[asterisk-users] changing the size of voice packets

[John Todd]

On Nov 10, 2008, at 3:30 AM, Igor Goncharovsky wrote:

> Hi!
>
> On Mon, Nov 10, 2008 at 5:16 PM, Pezhman Lali  
> <pezhman_lali at yahoo.com> 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
jtodd at digium.com        +1-256-428-6083
Asterisk Open Source Community Director