Network call quality using PJSUA

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



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:


Send pjsip mailing list submissions to
            pjsip at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>

To subscribe or unsubscribe via the World Wide Web, visit
            http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
or, via email, send a message with subject or body 'help' to
            pjsip-request at lists.pjsip.org<mailto:pjsip-request at lists.pjsip.org>

You can reach the person managing the list at
            pjsip-owner at lists.pjsip.org<mailto:pjsip-owner at lists.pjsip.org>

When replying, please edit your Subject line so it is more specific
than "Re: Contents of pjsip digest..."


Today's Topics:

  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>





_______________________________________________
Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20140129/29f94bc1/attachment.html>

------------------------------

Subject: Digest Footer

_______________________________________________
pjsip mailing list
pjsip at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org


------------------------------

End of pjsip Digest, Vol 77, Issue 115
**************************************

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20140131/638f8393/attachment-0001.html>


[Index of Archives]     [Asterisk Users]     [Asterisk App Development]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [Linux API]
  Powered by Linux