[Asterisk-Users] How to change the packet size
Keith Burns
kburns at porchlightcom.com
Wed Jan 19 19:04:05 MST 2005
Just beware of the effects of changing sample size for any codec.
We found that a sample size of "2" for G.711 (ie 2x20ms) allowed for
pretty robust interoperability between vendors. Not specifically with
Asterisk, but we did find that using a mixed CPE/gw environment with a
couple of Call Agent vendors that Smartbits PSQM scores varied wildly
with changed sample sizes but 2 samples yielded pretty consistent
multi-vendor results.
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Luki
> Sent: Wednesday, January 19, 2005 6:56 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] How to change the packet size
>
> Although this probably isn't the "right" way of doing it, you can
> change in the source code, globally for all calls using a codec:
>
> See the "smooter" creation statement in the function ast_rtp_write:
> rtp->smoother = ast_smoother_new(4 * 50);
>
> (I changed mine to 50 ms for G726 which did wonders for those slooooow
> DSL users to reduce the number of packet/sec, and the latency increase
> is virtually not noticeable to me).
>
> I'm sure we could make a patch to set it on a per-call basis from the
> dialplan... if someone cares to do so.
>
> --Luki
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