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 wrote:
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> 1. Re: Network call quality using PJSUA (Dennis Guse)
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> ----------------------------------------------------------------------
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> Message: 1
> Date: Wed, 29 Jan 2014 16:46:29 +0100
> From: Dennis Guse <dennis.guse@xxxxxxxxxxxxxxxxxxx>
> To: pjsip list <pjsip at lists.pjsip.org>
> Subject: Re: Network call quality using PJSUA
> Message-ID:
> <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> 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
>>
>>
>>
>>
>>
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