Hello Ray, thanks an awful lot for coming back to me. Next weekend I will certainly do some playing about. Problem is, as you might have guessed, that I know next to nothing about this technology. But we live and learn.
Regards, Jan. On Tue, Aug 21, 2012 at 8:07 PM, MJ Ray <[email protected]> wrote: > Jan de Kruyf <[email protected]> > > However, I am hardly able to use it, when the latency on the network > > is bad (like now) > > Even in stretches where there is no yellow or red warning avbout the > > line quality, nearly all sound packets are dropped. > > > > At the same time Skype performs without a hitch on the same line and > > at the same time of day! > > > > I have attaches a copy of the log, for 1 call, to illustrate the problem. > > Thanks for the log. I've left the bits I found interesting below. > > The line quality warnings aren't perfect. Also, you can probably > assume Skype is doing whatever the code decides is best for Skype, > rather than what you're telling it to do, and that it's using some > pretty aggressive codec which might be secret or licensed > restrictively. Linphone's not going to do much except what you tell > it and prefers freely-improvable codecs. > > Firstly, it looks like whatever you're calling supports only PCMU, > PCMA, GSM and telephone-event. None of those are great, but you > probably want to use GSM if you're having bandwidth problems. > Sadly, you seem to be using PCMU. Sounds good but high bandwidth. > Disable that and PCMA and see if things improve. > > Secondly, linphone says "Audio bandwidth for this call is 80". > Do you really have a sustained 80 (I think kbps)? If not, set > a lower quality limit in the settings or configuration file. > > Finally, there's a warning about alsa not supporting the period size. > I doubt that's a serious problem but it probably doesn't help make > it sound good when it does work. I'm not sure but it might suggest > a hardware problem or deficiency. > > Interesting log snippets only below: > > > linphone-message : Received message: > > SIP/2.0 200 OK > [...] > > m=audio 35350 RTP/AVP 0 8 3 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:101 telephone-event/8000 > > > > linphone-message : Call 0x8944f40: moving from state > LinphoneCallOutgoingProgress to LinphoneCallConnected > > linphone-message : Audio bandwidth for this call is 80 > > linphone-message : Payload's bitrate is 80000 > > linphone-message : Setting audio encoder network bitrate to 80000 > > linphone-message : MSUlawEnc: got ptime=20 > > > linphone-warning : alsa_set_params: The period size 256 is not supported > by your hardware. > > ==> Using 170 instead. > > > -- > MJ Ray (slef), member of www.software.coop, a for-more-than-profit co-op. > http://koha-community.org supporter, web and library systems developer. > In My Opinion Only: see http://mjr.towers.org.uk/email.html > Available for hire (including development) at http://www.software.coop/ > > _______________________________________________ > Linphone-users mailing list > [email protected] > https://lists.nongnu.org/mailman/listinfo/linphone-users >
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