On Thu, Mar 6, 2014 at 3:31 PM, George Joseph <[email protected]> wrote: > On Thu, Mar 6, 2014 at 1:22 PM, Scott Griepentrog <[email protected]> > wrote: >> >> First, a smidgen of background: >> >> The two sorcery options for pjsip.conf "allow" and "disallow" both accept >> a list of codecs and set the same table of codecs in behind the scenes. The >> difference being of course that the disallow field inverts the meaning. >> >> There is some potential confusion here as to why there is two lists of the >> exact same codecs (see >> https://issues.asterisk.org/jira/browse/ASTERISK-23092). I have a suggested >> patch (see https://reviewboard.asterisk.org/r/3193/) to make the disallow >> option disappear in a fashion. You can still use the disallow option in >> pjsip.conf, but when viewing the settings with pjsip show endpoint # only >> the allow list would appear. This is accomplished by marking the disallow >> field as an alias. >> >> An option to move away from SIP's convention of allow/disallow and have >> PJSIP use codecs=ulaw,etc has been suggested (and is coded in the review). >> The question then is: >> >> 1) Do we want to discontinue or alias both allow & disallow and move to >> codecs? >> >> >> 2) If yes, then which version should that be done in? 12? 13? > > > My vote...Move to codecs and alias allow/disallow in 12, discontinue > allow/disallow in 13. > >> >> Note that even if codecs is chosen, allow and disallow continue to work so >> no existing pjsip.conf is broken. >> > For me to be on-board with the change, we'd have to apply it to all channel drives that implement said codecs allow / disallow logic, so sip.conf, chan_ooh323.conf, gtalk.conf, h323.conf, iax.conf, jingle.conf.
That way all our documentation / functionality is consistent among channel drivers. -- Paul Belanger | PolyBeacon, Inc. Jabber: [email protected] | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
