If the packetisation durations are different between endpoints, the SDP offer/answer should fail with a "488 Not Acceptable Here." Right?
On Mon, November 10, 2008 2:17 pm, John Todd wrote: > On Nov 10, 2008, at 3:30 AM, Igor Goncharovsky wrote: > >> Hi! >> >> On Mon, Nov 10, 2008 at 5:16 PM, Pezhman Lali >> <[EMAIL PROTECTED]> wrote: >> Dear, >> is any way to change , the size of voice packets? >> I want to increase the quality by decreasing the size of each >> packets, because of bandwidth failure. >> >> You can specify size of voice packets in allow line of sip.conf peer >> configuration. >> ex.: allow=alaw:30,g729:50 >> >> For more information: >> http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.4+rtp-packetization.txt >> >> -- >> Best regards, >> Igor Goncharovsky > > > There was discussion recently (on -dev? on -users? on IRC?) about how > there are some shortcomings on RTP packetization/transcoding. It > appears, though I have not confirmed this, that trying to move a 20ms > G.711 stream from a client, though Asterisk, to a remote gateway using > 40ms G.711 will NOT work correctly. The 20ms packet size is passed > through without aggregating to 40ms, or vice versa - no change in > packetization (though I don't know which side takes precedence.) > Going the opposite directon for dis-aggregation (which is what you > want to do) I assume would fail in similar ways. I don't recall if > changing the codec made any difference on the packetization between > two bridged channels. > > For what it's worth, 10ms is the maximum rate for most codecs. This > creates twice as many packets as 20ms, three times as many as 30ms, > etc. - hopefully your network hardware has sufficient power or your > call volumes are reasonably low so as not to produce an overwhelming > number of Packets Per Second (PPS). Decreasing sampling interval also > gets you closer to reaching your NIC's threshhold of PPS, which often > is not huge. > > I seem to recall asking the person who reported that to open a bug in > Mantis, but I can't find it, though I didn't look exhaustively. If > you can verify this and/or it's relevant to you, please open a ticket > so that it at least will be reviewed. I'd open it myself, but I'm a > bit resource constrained at the moment in an airport lobby. > > JT > > --- > John Todd > [EMAIL PROTECTED] +1-256-428-6083 > Asterisk Open Source Community Director > > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
