Network call quality using PJSUA

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The problem with packet loss was that in case the network goes down the packet loss still shows 0, and after it comes back it starts reflecting packet loss even when the call is connected with great call quality. Even RTT is inconclusive, jitter seems like the best bet.

Thanks for the tip though, have to still optimise for edge.
-- 
Vinay Nair
vinay.nair at novanet.net




On 31-Jan-2014, at 5:42 pm, pjsip-request at lists.pjsip.org wrote:

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>   1. Re: Network call quality using PJSUA (Olle Frimanson)
> 
> 
> ----------------------------------------------------------------------
> 
> Message: 1
> Date: Fri, 31 Jan 2014 12:11:58 +0000
> From: Olle Frimanson <olle.frimanson@xxxxxxxxxxxx>
> To: pjsip list <pjsip at lists.pjsip.org>
> Subject: Re: Network call quality using PJSUA
> Message-ID:
> 	<185a8e72a6fd49e3b4827db58586abc1 at AM3PR04MB385.eurprd04.prod.outlook.com>
> 	
> Content-Type: text/plain; charset="iso-8859-1"
> 
> Hi , Vinay
> 
> you have the answer in front of you ;-)
> 
> Usually it's a combo of RTT, packet loss and jitter.
> 
> And a tip could be don't use 67 kbps on edge
> 
> BR/Olle
> 
> 
> Fr?n: pjsip [mailto:pjsip-bounces at lists.pjsip.org] F?r Vinay
> Skickat: den 31 januari 2014 07:27
> Till: pjsip at lists.pjsip.org
> ?mne: Re: [pjsip] Network call quality using PJSUA
> 
> Thanks! What I am looking for is to display the call quality while the user is on the call, like what Skype and Viber do.
> 
> I have implemented a method to call pjsua_call_dump every 3 seconds.
> 
> Here are two dumps, one on a good WiFi connection and one on edge:
> 
> 1) This was on a good wifi connection:
>    #0 audio opus @48kHz, sendrecv, peer=10.10.86.7:48819
>       SRTP status: Not active Crypto-suite: (null)
>       ICE role: Controlled, state: Negotiation Success, comp_cnt: 2
>          [0]: L:203.153.53.130:63428 (s) --> R:203.153.53.130:49543 (s)
>          [1]: L:203.153.53.130:49457 (s) --> R:203.153.53.130:58433 (s)
>       RX pt=124, last update:00h:00m:03.702s ago
>          total 401pkt 51.3KB (67.4KB +IP hdr) @avg=49.5Kbps/64.9Kbps
>          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   5.680   9.437   7.562   1.318
>       TX pt=124, ptime=20, last update:00h:00m:03.110s ago
>          total 416pkt 53.3KB (70.0KB +IP hdr) @avg=51.4Kbps/67.4Kbps
>          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   3.000   6.000   6.000   3.000
>       RTT msec      :  12.268  12.268  12.268  12.268   0.000
> 
> 2) This dump was on edge:
>                    Call time: 00h:00m:09s, 1st res in 1601 ms, conn in 2775ms
>                    #0 audio opus @48kHz, sendrecv, peer=10.10.86.7:48396
>                      SRTP status: Not active Crypto-suite: (null)
>                      ICE role: Controlled, state: Negotiation Success, comp_cnt: 2
>                          [0]: L:123.63.154.36:1462 (s) --> R:203.153.53.130:50353 (s)
>                          [1]: L:123.63.154.36:13584 (s) --> R:203.153.53.130:60799 (s)
>                      RX pt=124, last update:00h:00m:05.059s ago
>                          total 263pkt 33.7KB (44.2KB +IP hdr) @avg=26.5Kbps/34.8Kbps
>                          pkt loss=2 (0.8%), 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     :   1.229  41.802 121.000  32.895  14.663
>                      TX pt=124, ptime=20, last update:00h:00m:00.363s ago
>                          total 508pkt 65.1KB (85.4KB +IP hdr) @avg=51.1Kbps/67.1Kbps
>                          pkt loss=1 (0.2%), dup=0 (0.0%), reorder=0 (0.0%)
>                                (msec)    min     avg     max     last    dev
>                          loss period:  20.000  20.000  20.000  20.000   0.000
>                          jitter     :   0.000   6.219  12.437  12.437   6.218
>                      RTT msec      : 4229.000 4229.000 4229.000 4229.000   0.000
> 
> I have been reading the dumps, packet loss does not give an accurate picture, what parameters do I use to get a good idea of the ongoing call quality.
> 
> --
> Vinay Nair
> 
> 
> 
> On 29-Jan-2014, at 9:16 pm, pjsip-request at lists.pjsip.org<mailto:pjsip-request at lists.pjsip.org> wrote:
> 
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>  1. Re: Network call quality using PJSUA (Dennis Guse)
> 
> 
> ----------------------------------------------------------------------
> 
> Message: 1
> Date: Wed, 29 Jan 2014 16:46:29 +0100
> From: Dennis Guse <dennis.guse@xxxxxxxxxxxxxxxxxxx<mailto:dennis.guse at alumni.tu-berlin.de>>
> To: pjsip list <pjsip at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>>
> Subject: Re: Network call quality using PJSUA
> Message-ID:
>            <CAEeULf0eK+Cy=H9K8iU9f5xHKe77o=CnpAiKmA8KwQ8iNUxkcg at mail.gmail.com<mailto:CAEeULf0eK+Cy=H9K8iU9f5xHKe77o=CnpAiKmA8KwQ8iNUxkcg at mail.gmail.com>>
> Content-Type: text/plain; charset="utf-8"
> 
> Hi,
> 
> I am not aware of one... Are you just interested in packet-loss rates (also
> include jitter drops)?
> For this, you could regularly call pjsua_call_dump Y and parse the output
> manually.
> 
> Actually, I would love to have a callback in pjsua that is reporting packet
> loss in a regular basis (like one time per second).
> 
> Just my 2 cents....
> 
> 
> ---
> Dennis Guse
> 
> 
> On Wed, Jan 29, 2014 at 9:09 AM, Vinay <vinay.nair at novanet.net<mailto:vinay.nair at novanet.net>> wrote:
> 
> 
> Hi,
> 
> I would like to display a network quality indicator while the user is on
> call using pjsua.
> 
> --
> Vinay Nair
> vinay.nair at novanet.net<mailto:vinay.nair at novanet.net>
> 
> 
> 
> 
> 
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