[asterisk-users] changing the size of voice packets

[Alex Balashov]

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
>> <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
>
>
>
>
>
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-- 
Alex Balashov
Evariste Systems
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