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: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] 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 > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
