How do I change the bandwidth allocation and clock rate based on network speeds? I am using the opus codec. On Fri, Jan 31, 2014 at 6:07 PM, <pjsip-request at lists.pjsip.org> wrote: > Send pjsip mailing list submissions to > 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 > > You can reach the person managing the list at > 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 (Vinay) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 31 Jan 2014 18:07:10 +0530 > From: Vinay <vinay.nair@xxxxxxxxxxx> > To: pjsip at lists.pjsip.org > Subject: Re: Network call quality using PJSUA > Message-ID: <F791FC73-D941-4AF8-9EAC-5041BD8D19DA at novanet.net> > Content-Type: text/plain; charset="us-ascii" > > 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: > > > Send pjsip mailing list submissions to > > 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 > > > > You can reach the person managing the list at > > 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 (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: > > > > > > 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.html > > > > > > ------------------------------ > > > > Subject: Digest Footer > > > > _______________________________________________ > > pjsip mailing list > > pjsip at lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > ------------------------------ > > > > End of pjsip Digest, Vol 77, Issue 131 > > ************************************** > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20140131/73a34cbc/attachment.html > > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > ------------------------------ > > End of pjsip Digest, Vol 77, Issue 132 > ************************************** > -- Vinay Nair -------------- next part -------------- An HTML attachment was scrubbed... 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