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... 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