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
