[Asterisk-Users] Packetization

Matt mhoppes at gmail.com
Sun Apr 3 14:36:09 MST 2005


I have to admit this still doesn't make sence.. if sipura's default is .03ms 
and asterisk is 20ms.. why is the sipura dumping out around 60 frames/sec 
while the sipura is dumping out around 30 frames/sec??

Shouldn't the frames / packets per second go UP as the packetization gets 
smaller?

On Apr 3, 2005 5:25 PM, Matt <mhoppes at gmail.com> wrote:
> 
> Never mind... blah spoke before I thought :P
> 
> Found the setting....
> 
> 
> On Apr 3, 2005 5:23 PM, Matt <mhoppes at gmail.com> wrote:
> > 
> > Ok.. wow .03ms (if that's the default) is DEFINATELY part of the issue 
> > (audio going from the phone over wireless is slightly choppy).. while audio 
> > coming in (20ms) is ok... where do you change it on the sipura?
> > 
> > On Apr 3, 2005 4:07 PM, Bruce Komito <brucek at bagel.com> wrote:
> > > 
> > > The packet size is a function of the number of milliseconds of sound 
> > > sent
> > > in the RTP packet. I don't know how to force * to change this, but you
> > > *can* unilaterally change the RTP packet size on the Sipura. By doing
> > > this, RTP packets sent by the Sipura will be larger or smaller than 
> > > the
> > > default (.03 ms is the default), and I know * will swallow whatever 
> > > the
> > > Sipura sends it. So, I know it's possible to change this in at least 
> > > one
> > > direction if you are using a Sipura.
> > > 
> > > Bruce Komito
> > > High Sierra Networks, Inc.
> > > www.servers-r-us.com <http://www.servers-r-us.com>
> > > (775) 236-5815
> > > 
> > > 
> > > On Sun, 3 Apr 2005, Matt wrote:
> > > 
> > > > IAX is not an option as Sipura devices do not support AIX.
> > > > Yes, the sipura will handle the different packet sizes...
> > > >
> > > > Is it possible to reprogram asteris to do this?
> > > >
> > > > On Apr 3, 2005 1:55 AM, Steven Critchfield <critch at basesys.com> 
> > > wrote:
> > > > >
> > > > > On Sat, 2005-04-02 at 21:16 -0500, Matt wrote:
> > > > > > I'm aware that asterisk only supports 20ms packetization rates. 
> > > Due
> > > > > > to the fact that I will be using some voip devices on a wireless
> > > > > > network which is highly sensative to framerate.. is there any 
> > > way I
> > > > > > can hard code the packetization rate at say 30 or 40ms and then
> > > > > > compile astrisk? If so, can anyone in the know tell me what 
> > > variables
> > > > > > I need to look at to change?
> > > > >
> > > > > Are you sure your other devices support different packet sizes? 
> > > Are you
> > > > > sure the added delay in audio delivery can be handled decently and 
> > > not
> > > > > cause added echo?
> > > > >
> > > > > Have you considered what IAX trunking can do for you? It will 
> > > reduce
> > > > > frame rate as you add channels since each packet will then hold 
> > > the
> > > > > frames for each of the consecutive calls.
> > > > > --
> > > > > Steven Critchfield <critch at basesys.com>
> > > > >
> > > > >
> > > >
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