initialized
21:00:01.422 init_app.c WAV player conf port id: 1
21:00:01.422 init_app.c Setting clock rate to wav file
properties: 22050 Hz
21:00:01.422 init_app.c Conference port info: Port
#01[22KHz/20ms/1]
21:00:01.422 init_app.c Wav properties: Sample rate:
22KHz Bits per sample: 16 Channel count: 1 File length: 624546 bytes
21:00:01.422 init_app.c Estimated play duration: 14 seconds
21:00:01.422 init_app.c Enabling NULL audio
21:00:01.426 module.c ....INVITE: From: xxxxxxxxxxxx
21:00:01.426 callbacks.c .......Call 0 state changed to
CALLING
21:00:01.426 ev.c Initialize read event watcher
21:00:01.426 ev.c Start I/O event watcher
21:00:01.426 ev.c Init timeout event watcher
21:00:01.426 ev.c Init write event watcher
21:00:01.426 ev.c Start timeout event watcher
21:00:01.426 ev.c Start write event watcher
21:00:01.426 ev.c Start main event loop
21:00:01.436 module.c !.......INVITE: From: xxxxxxxxxxxxxx
21:00:01.436 module.c !.......INVITE: From: 0720258001
21:00:01.439 pjsua_acc.c
....sip:xxxxxxxxxxx@xxxxxxxxxxxxxxxx: registration success, status=200
(OK), will re-register in 600 seconds
21:00:05.200 callbacks.c ..Incoming call successfully
verified
21:00:05.200 callbacks.c ..Incoming call from
sip:xxxxxxxxxxxx@xxxxxxxxxxxxxxxxxxx
21:00:05.200 callbacks.c ..Call-ID: 1 Call-ID-string:
xxxxxxxxxxxxxxxxxxxxxxxxxxxxxx Active: 1
21:00:05.200 callbacks.c ..Local contact:
sip:xxxxxxxxx@xxx.xxx.xxx.xxx:5060;ob
21:00:05.200 callbacks.c ..Remote contact:
sip:xxxxxxxxxxxxxx@xxx.xxx.xxx.xxx
21:00:05.200 callbacks.c .......Media stream created: 1 0
21:00:05.201 callbacks.c .....Call media active: 1 Slot: 2
21:00:05.201 callbacks.c .........Call 1 state changed to
CONNECTING
21:00:05.223 callbacks.c ...Successfully connected wav
conf port: 1 to conf slot: 2
21:00:05.223 callbacks.c ...Call 1 state changed to CONFIRMED
21:00:05.292 callbacks.c .....Call 0 state changed to EARLY
21:00:05.496 callbacks.c .....Call 0 state changed to
CONNECTING
21:00:05.496 callbacks.c .......Media stream created: 0 0
21:00:05.497 callbacks.c .....Call media active: 0 Slot: 3
21:00:05.497 callbacks.c .....Call 0 state changed to
CONFIRMED
21:00:19.648 callbacks.c .....Call 1 is DISCONNECTED
[reason=200 (Normal call clearing)]
21:00:19.648 callbacks.c .....Call 1 disconnected, dumping
media stats..
21:00:19.648 common.c .....
[DISCONNCTD] t:
sip:xxxxxxxxxxxxxxx@xxxxxxxxxx;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
Call time: 00h:00m:14s, 1st res in 1 ms, conn in 24ms
#0 audio PCMA @8kHz, sendrecv, peer=xxx.xxx.xxx.xxx:37724
SRTP status: Not active Crypto-suite:
RX pt=8, last update:00h:00m:00.226s ago
total 717pkt 114.7KB (143.4KB +IP hdr) @avg=63.5Kbps/79.4Kbps
pkt loss=1 (0.1%), discrd=0 (0.0%), dup=0 (0.0%), reord=0
(0.0%)
(msec) min avg max last dev
loss period: 20.000 20.000 20.000 20.000 0.000
jitter : 0.000 0.011 0.125 0.000 0.035
TX pt=8, ptime=20, last update:never
total 714pkt 114.2KB (142.8KB +IP hdr) @avg=63.2Kbps/79.0Kbps
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
21:00:19.681 callbacks.c .....Call 0 is DISCONNECTED
[reason=200 (Normal call clearing)]
21:00:19.681 callbacks.c .....Call 0 disconnected, dumping
media stats..
21:00:19.681 common.c .....
[DISCONNCTD] t:
sip:xxxxxxxxxxxxxxxxxx@xxxxxxxxxxxxxx;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxx
Call time: 00h:00m:14s, 1st res in 3868 ms, conn in 4073ms
#0 audio PCMA @8kHz, sendrecv, peer=xxxxxxxxxxxxx:37764
SRTP status: Not active Crypto-suite:
RX pt=8, last update:00h:00m:03.418s ago
total 707pkt 113.1KB (141.4KB +IP hdr) @avg=63.7Kbps/79.7Kbps
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 0.285 0.500 0.375 0.124
TX pt=8, ptime=20, last update:00h:00m:05.044s ago
total 33pkt 5.2KB (6.6KB +IP hdr) @avg=2.9Kbps/3.7Kbps
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 : 12.802 16.518 20.233 12.802 3.715
21:00:20.428 ev.c !LOG DUMP DONE
21:00:20.428 common.c Connect duration (msec): 14184
Total durarion (msec): 18257 Setup time (seconds): 4.073
21:00:20.428 common.c Nagios output: connect time:
14.184s total time: 18.257s setup time: 4.073s jitter avg: 0.285 msecs
rtt avg: 16.518 msec loss: 0.0% codec: PCMA @8kHz
21:00:20.428 common.c warning: 15 Critical: 20
21:00:20.428 common.c Call Loop finished:: Success
[status=0]
21:00:20.428 ev.c Stopping outstanding event loops
21:00:20.428 common.c Releasing memory pool
21:00:20.428 common.c Destroy pjsua
21:00:20.442 pjsua_acc.c
.....xxxxxxxxxxx@xxxxxxxxxxxxxxxxx: unregistration success
Best regards
Franz
Von: pjsip pjsip-bounces@lists.pjsip.org im Auftrag von Mateusz
Viste mateusz@viste.fr
Gesendet: Donnerstag, 06. Februar 2020 21:21
An: pjsip@lists.pjsip.org
Betreff: Re: [pjsip] pjsua -- non-interactive mode for quality
monitoring?
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 :)
-
Is there a "proper" way to fetch call stats from pjsua? (other than
savagely grepping, seding and cutting its console output) -
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. -
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@lists.pjsip.org im Auftrag von Mateusz
Viste mateusz@viste.fr
Gesendet: Donnerstag, 06. Februar 2020 16:49
An: pjsip@lists.pjsip.org
Betreff: [pjsip] 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@sip.provider.pl' --realm='*' --username='mylogin'
--password='xxxxxx' --disable-stun --no-tcp --null-audio --no-vad
--max-calls=1 --duration=10 'sip:123@sip.provider.pl' --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
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org