[asterisk-users] changing the size of voice packets
jtodd at digium.com
Mon Nov 10 13:17:45 CST 2008
On Nov 10, 2008, at 3:30 AM, Igor Goncharovsky wrote:
> On Mon, Nov 10, 2008 at 5:16 PM, Pezhman Lali
> <pezhman_lali at yahoo.com> wrote:
> 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
> 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.
jtodd at digium.com +1-256-428-6083
Asterisk Open Source Community Director
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