On Nov 4, 2010, at 9:41 AM, C F wrote: > 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?
Out of curiosity, is there a reason why you want to exceed 20ms? The nice thing about 20ms is that in theory you can drop a packet and not notice (audibly). Over 20ms is supposed to be noticed by the human ear. This being said, doc/rtp-packetization will describe the acceptable payload sizes for different codecs. ---fred http://qxork.com -- _____________________________________________________________________ -- 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
