[asterisk-users] Change Packetization Time

Dovid B asteriskusers at dovid.net
Fri Aug 31 06:15:30 CDT 2007


Dan,
I sent this particular's traffic through a 1.4.X box to our 1.2.X box. Worked like a charm.

Thanks for the help.

Dovid
  ----- Original Message ----- 
  From: Dan Austin 
  To: Dovid B 
  Sent: Monday, August 20, 2007 9:22 AM
  Subject: RE: [asterisk-users] Change Packetization Time


  When the patch to provide the support was started, 1.2 had not yet been released,
  but the feature missed the cut-off for 1.2.

  It would be possible to backport the changes, or to updated the last patch for 1.2 in
  Mantis, but I would not expect it to be trivial.  And making such changes would lead
  to a version of 1.2 that would not be supported by the core developers.

  Dan



----------------------------------------------------------------------------
    From: Dovid B [mailto:asteriskusers at dovid.net] 
    Sent: Sunday, August 19, 2007 11:11 PM
    To: Dan Austin
    Subject: Re: [asterisk-users] Change Packetization Time



    ----- Original Message ----- 
    From: "Dan Austin" <Dan_Austin at Phoenix.com>
    To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
    <asterisk-users at lists.digium.com>
    Sent: Sunday, August 19, 2007 7:58 PM
    Subject: Re: [asterisk-users] Change Packetization Time


    > Dovid wrote:
    >
    >>     Does anyone know if it is possible to change the
    >> packetization time in Asterisk ? I was told by a client
    >> of mine that adjusting this with using G729 can greatly
    >> lower the amount of bandwidth used.
    >
    > Your client is correct.  Configurable packetization was added
    > introduced with the release of 1.4.0.  For details look at the
    > rtp-packetization.txt file in the doc directory for full details.
    >
    > The short answer is to append :<size> to any codec on your allow
    > directive that you want to change from the default of 20ms.
    > Ex.
    > Allow=g729:40
    >
    > Dan
    >
    >

    Dan,
    Can I make this change in 1.2.X ? (maybe in the source ?). I have not moved 
    to 1.4.X because of the lack of support. Currently using SpanDSP. 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070831/05230dd2/attachment.htm 


More information about the asterisk-users mailing list