You see the problem is that asterisk will send as many packets as its admin does on the list. There is no way to change that. I suggest you first change the amount of packets per second you send.
On Thu, Nov 4, 2010 at 5:38 AM, ali anjum <[email protected]> wrote: > Hi, > > (I have install trixbox2.8 with asterisk 1.6) > I am using speex codec for my Inter asterisk communication > > Question1: I want to configure speex on 2.15kbs and packetization of 60ms > seconds for that is have configured "codecs.conf" for desired result and > also placed a line in general section of "sip.conf" allow=speex:60 after > disallow=all line . > > I have also configure SIP trunk between two asterisk to use speex:60 > During debugging I have checked that both side accept speex as a codec for > call and ptime:60 but > > I am facing following unexpected results > > 1-> When I check the packet rate from one asterisk to other asterisk for one > call its not (1000/60 == 17)? > > 2-> When ever I change the softphone result changes i.e. data ratae chages ? > > 3-> How can I use my own codec "xyz" in asterisk to place calls ? > > 4->if I change the codecs.conf then no results appears in packet size which > is comming out of asterisk? > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
