I'm not sure there IS an issue, per se. There are lower bitrate codecs that will work fine for voice communications in both directions. But if you're trying to force a low-end codec to the upstream, that just means the downstream on the remote end is going to be stuck with a low-end codec. And if he's trying to force a low-end codec on his upstream, then you're still going to get a low-end codec on your downstream. If you're relying on the Asterisk box somewhere in the middle to transcode these two streams into a higher bitrate codec, you're still not going to get any higher quality than you send.
Your option is using a low bitrate codec for both directions and not bothering to try and min/max your streams. Excellent quality low bitrate codecs exist in the form of g729, iLBC, and Speex (in that order of prevalence). Just deal with the fact that, even if you min/maxed the streams, you still wouldn't be able to get any more streams than will fit in your upstream pipe, so let that be your guide for the technology involved. You may end up with some extra bandwidth on the downstream side, but trying to fill it up with something just to fill it up with something won't get you anywhere. N. Elliot Murdock wrote: > Hello Everyone! > > Thank you for all the information. > > I am wondering how the Asterisk community has been working on > solutions to deal with the asymmetric quality of ADSL. Voip is > becoming popular and a bottleneck does exists on the ADSL upload side. > > Elliot > > On Sun, Aug 2, 2009 at 3:17 PM, Kevin P. Fleming<kpflem...@digium.com> wrote: > >> Tim Panton wrote: >> >> >>> The protocol expects the 2 ends to agree a single symmetrical codec >>> as part of the connection setup, but it doesn't define what actually >>> happens >>> if the codec specified in the first (full frame) voice packet isn't what >>> was agreed. >>> >> Asterisk only supports symmetric codec configuration on its internal >> channels, so in Asterisk's IAX2 implementation, if a frame is received >> from the other endpoint that is not in the 'expected' format a warning >> is issued and the outbound direction is automatically switched to the >> same format. The same is done for any protocol using RTP in Asterisk. >> >> -- >> Kevin P. Fleming >> Digium, Inc. | Director of Software Technologies >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA >> skype: kpfleming | jabber: kpflem...@digium.com >> Check us out at www.digium.com & www.asterisk.org >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users